diff options
253 files changed, 22114 insertions, 8030 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 394d7d378dc7..841a9365d5fd 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -757,6 +757,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. model - force the model name position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF) probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) + probe_only - Only probing and no codec initialization (default=off); + Useful to check the initial codec status for debugging bdl_pos_adj - Specifies the DMA IRQ timing delay in samples. Passing -1 will make the driver to choose the appropriate value based on the controller chip. @@ -772,327 +774,23 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports multiple cards and autoprobe. + See Documentation/sound/alsa/HD-Audio.txt for more details about + HD-audio driver. + Each codec may have a model table for different configurations. If your machine isn't listed there, the default (usually minimal) configuration is set up. You can pass "model=<name>" option to specify a certain model in such a case. There are different - models depending on the codec chip. - - Model name Description - ---------- ----------- - ALC880 - 3stack 3-jack in back and a headphone out - 3stack-digout 3-jack in back, a HP out and a SPDIF out - 5stack 5-jack in back, 2-jack in front - 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out - 6stack 6-jack in back, 2-jack in front - 6stack-digout 6-jack with a SPDIF out - w810 3-jack - z71v 3-jack (HP shared SPDIF) - asus 3-jack (ASUS Mobo) - asus-w1v ASUS W1V - asus-dig ASUS with SPDIF out - asus-dig2 ASUS with SPDIF out (using GPIO2) - uniwill 3-jack - fujitsu Fujitsu Laptops (Pi1536) - F1734 2-jack - lg LG laptop (m1 express dual) - lg-lw LG LW20/LW25 laptop - tcl TCL S700 - clevo Clevo laptops (m520G, m665n) - medion Medion Rim 2150 - test for testing/debugging purpose, almost all controls can be - adjusted. Appearing only when compiled with - $CONFIG_SND_DEBUG=y - auto auto-config reading BIOS (default) - - ALC260 - hp HP machines - hp-3013 HP machines (3013-variant) - hp-dc7600 HP DC7600 - fujitsu Fujitsu S7020 - acer Acer TravelMate - will Will laptops (PB V7900) - replacer Replacer 672V - basic fixed pin assignment (old default model) - test for testing/debugging purpose, almost all controls can - adjusted. Appearing only when compiled with - $CONFIG_SND_DEBUG=y - auto auto-config reading BIOS (default) - - ALC262 - fujitsu Fujitsu Laptop - hp-bpc HP xw4400/6400/8400/9400 laptops - hp-bpc-d7000 HP BPC D7000 - hp-tc-t5735 HP Thin Client T5735 - hp-rp5700 HP RP5700 - benq Benq ED8 - benq-t31 Benq T31 - hippo Hippo (ATI) with jack detection, Sony UX-90s - hippo_1 Hippo (Benq) with jack detection - sony-assamd Sony ASSAMD - toshiba-s06 Toshiba S06 - toshiba-rx1 Toshiba RX1 - ultra Samsung Q1 Ultra Vista model - lenovo-3000 Lenovo 3000 y410 - nec NEC Versa S9100 - basic fixed pin assignment w/o SPDIF - auto auto-config reading BIOS (default) - - ALC267/268 - quanta-il1 Quanta IL1 mini-notebook - 3stack 3-stack model - toshiba Toshiba A205 - acer Acer laptops - acer-aspire Acer Aspire One - dell Dell OEM laptops (Vostro 1200) - zepto Zepto laptops - test for testing/debugging purpose, almost all controls can - adjusted. Appearing only when compiled with - $CONFIG_SND_DEBUG=y - auto auto-config reading BIOS (default) - - ALC269 - basic Basic preset - quanta Quanta FL1 - eeepc-p703 ASUS Eeepc P703 P900A - eeepc-p901 ASUS Eeepc P901 S101 - - ALC662/663 - 3stack-dig 3-stack (2-channel) with SPDIF - 3stack-6ch 3-stack (6-channel) - 3stack-6ch-dig 3-stack (6-channel) with SPDIF - 6stack-dig 6-stack with SPDIF - lenovo-101e Lenovo laptop - eeepc-p701 ASUS Eeepc P701 - eeepc-ep20 ASUS Eeepc EP20 - ecs ECS/Foxconn mobo - m51va ASUS M51VA - g71v ASUS G71V - h13 ASUS H13 - g50v ASUS G50V - asus-mode1 ASUS - asus-mode2 ASUS - asus-mode3 ASUS - asus-mode4 ASUS - asus-mode5 ASUS - asus-mode6 ASUS - auto auto-config reading BIOS (default) - - ALC882/885 - 3stack-dig 3-jack with SPDIF I/O - 6stack-dig 6-jack digital with SPDIF I/O - arima Arima W820Di1 - targa Targa T8, MSI-1049 T8 - asus-a7j ASUS A7J - asus-a7m ASUS A7M - macpro MacPro support - mbp3 Macbook Pro rev3 - imac24 iMac 24'' with jack detection - w2jc ASUS W2JC - auto auto-config reading BIOS (default) - - ALC883/888 - 3stack-dig 3-jack with SPDIF I/O - 6stack-dig 6-jack digital with SPDIF I/O - 3stack-6ch 3-jack 6-channel - 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O - 6stack-dig-demo 6-jack digital for Intel demo board - acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) - acer-aspire Acer Aspire 9810 - medion Medion Laptops - medion-md2 Medion MD2 - targa-dig Targa/MSI - targa-2ch-dig Targs/MSI with 2-channel - laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) - lenovo-101e Lenovo 101E - lenovo-nb0763 Lenovo NB0763 - lenovo-ms7195-dig Lenovo MS7195 - lenovo-sky Lenovo Sky - haier-w66 Haier W66 - 3stack-hp HP machines with 3stack (Lucknow, Samba boards) - 6stack-dell Dell machines with 6stack (Inspiron 530) - mitac Mitac 8252D - clevo-m720 Clevo M720 laptop series - fujitsu-pi2515 Fujitsu AMILO Pi2515 - 3stack-6ch-intel Intel DG33* boards - auto auto-config reading BIOS (default) - - ALC861/660 - 3stack 3-jack - 3stack-dig 3-jack with SPDIF I/O - 6stack-dig 6-jack with SPDIF I/O - 3stack-660 3-jack (for ALC660) - uniwill-m31 Uniwill M31 laptop - toshiba Toshiba laptop support - asus Asus laptop support - asus-laptop ASUS F2/F3 laptops - auto auto-config reading BIOS (default) - - ALC861VD/660VD - 3stack 3-jack - 3stack-dig 3-jack with SPDIF OUT - 6stack-dig 6-jack with SPDIF OUT - 3stack-660 3-jack (for ALC660VD) - 3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD) - lenovo Lenovo 3000 C200 - dallas Dallas laptops - hp HP TX1000 - auto auto-config reading BIOS (default) - - CMI9880 - minimal 3-jack in back - min_fp 3-jack in back, 2-jack in front - full 6-jack in back, 2-jack in front - full_dig 6-jack in back, 2-jack in front, SPDIF I/O - allout 5-jack in back, 2-jack in front, SPDIF out - auto auto-config reading BIOS (default) - - AD1882 / AD1882A - 3stack 3-stack mode (default) - 6stack 6-stack mode - - AD1884A / AD1883 / AD1984A / AD1984B - desktop 3-stack desktop (default) - laptop laptop with HP jack sensing - mobile mobile devices with HP jack sensing - thinkpad Lenovo Thinkpad X300 - - AD1884 - N/A - - AD1981 - basic 3-jack (default) - hp HP nx6320 - thinkpad Lenovo Thinkpad T60/X60/Z60 - toshiba Toshiba U205 - - AD1983 - N/A - - AD1984 - basic default configuration - thinkpad Lenovo Thinkpad T61/X61 - dell Dell T3400 - - AD1986A - 6stack 6-jack, separate surrounds (default) - 3stack 3-stack, shared surrounds - laptop 2-channel only (FSC V2060, Samsung M50) - laptop-eapd 2-channel with EAPD (Samsung R65, ASUS A6J) - laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100) - ultra 2-channel with EAPD (Samsung Ultra tablet PC) - - AD1988/AD1988B/AD1989A/AD1989B - 6stack 6-jack - 6stack-dig ditto with SPDIF - 3stack 3-jack - 3stack-dig ditto with SPDIF - laptop 3-jack with hp-jack automute - laptop-dig ditto with SPDIF - auto auto-config reading BIOS (default) - - Conexant 5045 - laptop-hpsense Laptop with HP sense (old model laptop) - laptop-micsense Laptop with Mic sense (old model fujitsu) - laptop-hpmicsense Laptop with HP and Mic senses - benq Benq R55E - test for testing/debugging purpose, almost all controls - can be adjusted. Appearing only when compiled with - $CONFIG_SND_DEBUG=y - - Conexant 5047 - laptop Basic Laptop config - laptop-hp Laptop config for some HP models (subdevice 30A5) - laptop-eapd Laptop config with EAPD support - test for testing/debugging purpose, almost all controls - can be adjusted. Appearing only when compiled with - $CONFIG_SND_DEBUG=y - - Conexant 5051 - laptop Basic Laptop config (default) - hp HP Spartan laptop - - STAC9200 - ref Reference board - dell-d21 Dell (unknown) - dell-d22 Dell (unknown) - dell-d23 Dell (unknown) - dell-m21 Dell Inspiron 630m, Dell Inspiron 640m - dell-m22 Dell Latitude D620, Dell Latitude D820 - dell-m23 Dell XPS M1710, Dell Precision M90 - dell-m24 Dell Latitude 120L - dell-m25 Dell Inspiron E1505n - dell-m26 Dell Inspiron 1501 - dell-m27 Dell Inspiron E1705/9400 - gateway Gateway laptops with EAPD control - panasonic Panasonic CF-74 - - STAC9205/9254 - ref Reference board - dell-m42 Dell (unknown) - dell-m43 Dell Precision - dell-m44 Dell Inspiron - - STAC9220/9221 - ref Reference board - 3stack D945 3stack - 5stack D945 5stack + SPDIF - intel-mac-v1 Intel Mac Type 1 - intel-mac-v2 Intel Mac Type 2 - intel-mac-v3 Intel Mac Type 3 - intel-mac-v4 Intel Mac Type 4 - intel-mac-v5 Intel Mac Type 5 - intel-mac-auto Intel Mac (detect type according to subsystem id) - macmini Intel Mac Mini (equivalent with type 3) - macbook Intel Mac Book (eq. type 5) - macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3) - macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3) - imac-intel Intel iMac (eq. type 2) - imac-intel-20 Intel iMac (newer version) (eq. type 3) - dell-d81 Dell (unknown) - dell-d82 Dell (unknown) - dell-m81 Dell (unknown) - dell-m82 Dell XPS M1210 - - STAC9202/9250/9251 - ref Reference board, base config - m2-2 Some Gateway MX series laptops - m6 Some Gateway NX series laptops - pa6 Gateway NX860 series - - STAC9227/9228/9229/927x - ref Reference board - ref-no-jd Reference board without HP/Mic jack detection - 3stack D965 3stack - 5stack D965 5stack + SPDIF - dell-3stack Dell Dimension E520 - dell-bios Fixes with Dell BIOS setup - - STAC92HD71B* - ref Reference board - dell-m4-1 Dell desktops - dell-m4-2 Dell desktops - dell-m4-3 Dell desktops - - STAC92HD73* - ref Reference board - no-jd BIOS setup but without jack-detection - dell-m6-amic Dell desktops/laptops with analog mics - dell-m6-dmic Dell desktops/laptops with digital mics - dell-m6 Dell desktops/laptops with both type of mics - - STAC9872 - vaio Setup for VAIO FE550G/SZ110 - vaio-ar Setup for VAIO AR + models depending on the codec chip. The list of available models + is found in HD-Audio-Models.txt The model name "genric" is treated as a special case. When this model is given, the driver uses the generic codec parser without "codec-patch". It's sometimes good for testing and debugging. If the default configuration doesn't work and one of the above - matches with your device, report it together with the PCI - subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel + matches with your device, report it together with alsa-info.sh + output (with --no-upload option) to kernel bugzilla or alsa-devel ML (see the section "Links and Addresses"). power_save and power_save_controller options are for power-saving @@ -1652,7 +1350,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * AuzenTech X-Meridian * Bgears b-Enspirer * Club3D Theatron DTS - * HT-Omega Claro + * HT-Omega Claro (plus) + * HT-Omega Claro halo (XT) * Razer Barracuda AC-1 * Sondigo Inferno @@ -2409,8 +2108,11 @@ Links and Addresses ALSA project homepage http://www.alsa-project.org - ALSA Bug Tracking System - https://bugtrack.alsa-project.org/bugs/ + Kernel Bugzilla + http://bugzilla.kernel.org/ ALSA Developers ML mailto:alsa-devel@alsa-project.org + + alsa-info.sh script + http://www.alsa-project.org/alsa-info.sh diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt new file mode 100644 index 000000000000..4b7ac21ea9eb --- /dev/null +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -0,0 +1,348 @@ + Model name Description + ---------- ----------- +ALC880 +====== + 3stack 3-jack in back and a headphone out + 3stack-digout 3-jack in back, a HP out and a SPDIF out + 5stack 5-jack in back, 2-jack in front + 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out + 6stack 6-jack in back, 2-jack in front + 6stack-digout 6-jack with a SPDIF out + w810 3-jack + z71v 3-jack (HP shared SPDIF) + asus 3-jack (ASUS Mobo) + asus-w1v ASUS W1V + asus-dig ASUS with SPDIF out + asus-dig2 ASUS with SPDIF out (using GPIO2) + uniwill 3-jack + fujitsu Fujitsu Laptops (Pi1536) + F1734 2-jack + lg LG laptop (m1 express dual) + lg-lw LG LW20/LW25 laptop + tcl TCL S700 + clevo Clevo laptops (m520G, m665n) + medion Medion Rim 2150 + test for testing/debugging purpose, almost all controls can be + adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + auto auto-config reading BIOS (default) + +ALC260 +====== + hp HP machines + hp-3013 HP machines (3013-variant) + hp-dc7600 HP DC7600 + fujitsu Fujitsu S7020 + acer Acer TravelMate + will Will laptops (PB V7900) + replacer Replacer 672V + basic fixed pin assignment (old default model) + test for testing/debugging purpose, almost all controls can + adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + auto auto-config reading BIOS (default) + +ALC262 +====== + fujitsu Fujitsu Laptop + hp-bpc HP xw4400/6400/8400/9400 laptops + hp-bpc-d7000 HP BPC D7000 + hp-tc-t5735 HP Thin Client T5735 + hp-rp5700 HP RP5700 + benq Benq ED8 + benq-t31 Benq T31 + hippo Hippo (ATI) with jack detection, Sony UX-90s + hippo_1 Hippo (Benq) with jack detection + sony-assamd Sony ASSAMD + toshiba-s06 Toshiba S06 + toshiba-rx1 Toshiba RX1 + ultra Samsung Q1 Ultra Vista model + lenovo-3000 Lenovo 3000 y410 + nec NEC Versa S9100 + basic fixed pin assignment w/o SPDIF + auto auto-config reading BIOS (default) + +ALC267/268 +========== + quanta-il1 Quanta IL1 mini-notebook + 3stack 3-stack model + toshiba Toshiba A205 + acer Acer laptops + acer-dmic Acer laptops with digital-mic + acer-aspire Acer Aspire One + dell Dell OEM laptops (Vostro 1200) + zepto Zepto laptops + test for testing/debugging purpose, almost all controls can + adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + auto auto-config reading BIOS (default) + +ALC269 +====== + basic Basic preset + quanta Quanta FL1 + eeepc-p703 ASUS Eeepc P703 P900A + eeepc-p901 ASUS Eeepc P901 S101 + fujitsu FSC Amilo + auto auto-config reading BIOS (default) + +ALC662/663 +========== + 3stack-dig 3-stack (2-channel) with SPDIF + 3stack-6ch 3-stack (6-channel) + 3stack-6ch-dig 3-stack (6-channel) with SPDIF + 6stack-dig 6-stack with SPDIF + lenovo-101e Lenovo laptop + eeepc-p701 ASUS Eeepc P701 + eeepc-ep20 ASUS Eeepc EP20 + ecs ECS/Foxconn mobo + m51va ASUS M51VA + g71v ASUS G71V + h13 ASUS H13 + g50v ASUS G50V + asus-mode1 ASUS + asus-mode2 ASUS + asus-mode3 ASUS + asus-mode4 ASUS + asus-mode5 ASUS + asus-mode6 ASUS + auto auto-config reading BIOS (default) + +ALC882/885 +========== + 3stack-dig 3-jack with SPDIF I/O + 6stack-dig 6-jack digital with SPDIF I/O + arima Arima W820Di1 + targa Targa T8, MSI-1049 T8 + asus-a7j ASUS A7J + asus-a7m ASUS A7M + macpro MacPro support + mbp3 Macbook Pro rev3 + imac24 iMac 24'' with jack detection + w2jc ASUS W2JC + auto auto-config reading BIOS (default) + +ALC883/888 +========== + 3stack-dig 3-jack with SPDIF I/O + 6stack-dig 6-jack digital with SPDIF I/O + 3stack-6ch 3-jack 6-channel + 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O + 6stack-dig-demo 6-jack digital for Intel demo board + acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) + acer-aspire Acer Aspire 9810 + acer-aspire-4930g Acer Aspire 4930G + medion Medion Laptops + medion-md2 Medion MD2 + targa-dig Targa/MSI + targa-2ch-dig Targs/MSI with 2-channel + laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) + lenovo-101e Lenovo 101E + lenovo-nb0763 Lenovo NB0763 + lenovo-ms7195-dig Lenovo MS7195 + lenovo-sky Lenovo Sky + haier-w66 Haier W66 + 3stack-hp HP machines with 3stack (Lucknow, Samba boards) + 6stack-dell Dell machines with 6stack (Inspiron 530) + mitac Mitac 8252D + clevo-m720 Clevo M720 laptop series + fujitsu-pi2515 Fujitsu AMILO Pi2515 + fujitsu-xa3530 Fujitsu AMILO XA3530 + 3stack-6ch-intel Intel DG33* boards + auto auto-config reading BIOS (default) + +ALC861/660 +========== + 3stack 3-jack + 3stack-dig 3-jack with SPDIF I/O + 6stack-dig 6-jack with SPDIF I/O + 3stack-660 3-jack (for ALC660) + uniwill-m31 Uniwill M31 laptop + toshiba Toshiba laptop support + asus Asus laptop support + asus-laptop ASUS F2/F3 laptops + auto auto-config reading BIOS (default) + +ALC861VD/660VD +============== + 3stack 3-jack + 3stack-dig 3-jack with SPDIF OUT + 6stack-dig 6-jack with SPDIF OUT + 3stack-660 3-jack (for ALC660VD) + 3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD) + lenovo Lenovo 3000 C200 + dallas Dallas laptops + hp HP TX1000 + asus-v1s ASUS V1Sn + auto auto-config reading BIOS (default) + +CMI9880 +======= + minimal 3-jack in back + min_fp 3-jack in back, 2-jack in front + full 6-jack in back, 2-jack in front + full_dig 6-jack in back, 2-jack in front, SPDIF I/O + allout 5-jack in back, 2-jack in front, SPDIF out + auto auto-config reading BIOS (default) + +AD1882 / AD1882A +================ + 3stack 3-stack mode (default) + 6stack 6-stack mode + +AD1884A / AD1883 / AD1984A / AD1984B +==================================== + desktop 3-stack desktop (default) + laptop laptop with HP jack sensing + mobile mobile devices with HP jack sensing + thinkpad Lenovo Thinkpad X300 + +AD1884 +====== + N/A + +AD1981 +====== + basic 3-jack (default) + hp HP nx6320 + thinkpad Lenovo Thinkpad T60/X60/Z60 + toshiba Toshiba U205 + +AD1983 +====== + N/A + +AD1984 +====== + basic default configuration + thinkpad Lenovo Thinkpad T61/X61 + dell Dell T3400 + +AD1986A +======= + 6stack 6-jack, separate surrounds (default) + 3stack 3-stack, shared surrounds + laptop 2-channel only (FSC V2060, Samsung M50) + laptop-eapd 2-channel with EAPD (ASUS A6J) + laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100) + ultra 2-channel with EAPD (Samsung Ultra tablet PC) + samsung 2-channel with EAPD (Samsung R65) + +AD1988/AD1988B/AD1989A/AD1989B +============================== + 6stack 6-jack + 6stack-dig ditto with SPDIF + 3stack 3-jack + 3stack-dig ditto with SPDIF + laptop 3-jack with hp-jack automute + laptop-dig ditto with SPDIF + auto auto-config reading BIOS (default) + +Conexant 5045 +============= + laptop-hpsense Laptop with HP sense (old model laptop) + laptop-micsense Laptop with Mic sense (old model fujitsu) + laptop-hpmicsense Laptop with HP and Mic senses + benq Benq R55E + test for testing/debugging purpose, almost all controls + can be adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + +Conexant 5047 +============= + laptop Basic Laptop config + laptop-hp Laptop config for some HP models (subdevice 30A5) + laptop-eapd Laptop config with EAPD support + test for testing/debugging purpose, almost all controls + can be adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + +Conexant 5051 +============= + laptop Basic Laptop config (default) + hp HP Spartan laptop + +STAC9200 +======== + ref Reference board + dell-d21 Dell (unknown) + dell-d22 Dell (unknown) + dell-d23 Dell (unknown) + dell-m21 Dell Inspiron 630m, Dell Inspiron 640m + dell-m22 Dell Latitude D620, Dell Latitude D820 + dell-m23 Dell XPS M1710, Dell Precision M90 + dell-m24 Dell Latitude 120L + dell-m25 Dell Inspiron E1505n + dell-m26 Dell Inspiron 1501 + dell-m27 Dell Inspiron E1705/9400 + gateway Gateway laptops with EAPD control + panasonic Panasonic CF-74 + +STAC9205/9254 +============= + ref Reference board + dell-m42 Dell (unknown) + dell-m43 Dell Precision + dell-m44 Dell Inspiron + +STAC9220/9221 +============= + ref Reference board + 3stack D945 3stack + 5stack D945 5stack + SPDIF + intel-mac-v1 Intel Mac Type 1 + intel-mac-v2 Intel Mac Type 2 + intel-mac-v3 Intel Mac Type 3 + intel-mac-v4 Intel Mac Type 4 + intel-mac-v5 Intel Mac Type 5 + intel-mac-auto Intel Mac (detect type according to subsystem id) + macmini Intel Mac Mini (equivalent with type 3) + macbook Intel Mac Book (eq. type 5) + macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3) + macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3) + imac-intel Intel iMac (eq. type 2) + imac-intel-20 Intel iMac (newer version) (eq. type 3) + dell-d81 Dell (unknown) + dell-d82 Dell (unknown) + dell-m81 Dell (unknown) + dell-m82 Dell XPS M1210 + +STAC9202/9250/9251 +================== + ref Reference board, base config + m2-2 Some Gateway MX series laptops + m6 Some Gateway NX series laptops + pa6 Gateway NX860 series + +STAC9227/9228/9229/927x +======================= + ref Reference board + ref-no-jd Reference board without HP/Mic jack detection + 3stack D965 3stack + 5stack D965 5stack + SPDIF + dell-3stack Dell Dimension E520 + dell-bios Fixes with Dell BIOS setup + +STAC92HD71B* +============ + ref Reference board + dell-m4-1 Dell desktops + dell-m4-2 Dell desktops + dell-m4-3 Dell desktops + +STAC92HD73* +=========== + ref Reference board + no-jd BIOS setup but without jack-detection + dell-m6-amic Dell desktops/laptops with analog mics + dell-m6-dmic Dell desktops/laptops with digital mics + dell-m6 Dell desktops/laptops with both type of mics + +STAC92HD83* +=========== + ref Reference board + +STAC9872 +======== + vaio Setup for VAIO FE550G/SZ110 + vaio-ar Setup for VAIO AR diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt new file mode 100644 index 000000000000..8d68fff71839 --- /dev/null +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -0,0 +1,577 @@ +MORE NOTES ON HD-AUDIO DRIVER +============================= + Takashi Iwai <tiwai@suse.de> + + +GENERAL +------- + +HD-audio is the new standard on-board audio component on modern PCs +after AC97. Although Linux has been supporting HD-audio since long +time ago, there are often problems with new machines. A part of the +problem is broken BIOS, and the rest is the driver implementation. +This document explains the brief trouble-shooting and debugging +methods for the HD-audio hardware. + +The HD-audio component consists of two parts: the controller chip and +the codec chips on the HD-audio bus. Linux provides a single driver +for all controllers, snd-hda-intel. Although the driver name contains +a word of a well-known harware vendor, it's not specific to it but for +all controller chips by other companies. Since the HD-audio +controllers are supposed to be compatible, the single snd-hda-driver +should work in most cases. But, not surprisingly, there are known +bugs and issues specific to each controller type. The snd-hda-intel +driver has a bunch of workarounds for these as described below. + +A controller may have multiple codecs. Usually you have one audio +codec and optionally one modem codec. In theory, there might be +multiple audio codecs, e.g. for analog and digital outputs, and the +driver might not work properly because of conflict of mixer elements. +This should be fixed in future if such hardware really exists. + +The snd-hda-intel driver has several different codec parsers depending +on the codec. It has a generic parser as a fallback, but this +functionality is fairly limited until now. Instead of the generic +parser, usually the codec-specific parser (coded in patch_*.c) is used +for the codec-specific implementations. The details about the +codec-specific problems are explained in the later sections. + +If you are interested in the deep debugging of HD-audio, read the +HD-audio specification at first. The specification is found on +Intel's web page, for example: + +- http://www.intel.com/standards/hdaudio/ + + +HD-AUDIO CONTROLLER +------------------- + +DMA-Position Problem +~~~~~~~~~~~~~~~~~~~~ +The most common problem of the controller is the inaccurate DMA +pointer reporting. The DMA pointer for playback and capture can be +read in two ways, either via a LPIB register or via a position-buffer +map. As default the driver tries to read from the io-mapped +position-buffer, and falls back to LPIB if the position-buffer appears +dead. However, this detection isn't perfect on some devices. In such +a case, you can change the default method via `position_fix` option. + +`position_fix=1` means to use LPIB method explicitly. +`position_fix=2` means to use the position-buffer. 0 is the default +value, the automatic check and fallback to LPIB as described in the +above. If you get a problem of repeated sounds, this option might +help. + +In addition to that, every controller is known to be broken regarding +the wake-up timing. It wakes up a few samples before actually +processing the data on the buffer. This caused a lot of problems, for +example, with ALSA dmix or JACK. Since 2.6.27 kernel, the driver puts +an artificial delay to the wake up timing. This delay is controlled +via `bdl_pos_adj` option. + +When `bdl_pos_adj` is a negative value (as default), it's assigned to +an appropriate value depending on the controller chip. For Intel +chips, it'd be 1 while it'd be 32 for others. Usually this works. +Only in case it doesn't work and you get warning messages, you should +change this parameter to other values. + + +Codec-Probing Problem +~~~~~~~~~~~~~~~~~~~~~ +A less often but a more severe problem is the codec probing. When +BIOS reports the available codec slots wrongly, the driver gets +confused and tries to access the non-existing codec slot. This often +results in the total screw-up, and destructs the further communication +with the codec chips. The symptom appears usually as error messages +like: +------------------------------------------------------------------------ + hda_intel: azx_get_response timeout, switching to polling mode: + last cmd=0x12345678 + hda_intel: azx_get_response timeout, switching to single_cmd mode: + last cmd=0x12345678 +------------------------------------------------------------------------ + +The first line is a warning, and this is usually relatively harmless. +It means that the codec response isn't notified via an IRQ. The +driver uses explicit polling method to read the response. It gives +very slight CPU overhead, but you'd unlikely notice it. + +The second line is, however, a fatal error. If this happens, usually +it means that something is really wrong. Most likely you are +accessing a non-existing codec slot. + +Thus, if the second error message appears, try to narrow the probed +codec slots via `probe_mask` option. It's a bitmask, and each bit +corresponds to the codec slot. For example, to probe only the first +slot, pass `probe_mask=1`. For the first and the third slots, pass +`probe_mask=5` (where 5 = 1 | 4), and so on. + +Since 2.6.29 kernel, the driver has a more robust probing method, so +this error might happen rarely, though. + + +Interrupt Handling +~~~~~~~~~~~~~~~~~~ +In rare but some cases, the interrupt isn't properly handled as +default. You would notice this by the DMA transfer error reported by +ALSA PCM core, for example. Using MSI might help in such a case. +Pass `enable_msi=1` option for enabling MSI. + + +HD-AUDIO CODEC +-------------- + +Model Option +~~~~~~~~~~~~ +The most common problem regarding the HD-audio driver is the +unsupported codec features or the mismatched device configuration. +Most of codec-specific code has several preset models, either to +override the BIOS setup or to provide more comprehensive features. + +The driver checks PCI SSID and looks through the static configuration +table until any matching entry is found. If you have a new machine, +you may see a message like below: +------------------------------------------------------------------------ + hda_codec: Unknown model for ALC880, trying auto-probe from BIOS... +------------------------------------------------------------------------ +Even if you see such a message, DON'T PANIC. Take a deep breath and +keep your towel. First of all, it's an informational message, no +warning, no error. This means that the PCI SSID of your device isn't +listed in the known preset model (white-)list. But, this doesn't mean +that the driver is broken. Many codec-drivers provide the automatic +configuration mechanism based on the BIOS setup. + +The HD-audio codec has usually "pin" widgets, and BIOS sets the default +configuration of each pin, which indicates the location, the +connection type, the jack color, etc. The HD-audio driver can guess +the right connection judging from these default configuration values. +However -- some codec-support codes, such as patch_analog.c, don't +support the automatic probing (yet as of 2.6.28). And, BIOS is often, +yes, pretty often broken. It sets up wrong values and screws up the +driver. + +The preset model is provided basically to overcome such a situation. +When the matching preset model is found in the white-list, the driver +assumes the static configuration of that preset and builds the mixer +elements and PCM streams based on the static information. Thus, if +you have a newer machine with a slightly different PCI SSID from the +existing one, you may have a good chance to re-use the same model. +You can pass the `model` option to specify the preset model instead of +PCI SSID look-up. + +What `model` option values are available depends on the codec chip. +Check your codec chip from the codec proc file (see "Codec Proc-File" +section below). It will show the vendor/product name of your codec +chip. Then, see Documentation/sound/alsa/HD-Audio-Modelstxt file, +the section of HD-audio driver. You can find a list of codecs +and `model` options belonging to each codec. For example, for Realtek +ALC262 codec chip, pass `model=ultra` for devices that are compatible +with Samsung Q1 Ultra. + +Thus, the first thing you can do for any brand-new, unsupported and +non-working HD-audio hardware is to check HD-audio codec and several +different `model` option values. If you have a luck, some of them +might suit with your device well. + +Some codecs such as ALC880 have a special model option `model=test`. +This configures the driver to provide as many mixer controls as +possible for every single pin feature except for the unsolicited +events (and maybe some other specials). Adjust each mixer element and +try the I/O in the way of trial-and-error until figuring out the whole +I/O pin mappings. + +Note that `model=generic` has a special meaning. It means to use the +generic parser regardless of the codec. Usually the codec-specific +parser is much better than the generic parser (as now). Thus this +option is more about the debugging purpose. + + +Speaker and Headphone Output +~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +One of the most frequent (and obvious) bugs with HD-audio is the +silent output from either or both of a built-in speaker and a +headphone jack. In general, you should try a headphone output at +first. A speaker output often requires more additional controls like +the external amplifier bits. Thus a headphone output has a slightly +better chance. + +Before making a bug report, double-check whether the mixer is set up +correctly. The recent version of snd-hda-intel driver provides mostly +"Master" volume control as well as "Front" volume (where Front +indicates the front-channels). In addition, there can be individual +"Headphone" and "Speaker" controls. + +Ditto for the speaker output. There can be "External Amplifier" +switch on some codecs. Turn on this if present. + +Another related problem is the automatic mute of speaker output by +headphone plugging. This feature is implemented in most cases, but +not on every preset model or codec-support code. + +In anyway, try a different model option if you have such a problem. +Some other models may match better and give you more matching +functionality. If none of the available models works, send a bug +report. See the bug report section for details. + +If you are masochistic enough to debug the driver problem, note the +following: + +- The speaker (and the headphone, too) output often requires the + external amplifier. This can be set usually via EAPD verb or a + certain GPIO. If the codec pin supports EAPD, you have a better + chance via SET_EAPD_BTL verb (0x70c). On others, GPIO pin (mostly + it's either GPIO0 or GPIO1) may turn on/off EAPD. +- Some Realtek codecs require special vendor-specific coefficients to + turn on the amplifier. See patch_realtek.c. +- IDT codecs may have extra power-enable/disable controls on each + analog pin. See patch_sigmatel.c. +- Very rare but some devices don't accept the pin-detection verb until + triggered. Issuing GET_PIN_SENSE verb (0xf09) may result in the + codec-communication stall. Some examples are found in + patch_realtek.c. + + +Capture Problems +~~~~~~~~~~~~~~~~ +The capture problems are often because of missing setups of mixers. +Thus, before submitting a bug report, make sure that you set up the +mixer correctly. For example, both "Capture Volume" and "Capture +Switch" have to be set properly in addition to the right "Capture +Source" or "Input Source" selection. Some devices have "Mic Boost" +volume or switch. + +When the PCM device is opened via "default" PCM (without pulse-audio +plugin), you'll likely have "Digital Capture Volume" control as well. +This is provided for the extra gain/attenuation of the signal in +software, especially for the inputs without the hardware volume +control such as digital microphones. Unless really needed, this +should be set to exactly 50%, corresponding to 0dB -- neither extra +gain nor attenuation. When you use "hw" PCM, i.e., a raw access PCM, +this control will have no influence, though. + +It's known that some codecs / devices have fairly bad analog circuits, +and the recorded sound contains a certain DC-offset. This is no bug +of the driver. + +Most of modern laptops have no analog CD-input connection. Thus, the +recording from CD input won't work in many cases although the driver +provides it as the capture source. Use CDDA instead. + +The automatic switching of the built-in and external mic per plugging +is implemented on some codec models but not on every model. Partly +because of my laziness but mostly lack of testers. Feel free to +submit the improvement patch to the author. + + +Direct Debugging +~~~~~~~~~~~~~~~~ +If no model option gives you a better result, and you are a tough guy +to fight against evil, try debugging via hitting the raw HD-audio +codec verbs to the device. Some tools are available: hda-emu and +hda-analyzer. The detailed description is found in the sections +below. You'd need to enable hwdep for using these tools. See "Kernel +Configuration" section. + + +OTHER ISSUES +------------ + +Kernel Configuration +~~~~~~~~~~~~~~~~~~~~ +In general, I recommend you to enable the sound debug option, +`CONFIG_SND_DEBUG=y`, no matter whether you are debugging or not. +This enables snd_printd() macro and others, and you'll get additional +kernel messages at probing. + +In addition, you can enable `CONFIG_SND_DEBUG_VERBOSE=y`. But this +will give you far more messages. Thus turn this on only when you are +sure to want it. + +Don't forget to turn on the appropriate `CONFIG_SND_HDA_CODEC_*` +options. Note that each of them corresponds to the codec chip, not +the controller chip. Thus, even if lspci shows the Nvidia controller, +you may need to choose the option for other vendors. If you are +unsure, just select all yes. + +`CONFIG_SND_HDA_HWDEP` is a useful option for debugging the driver. +When this is enabled, the driver creates hardware-dependent devices +(one per each codec), and you have a raw access to the device via +these device files. For example, `hwC0D2` will be created for the +codec slot #2 of the first card (#0). For debug-tools such as +hda-verb and hda-analyzer, the hwdep device has to be enabled. +Thus, it'd be better to turn this on always. + +`CONFIG_SND_HDA_RECONFIG` is a new option, and this depends on the +hwdep option above. When enabled, you'll have some sysfs files under +the corresponding hwdep directory. See "HD-audio reconfiguration" +section below. + +`CONFIG_SND_HDA_POWER_SAVE` option enables the power-saving feature. +See "Power-saving" section below. + + +Codec Proc-File +~~~~~~~~~~~~~~~ +The codec proc-file is a treasure-chest for debugging HD-audio. +It shows most of useful information of each codec widget. + +The proc file is located in /proc/asound/card*/codec#*, one file per +each codec slot. You can know the codec vendor, product id and +names, the type of each widget, capabilities and so on. +This file, however, doesn't show the jack sensing state, so far. This +is because the jack-sensing might be depending on the trigger state. + +This file will be picked up by the debug tools, and also it can be fed +to the emulator as the primary codec information. See the debug tools +section below. + +This proc file can be also used to check whether the generic parser is +used. When the generic parser is used, the vendor/product ID name +will appear as "Realtek ID 0262", instead of "Realtek ALC262". + + +HD-Audio Reconfiguration +~~~~~~~~~~~~~~~~~~~~~~~~ +This is an experimental feature to allow you re-configure the HD-audio +codec dynamically without reloading the driver. The following sysfs +files are available under each codec-hwdep device directory (e.g. +/sys/class/sound/hwC0D0): + +vendor_id:: + Shows the 32bit codec vendor-id hex number. You can change the + vendor-id value by writing to this file. +subsystem_id:: + Shows the 32bit codec subsystem-id hex number. You can change the + subsystem-id value by writing to this file. +revision_id:: + Shows the 32bit codec revision-id hex number. You can change the + revision-id value by writing to this file. +afg:: + Shows the AFG ID. This is read-only. +mfg:: + Shows the MFG ID. This is read-only. +name:: + Shows the codec name string. Can be changed by writing to this + file. +modelname:: + Shows the currently set `model` option. Can be changed by writing + to this file. +init_verbs:: + The extra verbs to execute at initialization. You can add a verb by + writing to this file. Pass tree numbers, nid, verb and parameter. +hints:: + Shows hint strings for codec parsers for any use. Right now it's + not used. +reconfig:: + Triggers the codec re-configuration. When any value is written to + this file, the driver re-initialize and parses the codec tree + again. All the changes done by the sysfs entries above are taken + into account. +clear:: + Resets the codec, removes the mixer elements and PCM stuff of the + specified codec, and clear all init verbs and hints. + + +Power-Saving +~~~~~~~~~~~~ +The power-saving is a kind of auto-suspend of the device. When the +device is inactive for a certain time, the device is automatically +turned off to save the power. The time to go down is specified via +`power_save` module option, and this option can be changed dynamically +via sysfs. + +The power-saving won't work when the analog loopback is enabled on +some codecs. Make sure that you mute all unneeded signal routes when +you want the power-saving. + +The power-saving feature might cause audible click noises at each +power-down/up depending on the device. Some of them might be +solvable, but some are hard, I'm afraid. Some distros such as +openSUSE enables the power-saving feature automatically when the power +cable is unplugged. Thus, if you hear noises, suspect first the +power-saving. See /sys/module/snd_hda_intel/parameters/power_save to +check the current value. If it's non-zero, the feature is turned on. + + +Development Tree +~~~~~~~~~~~~~~~~ +The latest development codes for HD-audio are found on sound git tree: + +- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git + +The master branch or for-next branches can be used as the main +development branches in general while the HD-audio specific patches +are committed in topic/hda branch. + +If you are using the latest Linus tree, it'd be better to pull the +above GIT tree onto it. If you are using the older kernels, an easy +way to try the latest ALSA code is to build from the snapshot +tarball. There are daily tarballs and the latest snapshot tarball. +All can be built just like normal alsa-driver release packages, that +is, installed via the usual spells: configure, make and make +install(-modules). See INSTALL in the package. The snapshot tarballs +are found at: + +- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ + + +Sending a Bug Report +~~~~~~~~~~~~~~~~~~~~ +If any model or module options don't work for your device, it's time +to send a bug report to the developers. Give the following in your +bug report: + +- Hardware vendor, product and model names +- Kernel version (and ALSA-driver version if you built externally) +- `alsa-info.sh` output; run with `--no-upload` option. See the + section below about alsa-info + +If it's a regression, at best, send alsa-info outputs of both working +and non-working kernels. This is really helpful because we can +compare the codec registers directly. + +Send a bug report either the followings: + +kernel-bugzilla:: + http://bugme.linux-foundation.org/ +alsa-devel ML:: + alsa-devel@alsa-project.org + + +DEBUG TOOLS +----------- + +This section describes some tools available for debugging HD-audio +problems. + +alsa-info +~~~~~~~~~ +The script `alsa-info.sh` is a very useful tool to gather the audio +device information. You can fetch the latest version from: + +- http://www.alsa-project.org/alsa-info.sh + +Run this script as root, and it will gather the important information +such as the module lists, module parameters, proc file contents +including the codec proc files, mixer outputs and the control +elements. As default, it will store the information onto a web server +on alsa-project.org. But, if you send a bug report, it'd be better to +run with `--no-upload` option, and attach the generated file. + +There are some other useful options. See `--help` option output for +details. + + +hda-verb +~~~~~~~~ +hda-verb is a tiny program that allows you to access the HD-audio +codec directly. You can execute a raw HD-audio codec verb with this. +This program accesses the hwdep device, thus you need to enable the +kernel config `CONFIG_SND_HDA_HWDEP=y` beforehand. + +The hda-verb program takes four arguments: the hwdep device file, the +widget NID, the verb and the parameter. When you access to the codec +on the slot 2 of the card 0, pass /dev/snd/hwC0D2 to the first +argument, typically. (However, the real path name depends on the +system.) + +The second parameter is the widget number-id to access. The third +parameter can be either a hex/digit number or a string corresponding +to a verb. Similarly, the last parameter is the value to write, or +can be a string for the parameter type. + +------------------------------------------------------------------------ + % hda-verb /dev/snd/hwC0D0 0x12 0x701 2 + nid = 0x12, verb = 0x701, param = 0x2 + value = 0x0 + + % hda-verb /dev/snd/hwC0D0 0x0 PARAMETERS VENDOR_ID + nid = 0x0, verb = 0xf00, param = 0x0 + value = 0x10ec0262 + + % hda-verb /dev/snd/hwC0D0 2 set_a 0xb080 + nid = 0x2, verb = 0x300, param = 0xb080 + value = 0x0 +------------------------------------------------------------------------ + +Although you can issue any verbs with this program, the driver state +won't be always updated. For example, the volume values are usually +cached in the driver, and thus changing the widget amp value directly +via hda-verb won't change the mixer value. + +The hda-verb program is found in the ftp directory: + +- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/ + +Also a git repository is available: + +- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-verb.git + +See README file in the tarball for more details about hda-verb +program. + + +hda-analyzer +~~~~~~~~~~~~ +hda-analyzer provides a graphical interface to access the raw HD-audio +control, based on pyGTK2 binding. It's a more powerful version of +hda-verb. The program gives you an easy-to-use GUI stuff for showing +the widget information and adjusting the amp values, as well as the +proc-compatible output. + +The hda-analyzer is a part of alsa.git repository in +alsa-project.org: + +- http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer + + +Codecgraph +~~~~~~~~~~ +Codecgraph is a utility program to generate a graph and visualizes the +codec-node connection of a codec chip. It's especially useful when +you analyze or debug a codec without a proper datasheet. The program +parses the given codec proc file and converts to SVG via graphiz +program. + +The tarball and GIT trees are found in the web page at: + +- http://helllabs.org/codecgraph/ + + +hda-emu +~~~~~~~ +hda-emu is an HD-audio emulator. The main purpose of this program is +to debug an HD-audio codec without the real hardware. Thus, it +doesn't emulate the behavior with the real audio I/O, but it just +dumps the codec register changes and the ALSA-driver internal changes +at probing and operating the HD-audio driver. + +The program requires a codec proc-file to simulate. Get a proc file +for the target codec beforehand, or pick up an example codec from the +codec proc collections in the tarball. Then, run the program with the +proc file, and the hda-emu program will start parsing the codec file +and simulates the HD-audio driver: + +------------------------------------------------------------------------ + % hda-emu codecs/stac9200-dell-d820-laptop + # Parsing.. + hda_codec: Unknown model for STAC9200, using BIOS defaults + hda_codec: pin nid 08 bios pin config 40c003fa + .... +------------------------------------------------------------------------ + +The program gives you only a very dumb command-line interface. You +can get a proc-file dump at the current state, get a list of control +(mixer) elements, set/get the control element value, simulate the PCM +operation, the jack plugging simulation, etc. + +The package is found in: + +- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/ + +A git repository is available: + +- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git + +See README file in the tarball for more details about hda-emu +program. diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt index f738b296440a..bba2dbb79d81 100644 --- a/Documentation/sound/alsa/Procfile.txt +++ b/Documentation/sound/alsa/Procfile.txt @@ -153,6 +153,16 @@ card*/codec#* Shows the general codec information and the attribute of each widget node. +card*/eld#* + Available for HDMI or DisplayPort interfaces. + Shows ELD(EDID Like Data) info retrieved from the attached HDMI sink, + and describes its audio capabilities and configurations. + + Some ELD fields may be modified by doing `echo name hex_value > eld#*`. + Only do this if you are sure the HDMI sink provided value is wrong. + And if that makes your HDMI audio work, please report to us so that we + can fix it in future kernel releases. + Sequencer Information --------------------- diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt index f370e7db86af..bab7711ce963 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/alsa/soc/machine.txt @@ -9,7 +9,7 @@ the audio subsystem with the kernel as a platform device and is represented by the following struct:- /* SoC machine */ -struct snd_soc_machine { +struct snd_soc_card { char *name; int (*probe)(struct platform_device *pdev); @@ -67,10 +67,10 @@ static struct snd_soc_dai_link corgi_dai = { .ops = &corgi_ops, }; -struct snd_soc_machine then sets up the machine with it's DAIs. e.g. +struct snd_soc_card then sets up the machine with it's DAIs. e.g. /* corgi audio machine driver */ -static struct snd_soc_machine snd_soc_machine_corgi = { +static struct snd_soc_card snd_soc_corgi = { .name = "Corgi", .dai_link = &corgi_dai, .num_links = 1, @@ -90,7 +90,7 @@ static struct wm8731_setup_data corgi_wm8731_setup = { /* corgi audio subsystem */ static struct snd_soc_device corgi_snd_devdata = { - .machine = &snd_soc_machine_corgi, + .machine = &snd_soc_corgi, .platform = &pxa2xx_soc_platform, .codec_dev = &soc_codec_dev_wm8731, .codec_data = &corgi_wm8731_setup, diff --git a/MAINTAINERS b/MAINTAINERS index fbc8fa58d56d..701025701514 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3977,7 +3977,7 @@ M: tiwai@suse.de L: alsa-devel@alsa-project.org (subscribers-only) S: Maintained -SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT +SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) P: Liam Girdwood M: lrg@slimlogic.co.uk P: Mark Brown diff --git a/arch/arm/mach-pxa/include/mach/palmasoc.h b/arch/arm/mach-pxa/include/mach/palmasoc.h new file mode 100644 index 000000000000..6c4b1f7de20a --- /dev/null +++ b/arch/arm/mach-pxa/include/mach/palmasoc.h @@ -0,0 +1,13 @@ +#ifndef _INCLUDE_PALMASOC_H_ +#define _INCLUDE_PALMASOC_H_ +struct palm27x_asoc_info { + int jack_gpio; +}; + +#ifdef CONFIG_SND_PXA2XX_SOC_PALM27X +void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data); +#else +static inline void palm27x_asoc_set_pdata(struct palm27x_asoc_info *data) {} +#endif + +#endif diff --git a/include/linux/input.h b/include/linux/input.h index 5341e8251f8c..9a6355f74db2 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -659,6 +659,8 @@ struct input_absinfo { #define SW_RADIO SW_RFKILL_ALL /* deprecated */ #define SW_MICROPHONE_INSERT 0x04 /* set = inserted */ #define SW_DOCK 0x05 /* set = plugged into dock */ +#define SW_LINEOUT_INSERT 0x06 /* set = inserted */ +#define SW_JACK_PHYSICAL_INSERT 0x07 /* set = mechanical switch set */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h index 217bb22ebb8e..af95a1d2f3a1 100644 --- a/include/linux/mfd/wm8350/audio.h +++ b/include/linux/mfd/wm8350/audio.h @@ -1,7 +1,7 @@ /* * audio.h -- Audio Driver for Wolfson WM8350 PMIC * - * Copyright 2007 Wolfson Microelectronics PLC + * Copyright 2007, 2008 Wolfson Microelectronics PLC * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -70,9 +70,9 @@ #define WM8350_CODEC_ISEL_0_5 3 /* x0.5 */ #define WM8350_VMID_OFF 0 -#define WM8350_VMID_500K 1 -#define WM8350_VMID_100K 2 -#define WM8350_VMID_10K 3 +#define WM8350_VMID_300K 1 +#define WM8350_VMID_50K 2 +#define WM8350_VMID_5K 3 /* * R40 (0x28) - Clock Control 1 @@ -591,8 +591,38 @@ #define WM8350_IRQ_CODEC_MICSCD 41 #define WM8350_IRQ_CODEC_MICD 42 +/* + * WM8350 Platform data. + * + * This must be initialised per platform for best audio performance. + * Please see WM8350 datasheet for information. + */ +struct wm8350_audio_platform_data { + int vmid_discharge_msecs; /* VMID --> OFF discharge time */ + int drain_msecs; /* OFF drain time */ + int cap_discharge_msecs; /* Cap ON (from OFF) discharge time */ + int vmid_charge_msecs; /* vmid power up time */ + u32 vmid_s_curve:2; /* vmid enable s curve speed */ + u32 dis_out4:2; /* out4 discharge speed */ + u32 dis_out3:2; /* out3 discharge speed */ + u32 dis_out2:2; /* out2 discharge speed */ + u32 dis_out1:2; /* out1 discharge speed */ + u32 vroi_out4:1; /* out4 tie off */ + u32 vroi_out3:1; /* out3 tie off */ + u32 vroi_out2:1; /* out2 tie off */ + u32 vroi_out1:1; /* out1 tie off */ + u32 vroi_enable:1; /* enable tie off */ + u32 codec_current_on:2; /* current level ON */ + u32 codec_current_standby:2; /* current level STANDBY */ + u32 codec_current_charge:2; /* codec current @ vmid charge */ +}; + +struct snd_soc_codec; + struct wm8350_codec { struct platform_device *pdev; + struct snd_soc_codec *codec; + struct wm8350_audio_platform_data *platform_data; }; #endif diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 9c309daf492b..251fc1cd5002 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -281,10 +281,12 @@ /* specific - Analog Devices */ #define AC97_AD_TEST 0x5a /* test register */ #define AC97_AD_TEST2 0x5c /* undocumented test register 2 */ +#define AC97_AD_HPFD_SHIFT 12 /* High Pass Filter Disable */ #define AC97_AD_CODEC_CFG 0x70 /* codec configuration */ #define AC97_AD_JACK_SPDIF 0x72 /* Jack Sense & S/PDIF */ #define AC97_AD_SERIAL_CFG 0x74 /* Serial Configuration */ #define AC97_AD_MISC 0x76 /* Misc Control Bits */ +#define AC97_AD_VREFD_SHIFT 2 /* V_REFOUT Disable (AD1888) */ /* specific - Cirrus Logic */ #define AC97_CSR_ACMODE 0x5e /* AC Mode Register */ diff --git a/include/sound/asound.h b/include/sound/asound.h index 2c4dc908a54a..1c02ed1d7c4a 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -575,6 +575,7 @@ enum { #define SNDRV_TIMER_GLOBAL_SYSTEM 0 #define SNDRV_TIMER_GLOBAL_RTC 1 #define SNDRV_TIMER_GLOBAL_HPET 2 +#define SNDRV_TIMER_GLOBAL_HRTIMER 3 /* info flags */ #define SNDRV_TIMER_FLG_SLAVE (1<<0) /* cannot be controlled */ diff --git a/include/sound/core.h b/include/sound/core.h index 1508c4ec1ba9..f632484bc743 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -353,7 +353,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) * snd_printk - printk wrapper * @fmt: format string * - * Works like print() but prints the file and the line of the caller + * Works like printk() but prints the file and the line of the caller * when configured with CONFIG_SND_VERBOSE_PRINTK. */ #define snd_printk(fmt, args...) \ @@ -380,18 +380,40 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) printk(fmt ,##args) #endif +/** + * snd_BUG - give a BUG warning message and stack trace + * + * Calls WARN() if CONFIG_SND_DEBUG is set. + * Ignored when CONFIG_SND_DEBUG is not set. + */ #define snd_BUG() WARN(1, "BUG?\n") + +/** + * snd_BUG_ON - debugging check macro + * @cond: condition to evaluate + * + * When CONFIG_SND_DEBUG is set, this macro evaluates the given condition, + * and call WARN() and returns the value if it's non-zero. + * + * When CONFIG_SND_DEBUG is not set, this just returns zero, and the given + * condition is ignored. + * + * NOTE: the argument won't be evaluated at all when CONFIG_SND_DEBUG=n. + * Thus, don't put any statement that influences on the code behavior, + * such as pre/post increment, to the argument of this macro. + * If you want to evaluate and give a warning, use standard WARN_ON(). + */ #define snd_BUG_ON(cond) WARN((cond), "BUG? (%s)\n", __stringify(cond)) #else /* !CONFIG_SND_DEBUG */ #define snd_printd(fmt, args...) do { } while (0) #define snd_BUG() do { } while (0) -static inline int __snd_bug_on(void) +static inline int __snd_bug_on(int cond) { return 0; } -#define snd_BUG_ON(cond) __snd_bug_on() /* always false */ +#define snd_BUG_ON(cond) __snd_bug_on(0 && (cond)) /* always false */ #endif /* CONFIG_SND_DEBUG */ diff --git a/include/sound/info.h b/include/sound/info.h index 8ae72e74f898..7c2ee1a21b00 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -40,30 +40,34 @@ struct snd_info_buffer { struct snd_info_entry; struct snd_info_entry_text { - void (*read) (struct snd_info_entry *entry, struct snd_info_buffer *buffer); - void (*write) (struct snd_info_entry *entry, struct snd_info_buffer *buffer); + void (*read)(struct snd_info_entry *entry, + struct snd_info_buffer *buffer); + void (*write)(struct snd_info_entry *entry, + struct snd_info_buffer *buffer); }; struct snd_info_entry_ops { - int (*open) (struct snd_info_entry *entry, - unsigned short mode, void **file_private_data); - int (*release) (struct snd_info_entry * entry, - unsigned short mode, void *file_private_data); - long (*read) (struct snd_info_entry *entry, void *file_private_data, - struct file * file, char __user *buf, + int (*open)(struct snd_info_entry *entry, + unsigned short mode, void **file_private_data); + int (*release)(struct snd_info_entry *entry, + unsigned short mode, void *file_private_data); + long (*read)(struct snd_info_entry *entry, void *file_private_data, + struct file *file, char __user *buf, + unsigned long count, unsigned long pos); + long (*write)(struct snd_info_entry *entry, void *file_private_data, + struct file *file, const char __user *buf, unsigned long count, unsigned long pos); - long (*write) (struct snd_info_entry *entry, void *file_private_data, - struct file * file, const char __user *buf, - unsigned long count, unsigned long pos); - long long (*llseek) (struct snd_info_entry *entry, void *file_private_data, - struct file * file, long long offset, int orig); - unsigned int (*poll) (struct snd_info_entry *entry, void *file_private_data, - struct file * file, poll_table * wait); - int (*ioctl) (struct snd_info_entry *entry, void *file_private_data, - struct file * file, unsigned int cmd, unsigned long arg); - int (*mmap) (struct snd_info_entry *entry, void *file_private_data, - struct inode * inode, struct file * file, - struct vm_area_struct * vma); + long long (*llseek)(struct snd_info_entry *entry, + void *file_private_data, struct file *file, + long long offset, int orig); + unsigned int(*poll)(struct snd_info_entry *entry, + void *file_private_data, struct file *file, + poll_table *wait); + int (*ioctl)(struct snd_info_entry *entry, void *file_private_data, + struct file *file, unsigned int cmd, unsigned long arg); + int (*mmap)(struct snd_info_entry *entry, void *file_private_data, + struct inode *inode, struct file *file, + struct vm_area_struct *vma); }; struct snd_info_entry { @@ -106,34 +110,37 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer); static inline void snd_card_info_read_oss(struct snd_info_buffer *buffer) {} #endif -int snd_iprintf(struct snd_info_buffer * buffer, char *fmt,...) __attribute__ ((format (printf, 2, 3))); +int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) \ + __attribute__ ((format (printf, 2, 3))); int snd_info_init(void); int snd_info_done(void); -int snd_info_get_line(struct snd_info_buffer * buffer, char *line, int len); +int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len); char *snd_info_get_str(char *dest, char *src, int len); -struct snd_info_entry *snd_info_create_module_entry(struct module * module, +struct snd_info_entry *snd_info_create_module_entry(struct module *module, const char *name, - struct snd_info_entry * parent); -struct snd_info_entry *snd_info_create_card_entry(struct snd_card * card, + struct snd_info_entry *parent); +struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, - struct snd_info_entry * parent); -void snd_info_free_entry(struct snd_info_entry * entry); -int snd_info_store_text(struct snd_info_entry * entry); -int snd_info_restore_text(struct snd_info_entry * entry); - -int snd_info_card_create(struct snd_card * card); -int snd_info_card_register(struct snd_card * card); -int snd_info_card_free(struct snd_card * card); -void snd_info_card_disconnect(struct snd_card * card); -int snd_info_register(struct snd_info_entry * entry); + struct snd_info_entry *parent); +void snd_info_free_entry(struct snd_info_entry *entry); +int snd_info_store_text(struct snd_info_entry *entry); +int snd_info_restore_text(struct snd_info_entry *entry); + +int snd_info_card_create(struct snd_card *card); +int snd_info_card_register(struct snd_card *card); +int snd_info_card_free(struct snd_card *card); +void snd_info_card_disconnect(struct snd_card *card); +void snd_info_card_id_change(struct snd_card *card); +int snd_info_register(struct snd_info_entry *entry); /* for card drivers */ -int snd_card_proc_new(struct snd_card *card, const char *name, struct snd_info_entry **entryp); +int snd_card_proc_new(struct snd_card *card, const char *name, + struct snd_info_entry **entryp); static inline void snd_info_set_text_ops(struct snd_info_entry *entry, - void *private_data, - void (*read)(struct snd_info_entry *, struct snd_info_buffer *)) + void *private_data, + void (*read)(struct snd_info_entry *, struct snd_info_buffer *)) { entry->private_data = private_data; entry->c.text.read = read; @@ -146,21 +153,22 @@ int snd_info_check_reserved_words(const char *str); #define snd_seq_root NULL #define snd_oss_root NULL -static inline int snd_iprintf(struct snd_info_buffer * buffer, char *fmt,...) { return 0; } +static inline int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) { return 0; } static inline int snd_info_init(void) { return 0; } static inline int snd_info_done(void) { return 0; } -static inline int snd_info_get_line(struct snd_info_buffer * buffer, char *line, int len) { return 0; } +static inline int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { return 0; } static inline char *snd_info_get_str(char *dest, char *src, int len) { return NULL; } -static inline struct snd_info_entry *snd_info_create_module_entry(struct module * module, const char *name, struct snd_info_entry * parent) { return NULL; } -static inline struct snd_info_entry *snd_info_create_card_entry(struct snd_card * card, const char *name, struct snd_info_entry * parent) { return NULL; } -static inline void snd_info_free_entry(struct snd_info_entry * entry) { ; } - -static inline int snd_info_card_create(struct snd_card * card) { return 0; } -static inline int snd_info_card_register(struct snd_card * card) { return 0; } -static inline int snd_info_card_free(struct snd_card * card) { return 0; } -static inline void snd_info_card_disconnect(struct snd_card * card) { } -static inline int snd_info_register(struct snd_info_entry * entry) { return 0; } +static inline struct snd_info_entry *snd_info_create_module_entry(struct module *module, const char *name, struct snd_info_entry *parent) { return NULL; } +static inline struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, struct snd_info_entry *parent) { return NULL; } +static inline void snd_info_free_entry(struct snd_info_entry *entry) { ; } + +static inline int snd_info_card_create(struct snd_card *card) { return 0; } +static inline int snd_info_card_register(struct snd_card *card) { return 0; } +static inline int snd_info_card_free(struct snd_card *card) { return 0; } +static inline void snd_info_card_disconnect(struct snd_card *card) { } +static inline void snd_info_card_id_change(struct snd_card *card) { } +static inline int snd_info_register(struct snd_info_entry *entry) { return 0; } static inline int snd_card_proc_new(struct snd_card *card, const char *name, struct snd_info_entry **entryp) { return -EINVAL; } diff --git a/include/sound/jack.h b/include/sound/jack.h index b1b2b8b59adb..2e0315cdd0d6 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -35,6 +35,8 @@ enum snd_jack_types { SND_JACK_HEADPHONE = 0x0001, SND_JACK_MICROPHONE = 0x0002, SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, + SND_JACK_LINEOUT = 0x0004, + SND_JACK_MECHANICAL = 0x0008, /* If detected separately */ }; struct snd_jack { diff --git a/include/sound/l3.h b/include/sound/l3.h new file mode 100644 index 000000000000..423a08f0f1b0 --- /dev/null +++ b/include/sound/l3.h @@ -0,0 +1,18 @@ +#ifndef _L3_H_ +#define _L3_H_ 1 + +struct l3_pins { + void (*setdat)(int); + void (*setclk)(int); + void (*setmode)(int); + int data_hold; + int data_setup; + int clock_high; + int mode_hold; + int mode; + int mode_setup; +}; + +int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len); + +#endif diff --git a/include/sound/s3c24xx_uda134x.h b/include/sound/s3c24xx_uda134x.h new file mode 100644 index 000000000000..33df4cb909d3 --- /dev/null +++ b/include/sound/s3c24xx_uda134x.h @@ -0,0 +1,14 @@ +#ifndef _S3C24XX_UDA134X_H_ +#define _S3C24XX_UDA134X_H_ 1 + +#include <sound/uda134x.h> + +struct s3c24xx_uda134x_platform_data { + int l3_clk; + int l3_mode; + int l3_data; + void (*power) (int); + int model; +}; + +#endif diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h new file mode 100644 index 000000000000..24247f763608 --- /dev/null +++ b/include/sound/soc-dai.h @@ -0,0 +1,231 @@ +/* + * linux/sound/soc-dai.h -- ALSA SoC Layer + * + * Copyright: 2005-2008 Wolfson Microelectronics. PLC. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Digital Audio Interface (DAI) API. + */ + +#ifndef __LINUX_SND_SOC_DAI_H +#define __LINUX_SND_SOC_DAI_H + + +#include <linux/list.h> + +struct snd_pcm_substream; + +/* + * DAI hardware audio formats. + * + * Describes the physical PCM data formating and clocking. Add new formats + * to the end. + */ +#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */ +#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */ +#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */ +#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */ +#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */ +#define SND_SOC_DAIFMT_AC97 5 /* AC97 */ + +/* left and right justified also known as MSB and LSB respectively */ +#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J +#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J + +/* + * DAI Clock gating. + * + * DAI bit clocks can be be gated (disabled) when not the DAI is not + * sending or receiving PCM data in a frame. This can be used to save power. + */ +#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */ +#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */ + +/* + * DAI Left/Right Clocks. + * + * Specifies whether the DAI can support different samples for similtanious + * playback and capture. This usually requires a seperate physical frame + * clock for playback and capture. + */ +#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */ +#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */ + +/* + * TDM + * + * Time Division Multiplexing. Allows PCM data to be multplexed with other + * data on the DAI. + */ +#define SND_SOC_DAIFMT_TDM (1 << 6) + +/* + * DAI hardware signal inversions. + * + * Specifies whether the DAI can also support inverted clocks for the specified + * format. + */ +#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ +#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */ +#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */ +#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */ + +/* + * DAI hardware clock masters. + * + * This is wrt the codec, the inverse is true for the interface + * i.e. if the codec is clk and frm master then the interface is + * clk and frame slave. + */ +#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */ +#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */ +#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ +#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */ + +#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f +#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 +#define SND_SOC_DAIFMT_INV_MASK 0x0f00 +#define SND_SOC_DAIFMT_MASTER_MASK 0xf000 + +/* + * Master Clock Directions + */ +#define SND_SOC_CLOCK_IN 0 +#define SND_SOC_CLOCK_OUT 1 + +struct snd_soc_dai_ops; +struct snd_soc_dai; +struct snd_ac97_bus_ops; + +/* Digital Audio Interface registration */ +int snd_soc_register_dai(struct snd_soc_dai *dai); +void snd_soc_unregister_dai(struct snd_soc_dai *dai); +int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count); +void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count); + +/* Digital Audio Interface clocking API.*/ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir); + +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div); + +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out); + +/* Digital Audio interface formatting */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); + +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots); + +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); + +/* Digital Audio Interface mute */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); + +/* + * Digital Audio Interface. + * + * Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97 + * operations an capabilities. Codec and platfom drivers will register a this + * structure for every DAI they have. + * + * This structure covers the clocking, formating and ALSA operations for each + * interface a + */ +struct snd_soc_dai_ops { + /* + * DAI clocking configuration, all optional. + * Called by soc_card drivers, normally in their hw_params. + */ + int (*set_sysclk)(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir); + int (*set_pll)(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out); + int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); + + /* + * DAI format configuration + * Called by soc_card drivers, normally in their hw_params. + */ + int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); + int (*set_tdm_slot)(struct snd_soc_dai *dai, + unsigned int mask, int slots); + int (*set_tristate)(struct snd_soc_dai *dai, int tristate); + + /* + * DAI digital mute - optional. + * Called by soc-core to minimise any pops. + */ + int (*digital_mute)(struct snd_soc_dai *dai, int mute); + + /* + * ALSA PCM audio operations - all optional. + * Called by soc-core during audio PCM operations. + */ + int (*startup)(struct snd_pcm_substream *, + struct snd_soc_dai *); + void (*shutdown)(struct snd_pcm_substream *, + struct snd_soc_dai *); + int (*hw_params)(struct snd_pcm_substream *, + struct snd_pcm_hw_params *, struct snd_soc_dai *); + int (*hw_free)(struct snd_pcm_substream *, + struct snd_soc_dai *); + int (*prepare)(struct snd_pcm_substream *, + struct snd_soc_dai *); + int (*trigger)(struct snd_pcm_substream *, int, + struct snd_soc_dai *); +}; + +/* + * Digital Audio Interface runtime data. + * + * Holds runtime data for a DAI. + */ +struct snd_soc_dai { + /* DAI description */ + char *name; + unsigned int id; + int ac97_control; + + struct device *dev; + + /* DAI callbacks */ + int (*probe)(struct platform_device *pdev, + struct snd_soc_dai *dai); + void (*remove)(struct platform_device *pdev, + struct snd_soc_dai *dai); + int (*suspend)(struct snd_soc_dai *dai); + int (*resume)(struct snd_soc_dai *dai); + + /* ops */ + struct snd_soc_dai_ops ops; + + /* DAI capabilities */ + struct snd_soc_pcm_stream capture; + struct snd_soc_pcm_stream playback; + + /* DAI runtime info */ + struct snd_pcm_runtime *runtime; + struct snd_soc_codec *codec; + unsigned int active; + unsigned char pop_wait:1; + void *dma_data; + + /* DAI private data */ + void *private_data; + + /* parent codec/platform */ + union { + struct snd_soc_codec *codec; + struct snd_soc_platform *platform; + }; + + struct list_head list; +}; + +#endif diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ca699a3017f3..7ee2f70ca42e 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -221,8 +221,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, int num); /* dapm path setup */ -int __deprecated snd_soc_dapm_connect_input(struct snd_soc_codec *codec, - const char *sink_name, const char *control_name, const char *src_name); int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec); void snd_soc_dapm_free(struct snd_soc_device *socdev); int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, diff --git a/include/sound/soc.h b/include/sound/soc.h index 5e0189876afd..f86e455d3828 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -21,8 +21,6 @@ #include <sound/control.h> #include <sound/ac97_codec.h> -#define SND_SOC_VERSION "0.13.2" - /* * Convenience kcontrol builders */ @@ -145,105 +143,31 @@ enum snd_soc_bias_level { SND_SOC_BIAS_OFF, }; -/* - * Digital Audio Interface (DAI) types - */ -#define SND_SOC_DAI_AC97 0x1 -#define SND_SOC_DAI_I2S 0x2 -#define SND_SOC_DAI_PCM 0x4 -#define SND_SOC_DAI_AC97_BUS 0x8 /* for custom i.e. non ac97_codec.c */ - -/* - * DAI hardware audio formats - */ -#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */ -#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right justified mode */ -#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */ -#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM or LRC */ -#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM or LRC */ -#define SND_SOC_DAIFMT_AC97 5 /* AC97 */ - -#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J -#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J - -/* - * DAI Gating - */ -#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */ -#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated when not Tx/Rx */ - -/* - * DAI Sync - * Synchronous LR (Left Right) clocks and Frame signals. - */ -#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */ -#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */ - -/* - * TDM - */ -#define SND_SOC_DAIFMT_TDM (1 << 6) - -/* - * DAI hardware signal inversions - */ -#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bclk + frm */ -#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */ -#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */ -#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */ - -/* - * DAI hardware clock masters - * This is wrt the codec, the inverse is true for the interface - * i.e. if the codec is clk and frm master then the interface is - * clk and frame slave. - */ -#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */ -#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */ -#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ -#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */ - -#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f -#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 -#define SND_SOC_DAIFMT_INV_MASK 0x0f00 -#define SND_SOC_DAIFMT_MASTER_MASK 0xf000 - - -/* - * Master Clock Directions - */ -#define SND_SOC_CLOCK_IN 0 -#define SND_SOC_CLOCK_OUT 1 - -/* - * AC97 codec ID's bitmask - */ -#define SND_SOC_DAI_AC97_ID0 (1 << 0) -#define SND_SOC_DAI_AC97_ID1 (1 << 1) -#define SND_SOC_DAI_AC97_ID2 (1 << 2) -#define SND_SOC_DAI_AC97_ID3 (1 << 3) - struct snd_soc_device; struct snd_soc_pcm_stream; struct snd_soc_ops; struct snd_soc_dai_mode; struct snd_soc_pcm_runtime; struct snd_soc_dai; +struct snd_soc_platform; struct snd_soc_codec; -struct snd_soc_machine_config; struct soc_enum; struct snd_soc_ac97_ops; -struct snd_soc_clock_info; typedef int (*hw_write_t)(void *,const char* ,int); typedef int (*hw_read_t)(void *,char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; +int snd_soc_register_platform(struct snd_soc_platform *platform); +void snd_soc_unregister_platform(struct snd_soc_platform *platform); +int snd_soc_register_codec(struct snd_soc_codec *codec); +void snd_soc_unregister_codec(struct snd_soc_codec *codec); + /* pcm <-> DAI connect */ void snd_soc_free_pcms(struct snd_soc_device *socdev); int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid); -int snd_soc_register_card(struct snd_soc_device *socdev); +int snd_soc_init_card(struct snd_soc_device *socdev); /* set runtime hw params */ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, @@ -263,27 +187,6 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); -/* Digital Audio Interface clocking API.*/ -int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, - unsigned int freq, int dir); - -int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, - int div_id, int div); - -int snd_soc_dai_set_pll(struct snd_soc_dai *dai, - int pll_id, unsigned int freq_in, unsigned int freq_out); - -/* Digital Audio interface formatting */ -int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); - -int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); - -int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); - -/* Digital Audio Interface mute */ -int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); - /* *Controls */ @@ -341,66 +244,14 @@ struct snd_soc_ops { int (*trigger)(struct snd_pcm_substream *, int); }; -/* ASoC DAI ops */ -struct snd_soc_dai_ops { - /* DAI clocking configuration */ - int (*set_sysclk)(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir); - int (*set_pll)(struct snd_soc_dai *dai, - int pll_id, unsigned int freq_in, unsigned int freq_out); - int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); - - /* DAI format configuration */ - int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); - int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); - int (*set_tristate)(struct snd_soc_dai *dai, int tristate); - - /* digital mute */ - int (*digital_mute)(struct snd_soc_dai *dai, int mute); -}; - -/* SoC DAI (Digital Audio Interface) */ -struct snd_soc_dai { - /* DAI description */ - char *name; - unsigned int id; - unsigned char type; - - /* DAI callbacks */ - int (*probe)(struct platform_device *pdev, - struct snd_soc_dai *dai); - void (*remove)(struct platform_device *pdev, - struct snd_soc_dai *dai); - int (*suspend)(struct platform_device *pdev, - struct snd_soc_dai *dai); - int (*resume)(struct platform_device *pdev, - struct snd_soc_dai *dai); - - /* ops */ - struct snd_soc_ops ops; - struct snd_soc_dai_ops dai_ops; - - /* DAI capabilities */ - struct snd_soc_pcm_stream capture; - struct snd_soc_pcm_stream playback; - - /* DAI runtime info */ - struct snd_pcm_runtime *runtime; - struct snd_soc_codec *codec; - unsigned int active; - unsigned char pop_wait:1; - void *dma_data; - - /* DAI private data */ - void *private_data; -}; - /* SoC Audio Codec */ struct snd_soc_codec { char *name; struct module *owner; struct mutex mutex; + struct device *dev; + + struct list_head list; /* callbacks */ int (*set_bias_level)(struct snd_soc_codec *, @@ -426,6 +277,7 @@ struct snd_soc_codec { short reg_cache_step; /* dapm */ + u32 pop_time; struct list_head dapm_widgets; struct list_head dapm_paths; enum snd_soc_bias_level bias_level; @@ -435,6 +287,11 @@ struct snd_soc_codec { /* codec DAI's */ struct snd_soc_dai *dai; unsigned int num_dai; + +#ifdef CONFIG_DEBUG_FS + struct dentry *debugfs_reg; + struct dentry *debugfs_pop_time; +#endif }; /* codec device */ @@ -448,13 +305,12 @@ struct snd_soc_codec_device { /* SoC platform interface */ struct snd_soc_platform { char *name; + struct list_head list; int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); - int (*suspend)(struct platform_device *pdev, - struct snd_soc_dai *dai); - int (*resume)(struct platform_device *pdev, - struct snd_soc_dai *dai); + int (*suspend)(struct snd_soc_dai *dai); + int (*resume)(struct snd_soc_dai *dai); /* pcm creation and destruction */ int (*pcm_new)(struct snd_card *, struct snd_soc_dai *, @@ -484,9 +340,14 @@ struct snd_soc_dai_link { struct snd_pcm *pcm; }; -/* SoC machine */ -struct snd_soc_machine { +/* SoC card */ +struct snd_soc_card { char *name; + struct device *dev; + + struct list_head list; + + int instantiated; int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); @@ -499,23 +360,26 @@ struct snd_soc_machine { int (*resume_post)(struct platform_device *pdev); /* callbacks */ - int (*set_bias_level)(struct snd_soc_machine *, + int (*set_bias_level)(struct snd_soc_card *, enum snd_soc_bias_level level); /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; + + struct snd_soc_device *socdev; + + struct snd_soc_platform *platform; + struct delayed_work delayed_work; + struct work_struct deferred_resume_work; }; /* SoC Device - the audio subsystem */ struct snd_soc_device { struct device *dev; - struct snd_soc_machine *machine; - struct snd_soc_platform *platform; + struct snd_soc_card *card; struct snd_soc_codec *codec; struct snd_soc_codec_device *codec_dev; - struct delayed_work delayed_work; - struct work_struct deferred_resume_work; void *codec_data; }; @@ -542,4 +406,6 @@ struct soc_enum { void *dapm; }; +#include <sound/soc-dai.h> + #endif diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h new file mode 100644 index 000000000000..475ef8bb7dcd --- /dev/null +++ b/include/sound/uda134x.h @@ -0,0 +1,26 @@ +/* + * uda134x.h -- UDA134x ALSA SoC Codec driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _UDA134X_H +#define _UDA134X_H + +#include <sound/l3.h> + +struct uda134x_platform_data { + struct l3_pins l3; + void (*power) (int); + int model; +#define UDA134X_UDA1340 1 +#define UDA134X_UDA1341 2 +#define UDA134X_UDA1344 3 +}; + +#endif /* _UDA134X_H */ diff --git a/include/sound/version.h b/include/sound/version.h index 4aafeda88634..2b48237e23bf 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.18rc3" +#define CONFIG_SND_VERSION "1.0.18a" #define CONFIG_SND_DATE "" diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c index 7fa37e15f196..a351dd0a09c7 100644 --- a/sound/ac97_bus.c +++ b/sound/ac97_bus.c @@ -15,6 +15,7 @@ #include <linux/init.h> #include <linux/device.h> #include <linux/string.h> +#include <sound/ac97_codec.h> /* * Let drivers decide whether they want to support given codec from their diff --git a/sound/aoa/codecs/Makefile b/sound/aoa/codecs/Makefile index 31cbe68fd42f..c3ee77fc4b2d 100644 --- a/sound/aoa/codecs/Makefile +++ b/sound/aoa/codecs/Makefile @@ -1,3 +1,7 @@ +snd-aoa-codec-onyx-objs := onyx.o +snd-aoa-codec-tas-objs := tas.o +snd-aoa-codec-toonie-objs := toonie.o + obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o obj-$(CONFIG_SND_AOA_TOONIE) += snd-aoa-codec-toonie.o diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.c b/sound/aoa/codecs/onyx.c index 6a3837d480e5..15500b9d2da0 100644 --- a/sound/aoa/codecs/snd-aoa-codec-onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -37,7 +37,7 @@ MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa"); -#include "snd-aoa-codec-onyx.h" +#include "onyx.h" #include "../aoa.h" #include "../soundbus/soundbus.h" @@ -292,7 +292,7 @@ static int onyx_snd_capture_source_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new capture_source_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* If we name this 'Input Source', it properly shows up in - * alsamixer as a selection, * but it's shown under the + * alsamixer as a selection, * but it's shown under the * 'Playback' category. * If I name it 'Capture Source', it shows up in strange * ways (two bools of which one can be selected at a @@ -477,7 +477,7 @@ static int onyx_spdif_mask_get(struct snd_kcontrol *kcontrol, ucontrol->value.iec958.status[3] = 0x3f; ucontrol->value.iec958.status[4] = 0x0f; - + return 0; } @@ -682,7 +682,7 @@ static int onyx_usable(struct codec_info_item *cii, onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v); spdif_enabled = !!(v & ONYX_SPDIF_ENABLE); onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v); - analog_enabled = + analog_enabled = (v & (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT)) != (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT); mutex_unlock(&onyx->mutex); @@ -882,7 +882,7 @@ static int onyx_init_codec(struct aoa_codec *codec) msleep(1); onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0); msleep(1); - + if (onyx_register_init(onyx)) { printk(KERN_ERR PFX "failed to initialise onyx registers\n"); return -ENODEV; @@ -1069,7 +1069,7 @@ static int onyx_i2c_attach(struct i2c_adapter *adapter) /* if that didn't work, try desperate mode for older * machines that have stuff missing from the device tree */ - + if (!of_device_is_compatible(busnode, "k2-i2c")) return -ENODEV; diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.h b/sound/aoa/codecs/onyx.h index ffd20254ff76..ffd20254ff76 100644 --- a/sound/aoa/codecs/snd-aoa-codec-onyx.h +++ b/sound/aoa/codecs/onyx.h diff --git a/sound/aoa/codecs/snd-aoa-codec-tas-basstreble.h b/sound/aoa/codecs/tas-basstreble.h index 69b61136fd54..69b61136fd54 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas-basstreble.h +++ b/sound/aoa/codecs/tas-basstreble.h diff --git a/sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h b/sound/aoa/codecs/tas-gain-table.h index 4cfa6757715e..4cfa6757715e 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h +++ b/sound/aoa/codecs/tas-gain-table.h diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c b/sound/aoa/codecs/tas.c index 6c515b2b8bbd..008e0f85097d 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas.c +++ b/sound/aoa/codecs/tas.c @@ -71,9 +71,9 @@ MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("tas codec driver for snd-aoa"); -#include "snd-aoa-codec-tas.h" -#include "snd-aoa-codec-tas-gain-table.h" -#include "snd-aoa-codec-tas-basstreble.h" +#include "tas.h" +#include "tas-gain-table.h" +#include "tas-basstreble.h" #include "../aoa.h" #include "../soundbus/soundbus.h" @@ -880,7 +880,7 @@ static void tas_exit_codec(struct aoa_codec *codec) return; tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas); } - + static struct i2c_driver tas_driver; diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.h b/sound/aoa/codecs/tas.h index ae177e3466e6..ae177e3466e6 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas.h +++ b/sound/aoa/codecs/tas.h diff --git a/sound/aoa/codecs/snd-aoa-codec-toonie.c b/sound/aoa/codecs/toonie.c index 3c7d1d8a9a6f..f13827e17562 100644 --- a/sound/aoa/codecs/snd-aoa-codec-toonie.c +++ b/sound/aoa/codecs/toonie.c @@ -131,7 +131,7 @@ static int __init toonie_init(void) toonie->codec.owner = THIS_MODULE; toonie->codec.init = toonie_init_codec; toonie->codec.exit = toonie_exit_codec; - + if (aoa_codec_register(&toonie->codec)) { kfree(toonie); return -EINVAL; diff --git a/sound/aoa/core/Makefile b/sound/aoa/core/Makefile index 62dc7287f663..a1596e88c718 100644 --- a/sound/aoa/core/Makefile +++ b/sound/aoa/core/Makefile @@ -1,5 +1,5 @@ obj-$(CONFIG_SND_AOA) += snd-aoa.o -snd-aoa-objs := snd-aoa-core.o \ - snd-aoa-alsa.o \ - snd-aoa-gpio-pmf.o \ - snd-aoa-gpio-feature.o +snd-aoa-objs := core.o \ + alsa.o \ + gpio-pmf.o \ + gpio-feature.o diff --git a/sound/aoa/core/snd-aoa-alsa.c b/sound/aoa/core/alsa.c index 17fe689ed287..617850463582 100644 --- a/sound/aoa/core/snd-aoa-alsa.c +++ b/sound/aoa/core/alsa.c @@ -6,7 +6,7 @@ * GPL v2, can be found in COPYING. */ #include <linux/module.h> -#include "snd-aoa-alsa.h" +#include "alsa.h" static int index = -1; module_param(index, int, 0444); @@ -64,7 +64,7 @@ int aoa_snd_device_new(snd_device_type_t type, { struct snd_card *card = aoa_get_card(); int err; - + if (!card) return -ENOMEM; err = snd_device_new(card, type, device_data, ops); diff --git a/sound/aoa/core/snd-aoa-alsa.h b/sound/aoa/core/alsa.h index 9669e4489cab..9669e4489cab 100644 --- a/sound/aoa/core/snd-aoa-alsa.h +++ b/sound/aoa/core/alsa.h diff --git a/sound/aoa/core/snd-aoa-core.c b/sound/aoa/core/core.c index 19fdae400687..10bec6c61382 100644 --- a/sound/aoa/core/snd-aoa-core.c +++ b/sound/aoa/core/core.c @@ -10,7 +10,7 @@ #include <linux/module.h> #include <linux/list.h> #include "../aoa.h" -#include "snd-aoa-alsa.h" +#include "alsa.h" MODULE_DESCRIPTION("Apple Onboard Audio Sound Driver"); MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>"); diff --git a/sound/aoa/core/snd-aoa-gpio-feature.c b/sound/aoa/core/gpio-feature.c index 805dcbff2257..c93ad5dec66b 100644 --- a/sound/aoa/core/snd-aoa-gpio-feature.c +++ b/sound/aoa/core/gpio-feature.c @@ -5,7 +5,7 @@ * * GPL v2, can be found in COPYING. * - * This file contains the GPIO control routines for + * This file contains the GPIO control routines for * direct (through feature calls) access to the GPIO * registers. */ diff --git a/sound/aoa/core/snd-aoa-gpio-pmf.c b/sound/aoa/core/gpio-pmf.c index 5ca2220eac7d..5ca2220eac7d 100644 --- a/sound/aoa/core/snd-aoa-gpio-pmf.c +++ b/sound/aoa/core/gpio-pmf.c diff --git a/sound/aoa/fabrics/Makefile b/sound/aoa/fabrics/Makefile index 55fc5e7e52cf..da37c10eca51 100644 --- a/sound/aoa/fabrics/Makefile +++ b/sound/aoa/fabrics/Makefile @@ -1 +1,3 @@ +snd-aoa-fabric-layout-objs += layout.o + obj-$(CONFIG_SND_AOA_FABRIC_LAYOUT) += snd-aoa-fabric-layout.o diff --git a/sound/aoa/fabrics/snd-aoa-fabric-layout.c b/sound/aoa/fabrics/layout.c index dea7abb082cd..ad60f5d10e82 100644 --- a/sound/aoa/fabrics/snd-aoa-fabric-layout.c +++ b/sound/aoa/fabrics/layout.c @@ -66,7 +66,7 @@ struct layout { unsigned int layout_id; struct codec_connect_info codecs[MAX_CODECS_PER_BUS]; int flags; - + /* if busname is not assigned, we use 'Master' below, * so that our layout table doesn't need to be filled * too much. diff --git a/sound/aoa/soundbus/i2sbus/Makefile b/sound/aoa/soundbus/i2sbus/Makefile index e57a5cf65655..1b949b2a4028 100644 --- a/sound/aoa/soundbus/i2sbus/Makefile +++ b/sound/aoa/soundbus/i2sbus/Makefile @@ -1,2 +1,2 @@ obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += snd-aoa-i2sbus.o -snd-aoa-i2sbus-objs := i2sbus-core.o i2sbus-pcm.o i2sbus-control.o +snd-aoa-i2sbus-objs := core.o pcm.o control.o diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-control.c b/sound/aoa/soundbus/i2sbus/control.c index 87beb4ad4d63..87beb4ad4d63 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-control.c +++ b/sound/aoa/soundbus/i2sbus/control.c diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/core.c index b4590df07466..be468edf3ecb 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -64,7 +64,7 @@ static void free_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev, struct dbdma_command_mem *r) { if (!r->space) return; - + dma_free_coherent(&macio_get_pci_dev(i2sdev->macio)->dev, r->size, r->space, r->bus_addr); } @@ -247,7 +247,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, * but request_resource doesn't know about parents and * contained resources... */ - dev->allocated_resource[i] = + dev->allocated_resource[i] = request_mem_region(dev->resources[i].start, dev->resources[i].end - dev->resources[i].start + 1, diff --git a/sound/aoa/soundbus/i2sbus/i2sbus.h b/sound/aoa/soundbus/i2sbus/i2sbus.h index ff29654782c9..befefd99e271 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus.h +++ b/sound/aoa/soundbus/i2sbus/i2sbus.h @@ -18,7 +18,7 @@ #include <asm/pmac_feature.h> #include <asm/dbdma.h> -#include "i2sbus-interface.h" +#include "interface.h" #include "../soundbus.h" struct i2sbus_control { diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-interface.h b/sound/aoa/soundbus/i2sbus/interface.h index c6b5f5452d20..c6b5f5452d20 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-interface.h +++ b/sound/aoa/soundbus/i2sbus/interface.h diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 59bacd365733..59bacd365733 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 66348c92f88d..7bbdda041a99 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -95,6 +95,26 @@ config SND_SEQUENCER_OSS this will be compiled as a module. The module will be called snd-seq-oss. +config SND_HRTIMER + tristate "HR-timer backend support" + depends on HIGH_RES_TIMERS + select SND_TIMER + help + Say Y here to enable HR-timer backend for ALSA timer. ALSA uses + the hrtimer as a precise timing source. The ALSA sequencer code + also can use this timing source. + + To compile this driver as a module, choose M here: the module + will be called snd-hrtimer. + +config SND_SEQ_HRTIMER_DEFAULT + bool "Use HR-timer as default sequencer timer" + depends on SND_HRTIMER && SND_SEQUENCER + default y + help + Say Y here to use the HR-timer backend as the default sequencer + timer. + config SND_RTCTIMER tristate "RTC Timer support" depends on RTC @@ -114,6 +134,7 @@ config SND_RTCTIMER config SND_SEQ_RTCTIMER_DEFAULT bool "Use RTC as default sequencer timer" depends on SND_RTCTIMER && SND_SEQUENCER + depends on !SND_SEQ_HRTIMER_DEFAULT default y help Say Y here to use the RTC timer as the default sequencer diff --git a/sound/core/Makefile b/sound/core/Makefile index d57125a5687d..4229052e7b91 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -17,12 +17,14 @@ snd-page-alloc-$(CONFIG_HAS_DMA) += sgbuf.o snd-rawmidi-objs := rawmidi.o snd-timer-objs := timer.o +snd-hrtimer-objs := hrtimer.o snd-rtctimer-objs := rtctimer.o snd-hwdep-objs := hwdep.o obj-$(CONFIG_SND) += snd.o obj-$(CONFIG_SND_HWDEP) += snd-hwdep.o obj-$(CONFIG_SND_TIMER) += snd-timer.o +obj-$(CONFIG_SND_HRTIMER) += snd-hrtimer.o obj-$(CONFIG_SND_RTCTIMER) += snd-rtctimer.o obj-$(CONFIG_SND_PCM) += snd-pcm.o snd-page-alloc.o obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o diff --git a/sound/core/device.c b/sound/core/device.c index c58d8227254c..a67dfac08c03 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -98,7 +98,7 @@ int snd_device_free(struct snd_card *card, void *device_data) kfree(dev); return 0; } - snd_printd("device free %p (from %p), not found\n", device_data, + snd_printd("device free %p (from %pF), not found\n", device_data, __builtin_return_address(0)); return -ENXIO; } @@ -135,7 +135,7 @@ int snd_device_disconnect(struct snd_card *card, void *device_data) } return 0; } - snd_printd("device disconnect %p (from %p), not found\n", device_data, + snd_printd("device disconnect %p (from %pF), not found\n", device_data, __builtin_return_address(0)); return -ENXIO; } diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c new file mode 100644 index 000000000000..c1d285921f80 --- /dev/null +++ b/sound/core/hrtimer.c @@ -0,0 +1,155 @@ +/* + * ALSA timer back-end using hrtimer + * Copyright (C) 2008 Takashi Iwai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/hrtimer.h> +#include <sound/core.h> +#include <sound/timer.h> + +MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>"); +MODULE_DESCRIPTION("ALSA hrtimer backend"); +MODULE_LICENSE("GPL"); + +MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_HRTIMER)); + +#define NANO_SEC 1000000000UL /* 10^9 in sec */ +static unsigned int resolution; + +struct snd_hrtimer { + struct snd_timer *timer; + struct hrtimer hrt; +}; + +static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) +{ + struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt); + struct snd_timer *t = stime->timer; + hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); + snd_timer_interrupt(stime->timer, t->sticks); + return HRTIMER_RESTART; +} + +static int snd_hrtimer_open(struct snd_timer *t) +{ + struct snd_hrtimer *stime; + + stime = kmalloc(sizeof(*stime), GFP_KERNEL); + if (!stime) + return -ENOMEM; + hrtimer_init(&stime->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + stime->timer = t; + stime->hrt.cb_mode = HRTIMER_CB_IRQSAFE_UNLOCKED; + stime->hrt.function = snd_hrtimer_callback; + t->private_data = stime; + return 0; +} + +static int snd_hrtimer_close(struct snd_timer *t) +{ + struct snd_hrtimer *stime = t->private_data; + + if (stime) { + hrtimer_cancel(&stime->hrt); + kfree(stime); + t->private_data = NULL; + } + return 0; +} + +static int snd_hrtimer_start(struct snd_timer *t) +{ + struct snd_hrtimer *stime = t->private_data; + + hrtimer_start(&stime->hrt, ns_to_ktime(t->sticks * resolution), + HRTIMER_MODE_REL); + return 0; +} + +static int snd_hrtimer_stop(struct snd_timer *t) +{ + struct snd_hrtimer *stime = t->private_data; + + hrtimer_cancel(&stime->hrt); + return 0; +} + +static struct snd_timer_hardware hrtimer_hw = { + .flags = SNDRV_TIMER_HW_AUTO, + .open = snd_hrtimer_open, + .close = snd_hrtimer_close, + .start = snd_hrtimer_start, + .stop = snd_hrtimer_stop, +}; + +/* + * entry functions + */ + +static struct snd_timer *mytimer; + +static int __init snd_hrtimer_init(void) +{ + struct snd_timer *timer; + struct timespec tp; + int err; + + hrtimer_get_res(CLOCK_MONOTONIC, &tp); + if (tp.tv_sec > 0 || !tp.tv_nsec) { + snd_printk(KERN_ERR + "snd-hrtimer: Invalid resolution %u.%09u", + (unsigned)tp.tv_sec, (unsigned)tp.tv_nsec); + return -EINVAL; + } + resolution = tp.tv_nsec; + + /* Create a new timer and set up the fields */ + err = snd_timer_global_new("hrtimer", SNDRV_TIMER_GLOBAL_HRTIMER, + &timer); + if (err < 0) + return err; + + timer->module = THIS_MODULE; + strcpy(timer->name, "HR timer"); + timer->hw = hrtimer_hw; + timer->hw.resolution = resolution; + timer->hw.ticks = NANO_SEC / resolution; + + err = snd_timer_global_register(timer); + if (err < 0) { + snd_timer_global_free(timer); + return err; + } + mytimer = timer; /* remember this */ + + return 0; +} + +static void __exit snd_hrtimer_exit(void) +{ + if (mytimer) { + snd_timer_global_free(mytimer); + mytimer = NULL; + } +} + +module_init(snd_hrtimer_init); +module_exit(snd_hrtimer_exit); diff --git a/sound/core/info.c b/sound/core/info.c index 527b207462b0..70fa87189f36 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -653,6 +653,23 @@ int snd_info_card_register(struct snd_card *card) } /* + * called on card->id change + */ +void snd_info_card_id_change(struct snd_card *card) +{ + mutex_lock(&info_mutex); + if (card->proc_root_link) { + snd_remove_proc_entry(snd_proc_root, card->proc_root_link); + card->proc_root_link = NULL; + } + if (strcmp(card->id, card->proc_root->name)) + card->proc_root_link = proc_symlink(card->id, + snd_proc_root, + card->proc_root->name); + mutex_unlock(&info_mutex); +} + +/* * de-register the card proc file * called from init.c */ diff --git a/sound/core/init.c b/sound/core/init.c index b47ff8b44be8..0d5520c415d3 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -533,6 +533,65 @@ static void choose_default_id(struct snd_card *card) } } +#ifndef CONFIG_SYSFS_DEPRECATED +static ssize_t +card_id_show_attr(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_card *card = dev_get_drvdata(dev); + return snprintf(buf, PAGE_SIZE, "%s\n", card ? card->id : "(null)"); +} + +static ssize_t +card_id_store_attr(struct device *dev, struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_card *card = dev_get_drvdata(dev); + char buf1[sizeof(card->id)]; + size_t copy = count > sizeof(card->id) - 1 ? + sizeof(card->id) - 1 : count; + size_t idx; + int c; + + for (idx = 0; idx < copy; idx++) { + c = buf[idx]; + if (!isalnum(c) && c != '_' && c != '-') + return -EINVAL; + } + memcpy(buf1, buf, copy); + buf1[copy] = '\0'; + mutex_lock(&snd_card_mutex); + if (!snd_info_check_reserved_words(buf1)) { + __exist: + mutex_unlock(&snd_card_mutex); + return -EEXIST; + } + for (idx = 0; idx < snd_ecards_limit; idx++) { + if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1)) + goto __exist; + } + strcpy(card->id, buf1); + snd_info_card_id_change(card); + mutex_unlock(&snd_card_mutex); + + return count; +} + +static struct device_attribute card_id_attrs = + __ATTR(id, S_IRUGO | S_IWUSR, card_id_show_attr, card_id_store_attr); + +static ssize_t +card_number_show_attr(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_card *card = dev_get_drvdata(dev); + return snprintf(buf, PAGE_SIZE, "%i\n", card ? card->number : -1); +} + +static struct device_attribute card_number_attrs = + __ATTR(number, S_IRUGO, card_number_show_attr, NULL); +#endif /* CONFIG_SYSFS_DEPRECATED */ + /** * snd_card_register - register the soundcard * @card: soundcard structure @@ -553,7 +612,7 @@ int snd_card_register(struct snd_card *card) #ifndef CONFIG_SYSFS_DEPRECATED if (!card->card_dev) { card->card_dev = device_create(sound_class, card->dev, - MKDEV(0, 0), NULL, + MKDEV(0, 0), card, "card%i", card->number); if (IS_ERR(card->card_dev)) card->card_dev = NULL; @@ -576,6 +635,16 @@ int snd_card_register(struct snd_card *card) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_REGISTER); #endif +#ifndef CONFIG_SYSFS_DEPRECATED + if (card->card_dev) { + err = device_create_file(card->card_dev, &card_id_attrs); + if (err < 0) + return err; + err = device_create_file(card->card_dev, &card_number_attrs); + if (err < 0) + return err; + } +#endif return 0; } diff --git a/sound/core/jack.c b/sound/core/jack.c index bd2d9e6b55e9..dd4a12dc09aa 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -34,6 +34,7 @@ static int snd_jack_dev_free(struct snd_device *device) else input_free_device(jack->input_dev); + kfree(jack->id); kfree(jack); return 0; @@ -87,7 +88,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (jack == NULL) return -ENOMEM; - jack->id = id; + jack->id = kstrdup(id, GFP_KERNEL); jack->input_dev = input_allocate_device(); if (jack->input_dev == NULL) { @@ -102,9 +103,15 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (type & SND_JACK_HEADPHONE) input_set_capability(jack->input_dev, EV_SW, SW_HEADPHONE_INSERT); + if (type & SND_JACK_LINEOUT) + input_set_capability(jack->input_dev, EV_SW, + SW_LINEOUT_INSERT); if (type & SND_JACK_MICROPHONE) input_set_capability(jack->input_dev, EV_SW, SW_MICROPHONE_INSERT); + if (type & SND_JACK_MECHANICAL) + input_set_capability(jack->input_dev, EV_SW, + SW_JACK_PHYSICAL_INSERT); err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -153,9 +160,15 @@ void snd_jack_report(struct snd_jack *jack, int status) if (jack->type & SND_JACK_HEADPHONE) input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, status & SND_JACK_HEADPHONE); + if (jack->type & SND_JACK_LINEOUT) + input_report_switch(jack->input_dev, SW_LINEOUT_INSERT, + status & SND_JACK_LINEOUT); if (jack->type & SND_JACK_MICROPHONE) input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, status & SND_JACK_MICROPHONE); + if (jack->type & SND_JACK_MECHANICAL) + input_report_switch(jack->input_dev, SW_JACK_PHYSICAL_INSERT, + status & SND_JACK_MECHANICAL); input_sync(jack->input_dev); } diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 39672f68ce5d..002777ba336a 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -151,7 +151,7 @@ static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *subs if (!substream->opened) return; if (up) { - tasklet_hi_schedule(&substream->runtime->tasklet); + tasklet_schedule(&substream->runtime->tasklet); } else { tasklet_kill(&substream->runtime->tasklet); substream->ops->trigger(substream, 0); @@ -908,7 +908,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, } if (result > 0) { if (runtime->event) - tasklet_hi_schedule(&runtime->tasklet); + tasklet_schedule(&runtime->tasklet); else if (snd_rawmidi_ready(substream)) wake_up(&runtime->sleep); } diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c index 51e64e30dd3b..0851cd13e303 100644 --- a/sound/core/rtctimer.c +++ b/sound/core/rtctimer.c @@ -118,7 +118,7 @@ static void rtctimer_tasklet(unsigned long data) */ static void rtctimer_interrupt(void *private_data) { - tasklet_hi_schedule(private_data); + tasklet_schedule(private_data); } diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index ee0f8405ab35..bf09a5ad1865 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -43,7 +43,9 @@ int seq_default_timer_class = SNDRV_TIMER_CLASS_GLOBAL; int seq_default_timer_sclass = SNDRV_TIMER_SCLASS_NONE; int seq_default_timer_card = -1; int seq_default_timer_device = -#ifdef CONFIG_SND_SEQ_RTCTIMER_DEFAULT +#ifdef CONFIG_SND_SEQ_HRTIMER_DEFAULT + SNDRV_TIMER_GLOBAL_HRTIMER +#elif defined(CONFIG_SND_SEQ_RTCTIMER_DEFAULT) SNDRV_TIMER_GLOBAL_RTC #else SNDRV_TIMER_GLOBAL_SYSTEM diff --git a/sound/core/timer.c b/sound/core/timer.c index c584408c9f17..796532081e81 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -743,7 +743,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) spin_unlock_irqrestore(&timer->lock, flags); if (use_tasklet) - tasklet_hi_schedule(&timer->task_queue); + tasklet_schedule(&timer->task_queue); } /* diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 255fd18b9aec..0bcf14640fde 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -163,7 +163,7 @@ config SND_ML403_AC97CR config SND_AC97_POWER_SAVE bool "AC97 Power-Saving Mode" - depends on SND_AC97_CODEC && EXPERIMENTAL + depends on SND_AC97_CODEC default n help Say Y here to enable the aggressive power-saving support of diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 1899cf0685bc..2a02f704f366 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -96,7 +96,7 @@ static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev) return -EINVAL; hrtimer_init(&pcsp_chip.timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); - pcsp_chip.timer.cb_mode = HRTIMER_CB_SOFTIRQ; + pcsp_chip.timer.cb_mode = HRTIMER_CB_IRQSAFE_UNLOCKED; pcsp_chip.timer.function = pcsp_do_timer; card = snd_card_new(index, id, THIS_MODULE, 0); @@ -188,10 +188,8 @@ static int __devexit pcsp_remove(struct platform_device *dev) static void pcsp_stop_beep(struct snd_pcsp *chip) { - spin_lock_irq(&chip->substream_lock); - if (!chip->playback_substream) - pcspkr_stop_sound(); - spin_unlock_irq(&chip->substream_lock); + pcsp_sync_stop(chip); + pcspkr_stop_sound(); } #ifdef CONFIG_PM diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h index 1d661f795e8c..cdef2664218f 100644 --- a/sound/drivers/pcsp/pcsp.h +++ b/sound/drivers/pcsp/pcsp.h @@ -62,6 +62,8 @@ struct snd_pcsp { unsigned short port, irq, dma; spinlock_t substream_lock; struct snd_pcm_substream *playback_substream; + unsigned int fmt_size; + unsigned int is_signed; size_t playback_ptr; size_t period_ptr; atomic_t timer_active; @@ -77,6 +79,7 @@ struct snd_pcsp { extern struct snd_pcsp pcsp_chip; extern enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle); +extern void pcsp_sync_stop(struct snd_pcsp *chip); extern int snd_pcsp_new_pcm(struct snd_pcsp *chip); extern int snd_pcsp_new_mixer(struct snd_pcsp *chip); diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index 1f42e4063118..84cc2658c05b 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/interrupt.h> #include <sound/pcm.h> #include <asm/io.h> #include "pcsp.h" @@ -19,61 +20,57 @@ MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround " #define DMIX_WANTS_S16 1 -enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) +/* + * Call snd_pcm_period_elapsed in a tasklet + * This avoids spinlock messes and long-running irq contexts + */ +static void pcsp_call_pcm_elapsed(unsigned long priv) +{ + if (atomic_read(&pcsp_chip.timer_active)) { + struct snd_pcm_substream *substream; + substream = pcsp_chip.playback_substream; + if (substream) + snd_pcm_period_elapsed(substream); + } +} + +static DECLARE_TASKLET(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed, 0); + +/* write the port and returns the next expire time in ns; + * called at the trigger-start and in hrtimer callback + */ +static unsigned long pcsp_timer_update(struct hrtimer *handle) { unsigned char timer_cnt, val; - int fmt_size, periods_elapsed; u64 ns; - size_t period_bytes, buffer_bytes; struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); + unsigned long flags; if (chip->thalf) { outb(chip->val61, 0x61); chip->thalf = 0; if (!atomic_read(&chip->timer_active)) - return HRTIMER_NORESTART; - hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer), - ktime_set(0, chip->ns_rem)); - return HRTIMER_RESTART; + return 0; + return chip->ns_rem; } - spin_lock_irq(&chip->substream_lock); - /* Takashi Iwai says regarding this extra lock: - - If the irq handler handles some data on the DMA buffer, it should - do snd_pcm_stream_lock(). - That protects basically against all races among PCM callbacks, yes. - However, there are two remaining issues: - 1. The substream pointer you try to lock isn't protected _before_ - this lock yet. - 2. snd_pcm_period_elapsed() itself acquires the lock. - The requirement of another lock is because of 1. When you get - chip->playback_substream, it's not protected. - Keeping this lock while snd_pcm_period_elapsed() assures the substream - is still protected (at least, not released). And the other status is - handled properly inside snd_pcm_stream_lock() in - snd_pcm_period_elapsed(). - - */ - if (!chip->playback_substream) - goto exit_nr_unlock1; - substream = chip->playback_substream; - snd_pcm_stream_lock(substream); if (!atomic_read(&chip->timer_active)) - goto exit_nr_unlock2; + return 0; + substream = chip->playback_substream; + if (!substream) + return 0; runtime = substream->runtime; - fmt_size = snd_pcm_format_physical_width(runtime->format) >> 3; /* assume it is mono! */ - val = runtime->dma_area[chip->playback_ptr + fmt_size - 1]; - if (snd_pcm_format_signed(runtime->format)) + val = runtime->dma_area[chip->playback_ptr + chip->fmt_size - 1]; + if (chip->is_signed) val ^= 0x80; timer_cnt = val * CUR_DIV() / 256; if (timer_cnt && chip->enable) { - spin_lock(&i8253_lock); + spin_lock_irqsave(&i8253_lock, flags); if (!nforce_wa) { outb_p(chip->val61, 0x61); outb_p(timer_cnt, 0x42); @@ -82,12 +79,39 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) outb(chip->val61 ^ 2, 0x61); chip->thalf = 1; } - spin_unlock(&i8253_lock); + spin_unlock_irqrestore(&i8253_lock, flags); } + chip->ns_rem = PCSP_PERIOD_NS(); + ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem); + chip->ns_rem -= ns; + return ns; +} + +enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) +{ + struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); + struct snd_pcm_substream *substream; + int periods_elapsed, pointer_update; + size_t period_bytes, buffer_bytes; + unsigned long ns; + unsigned long flags; + + pointer_update = !chip->thalf; + ns = pcsp_timer_update(handle); + if (!ns) + return HRTIMER_NORESTART; + + /* update the playback position */ + substream = chip->playback_substream; + if (!substream) + return HRTIMER_NORESTART; + period_bytes = snd_pcm_lib_period_bytes(substream); buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - chip->playback_ptr += PCSP_INDEX_INC() * fmt_size; + + spin_lock_irqsave(&chip->substream_lock, flags); + chip->playback_ptr += PCSP_INDEX_INC() * chip->fmt_size; periods_elapsed = chip->playback_ptr - chip->period_ptr; if (periods_elapsed < 0) { #if PCSP_DEBUG @@ -102,41 +126,30 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) * or ALSA will BUG on us. */ chip->playback_ptr %= buffer_bytes; - snd_pcm_stream_unlock(substream); - if (periods_elapsed) { - snd_pcm_period_elapsed(substream); chip->period_ptr += periods_elapsed * period_bytes; chip->period_ptr %= buffer_bytes; } + spin_unlock_irqrestore(&chip->substream_lock, flags); - spin_unlock_irq(&chip->substream_lock); + if (periods_elapsed) + tasklet_schedule(&pcsp_pcm_tasklet); - if (!atomic_read(&chip->timer_active)) - return HRTIMER_NORESTART; + hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns)); - chip->ns_rem = PCSP_PERIOD_NS(); - ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem); - chip->ns_rem -= ns; - hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer), - ktime_set(0, ns)); return HRTIMER_RESTART; - -exit_nr_unlock2: - snd_pcm_stream_unlock(substream); -exit_nr_unlock1: - spin_unlock_irq(&chip->substream_lock); - return HRTIMER_NORESTART; } -static void pcsp_start_playing(struct snd_pcsp *chip) +static int pcsp_start_playing(struct snd_pcsp *chip) { + unsigned long ns; + #if PCSP_DEBUG printk(KERN_INFO "PCSP: start_playing called\n"); #endif if (atomic_read(&chip->timer_active)) { printk(KERN_ERR "PCSP: Timer already active\n"); - return; + return -EIO; } spin_lock(&i8253_lock); @@ -146,7 +159,12 @@ static void pcsp_start_playing(struct snd_pcsp *chip) atomic_set(&chip->timer_active, 1); chip->thalf = 0; - hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL); + ns = pcsp_timer_update(&pcsp_chip.timer); + if (!ns) + return -EIO; + + hrtimer_start(&pcsp_chip.timer, ktime_set(0, ns), HRTIMER_MODE_REL); + return 0; } static void pcsp_stop_playing(struct snd_pcsp *chip) @@ -165,26 +183,35 @@ static void pcsp_stop_playing(struct snd_pcsp *chip) spin_unlock(&i8253_lock); } +/* + * Force to stop and sync the stream + */ +void pcsp_sync_stop(struct snd_pcsp *chip) +{ + local_irq_disable(); + pcsp_stop_playing(chip); + local_irq_enable(); + hrtimer_cancel(&chip->timer); + tasklet_kill(&pcsp_pcm_tasklet); +} + static int snd_pcsp_playback_close(struct snd_pcm_substream *substream) { struct snd_pcsp *chip = snd_pcm_substream_chip(substream); #if PCSP_DEBUG printk(KERN_INFO "PCSP: close called\n"); #endif - if (atomic_read(&chip->timer_active)) { - printk(KERN_ERR "PCSP: timer still active\n"); - pcsp_stop_playing(chip); - } - spin_lock_irq(&chip->substream_lock); + pcsp_sync_stop(chip); chip->playback_substream = NULL; - spin_unlock_irq(&chip->substream_lock); return 0; } static int snd_pcsp_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { + struct snd_pcsp *chip = snd_pcm_substream_chip(substream); int err; + pcsp_sync_stop(chip); err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) @@ -194,9 +221,11 @@ static int snd_pcsp_playback_hw_params(struct snd_pcm_substream *substream, static int snd_pcsp_playback_hw_free(struct snd_pcm_substream *substream) { + struct snd_pcsp *chip = snd_pcm_substream_chip(substream); #if PCSP_DEBUG printk(KERN_INFO "PCSP: hw_free called\n"); #endif + pcsp_sync_stop(chip); return snd_pcm_lib_free_pages(substream); } @@ -212,8 +241,12 @@ static int snd_pcsp_playback_prepare(struct snd_pcm_substream *substream) snd_pcm_lib_period_bytes(substream), substream->runtime->periods); #endif + pcsp_sync_stop(chip); chip->playback_ptr = 0; chip->period_ptr = 0; + chip->fmt_size = + snd_pcm_format_physical_width(substream->runtime->format) >> 3; + chip->is_signed = snd_pcm_format_signed(substream->runtime->format); return 0; } @@ -226,8 +259,7 @@ static int snd_pcsp_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: - pcsp_start_playing(chip); - break; + return pcsp_start_playing(chip); case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: pcsp_stop_playing(chip); @@ -242,7 +274,11 @@ static snd_pcm_uframes_t snd_pcsp_playback_pointer(struct snd_pcm_substream *substream) { struct snd_pcsp *chip = snd_pcm_substream_chip(substream); - return bytes_to_frames(substream->runtime, chip->playback_ptr); + unsigned int pos; + spin_lock(&chip->substream_lock); + pos = chip->playback_ptr; + spin_unlock(&chip->substream_lock); + return bytes_to_frames(substream->runtime, pos); } static struct snd_pcm_hardware snd_pcsp_playback = { @@ -279,9 +315,7 @@ static int snd_pcsp_playback_open(struct snd_pcm_substream *substream) return -EBUSY; } runtime->hw = snd_pcsp_playback; - spin_lock_irq(&chip->substream_lock); chip->playback_substream = substream; - spin_unlock_irq(&chip->substream_lock); return 0; } diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index 473b07f6ae85..14e3354be43a 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -548,7 +548,7 @@ irqreturn_t snd_vx_irq_handler(int irq, void *dev) (chip->chip_status & VX_STAT_IS_STALE)) return IRQ_NONE; if (! vx_test_and_ack(chip)) - tasklet_hi_schedule(&chip->tq); + tasklet_schedule(&chip->tq); return IRQ_HANDLED; } diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 27de574c08f7..6644d0034fba 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -823,7 +823,7 @@ static int vx_pcm_trigger(struct snd_pcm_substream *subs, int cmd) * we trigger the pipe using tasklet, so that the interrupts are * issued surely after the trigger is completed. */ - tasklet_hi_schedule(&pipe->start_tq); + tasklet_schedule(&pipe->start_tq); chip->pcm_running++; pipe->running = 1; break; diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 667eccc676a4..ea06877be4b1 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -140,8 +140,10 @@ static int __devinit snd_sb8_probe(struct device *pdev, unsigned int dev) break; } } - if (i >= ARRAY_SIZE(possible_ports)) + if (i >= ARRAY_SIZE(possible_ports)) { + err = -EINVAL; goto _err; + } } acard->chip = chip; diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7003711f4fcc..6e3a1848447c 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -208,7 +208,8 @@ config SND_OXYGEN * AuzenTech X-Meridian * Bgears b-Enspirer * Club3D Theatron DTS - * HT-Omega Claro + * HT-Omega Claro (plus) + * HT-Omega Claro halo (XT) * Razer Barracuda AC-1 * Sondigo Inferno @@ -497,129 +498,7 @@ config SND_FM801_TEA575X depends on SND_FM801_TEA575X_BOOL default SND_FM801 -config SND_HDA_INTEL - tristate "Intel HD Audio" - select SND_PCM - select SND_VMASTER - help - Say Y here to include support for Intel "High Definition - Audio" (Azalia) motherboard devices. - - To compile this driver as a module, choose M here: the module - will be called snd-hda-intel. - -config SND_HDA_HWDEP - bool "Build hwdep interface for HD-audio driver" - depends on SND_HDA_INTEL - select SND_HWDEP - help - Say Y here to build a hwdep interface for HD-audio driver. - This interface can be used for out-of-band communication - with codecs for debugging purposes. - -config SND_HDA_INPUT_BEEP - bool "Support digital beep via input layer" - depends on SND_HDA_INTEL - depends on INPUT=y || INPUT=SND_HDA_INTEL - help - Say Y here to build a digital beep interface for HD-audio - driver. This interface is used to generate digital beeps. - -config SND_HDA_CODEC_REALTEK - bool "Build Realtek HD-audio codec support" - depends on SND_HDA_INTEL - default y - help - Say Y here to include Realtek HD-audio codec support in - snd-hda-intel driver, such as ALC880. - -config SND_HDA_CODEC_ANALOG - bool "Build Analog Device HD-audio codec support" - depends on SND_HDA_INTEL - default y - help - Say Y here to include Analog Device HD-audio codec support in - snd-hda-intel driver, such as AD1986A. - -config SND_HDA_CODEC_SIGMATEL - bool "Build IDT/Sigmatel HD-audio codec support" - depends on SND_HDA_INTEL - default y - help - Say Y here to include IDT (Sigmatel) HD-audio codec support in - snd-hda-intel driver, such as STAC9200. - -config SND_HDA_CODEC_VIA - bool "Build VIA HD-audio codec support" - depends on SND_HDA_INTEL - default y - help - Say Y here to include VIA HD-audio codec support in - snd-hda-intel driver, such as VT1708. - -config SND_HDA_CODEC_ATIHDMI - bool "Build ATI HDMI HD-audio codec support" - depends on SND_HDA_INTEL - default y - help - Say Y here to include ATI HDMI HD-audio codec support in - snd-hda-intel driver, such as ATI RS600 HDMI. - -config SND_HDA_CODEC_NVHDMI - bool "Build NVIDIA HDMI HD-audio codec support" - depends on SND_HDA_INTEL - default y - help - Say Y here to include NVIDIA HDMI HD-audio codec support in - snd-hda-intel driver, such as NVIDIA MCP78 HDMI. - -config SND_HDA_CODEC_CONEXANT - bool "Build Conexant HD-audio codec support" - depends on SND_HDA_INTEL - default y - help - Say Y here to include Conexant HD-audio codec support in - snd-hda-intel driver, such as CX20549. - -config SND_HDA_CODEC_CMEDIA - bool "Build C-Media HD-audio codec support" - depends on SND_HDA_INTEL - default y - help - Say Y here to include C-Media HD-audio codec support in - snd-hda-intel driver, such as CMI9880. - -config SND_HDA_CODEC_SI3054 - bool "Build Silicon Labs 3054 HD-modem codec support" - depends on SND_HDA_INTEL - default y - help - Say Y here to include Silicon Labs 3054 HD-modem codec - (and compatibles) support in snd-hda-intel driver. - -config SND_HDA_GENERIC - bool "Enable generic HD-audio codec parser" - depends on SND_HDA_INTEL - default y - help - Say Y here to enable the generic HD-audio codec parser - in snd-hda-intel driver. - -config SND_HDA_POWER_SAVE - bool "Aggressive power-saving on HD-audio" - depends on SND_HDA_INTEL && EXPERIMENTAL - help - Say Y here to enable more aggressive power-saving mode on - HD-audio driver. The power-saving timeout can be configured - via power_save option or over sysfs on-the-fly. - -config SND_HDA_POWER_SAVE_DEFAULT - int "Default time-out for HD-audio power-save mode" - depends on SND_HDA_POWER_SAVE - default 0 - help - The default time-out value in seconds for HD-audio automatic - power-save mode. 0 means to disable the power-save mode. +source "sound/pci/hda/Kconfig" config SND_HDSP tristate "RME Hammerfall DSP Audio" diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index bd510eceff1f..e2b843b4f9d0 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -175,7 +175,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x574d4C04, 0xffffffff, "WM9704M,WM9704Q", patch_wolfson04, NULL}, { 0x574d4C05, 0xffffffff, "WM9705,WM9710", patch_wolfson05, NULL}, { 0x574d4C09, 0xffffffff, "WM9709", NULL, NULL}, -{ 0x574d4C12, 0xffffffff, "WM9711,WM9712", patch_wolfson11, NULL}, +{ 0x574d4C12, 0xffffffff, "WM9711,WM9712,WM9715", patch_wolfson11, NULL}, { 0x574d4c13, 0xffffffff, "WM9713,WM9714", patch_wolfson13, NULL, AC97_DEFAULT_POWER_OFF}, { 0x594d4800, 0xffffffff, "YMF743", patch_yamaha_ymf743, NULL }, { 0x594d4802, 0xffffffff, "YMF752", NULL, NULL }, diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 6e831aff1bd0..81bc93e5f1e3 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -2054,8 +2054,9 @@ static const struct snd_kcontrol_new snd_ac97_ad1888_controls[] = { .get = snd_ac97_ad1888_lohpsel_get, .put = snd_ac97_ad1888_lohpsel_put }, - AC97_SINGLE("V_REFOUT Enable", AC97_AD_MISC, 2, 1, 1), - AC97_SINGLE("High Pass Filter Enable", AC97_AD_TEST2, 12, 1, 1), + AC97_SINGLE("V_REFOUT Enable", AC97_AD_MISC, AC97_AD_VREFD_SHIFT, 1, 1), + AC97_SINGLE("High Pass Filter Enable", AC97_AD_TEST2, + AC97_AD_HPFD_SHIFT, 1, 1), AC97_SINGLE("Spread Front to Surround and Center/LFE", AC97_AD_MISC, 7, 1, 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -2832,6 +2833,8 @@ static int patch_alc655(struct snd_ac97 * ac97) val &= ~(1 << 1); /* Pin 47 is EAPD (for internal speaker) */ else val |= (1 << 1); /* Pin 47 is spdif input pin */ + /* this seems missing on some hardwares */ + ac97->ext_id |= AC97_EI_SPDIF; } val &= ~(1 << 12); /* vref enable */ snd_ac97_write_cache(ac97, 0x7a, val); diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index 74175fc80c7f..14b8d9a91aae 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -664,10 +664,14 @@ struct snd_ca0106_pcm { struct snd_ca0106_details { u32 serial; char * name; - int ac97; - int gpio_type; - int i2c_adc; - int spi_dac; + int ac97; /* ac97 = 0 -> Select MIC, Line in, TAD in, AUX in. + ac97 = 1 -> Default to AC97 in. */ + int gpio_type; /* gpio_type = 1 -> shared mic-in/line-in + gpio_type = 2 -> shared side-out/line-in. */ + int i2c_adc; /* with i2c_adc=1, the driver adds some capture volume + controls, phone, mic, line-in and aux. */ + int spi_dac; /* spi_dac=1 adds the mute switch for each analog + output, front, rear, etc. */ }; // definition of the chip-specific record @@ -686,11 +690,12 @@ struct snd_ca0106 { spinlock_t emu_lock; struct snd_ac97 *ac97; - struct snd_pcm *pcm; + struct snd_pcm *pcm[4]; struct snd_ca0106_channel playback_channels[4]; struct snd_ca0106_channel capture_channels[4]; - u32 spdif_bits[4]; /* s/pdif out setup */ + u32 spdif_bits[4]; /* s/pdif out default setup */ + u32 spdif_str_bits[4]; /* s/pdif out per-stream setup */ int spdif_enable; int capture_source; int i2c_capture_source; @@ -703,6 +708,11 @@ struct snd_ca0106 { struct snd_ca_midi midi2; u16 spi_dac_reg[16]; + +#ifdef CONFIG_PM +#define NUM_SAVED_VOLUMES 9 + unsigned int saved_vol[NUM_SAVED_VOLUMES]; +#endif }; int snd_ca0106_mixer(struct snd_ca0106 *emu); @@ -721,3 +731,11 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value); int snd_ca0106_spi_write(struct snd_ca0106 * emu, unsigned int data); + +#ifdef CONFIG_PM +void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip); +void snd_ca0106_mixer_resume(struct snd_ca0106 *chip); +#else +#define snd_ca0106_mixer_suspend(chip) do { } while (0) +#define snd_ca0106_mixer_resume(chip) do { } while (0) +#endif diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 88fbf285d2b7..0e62205d4081 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -254,7 +254,7 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "MSI K8N Diamond MB", .gpio_type = 2, .i2c_adc = 1, - .spi_dac = 2 } , + .spi_dac = 1 } , /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". @@ -305,9 +305,15 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = { SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, +#if 0 /* FIXME: looks like 44.1kHz capture causes noisy output on 48kHz */ .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), .rate_min = 44100, +#else + .rates = (SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), + .rate_min = 48000, +#endif /* FIXME */ .rate_max = 192000, .channels_min = 2, .channels_max = 2, @@ -479,6 +485,15 @@ static const int spi_dacd_bit[] = { [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_BIT, }; +static void restore_spdif_bits(struct snd_ca0106 *chip, int idx) +{ + if (chip->spdif_str_bits[idx] != chip->spdif_bits[idx]) { + chip->spdif_str_bits[idx] = chip->spdif_bits[idx]; + snd_ca0106_ptr_write(chip, SPCS0 + idx, 0, + chip->spdif_str_bits[idx]); + } +} + /* open_playback callback */ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substream, int channel_id) @@ -524,6 +539,9 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr if (err < 0) return err; } + + restore_spdif_bits(chip, channel_id); + return 0; } @@ -535,6 +553,8 @@ static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream) struct snd_ca0106_pcm *epcm = runtime->private_data; chip->playback_channels[epcm->channel_id].use = 0; + restore_spdif_bits(chip, epcm->channel_id); + if (chip->details->spi_dac && epcm->channel_id != PCM_FRONT_CHANNEL) { const int reg = spi_dacd_reg[epcm->channel_id]; @@ -847,15 +867,18 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream, struct snd_pcm_substream *s; u32 basic = 0; u32 extended = 0; - int running=0; + u32 bits; + int running = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: - running=1; + case SNDRV_PCM_TRIGGER_RESUME: + running = 1; break; case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: default: - running=0; + running = 0; break; } snd_pcm_group_for_each_entry(s, substream) { @@ -865,22 +888,32 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream, runtime = s->runtime; epcm = runtime->private_data; channel = epcm->channel_id; - //snd_printk("channel=%d\n",channel); + /* snd_printk("channel=%d\n",channel); */ epcm->running = running; - basic |= (0x1<<channel); - extended |= (0x10<<channel); + basic |= (0x1 << channel); + extended |= (0x10 << channel); snd_pcm_trigger_done(s, substream); } - //snd_printk("basic=0x%x, extended=0x%x\n",basic, extended); + /* snd_printk("basic=0x%x, extended=0x%x\n",basic, extended); */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: - snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) | (extended)); - snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0)|(basic)); + case SNDRV_PCM_TRIGGER_RESUME: + bits = snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0); + bits |= extended; + snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, bits); + bits = snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0); + bits |= basic; + snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, bits); break; case SNDRV_PCM_TRIGGER_STOP: - snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0) & ~(basic)); - snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) & ~(extended)); + case SNDRV_PCM_TRIGGER_SUSPEND: + bits = snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0); + bits &= ~basic; + snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, bits); + bits = snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0); + bits &= ~extended; + snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, bits); break; default: result = -EINVAL; @@ -1103,21 +1136,13 @@ static int snd_ca0106_ac97(struct snd_ca0106 *chip) return snd_ac97_mixer(pbus, &ac97, &chip->ac97); } +static void ca0106_stop_chip(struct snd_ca0106 *chip); + static int snd_ca0106_free(struct snd_ca0106 *chip) { - if (chip->res_port != NULL) { /* avoid access to already used hardware */ - // disable interrupts - snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0); - outl(0, chip->port + INTE); - snd_ca0106_ptr_write(chip, EXTENDED_INT_MASK, 0, 0); - udelay(1000); - // disable audio - //outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG); - outl(0, chip->port + HCFG); - /* FIXME: We need to stop and DMA transfers here. - * But as I am not sure how yet, we cannot from the dma pages. - * So we can fix: snd-malloc: Memory leak? pages not freed = 8 - */ + if (chip->res_port != NULL) { + /* avoid access to already used hardware */ + ca0106_stop_chip(chip); } if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -1203,15 +1228,14 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } -static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct snd_pcm **rpcm) +static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device) { struct snd_pcm *pcm; struct snd_pcm_substream *substream; int err; - if (rpcm) - *rpcm = NULL; - if ((err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm)) < 0) + err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm); + if (err < 0) return err; pcm->private_data = emu; @@ -1238,7 +1262,6 @@ static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct s pcm->info_flags = 0; pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX; strcpy(pcm->name, "CA0106"); - emu->pcm = pcm; for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; substream; @@ -1260,8 +1283,7 @@ static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct s return err; } - if (rpcm) - *rpcm = pcm; + emu->pcm[device] = pcm; return 0; } @@ -1301,89 +1323,10 @@ static unsigned int i2c_adc_init[][2] = { { 0x15, ADC_MUX_LINEIN }, /* ADC Mixer control */ }; -static int __devinit snd_ca0106_create(int dev, struct snd_card *card, - struct pci_dev *pci, - struct snd_ca0106 **rchip) +static void ca0106_init_chip(struct snd_ca0106 *chip, int resume) { - struct snd_ca0106 *chip; - struct snd_ca0106_details *c; - int err; int ch; - static struct snd_device_ops ops = { - .dev_free = snd_ca0106_dev_free, - }; - - *rchip = NULL; - - if ((err = pci_enable_device(pci)) < 0) - return err; - if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) { - printk(KERN_ERR "error to set 32bit mask DMA\n"); - pci_disable_device(pci); - return -ENXIO; - } - - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) { - pci_disable_device(pci); - return -ENOMEM; - } - - chip->card = card; - chip->pci = pci; - chip->irq = -1; - - spin_lock_init(&chip->emu_lock); - - chip->port = pci_resource_start(pci, 0); - if ((chip->res_port = request_region(chip->port, 0x20, - "snd_ca0106")) == NULL) { - snd_ca0106_free(chip); - printk(KERN_ERR "cannot allocate the port\n"); - return -EBUSY; - } - - if (request_irq(pci->irq, snd_ca0106_interrupt, - IRQF_SHARED, "snd_ca0106", chip)) { - snd_ca0106_free(chip); - printk(KERN_ERR "cannot grab irq\n"); - return -EBUSY; - } - chip->irq = pci->irq; - - /* This stores the periods table. */ - if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &chip->buffer) < 0) { - snd_ca0106_free(chip); - return -ENOMEM; - } - - pci_set_master(pci); - /* read serial */ - pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial); - pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model); -#if 1 - printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n", chip->model, - pci->revision, chip->serial); -#endif - strcpy(card->driver, "CA0106"); - strcpy(card->shortname, "CA0106"); - - for (c = ca0106_chip_details; c->serial; c++) { - if (subsystem[dev]) { - if (c->serial == subsystem[dev]) - break; - } else if (c->serial == chip->serial) - break; - } - chip->details = c; - if (subsystem[dev]) { - printk(KERN_INFO "snd-ca0106: Sound card name=%s, subsystem=0x%x. Forced to subsystem=0x%x\n", - c->name, chip->serial, subsystem[dev]); - } - - sprintf(card->longname, "%s at 0x%lx irq %i", - c->name, chip->port, chip->irq); + unsigned int def_bits; outl(0, chip->port + INTE); @@ -1401,31 +1344,22 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, * AN = 0 (Audio data) * P = 0 (Consumer) */ - snd_ca0106_ptr_write(chip, SPCS0, 0, - chip->spdif_bits[0] = - SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | - SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | - SPCS_GENERATIONSTATUS | 0x00001200 | - 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); + def_bits = + SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | + SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | + SPCS_GENERATIONSTATUS | 0x00001200 | + 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT; + if (!resume) { + chip->spdif_str_bits[0] = chip->spdif_bits[0] = def_bits; + chip->spdif_str_bits[1] = chip->spdif_bits[1] = def_bits; + chip->spdif_str_bits[2] = chip->spdif_bits[2] = def_bits; + chip->spdif_str_bits[3] = chip->spdif_bits[3] = def_bits; + } /* Only SPCS1 has been tested */ - snd_ca0106_ptr_write(chip, SPCS1, 0, - chip->spdif_bits[1] = - SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | - SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | - SPCS_GENERATIONSTATUS | 0x00001200 | - 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); - snd_ca0106_ptr_write(chip, SPCS2, 0, - chip->spdif_bits[2] = - SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | - SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | - SPCS_GENERATIONSTATUS | 0x00001200 | - 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); - snd_ca0106_ptr_write(chip, SPCS3, 0, - chip->spdif_bits[3] = - SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | - SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | - SPCS_GENERATIONSTATUS | 0x00001200 | - 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); + snd_ca0106_ptr_write(chip, SPCS1, 0, chip->spdif_str_bits[1]); + snd_ca0106_ptr_write(chip, SPCS0, 0, chip->spdif_str_bits[0]); + snd_ca0106_ptr_write(chip, SPCS2, 0, chip->spdif_str_bits[2]); + snd_ca0106_ptr_write(chip, SPCS3, 0, chip->spdif_str_bits[3]); snd_ca0106_ptr_write(chip, PLAYBACK_MUTE, 0, 0x00fc0000); snd_ca0106_ptr_write(chip, CAPTURE_MUTE, 0, 0x00fc0000); @@ -1433,92 +1367,124 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, /* Write 0x8000 to AC97_REC_GAIN to mute it. */ outb(AC97_REC_GAIN, chip->port + AC97ADDRESS); outw(0x8000, chip->port + AC97DATA); -#if 0 +#if 0 /* FIXME: what are these? */ snd_ca0106_ptr_write(chip, SPCS0, 0, 0x2108006); snd_ca0106_ptr_write(chip, 0x42, 0, 0x2108006); snd_ca0106_ptr_write(chip, 0x43, 0, 0x2108006); snd_ca0106_ptr_write(chip, 0x44, 0, 0x2108006); #endif - //snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); /* OSS drivers set this. */ + /* OSS drivers set this. */ + /* snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); */ + /* Analog or Digital output */ snd_ca0106_ptr_write(chip, SPDIF_SELECT1, 0, 0xf); - snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000f0000); /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers. Use 0x000f0000 for surround71 */ + /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers. + * Use 0x000f0000 for surround71 + */ + snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000f0000); + chip->spdif_enable = 0; /* Set digital SPDIF output off */ - //snd_ca0106_ptr_write(chip, 0x45, 0, 0); /* Analogue out */ - //snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00); /* Digital out */ + /*snd_ca0106_ptr_write(chip, 0x45, 0, 0);*/ /* Analogue out */ + /*snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00);*/ /* Digital out */ + + /* goes to 0x40c80000 when doing SPDIF IN/OUT */ + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000); + /* (Mute) CAPTURE feedback into PLAYBACK volume. + * Only lower 16 bits matter. + */ + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff); + /* SPDIF IN Volume */ + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000); + /* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */ + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000); - snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000); /* goes to 0x40c80000 when doing SPDIF IN/OUT */ - snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff); /* (Mute) CAPTURE feedback into PLAYBACK volume. Only lower 16 bits matter. */ - snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000); /* SPDIF IN Volume */ - snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000); /* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */ snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING1, 0, 0x32765410); snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING2, 0, 0x76767676); snd_ca0106_ptr_write(chip, CAPTURE_ROUTING1, 0, 0x32765410); snd_ca0106_ptr_write(chip, CAPTURE_ROUTING2, 0, 0x76767676); - for(ch = 0; ch < 4; ch++) { - snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030); /* Only high 16 bits matter */ + + for (ch = 0; ch < 4; ch++) { + /* Only high 16 bits matter */ + snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030); snd_ca0106_ptr_write(chip, CAPTURE_VOLUME2, ch, 0x30303030); - //snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040); /* Mute */ - //snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040); /* Mute */ - snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff); /* Mute */ - snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff); /* Mute */ +#if 0 /* Mute */ + snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040); + snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040); + snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff); + snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff); +#endif } if (chip->details->i2c_adc == 1) { /* Select MIC, Line in, TAD in, AUX in */ snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Default to CAPTURE_SOURCE to i2s in */ - chip->capture_source = 3; + if (!resume) + chip->capture_source = 3; } else if (chip->details->ac97 == 1) { /* Default to AC97 in */ snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x444400e4); /* Default to CAPTURE_SOURCE to AC97 in */ - chip->capture_source = 4; + if (!resume) + chip->capture_source = 4; } else { /* Select MIC, Line in, TAD in, AUX in */ snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Default to Set CAPTURE_SOURCE to i2s in */ - chip->capture_source = 3; + if (!resume) + chip->capture_source = 3; } - if (chip->details->gpio_type == 2) { /* The SB0438 use GPIO differently. */ - /* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */ + if (chip->details->gpio_type == 2) { + /* The SB0438 use GPIO differently. */ + /* FIXME: Still need to find out what the other GPIO bits do. + * E.g. For digital spdif out. + */ outl(0x0, chip->port+GPIO); - //outl(0x00f0e000, chip->port+GPIO); /* Analog */ + /* outl(0x00f0e000, chip->port+GPIO); */ /* Analog */ outl(0x005f5301, chip->port+GPIO); /* Analog */ - } else if (chip->details->gpio_type == 1) { /* The SB0410 and SB0413 use GPIO differently. */ - /* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */ + } else if (chip->details->gpio_type == 1) { + /* The SB0410 and SB0413 use GPIO differently. */ + /* FIXME: Still need to find out what the other GPIO bits do. + * E.g. For digital spdif out. + */ outl(0x0, chip->port+GPIO); - //outl(0x00f0e000, chip->port+GPIO); /* Analog */ + /* outl(0x00f0e000, chip->port+GPIO); */ /* Analog */ outl(0x005f5301, chip->port+GPIO); /* Analog */ } else { outl(0x0, chip->port+GPIO); outl(0x005f03a3, chip->port+GPIO); /* Analog */ - //outl(0x005f02a2, chip->port+GPIO); /* SPDIF */ + /* outl(0x005f02a2, chip->port+GPIO); */ /* SPDIF */ } snd_ca0106_intr_enable(chip, 0x105); /* Win2000 uses 0x1e0 */ - //outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG); - //outl(0x00001409, chip->port+HCFG); /* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */ - //outl(0x00000009, chip->port+HCFG); - outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); /* AC97 2.0, Enable outputs. */ + /* outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG); */ + /* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */ + /* outl(0x00001409, chip->port+HCFG); */ + /* outl(0x00000009, chip->port+HCFG); */ + /* AC97 2.0, Enable outputs. */ + outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); - if (chip->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */ + if (chip->details->i2c_adc == 1) { + /* The SB0410 and SB0413 use I2C to control ADC. */ int size, n; size = ARRAY_SIZE(i2c_adc_init); - //snd_printk("I2C:array size=0x%x\n", size); - for (n=0; n < size; n++) { - snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], i2c_adc_init[n][1]); - } - for (n=0; n < 4; n++) { - chip->i2c_capture_volume[n][0]= 0xcf; - chip->i2c_capture_volume[n][1]= 0xcf; + /* snd_printk("I2C:array size=0x%x\n", size); */ + for (n = 0; n < size; n++) + snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], + i2c_adc_init[n][1]); + for (n = 0; n < 4; n++) { + chip->i2c_capture_volume[n][0] = 0xcf; + chip->i2c_capture_volume[n][1] = 0xcf; } - chip->i2c_capture_source=2; /* Line in */ - //snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); /* Enable Line-in capture. MIC in currently untested. */ + chip->i2c_capture_source = 2; /* Line in */ + /* Enable Line-in capture. MIC in currently untested. */ + /* snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); */ } - if (chip->details->spi_dac == 1) { /* The SB0570 use SPI to control DAC. */ + + if (chip->details->spi_dac == 1) { + /* The SB0570 use SPI to control DAC. */ int size, n; size = ARRAY_SIZE(spi_dac_init); @@ -1530,9 +1496,112 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, chip->spi_dac_reg[reg] = spi_dac_init[n]; } } +} - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, - chip, &ops)) < 0) { +static void ca0106_stop_chip(struct snd_ca0106 *chip) +{ + /* disable interrupts */ + snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0); + outl(0, chip->port + INTE); + snd_ca0106_ptr_write(chip, EXTENDED_INT_MASK, 0, 0); + udelay(1000); + /* disable audio */ + /* outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG); */ + outl(0, chip->port + HCFG); + /* FIXME: We need to stop and DMA transfers here. + * But as I am not sure how yet, we cannot from the dma pages. + * So we can fix: snd-malloc: Memory leak? pages not freed = 8 + */ +} + +static int __devinit snd_ca0106_create(int dev, struct snd_card *card, + struct pci_dev *pci, + struct snd_ca0106 **rchip) +{ + struct snd_ca0106 *chip; + struct snd_ca0106_details *c; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_ca0106_dev_free, + }; + + *rchip = NULL; + + err = pci_enable_device(pci); + if (err < 0) + return err; + if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || + pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) { + printk(KERN_ERR "error to set 32bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + chip->card = card; + chip->pci = pci; + chip->irq = -1; + + spin_lock_init(&chip->emu_lock); + + chip->port = pci_resource_start(pci, 0); + chip->res_port = request_region(chip->port, 0x20, "snd_ca0106"); + if (!chip->res_port) { + snd_ca0106_free(chip); + printk(KERN_ERR "cannot allocate the port\n"); + return -EBUSY; + } + + if (request_irq(pci->irq, snd_ca0106_interrupt, + IRQF_SHARED, "snd_ca0106", chip)) { + snd_ca0106_free(chip); + printk(KERN_ERR "cannot grab irq\n"); + return -EBUSY; + } + chip->irq = pci->irq; + + /* This stores the periods table. */ + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + 1024, &chip->buffer) < 0) { + snd_ca0106_free(chip); + return -ENOMEM; + } + + pci_set_master(pci); + /* read serial */ + pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial); + pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model); + printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n", + chip->model, pci->revision, chip->serial); + strcpy(card->driver, "CA0106"); + strcpy(card->shortname, "CA0106"); + + for (c = ca0106_chip_details; c->serial; c++) { + if (subsystem[dev]) { + if (c->serial == subsystem[dev]) + break; + } else if (c->serial == chip->serial) + break; + } + chip->details = c; + if (subsystem[dev]) { + printk(KERN_INFO "snd-ca0106: Sound card name=%s, " + "subsystem=0x%x. Forced to subsystem=0x%x\n", + c->name, chip->serial, subsystem[dev]); + } + + sprintf(card->longname, "%s at 0x%lx irq %i", + c->name, chip->port, chip->irq); + + ca0106_init_chip(chip, 0); + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { snd_ca0106_free(chip); return err; } @@ -1629,7 +1698,7 @@ static int __devinit snd_ca0106_probe(struct pci_dev *pci, static int dev; struct snd_card *card; struct snd_ca0106 *chip; - int err; + int i, err; if (dev >= SNDRV_CARDS) return -ENODEV; @@ -1642,44 +1711,31 @@ static int __devinit snd_ca0106_probe(struct pci_dev *pci, if (card == NULL) return -ENOMEM; - if ((err = snd_ca0106_create(dev, card, pci, &chip)) < 0) { - snd_card_free(card); - return err; - } + err = snd_ca0106_create(dev, card, pci, &chip); + if (err < 0) + goto error; + card->private_data = chip; - if ((err = snd_ca0106_pcm(chip, 0, NULL)) < 0) { - snd_card_free(card); - return err; - } - if ((err = snd_ca0106_pcm(chip, 1, NULL)) < 0) { - snd_card_free(card); - return err; - } - if ((err = snd_ca0106_pcm(chip, 2, NULL)) < 0) { - snd_card_free(card); - return err; - } - if ((err = snd_ca0106_pcm(chip, 3, NULL)) < 0) { - snd_card_free(card); - return err; - } - if (chip->details->ac97 == 1) { /* The SB0410 and SB0413 do not have an AC97 chip. */ - if ((err = snd_ca0106_ac97(chip)) < 0) { - snd_card_free(card); - return err; - } + for (i = 0; i < 4; i++) { + err = snd_ca0106_pcm(chip, i); + if (err < 0) + goto error; } - if ((err = snd_ca0106_mixer(chip)) < 0) { - snd_card_free(card); - return err; + + if (chip->details->ac97 == 1) { + /* The SB0410 and SB0413 do not have an AC97 chip. */ + err = snd_ca0106_ac97(chip); + if (err < 0) + goto error; } + err = snd_ca0106_mixer(chip); + if (err < 0) + goto error; snd_printdd("ca0106: probe for MIDI channel A ..."); - if ((err = snd_ca0106_midi(chip,CA0106_MIDI_CHAN_A)) < 0) { - snd_card_free(card); - snd_printdd(" failed, err=0x%x\n",err); - return err; - } + err = snd_ca0106_midi(chip, CA0106_MIDI_CHAN_A); + if (err < 0) + goto error; snd_printdd(" done.\n"); #ifdef CONFIG_PROC_FS @@ -1688,14 +1744,17 @@ static int __devinit snd_ca0106_probe(struct pci_dev *pci, snd_card_set_dev(card, &pci->dev); - if ((err = snd_card_register(card)) < 0) { - snd_card_free(card); - return err; - } + err = snd_card_register(card); + if (err < 0) + goto error; pci_set_drvdata(pci, card); dev++; return 0; + + error: + snd_card_free(card); + return err; } static void __devexit snd_ca0106_remove(struct pci_dev *pci) @@ -1704,6 +1763,59 @@ static void __devexit snd_ca0106_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } +#ifdef CONFIG_PM +static int snd_ca0106_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct snd_ca0106 *chip = card->private_data; + int i; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + for (i = 0; i < 4; i++) + snd_pcm_suspend_all(chip->pcm[i]); + if (chip->details->ac97) + snd_ac97_suspend(chip->ac97); + snd_ca0106_mixer_suspend(chip); + + ca0106_stop_chip(chip); + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} + +static int snd_ca0106_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct snd_ca0106 *chip = card->private_data; + int i; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + + if (pci_enable_device(pci) < 0) { + snd_card_disconnect(card); + return -EIO; + } + + pci_set_master(pci); + + ca0106_init_chip(chip, 1); + + if (chip->details->ac97) + snd_ac97_resume(chip->ac97); + snd_ca0106_mixer_resume(chip); + if (chip->details->spi_dac) { + for (i = 0; i < ARRAY_SIZE(chip->spi_dac_reg); i++) + snd_ca0106_spi_write(chip, chip->spi_dac_reg[i]); + } + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + // PCI IDs static struct pci_device_id snd_ca0106_ids[] = { { 0x1102, 0x0007, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* Audigy LS or Live 24bit */ @@ -1717,6 +1829,10 @@ static struct pci_driver driver = { .id_table = snd_ca0106_ids, .probe = snd_ca0106_probe, .remove = __devexit_p(snd_ca0106_remove), +#ifdef CONFIG_PM + .suspend = snd_ca0106_suspend, + .resume = snd_ca0106_resume, +#endif }; // initialization of the module diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 3025ed1b6e1e..ad2888705d2a 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -75,6 +75,84 @@ #include "ca0106.h" +static void ca0106_spdif_enable(struct snd_ca0106 *emu) +{ + unsigned int val; + + if (emu->spdif_enable) { + /* Digital */ + snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf); + snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000); + val = snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000; + snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, val); + val = inl(emu->port + GPIO) & ~0x101; + outl(val, emu->port + GPIO); + + } else { + /* Analog */ + snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf); + snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000f0000); + val = snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000; + snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, val); + val = inl(emu->port + GPIO) | 0x101; + outl(val, emu->port + GPIO); + } +} + +static void ca0106_set_capture_source(struct snd_ca0106 *emu) +{ + unsigned int val = emu->capture_source; + unsigned int source, mask; + source = (val << 28) | (val << 24) | (val << 20) | (val << 16); + mask = snd_ca0106_ptr_read(emu, CAPTURE_SOURCE, 0) & 0xffff; + snd_ca0106_ptr_write(emu, CAPTURE_SOURCE, 0, source | mask); +} + +static void ca0106_set_i2c_capture_source(struct snd_ca0106 *emu, + unsigned int val, int force) +{ + unsigned int ngain, ogain; + u32 source; + + snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */ + ngain = emu->i2c_capture_volume[val][0]; /* Left */ + ogain = emu->i2c_capture_volume[emu->i2c_capture_source][0]; /* Left */ + if (force || ngain != ogain) + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ngain & 0xff); + ngain = emu->i2c_capture_volume[val][1]; /* Right */ + ogain = emu->i2c_capture_volume[emu->i2c_capture_source][1]; /* Right */ + if (force || ngain != ogain) + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ngain & 0xff); + source = 1 << val; + snd_ca0106_i2c_write(emu, ADC_MUX, source); /* Set source */ + emu->i2c_capture_source = val; +} + +static void ca0106_set_capture_mic_line_in(struct snd_ca0106 *emu) +{ + u32 tmp; + + if (emu->capture_mic_line_in) { + /* snd_ca0106_i2c_write(emu, ADC_MUX, 0); */ /* Mute input */ + tmp = inl(emu->port+GPIO) & ~0x400; + tmp = tmp | 0x400; + outl(tmp, emu->port+GPIO); + /* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC); */ + } else { + /* snd_ca0106_i2c_write(emu, ADC_MUX, 0); */ /* Mute input */ + tmp = inl(emu->port+GPIO) & ~0x400; + outl(tmp, emu->port+GPIO); + /* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN); */ + } +} + +static void ca0106_set_spdif_bits(struct snd_ca0106 *emu, int idx) +{ + snd_ca0106_ptr_write(emu, SPCS0 + idx, 0, emu->spdif_str_bits[idx]); +} + +/* + */ static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale1, -5175, 25, 1); static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1); @@ -95,30 +173,12 @@ static int snd_ca0106_shared_spdif_put(struct snd_kcontrol *kcontrol, struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); unsigned int val; int change = 0; - u32 mask; val = !!ucontrol->value.integer.value[0]; change = (emu->spdif_enable != val); if (change) { emu->spdif_enable = val; - if (val) { - /* Digital */ - snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf); - snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000); - snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, - snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000); - mask = inl(emu->port + GPIO) & ~0x101; - outl(mask, emu->port + GPIO); - - } else { - /* Analog */ - snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf); - snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000f0000); - snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, - snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000); - mask = inl(emu->port + GPIO) | 0x101; - outl(mask, emu->port + GPIO); - } + ca0106_spdif_enable(emu); } return change; } @@ -154,8 +214,6 @@ static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol, struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); unsigned int val; int change = 0; - u32 mask; - u32 source; val = ucontrol->value.enumerated.item[0] ; if (val >= 6) @@ -163,9 +221,7 @@ static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol, change = (emu->capture_source != val); if (change) { emu->capture_source = val; - source = (val << 28) | (val << 24) | (val << 20) | (val << 16); - mask = snd_ca0106_ptr_read(emu, CAPTURE_SOURCE, 0) & 0xffff; - snd_ca0106_ptr_write(emu, CAPTURE_SOURCE, 0, source | mask); + ca0106_set_capture_source(emu); } return change; } @@ -200,9 +256,7 @@ static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol, { struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); unsigned int source_id; - unsigned int ngain, ogain; int change = 0; - u32 source; /* If the capture source has changed, * update the capture volume from the cached value * for the particular source. @@ -212,18 +266,7 @@ static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol, return -EINVAL; change = (emu->i2c_capture_source != source_id); if (change) { - snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */ - ngain = emu->i2c_capture_volume[source_id][0]; /* Left */ - ogain = emu->i2c_capture_volume[emu->i2c_capture_source][0]; /* Left */ - if (ngain != ogain) - snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff)); - ngain = emu->i2c_capture_volume[source_id][1]; /* Left */ - ogain = emu->i2c_capture_volume[emu->i2c_capture_source][1]; /* Left */ - if (ngain != ogain) - snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff)); - source = 1 << source_id; - snd_ca0106_i2c_write(emu, ADC_MUX, source); /* Set source */ - emu->i2c_capture_source = source_id; + ca0106_set_i2c_capture_source(emu, source_id, 0); } return change; } @@ -271,7 +314,6 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol, struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); unsigned int val; int change = 0; - u32 tmp; val = ucontrol->value.enumerated.item[0] ; if (val > 1) @@ -279,18 +321,7 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol, change = (emu->capture_mic_line_in != val); if (change) { emu->capture_mic_line_in = val; - if (val) { - //snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */ - tmp = inl(emu->port+GPIO) & ~0x400; - tmp = tmp | 0x400; - outl(tmp, emu->port+GPIO); - //snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC); - } else { - //snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */ - tmp = inl(emu->port+GPIO) & ~0x400; - outl(tmp, emu->port+GPIO); - //snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN); - } + ca0106_set_capture_mic_line_in(emu); } return change; } @@ -322,16 +353,33 @@ static int snd_ca0106_spdif_info(struct snd_kcontrol *kcontrol, return 0; } -static int snd_ca0106_spdif_get(struct snd_kcontrol *kcontrol, +static void decode_spdif_bits(unsigned char *status, unsigned int bits) +{ + status[0] = (bits >> 0) & 0xff; + status[1] = (bits >> 8) & 0xff; + status[2] = (bits >> 16) & 0xff; + status[3] = (bits >> 24) & 0xff; +} + +static int snd_ca0106_spdif_get_default(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - ucontrol->value.iec958.status[0] = (emu->spdif_bits[idx] >> 0) & 0xff; - ucontrol->value.iec958.status[1] = (emu->spdif_bits[idx] >> 8) & 0xff; - ucontrol->value.iec958.status[2] = (emu->spdif_bits[idx] >> 16) & 0xff; - ucontrol->value.iec958.status[3] = (emu->spdif_bits[idx] >> 24) & 0xff; + decode_spdif_bits(ucontrol->value.iec958.status, + emu->spdif_bits[idx]); + return 0; +} + +static int snd_ca0106_spdif_get_stream(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + + decode_spdif_bits(ucontrol->value.iec958.status, + emu->spdif_str_bits[idx]); return 0; } @@ -345,24 +393,48 @@ static int snd_ca0106_spdif_get_mask(struct snd_kcontrol *kcontrol, return 0; } -static int snd_ca0106_spdif_put(struct snd_kcontrol *kcontrol, +static unsigned int encode_spdif_bits(unsigned char *status) +{ + return ((unsigned int)status[0] << 0) | + ((unsigned int)status[1] << 8) | + ((unsigned int)status[2] << 16) | + ((unsigned int)status[3] << 24); +} + +static int snd_ca0106_spdif_put_default(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - int change; unsigned int val; - val = (ucontrol->value.iec958.status[0] << 0) | - (ucontrol->value.iec958.status[1] << 8) | - (ucontrol->value.iec958.status[2] << 16) | - (ucontrol->value.iec958.status[3] << 24); - change = val != emu->spdif_bits[idx]; - if (change) { - snd_ca0106_ptr_write(emu, SPCS0 + idx, 0, val); + val = encode_spdif_bits(ucontrol->value.iec958.status); + if (val != emu->spdif_bits[idx]) { emu->spdif_bits[idx] = val; + /* FIXME: this isn't safe, but needed to keep the compatibility + * with older alsa-lib config + */ + emu->spdif_str_bits[idx] = val; + ca0106_set_spdif_bits(emu, idx); + return 1; } - return change; + return 0; +} + +static int snd_ca0106_spdif_put_stream(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + unsigned int val; + + val = encode_spdif_bits(ucontrol->value.iec958.status); + if (val != emu->spdif_str_bits[idx]) { + emu->spdif_str_bits[idx] = val; + ca0106_set_spdif_bits(emu, idx); + return 1; + } + return 0; } static int snd_ca0106_volume_info(struct snd_kcontrol *kcontrol, @@ -573,8 +645,16 @@ static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = { .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), .count = 4, .info = snd_ca0106_spdif_info, - .get = snd_ca0106_spdif_get, - .put = snd_ca0106_spdif_put + .get = snd_ca0106_spdif_get_default, + .put = snd_ca0106_spdif_put_default + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM), + .count = 4, + .info = snd_ca0106_spdif_info, + .get = snd_ca0106_spdif_get_stream, + .put = snd_ca0106_spdif_put_stream }, }; @@ -773,3 +853,50 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) return 0; } +#ifdef CONFIG_PM +struct ca0106_vol_tbl { + unsigned int channel_id; + unsigned int reg; +}; + +static struct ca0106_vol_tbl saved_volumes[NUM_SAVED_VOLUMES] = { + { CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME2 }, + { CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME2 }, + { CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME2 }, + { CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME2 }, + { CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME1 }, + { CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME1 }, + { CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME1 }, + { CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME1 }, + { 1, CAPTURE_CONTROL }, +}; + +void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip) +{ + int i; + + /* save volumes */ + for (i = 0; i < NUM_SAVED_VOLUMES; i++) + chip->saved_vol[i] = + snd_ca0106_ptr_read(chip, saved_volumes[i].reg, + saved_volumes[i].channel_id); +} + +void snd_ca0106_mixer_resume(struct snd_ca0106 *chip) +{ + int i; + + for (i = 0; i < NUM_SAVED_VOLUMES; i++) + snd_ca0106_ptr_write(chip, saved_volumes[i].reg, + saved_volumes[i].channel_id, + chip->saved_vol[i]); + + ca0106_spdif_enable(chip); + ca0106_set_capture_source(chip); + ca0106_set_i2c_capture_source(chip, chip->i2c_capture_source, 1); + for (i = 0; i < 4; i++) + ca0106_set_spdif_bits(chip, i); + if (chip->details->i2c_adc) + ca0106_set_capture_mic_line_in(chip); +} +#endif /* CONFIG_PM */ diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index fb6dc3980257..8ab07aa63652 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -3640,7 +3640,10 @@ int snd_cs46xx_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct snd_cs46xx *chip = card->private_data; - int i, amp_saved; + int amp_saved; +#ifdef CONFIG_SND_CS46XX_NEW_DSP + int i; +#endif pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); diff --git a/sound/pci/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile index bb3d57e6a3cb..fda7a94c992f 100644 --- a/sound/pci/cs5535audio/Makefile +++ b/sound/pci/cs5535audio/Makefile @@ -4,6 +4,9 @@ snd-cs5535audio-y := cs5535audio.o cs5535audio_pcm.o snd-cs5535audio-$(CONFIG_PM) += cs5535audio_pm.o +ifdef CONFIG_MGEODE_LX +snd-cs5535audio-$(CONFIG_OLPC) += cs5535audio_olpc.o +endif # Toplevel Module Dependency obj-$(CONFIG_SND_CS5535AUDIO) += snd-cs5535audio.o diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 1d8b16052535..826e6dec2e97 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -159,10 +159,14 @@ static int __devinit snd_cs5535audio_mixer(struct cs5535audio *cs5535au) return err; memset(&ac97, 0, sizeof(ac97)); - ac97.scaps = AC97_SCAP_AUDIO|AC97_SCAP_SKIP_MODEM; + ac97.scaps = AC97_SCAP_AUDIO | AC97_SCAP_SKIP_MODEM + | AC97_SCAP_POWER_SAVE; ac97.private_data = cs5535au; ac97.pci = cs5535au->pci; + /* set any OLPC-specific scaps */ + olpc_prequirks(card, &ac97); + if ((err = snd_ac97_mixer(pbus, &ac97, &cs5535au->ac97)) < 0) { snd_printk(KERN_ERR "mixer failed\n"); return err; @@ -170,6 +174,12 @@ static int __devinit snd_cs5535audio_mixer(struct cs5535audio *cs5535au) snd_ac97_tune_hardware(cs5535au->ac97, ac97_quirks, ac97_quirk); + err = olpc_quirks(card, cs5535au->ac97); + if (err < 0) { + snd_printk(KERN_ERR "olpc quirks failed\n"); + return err; + } + return 0; } diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h index 66bae7664193..7a298ac662e3 100644 --- a/sound/pci/cs5535audio/cs5535audio.h +++ b/sound/pci/cs5535audio/cs5535audio.h @@ -78,6 +78,7 @@ struct cs5535audio_dma { unsigned int buf_addr, buf_bytes; unsigned int period_bytes, periods; u32 saved_prd; + int pcm_open_flag; }; struct cs5535audio { @@ -93,8 +94,46 @@ struct cs5535audio { struct cs5535audio_dma dmas[NUM_CS5535AUDIO_DMAS]; }; +#ifdef CONFIG_PM int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state); int snd_cs5535audio_resume(struct pci_dev *pci); +#endif + +#if defined(CONFIG_OLPC) && defined(CONFIG_MGEODE_LX) +void __devinit olpc_prequirks(struct snd_card *card, + struct snd_ac97_template *ac97); +int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97); +void olpc_analog_input(struct snd_ac97 *ac97, int on); +void olpc_mic_bias(struct snd_ac97 *ac97, int on); + +static inline void olpc_capture_open(struct snd_ac97 *ac97) +{ + /* default to Analog Input off */ + olpc_analog_input(ac97, 0); + /* enable MIC Bias for recording */ + olpc_mic_bias(ac97, 1); +} + +static inline void olpc_capture_close(struct snd_ac97 *ac97) +{ + /* disable Analog Input */ + olpc_analog_input(ac97, 0); + /* disable the MIC Bias (so the recording LED turns off) */ + olpc_mic_bias(ac97, 0); +} +#else +static inline void olpc_prequirks(struct snd_card *card, + struct snd_ac97_template *ac97) { } +static inline int olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97) +{ + return 0; +} +static inline void olpc_analog_input(struct snd_ac97 *ac97, int on) { } +static inline void olpc_mic_bias(struct snd_ac97 *ac97, int on) { } +static inline void olpc_capture_open(struct snd_ac97 *ac97) { } +static inline void olpc_capture_close(struct snd_ac97 *ac97) { } +#endif + int __devinit snd_cs5535audio_pcm(struct cs5535audio *cs5535audio); #endif /* __SOUND_CS5535AUDIO_H */ diff --git a/sound/pci/cs5535audio/cs5535audio_olpc.c b/sound/pci/cs5535audio/cs5535audio_olpc.c new file mode 100644 index 000000000000..5c6814335cd7 --- /dev/null +++ b/sound/pci/cs5535audio/cs5535audio_olpc.c @@ -0,0 +1,179 @@ +/* + * OLPC XO-1 additional sound features + * + * Copyright © 2006 Jaya Kumar <jayakumar.lkml@gmail.com> + * Copyright © 2007-2008 Andres Salomon <dilinger@debian.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ +#include <sound/core.h> +#include <sound/info.h> +#include <sound/control.h> +#include <sound/ac97_codec.h> + +#include <asm/olpc.h> +#include "cs5535audio.h" + +/* + * OLPC has an additional feature on top of the regular AD1888 codec features. + * It has an Analog Input mode that is switched into (after disabling the + * High Pass Filter) via GPIO. It is supported on B2 and later models. + */ +void olpc_analog_input(struct snd_ac97 *ac97, int on) +{ + int err; + + if (!machine_is_olpc()) + return; + + /* update the High Pass Filter (via AC97_AD_TEST2) */ + err = snd_ac97_update_bits(ac97, AC97_AD_TEST2, + 1 << AC97_AD_HPFD_SHIFT, on << AC97_AD_HPFD_SHIFT); + if (err < 0) { + snd_printk(KERN_ERR "setting High Pass Filter - %d\n", err); + return; + } + + /* set Analog Input through GPIO */ + if (on) + geode_gpio_set(OLPC_GPIO_MIC_AC, GPIO_OUTPUT_VAL); + else + geode_gpio_clear(OLPC_GPIO_MIC_AC, GPIO_OUTPUT_VAL); +} + +/* + * OLPC XO-1's V_REFOUT is a mic bias enable. + */ +void olpc_mic_bias(struct snd_ac97 *ac97, int on) +{ + int err; + + if (!machine_is_olpc()) + return; + + on = on ? 0 : 1; + err = snd_ac97_update_bits(ac97, AC97_AD_MISC, + 1 << AC97_AD_VREFD_SHIFT, on << AC97_AD_VREFD_SHIFT); + if (err < 0) + snd_printk(KERN_ERR "setting MIC Bias - %d\n", err); +} + +static int olpc_dc_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int olpc_dc_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v) +{ + v->value.integer.value[0] = geode_gpio_isset(OLPC_GPIO_MIC_AC, + GPIO_OUTPUT_VAL); + return 0; +} + +static int olpc_dc_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v) +{ + struct cs5535audio *cs5535au = snd_kcontrol_chip(kctl); + + olpc_analog_input(cs5535au->ac97, v->value.integer.value[0]); + return 1; +} + +static int olpc_mic_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int olpc_mic_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v) +{ + struct cs5535audio *cs5535au = snd_kcontrol_chip(kctl); + struct snd_ac97 *ac97 = cs5535au->ac97; + int i; + + i = (snd_ac97_read(ac97, AC97_AD_MISC) >> AC97_AD_VREFD_SHIFT) & 0x1; + v->value.integer.value[0] = i ? 0 : 1; + return 0; +} + +static int olpc_mic_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v) +{ + struct cs5535audio *cs5535au = snd_kcontrol_chip(kctl); + + olpc_mic_bias(cs5535au->ac97, v->value.integer.value[0]); + return 1; +} + +static struct snd_kcontrol_new olpc_cs5535audio_ctls[] __devinitdata = { +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DC Mode Enable", + .info = olpc_dc_info, + .get = olpc_dc_get, + .put = olpc_dc_put, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "MIC Bias Enable", + .info = olpc_mic_info, + .get = olpc_mic_get, + .put = olpc_mic_put, + .private_value = 0, +}, +}; + +void __devinit olpc_prequirks(struct snd_card *card, + struct snd_ac97_template *ac97) +{ + if (!machine_is_olpc()) + return; + + /* invert EAPD if on an OLPC B3 or higher */ + if (olpc_board_at_least(olpc_board_pre(0xb3))) + ac97->scaps |= AC97_SCAP_INV_EAPD; +} + +int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97) +{ + struct snd_ctl_elem_id elem; + int i, err; + + if (!machine_is_olpc()) + return 0; + + /* drop the original AD1888 HPF control */ + memset(&elem, 0, sizeof(elem)); + elem.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strncpy(elem.name, "High Pass Filter Enable", sizeof(elem.name)); + snd_ctl_remove_id(card, &elem); + + /* drop the original V_REFOUT control */ + memset(&elem, 0, sizeof(elem)); + elem.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strncpy(elem.name, "V_REFOUT Enable", sizeof(elem.name)); + snd_ctl_remove_id(card, &elem); + + /* add the OLPC-specific controls */ + for (i = 0; i < ARRAY_SIZE(olpc_cs5535audio_ctls); i++) { + err = snd_ctl_add(card, snd_ctl_new1(&olpc_cs5535audio_ctls[i], + ac97->private_data)); + if (err < 0) + return err; + } + + /* turn off the mic by default */ + olpc_mic_bias(ac97, 0); + return 0; +} diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index cdcda87116c3..0f48a871f17b 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -260,6 +260,9 @@ static int snd_cs5535audio_hw_params(struct snd_pcm_substream *substream, err = cs5535audio_build_dma_packets(cs5535au, dma, substream, params_periods(hw_params), params_period_bytes(hw_params)); + if (!err) + dma->pcm_open_flag = 1; + return err; } @@ -268,6 +271,15 @@ static int snd_cs5535audio_hw_free(struct snd_pcm_substream *substream) struct cs5535audio *cs5535au = snd_pcm_substream_chip(substream); struct cs5535audio_dma *dma = substream->runtime->private_data; + if (dma->pcm_open_flag) { + if (substream == cs5535au->playback_substream) + snd_ac97_update_power(cs5535au->ac97, + AC97_PCM_FRONT_DAC_RATE, 0); + else + snd_ac97_update_power(cs5535au->ac97, + AC97_PCM_LR_ADC_RATE, 0); + dma->pcm_open_flag = 0; + } cs5535audio_clear_dma_packets(cs5535au, dma, substream); return snd_pcm_lib_free_pages(substream); } @@ -351,11 +363,14 @@ static int snd_cs5535audio_capture_open(struct snd_pcm_substream *substream) if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; + olpc_capture_open(cs5535au->ac97); return 0; } static int snd_cs5535audio_capture_close(struct snd_pcm_substream *substream) { + struct cs5535audio *cs5535au = snd_pcm_substream_chip(substream); + olpc_capture_close(cs5535au->ac97); return 0; } diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index de5ee8f097f6..7958006a1d66 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -69,7 +69,7 @@ MODULE_FIRMWARE(EMU1010_NOTEBOOK_FILENAME); * EMU10K1 init / done *************************************************************************/ -void snd_emu10k1_voice_init(struct snd_emu10k1 * emu, int ch) +void snd_emu10k1_voice_init(struct snd_emu10k1 *emu, int ch) { snd_emu10k1_ptr_write(emu, DCYSUSV, ch, 0); snd_emu10k1_ptr_write(emu, IP, ch, 0); @@ -151,9 +151,9 @@ static unsigned int i2c_adc_init[][2] = { { 0x12, 0x32 }, /* ALC Control 3 */ { 0x13, 0x00 }, /* Noise gate control */ { 0x14, 0xa6 }, /* Limiter control */ - { 0x15, ADC_MUX_2 }, /* ADC Mixer control. Mic for Audigy 2 ZS Notebook */ + { 0x15, ADC_MUX_2 }, /* ADC Mixer control. Mic for A2ZS Notebook */ }; - + static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) { unsigned int silent_page; @@ -161,8 +161,8 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) u32 tmp; /* disable audio and lock cache */ - outl(HCFG_LOCKSOUNDCACHE | HCFG_LOCKTANKCACHE_MASK | HCFG_MUTEBUTTONENABLE, - emu->port + HCFG); + outl(HCFG_LOCKSOUNDCACHE | HCFG_LOCKTANKCACHE_MASK | + HCFG_MUTEBUTTONENABLE, emu->port + HCFG); /* reset recording buffers */ snd_emu10k1_ptr_write(emu, MICBS, 0, ADCBS_BUFSIZE_NONE); @@ -179,7 +179,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) snd_emu10k1_ptr_write(emu, SOLEL, 0, 0); snd_emu10k1_ptr_write(emu, SOLEH, 0, 0); - if (emu->audigy){ + if (emu->audigy) { /* set SPDIF bypass mode */ snd_emu10k1_ptr_write(emu, SPBYPASS, 0, SPBYPASS_FORMAT); /* enable rear left + rear right AC97 slots */ @@ -197,12 +197,12 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) if (emu->card_capabilities->ca0151_chip) { /* audigy2 */ /* Hacks for Alice3 to work independent of haP16V driver */ - //Setup SRCMulti_I2S SamplingRate + /* Setup SRCMulti_I2S SamplingRate */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0); tmp &= 0xfffff1ff; tmp |= (0x2<<9); snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, 0, tmp); - + /* Setup SRCSel (Enable Spdif,I2S SRCMulti) */ snd_emu10k1_ptr20_write(emu, SRCSel, 0, 0x14); /* Setup SRCMulti Input Audio Enable */ @@ -217,7 +217,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) if (emu->card_capabilities->ca0108_chip) { /* audigy2 Value */ /* Hacks for Alice3 to work independent of haP16V driver */ snd_printk(KERN_INFO "Audigy2 value: Special config.\n"); - //Setup SRCMulti_I2S SamplingRate + /* Setup SRCMulti_I2S SamplingRate */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0); tmp &= 0xfffff1ff; tmp |= (0x2<<9); @@ -270,13 +270,13 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) size = ARRAY_SIZE(i2c_adc_init); for (n = 0; n < size; n++) snd_emu10k1_i2c_write(emu, i2c_adc_init[n][0], i2c_adc_init[n][1]); - for (n=0; n < 4; n++) { - emu->i2c_capture_volume[n][0]= 0xcf; - emu->i2c_capture_volume[n][1]= 0xcf; + for (n = 0; n < 4; n++) { + emu->i2c_capture_volume[n][0] = 0xcf; + emu->i2c_capture_volume[n][1] = 0xcf; } } - + snd_emu10k1_ptr_write(emu, PTB, 0, emu->ptb_pages.addr); snd_emu10k1_ptr_write(emu, TCB, 0, 0); /* taken from original driver */ snd_emu10k1_ptr_write(emu, TCBS, 0, 4); /* taken from original driver */ @@ -313,7 +313,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) (emu->model == 0x21 && emu->revision < 6)) outl(HCFG_LOCKTANKCACHE_MASK | HCFG_AUTOMUTE, emu->port + HCFG); else - // With on-chip joystick + /* With on-chip joystick */ outl(HCFG_LOCKTANKCACHE_MASK | HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG); if (enable_ir) { /* enable IR for SB Live */ @@ -335,9 +335,9 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) outl(reg | HCFG_GPOUT1 | HCFG_GPOUT2, emu->port + HCFG); udelay(100); outl(reg, emu->port + HCFG); - } + } } - + if (emu->card_capabilities->emu_model) { ; /* Disable all access to A_IOCFG for the emu1010 */ } else if (emu->card_capabilities->i2c_adc) { @@ -364,7 +364,7 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu) ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */ } else if (emu->audigy) { outl(inl(emu->port + A_IOCFG) & ~0x44, emu->port + A_IOCFG); - + if (emu->card_capabilities->ca0151_chip) { /* audigy2 */ /* Unmute Analog now. Set GPO6 to 1 for Apollo. * This has to be done after init ALice3 I2SOut beyond 48KHz. @@ -378,12 +378,12 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu) outl(inl(emu->port + A_IOCFG) | 0x0080, emu->port + A_IOCFG); } } - + #if 0 { unsigned int tmp; /* FIXME: the following routine disables LiveDrive-II !! */ - // TOSLink detection + /* TOSLink detection */ emu->tos_link = 0; tmp = inl(emu->port + HCFG); if (tmp & (HCFG_GPINPUT0 | HCFG_GPINPUT1)) { @@ -400,7 +400,7 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu) snd_emu10k1_intr_enable(emu, INTE_PCIERRORENABLE); } -int snd_emu10k1_done(struct snd_emu10k1 * emu) +int snd_emu10k1_done(struct snd_emu10k1 *emu) { int ch; @@ -495,7 +495,7 @@ int snd_emu10k1_done(struct snd_emu10k1 * emu) #define EC_LAST_PROMFILE_ADDR 0x2f -#define EC_SERIALNUM_ADDR 0x30 /* First word of serial number. The +#define EC_SERIALNUM_ADDR 0x30 /* First word of serial number. The * can be up to 30 characters in length * and is stored as a NULL-terminated * ASCII string. Any unused bytes must be @@ -503,8 +503,8 @@ int snd_emu10k1_done(struct snd_emu10k1 * emu) #define EC_CHECKSUM_ADDR 0x3f /* Location at which checksum is stored */ -/* Most of this stuff is pretty self-evident. According to the hardware - * dudes, we need to leave the ADCCAL bit low in order to avoid a DC +/* Most of this stuff is pretty self-evident. According to the hardware + * dudes, we need to leave the ADCCAL bit low in order to avoid a DC * offset problem. Weird. */ #define EC_RAW_RUN_MODE (EC_DACMUTEN | EC_ADCRSTN | EC_TRIM_MUTEN | \ @@ -523,7 +523,7 @@ int snd_emu10k1_done(struct snd_emu10k1 * emu) * register. */ -static void snd_emu10k1_ecard_write(struct snd_emu10k1 * emu, unsigned int value) +static void snd_emu10k1_ecard_write(struct snd_emu10k1 *emu, unsigned int value) { unsigned short count; unsigned int data; @@ -561,7 +561,7 @@ static void snd_emu10k1_ecard_write(struct snd_emu10k1 * emu, unsigned int value * channel. */ -static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 * emu, +static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 *emu, unsigned short gain) { unsigned int bit; @@ -574,7 +574,7 @@ static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 * emu, for (bit = (1 << 15); bit; bit >>= 1) { unsigned int value; - + value = emu->ecard_ctrl & ~(EC_TRIM_CSN | EC_TRIM_SDATA); if (gain & bit) @@ -589,7 +589,7 @@ static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 * emu, snd_emu10k1_ecard_write(emu, emu->ecard_ctrl); } -static int snd_emu10k1_ecard_init(struct snd_emu10k1 * emu) +static int snd_emu10k1_ecard_init(struct snd_emu10k1 *emu) { unsigned int hc_value; @@ -598,7 +598,7 @@ static int snd_emu10k1_ecard_init(struct snd_emu10k1 * emu) EC_SPDIF0_SELECT(EC_DEFAULT_SPDIF0_SEL) | EC_SPDIF1_SELECT(EC_DEFAULT_SPDIF1_SEL); - /* Step 0: Set the codec type in the hardware control register + /* Step 0: Set the codec type in the hardware control register * and enable audio output */ hc_value = inl(emu->port + HCFG); outl(hc_value | HCFG_AUDIOENABLE | HCFG_CODECFORMAT_I2S, emu->port + HCFG); @@ -629,7 +629,7 @@ static int snd_emu10k1_ecard_init(struct snd_emu10k1 * emu) return 0; } -static int snd_emu10k1_cardbus_init(struct snd_emu10k1 * emu) +static int snd_emu10k1_cardbus_init(struct snd_emu10k1 *emu) { unsigned long special_port; unsigned int value; @@ -656,7 +656,7 @@ static int snd_emu10k1_cardbus_init(struct snd_emu10k1 * emu) return 0; } -static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * filename) +static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu, const char *filename) { int err; int n, i; @@ -666,11 +666,12 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file unsigned long flags; const struct firmware *fw_entry; - if ((err = request_firmware(&fw_entry, filename, &emu->pci->dev)) != 0) { - snd_printk(KERN_ERR "firmware: %s not found. Err=%d\n",filename, err); + err = request_firmware(&fw_entry, filename, &emu->pci->dev); + if (err != 0) { + snd_printk(KERN_ERR "firmware: %s not found. Err = %d\n", filename, err); return err; } - snd_printk(KERN_INFO "firmware size=0x%zx\n", fw_entry->size); + snd_printk(KERN_INFO "firmware size = 0x%zx\n", fw_entry->size); /* The FPGA is a Xilinx Spartan IIE XC2S50E */ /* GPIO7 -> FPGA PGMN @@ -685,13 +686,13 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file outl(0x80, emu->port + A_IOCFG); /* Leave bit 7 set during netlist setup. */ write_post = inl(emu->port + A_IOCFG); udelay(100); /* Allow FPGA memory to clean */ - for(n = 0; n < fw_entry->size; n++) { - value=fw_entry->data[n]; - for(i = 0; i < 8; i++) { + for (n = 0; n < fw_entry->size; n++) { + value = fw_entry->data[n]; + for (i = 0; i < 8; i++) { reg = 0x80; if (value & 0x1) reg = reg | 0x20; - value = value >> 1; + value = value >> 1; outl(reg, emu->port + A_IOCFG); write_post = inl(emu->port + A_IOCFG); outl(reg | 0x40, emu->port + A_IOCFG); @@ -703,14 +704,14 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file write_post = inl(emu->port + A_IOCFG); spin_unlock_irqrestore(&emu->emu_lock, flags); - release_firmware(fw_entry); + release_firmware(fw_entry); return 0; } static int emu1010_firmware_thread(void *data) { - struct snd_emu10k1 * emu = data; - int tmp,tmp2; + struct snd_emu10k1 *emu = data; + int tmp, tmp2; int reg; int err; @@ -719,50 +720,50 @@ static int emu1010_firmware_thread(void *data) msleep_interruptible(1000); if (kthread_should_stop()) break; - snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */ - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); /* OPTIONS: Which cards are attached to the EMU */ + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp); /* IRQ Status */ + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ®); /* OPTIONS: Which cards are attached to the EMU */ if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) { /* Audio Dock attached */ /* Return to Audio Dock programming mode */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n"); - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK ); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK); if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1010) { - if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { + err = snd_emu1010_load_firmware(emu, DOCK_FILENAME); + if (err != 0) continue; - } } else if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1010B) { - if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME); + if (err != 0) continue; - } } else if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616) { - if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME); + if (err != 0) continue; - } } - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 ); - snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, ® ); - snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0); + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, ®); + snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS = 0x%x\n", reg); /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ - snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); - snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg); + snd_emu1010_fpga_read(emu, EMU_HANA_ID, ®); + snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID = 0x%x\n", reg); if ((reg & 0x1f) != 0x15) { /* FPGA failed to be programmed */ - snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg); + snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg = 0x%x\n", reg); continue; } snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n"); - snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp ); - snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2 ); - snd_printk("Audio Dock ver:%d.%d\n",tmp ,tmp2); + snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp); + snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2); + snd_printk("Audio Dock ver:%d.%d\n", tmp, tmp2); /* Sync clocking between 1010 and Dock */ /* Allow DLL to settle */ msleep(10); /* Unmute all. Default is muted after a firmware load */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } } snd_printk(KERN_INFO "emu1010: firmware thread stopping\n"); @@ -800,10 +801,10 @@ static int emu1010_firmware_thread(void *data) * 16 x 16-bit playback - snd_emu10k1_fx8010_playback_ops * 16 x 32-bit capture - snd_emu10k1_capture_efx_ops */ -static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) +static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) { unsigned int i; - int tmp,tmp2; + int tmp, tmp2; int reg; int err; const char *filename = NULL; @@ -818,7 +819,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) * Lock Tank Memory Cache, * Mute all codecs. */ - outl(0x0005a004, emu->port + HCFG); + outl(0x0005a004, emu->port + HCFG); /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave, * Mute all codecs. */ @@ -829,25 +830,25 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) outl(0x0005a000, emu->port + HCFG); /* Disable 48Volt power to Audio Dock */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ - snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); - snd_printdd("reg1=0x%x\n",reg); + snd_emu1010_fpga_read(emu, EMU_HANA_ID, ®); + snd_printdd("reg1 = 0x%x\n", reg); if ((reg & 0x3f) == 0x15) { /* FPGA netlist already present so clear it */ /* Return to programming mode */ - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0x02 ); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0x02); } - snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); - snd_printdd("reg2=0x%x\n",reg); + snd_emu1010_fpga_read(emu, EMU_HANA_ID, ®); + snd_printdd("reg2 = 0x%x\n", reg); if ((reg & 0x3f) == 0x15) { /* FPGA failed to return to programming mode */ snd_printk(KERN_INFO "emu1010: FPGA failed to return to programming mode\n"); return -ENODEV; } - snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg); + snd_printk(KERN_INFO "emu1010: EMU_HANA_ID = 0x%x\n", reg); switch (emu->card_capabilities->emu_model) { case EMU_MODEL_EMU1010: filename = HANA_FILENAME; @@ -876,25 +877,25 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) } /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ - snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); + snd_emu1010_fpga_read(emu, EMU_HANA_ID, ®); if ((reg & 0x3f) != 0x15) { /* FPGA failed to be programmed */ - snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg=0x%x\n", reg); + snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg = 0x%x\n", reg); return -ENODEV; } snd_printk(KERN_INFO "emu1010: Hana Firmware loaded\n"); - snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp ); - snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2 ); - snd_printk("Hana ver:%d.%d\n",tmp ,tmp2); + snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp); + snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2); + snd_printk("emu1010: Hana version: %d.%d\n", tmp, tmp2); /* Enable 48Volt power to Audio Dock */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); - snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); - snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp ); + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ®); + snd_printk(KERN_INFO "emu1010: Card options = 0x%x\n", reg); + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ®); + snd_printk(KERN_INFO "emu1010: Card options = 0x%x\n", reg); + snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp); /* Optical -> ADAT I/O */ /* 0 : SPDIF * 1 : ADAT @@ -904,41 +905,42 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) tmp = 0; tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); - snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp ); - snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp ); + snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); + snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp); /* Set no attenuation on Audio Dock pads. */ - snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00 ); + snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00); emu->emu1010.adc_pads = 0x00; - snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp ); + snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp); /* Unmute Audio dock DACs, Headphone source DAC-4. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30 ); - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12 ); - snd_emu1010_fpga_read(emu, EMU_HANA_DAC_PADS, &tmp ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12); + snd_emu1010_fpga_read(emu, EMU_HANA_DAC_PADS, &tmp); /* DAC PADs. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, 0x0f ); + snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, 0x0f); emu->emu1010.dac_pads = 0x0f; - snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp ); - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30 ); - snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp ); + snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30); + snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp); /* SPDIF Format. Set Consumer mode, 24bit, copy enable */ - snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); + snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10); /* MIDI routing */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); /* Unknown. */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); - /* snd_emu1010_fpga_write(emu, 0x09, 0x0f ); // IRQ Enable: All on */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); + /* IRQ Enable: Alll on */ + /* snd_emu1010_fpga_write(emu, 0x09, 0x0f ); */ /* IRQ Enable: All off */ - snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x00 ); + snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x00); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); - snd_printk(KERN_INFO "emu1010: Card options3=0x%x\n",reg); + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ®); + snd_printk(KERN_INFO "emu1010: Card options3 = 0x%x\n", reg); /* Default WCLK set to 48kHz. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x00 ); + snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x00); /* Word Clock source, Internal 48kHz x1 */ - snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K ); - //snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X ); + snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K); + /* snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X); */ /* Audio Dock LEDs. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12 ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12); #if 0 /* For 96kHz */ @@ -992,7 +994,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) * Defaults only, users will set their own values anyways, let's * just copy/paste. */ - + snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE2_EMU32_8, EMU_SRC_DOCK_MIC_A1); snd_emu1010_fpga_link_dst_src_write(emu, @@ -1037,19 +1039,19 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE2_EMU32_F, EMU_SRC_HAMOA_ADC_LEFT2); #endif - for (i = 0;i < 0x20; i++ ) { - /* AudioDock Elink <- Silence */ - snd_emu1010_fpga_link_dst_src_write(emu, 0x0100+i, EMU_SRC_SILENCE); + for (i = 0; i < 0x20; i++) { + /* AudioDock Elink <- Silence */ + snd_emu1010_fpga_link_dst_src_write(emu, 0x0100 + i, EMU_SRC_SILENCE); } - for (i = 0;i < 4; i++) { + for (i = 0; i < 4; i++) { /* Hana SPDIF Out <- Silence */ - snd_emu1010_fpga_link_dst_src_write(emu, 0x0200+i, EMU_SRC_SILENCE); + snd_emu1010_fpga_link_dst_src_write(emu, 0x0200 + i, EMU_SRC_SILENCE); } - for (i = 0;i < 7; i++) { + for (i = 0; i < 7; i++) { /* Hamoa DAC <- Silence */ - snd_emu1010_fpga_link_dst_src_write(emu, 0x0300+i, EMU_SRC_SILENCE); + snd_emu1010_fpga_link_dst_src_write(emu, 0x0300 + i, EMU_SRC_SILENCE); } - for (i = 0;i < 7; i++) { + for (i = 0; i < 7; i++) { /* Hana ADAT Out <- Silence */ snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_HANA_ADAT + i, EMU_SRC_SILENCE); } @@ -1065,30 +1067,30 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) EMU_DST_ALICE_I2S2_LEFT, EMU_SRC_DOCK_ADC3_LEFT1); snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE_I2S2_RIGHT, EMU_SRC_DOCK_ADC3_RIGHT1); - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x01 ); // Unmute all + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x01); /* Unmute all */ + + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp ); - /* AC97 1.03, Any 32Meg of 2Gig address, Auto-Mute, EMU32 Slave, * Lock Sound Memory Cache, Lock Tank Memory Cache, * Mute all codecs. */ - outl(0x0000a000, emu->port + HCFG); + outl(0x0000a000, emu->port + HCFG); /* AC97 1.03, Any 32Meg of 2Gig address, Auto-Mute, EMU32 Slave, * Lock Sound Memory Cache, Lock Tank Memory Cache, * Un-Mute all codecs. */ outl(0x0000a001, emu->port + HCFG); - + /* Initial boot complete. Now patches */ - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp ); - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); /* MIDI Route */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); /* Unknown */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); /* MIDI Route */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); /* Unknown */ - snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp ); - snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); /* SPDIF Format spdif (or 0x11 for aes/ebu) */ + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp); + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); /* MIDI Route */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); /* Unknown */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); /* MIDI Route */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); /* Unknown */ + snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp); + snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10); /* SPDIF Format spdif (or 0x11 for aes/ebu) */ /* Start Micro/Audio Dock firmware loader thread */ if (!emu->emu1010.firmware_thread) { @@ -1218,20 +1220,20 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) emu->emu1010.output_source[23] = 28; } /* TEMP: Select SPDIF in/out */ - //snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */ + /* snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); */ /* Output spdif */ /* TEMP: Select 48kHz SPDIF out */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x0); /* Mute all */ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x0); /* Default fallback clock 48kHz */ /* Word Clock source, Internal 48kHz x1 */ - snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K ); - //snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X ); + snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K); + /* snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X); */ emu->emu1010.internal_clock = 1; /* 48000 */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12);/* Set LEDs on Audio Dock */ + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12); /* Set LEDs on Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x1); /* Unmute all */ - //snd_emu1010_fpga_write(emu, 0x7, 0x0); /* Mute all */ - //snd_emu1010_fpga_write(emu, 0x7, 0x1); /* Unmute all */ - //snd_emu1010_fpga_write(emu, 0xe, 0x12); /* Set LEDs on Audio Dock */ + /* snd_emu1010_fpga_write(emu, 0x7, 0x0); */ /* Mute all */ + /* snd_emu1010_fpga_write(emu, 0x7, 0x1); */ /* Unmute all */ + /* snd_emu1010_fpga_write(emu, 0xe, 0x12); */ /* Set LEDs on Audio Dock */ return 0; } @@ -1247,13 +1249,13 @@ static void free_pm_buffer(struct snd_emu10k1 *emu); static int snd_emu10k1_free(struct snd_emu10k1 *emu) { if (emu->port) { /* avoid access to already used hardware */ - snd_emu10k1_fx8010_tram_setup(emu, 0); + snd_emu10k1_fx8010_tram_setup(emu, 0); snd_emu10k1_done(emu); snd_emu10k1_free_efx(emu); - } + } if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1010) { /* Disable 48Volt power to Audio Dock */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); } if (emu->emu1010.firmware_thread) kthread_stop(emu->emu1010.firmware_thread); @@ -1278,7 +1280,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) #endif if (emu->port) pci_release_regions(emu->pci); - if (emu->card_capabilities->ca0151_chip) /* P16V */ + if (emu->card_capabilities->ca0151_chip) /* P16V */ snd_p16v_free(emu); pci_disable_device(emu->pci); kfree(emu); @@ -1292,21 +1294,6 @@ static int snd_emu10k1_dev_free(struct snd_device *device) } static struct snd_emu_chip_details emu_chip_details[] = { - /* Audigy 2 Value AC3 out does not work yet. Need to find out how to turn off interpolators.*/ - /* Tested by James@superbug.co.uk 3rd July 2005 */ - /* DSP: CA0108-IAT - * DAC: CS4382-KQ - * ADC: Philips 1361T - * AC97: STAC9750 - * CA0151: None - */ - {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10011102, - .driver = "Audigy2", .name = "Audigy 2 Value [SB0400]", - .id = "Audigy2", - .emu10k2_chip = 1, - .ca0108_chip = 1, - .spk71 = 1, - .ac97_chip = 1} , /* Audigy4 (Not PRO) SB0610 */ /* Tested by James@superbug.co.uk 4th April 2006 */ /* A_IOCFG bits @@ -1346,20 +1333,37 @@ static struct snd_emu_chip_details emu_chip_details[] = { * CA0151: None */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10211102, - .driver = "Audigy2", .name = "Audigy 4 [SB0610]", + .driver = "Audigy2", .name = "SB Audigy 4 [SB0610]", .id = "Audigy2", .emu10k2_chip = 1, .ca0108_chip = 1, .spk71 = 1, .adc_1361t = 1, /* 24 bit capture instead of 16bit */ .ac97_chip = 1} , + /* Audigy 2 Value AC3 out does not work yet. + * Need to find out how to turn off interpolators. + */ + /* Tested by James@superbug.co.uk 3rd July 2005 */ + /* DSP: CA0108-IAT + * DAC: CS4382-KQ + * ADC: Philips 1361T + * AC97: STAC9750 + * CA0151: None + */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10011102, + .driver = "Audigy2", .name = "SB Audigy 2 Value [SB0400]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .ac97_chip = 1} , /* Audigy 2 ZS Notebook Cardbus card.*/ /* Tested by James@superbug.co.uk 6th November 2006 */ /* Audio output 7.1/Headphones working. * Digital output working. (AC3 not checked, only PCM) * Audio Mic/Line inputs working. * Digital input not tested. - */ + */ /* DSP: Tina2 * DAC: Wolfson WM8768/WM8568 * ADC: Wolfson WM8775 @@ -1386,7 +1390,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { * */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x20011102, - .driver = "Audigy2", .name = "Audigy 2 ZS Notebook [SB0530]", + .driver = "Audigy2", .name = "SB Audigy 2 ZS Notebook [SB0530]", .id = "Audigy2", .emu10k2_chip = 1, .ca0108_chip = 1, @@ -1396,7 +1400,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spk71 = 1} , /* Tested by James@superbug.co.uk 4th Nov 2007. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x42011102, - .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]", + .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]", .id = "EMU1010", .emu10k2_chip = 1, .ca0108_chip = 1, @@ -1404,47 +1408,49 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spk71 = 1 , .emu_model = EMU_MODEL_EMU1616}, /* Tested by James@superbug.co.uk 4th Nov 2007. */ + /* This is MAEM8960, 0202 is MAEM 8980 */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102, - .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", + .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM8960]", .id = "EMU1010", .emu10k2_chip = 1, .ca0108_chip = 1, .spk71 = 1, - .emu_model = EMU_MODEL_EMU1010B}, + .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 new revision */ /* Tested by James@superbug.co.uk 8th July 2005. */ + /* This is MAEM8810, 0202 is MAEM8820 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102, - .driver = "Audigy2", .name = "E-mu 1010 [4001]", + .driver = "Audigy2", .name = "E-mu 1010 [MAEM8810]", .id = "EMU1010", .emu10k2_chip = 1, .ca0102_chip = 1, .spk71 = 1, - .emu_model = EMU_MODEL_EMU1010}, /* Emu 1010 */ + .emu_model = EMU_MODEL_EMU1010}, /* EMU 1010 old revision */ /* EMU0404b */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40021102, - .driver = "Audigy2", .name = "E-mu 0404b [4002]", + .driver = "Audigy2", .name = "E-mu 0404b PCI [MAEM8852]", .id = "EMU0404", .emu10k2_chip = 1, .ca0108_chip = 1, .spk71 = 1, - .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ + .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 new revision */ /* Tested by James@superbug.co.uk 20-3-2007. */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40021102, - .driver = "Audigy2", .name = "E-mu 0404 [4002]", + .driver = "Audigy2", .name = "E-mu 0404 [MAEM8850]", .id = "EMU0404", .emu10k2_chip = 1, .ca0102_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ - /* Audigy4 (Not PRO) SB0610 */ - {.vendor = 0x1102, .device = 0x0008, - .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]", + /* Note that all E-mu cards require kernel 2.6 or newer. */ + {.vendor = 0x1102, .device = 0x0008, + .driver = "Audigy2", .name = "SB Audigy 2 Value [Unknown]", .id = "Audigy2", .emu10k2_chip = 1, .ca0108_chip = 1, .ac97_chip = 1} , /* Tested by James@superbug.co.uk 3rd July 2005 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20071102, - .driver = "Audigy2", .name = "Audigy 4 PRO [SB0380]", + .driver = "Audigy2", .name = "SB Audigy 4 PRO [SB0380]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1457,7 +1463,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { * Just like 0x20021102 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20061102, - .driver = "Audigy2", .name = "Audigy 2 [SB0350b]", + .driver = "Audigy2", .name = "SB Audigy 2 [SB0350b]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1467,7 +1473,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20021102, - .driver = "Audigy2", .name = "Audigy 2 ZS [SB0350]", + .driver = "Audigy2", .name = "SB Audigy 2 ZS [SB0350]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1477,7 +1483,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20011102, - .driver = "Audigy2", .name = "Audigy 2 ZS [2001]", + .driver = "Audigy2", .name = "SB Audigy 2 ZS [SB0360]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1495,7 +1501,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { * CA0151: Yes */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10071102, - .driver = "Audigy2", .name = "Audigy 2 [SB0240]", + .driver = "Audigy2", .name = "SB Audigy 2 [SB0240]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1505,7 +1511,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .adc_1361t = 1, /* 24 bit capture instead of 16bit */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10051102, - .driver = "Audigy2", .name = "Audigy 2 EX [1005]", + .driver = "Audigy2", .name = "SB Audigy 2 Platinum EX [SB0280]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1515,7 +1521,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { /* Dell OEM/Creative Labs Audigy 2 ZS */ /* See ALSA bug#1365 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10031102, - .driver = "Audigy2", .name = "Audigy 2 ZS [SB0353]", + .driver = "Audigy2", .name = "SB Audigy 2 ZS [SB0353]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1524,7 +1530,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spdif_bug = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102, - .driver = "Audigy2", .name = "Audigy 2 Platinum [SB0240P]", + .driver = "Audigy2", .name = "SB Audigy 2 Platinum [SB0240P]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1535,7 +1541,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .adc_1361t = 1, /* 24 bit capture instead of 16bit. Fixes ALSA bug#324 */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .revision = 0x04, - .driver = "Audigy2", .name = "Audigy 2 [Unknown]", + .driver = "Audigy2", .name = "SB Audigy 2 [Unknown]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1543,78 +1549,79 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spdif_bug = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00531102, - .driver = "Audigy", .name = "Audigy 1 [SB0090]", + .driver = "Audigy", .name = "SB Audigy 1 [SB0092]", .id = "Audigy", .emu10k2_chip = 1, .ca0102_chip = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00521102, - .driver = "Audigy", .name = "Audigy 1 ES [SB0160]", + .driver = "Audigy", .name = "SB Audigy 1 ES [SB0160]", .id = "Audigy", .emu10k2_chip = 1, .ca0102_chip = 1, .spdif_bug = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00511102, - .driver = "Audigy", .name = "Audigy 1 [SB0090]", + .driver = "Audigy", .name = "SB Audigy 1 [SB0090]", .id = "Audigy", .emu10k2_chip = 1, .ca0102_chip = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, - .driver = "Audigy", .name = "Audigy 1 [Unknown]", + .driver = "Audigy", .name = "Audigy 1 [Unknown]", .id = "Audigy", .emu10k2_chip = 1, .ca0102_chip = 1, .ac97_chip = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806B1102, - .driver = "EMU10K1", .name = "SBLive! [SB0105]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x100a1102, + .driver = "EMU10K1", .name = "SB Live! 5.1 [SB0220]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806b1102, + .driver = "EMU10K1", .name = "SB Live! [SB0105]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806A1102, - .driver = "EMU10K1", .name = "SBLive! Value [SB0103]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806a1102, + .driver = "EMU10K1", .name = "SB Live! Value [SB0103]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80691102, - .driver = "EMU10K1", .name = "SBLive! Value [SB0101]", + .driver = "EMU10K1", .name = "SB Live! Value [SB0101]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , /* Tested by ALSA bug#1680 26th December 2005 */ - /* note: It really has SB0220 written on the card. */ + /* note: It really has SB0220 written on the card, */ + /* but it's SB0228 according to kx.inf */ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80661102, - .driver = "EMU10K1", .name = "SB Live 5.1 Dell OEM [SB0220]", + .driver = "EMU10K1", .name = "SB Live! 5.1 Dell OEM [SB0228]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , /* Tested by Thomas Zehetbauer 27th Aug 2005 */ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80651102, - .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]", - .id = "Live", - .emu10k1_chip = 1, - .ac97_chip = 1, - .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x100a1102, - .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]", + .driver = "EMU10K1", .name = "SB Live! 5.1 [SB0220]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80641102, - .driver = "EMU10K1", .name = "SB Live 5.1", + .driver = "EMU10K1", .name = "SB Live! 5.1", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , /* Tested by alsa bugtrack user "hus" bug #1297 12th Aug 2005 */ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80611102, - .driver = "EMU10K1", .name = "SBLive 5.1 [SB0060]", + .driver = "EMU10K1", .name = "SB Live! 5.1 [SB0060]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 2, /* ac97 is optional; both SBLive 5.1 and platinum @@ -1622,78 +1629,78 @@ static struct snd_emu_chip_details emu_chip_details[] = { */ .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80511102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4850]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4850]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80401102, - .driver = "EMU10K1", .name = "SBLive! Platinum [CT4760P]", + .driver = "EMU10K1", .name = "SB Live! Platinum [CT4760P]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80321102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4871]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4871]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80311102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4831]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4831]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80281102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4870]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4870]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , /* Tested by James@superbug.co.uk 3rd July 2005 */ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80271102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4832]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4832]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80261102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4830]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4830]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80231102, - .driver = "EMU10K1", .name = "SB PCI512 [CT4790]", + .driver = "EMU10K1", .name = "SB PCI512 [CT4790]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80221102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4780]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4780]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x40011102, - .driver = "EMU10K1", .name = "E-mu APS [4001]", + .driver = "EMU10K1", .name = "E-mu APS [PC545]", .id = "APS", .emu10k1_chip = 1, .ecard = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00211102, - .driver = "EMU10K1", .name = "SBLive! [CT4620]", + .driver = "EMU10K1", .name = "SB Live! [CT4620]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00201102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4670]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4670]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, - .driver = "EMU10K1", .name = "SB Live [Unknown]", + .driver = "EMU10K1", .name = "SB Live! [Unknown]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, @@ -1702,13 +1709,13 @@ static struct snd_emu_chip_details emu_chip_details[] = { }; int __devinit snd_emu10k1_create(struct snd_card *card, - struct pci_dev * pci, + struct pci_dev *pci, unsigned short extin_mask, unsigned short extout_mask, long max_cache_bytes, int enable_ir, uint subsystem, - struct snd_emu10k1 ** remu) + struct snd_emu10k1 **remu) { struct snd_emu10k1 *emu; int idx, err; @@ -1718,11 +1725,12 @@ int __devinit snd_emu10k1_create(struct snd_card *card, static struct snd_device_ops ops = { .dev_free = snd_emu10k1_dev_free, }; - + *remu = NULL; /* enable PCI device */ - if ((err = pci_enable_device(pci)) < 0) + err = pci_enable_device(pci); + if (err < 0) return err; emu = kzalloc(sizeof(*emu), GFP_KERNEL); @@ -1749,16 +1757,17 @@ int __devinit snd_emu10k1_create(struct snd_card *card, emu->revision = pci->revision; pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &emu->serial); pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &emu->model); - snd_printdd("vendor=0x%x, device=0x%x, subsystem_vendor_id=0x%x, subsystem_id=0x%x\n",pci->vendor, pci->device, emu->serial, emu->model); + snd_printdd("vendor = 0x%x, device = 0x%x, subsystem_vendor_id = 0x%x, subsystem_id = 0x%x\n", pci->vendor, pci->device, emu->serial, emu->model); for (c = emu_chip_details; c->vendor; c++) { if (c->vendor == pci->vendor && c->device == pci->device) { if (subsystem) { - if (c->subsystem && (c->subsystem == subsystem) ) { + if (c->subsystem && (c->subsystem == subsystem)) break; - } else continue; + else + continue; } else { - if (c->subsystem && (c->subsystem != emu->serial) ) + if (c->subsystem && (c->subsystem != emu->serial)) continue; if (c->revision && c->revision != emu->revision) continue; @@ -1774,14 +1783,18 @@ int __devinit snd_emu10k1_create(struct snd_card *card, } emu->card_capabilities = c; if (c->subsystem && !subsystem) - snd_printdd("Sound card name=%s\n", c->name); - else if (subsystem) - snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x. Forced to subsytem=0x%x\n", - c->name, pci->vendor, pci->device, emu->serial, c->subsystem); - else - snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x.\n", - c->name, pci->vendor, pci->device, emu->serial); - + snd_printdd("Sound card name = %s\n", c->name); + else if (subsystem) + snd_printdd("Sound card name = %s, " + "vendor = 0x%x, device = 0x%x, subsystem = 0x%x. " + "Forced to subsytem = 0x%x\n", c->name, + pci->vendor, pci->device, emu->serial, c->subsystem); + else + snd_printdd("Sound card name = %s, " + "vendor = 0x%x, device = 0x%x, subsystem = 0x%x.\n", + c->name, pci->vendor, pci->device, + emu->serial); + if (!*card->id && c->id) { int i, n = 0; strlcpy(card->id, c->id, sizeof(card->id)); @@ -1815,7 +1828,8 @@ int __devinit snd_emu10k1_create(struct snd_card *card, else emu->gpr_base = FXGPREGBASE; - if ((err = pci_request_regions(pci, "EMU10K1")) < 0) { + err = pci_request_regions(pci, "EMU10K1"); + if (err < 0) { kfree(emu); pci_disable_device(pci); return err; @@ -1862,21 +1876,25 @@ int __devinit snd_emu10k1_create(struct snd_card *card, emu->enable_ir = enable_ir; if (emu->card_capabilities->ca_cardbus_chip) { - if ((err = snd_emu10k1_cardbus_init(emu)) < 0) + err = snd_emu10k1_cardbus_init(emu); + if (err < 0) goto error; } if (emu->card_capabilities->ecard) { - if ((err = snd_emu10k1_ecard_init(emu)) < 0) + err = snd_emu10k1_ecard_init(emu); + if (err < 0) goto error; } else if (emu->card_capabilities->emu_model) { - if ((err = snd_emu10k1_emu1010_init(emu)) < 0) { - snd_emu10k1_free(emu); - return err; - } + err = snd_emu10k1_emu1010_init(emu); + if (err < 0) { + snd_emu10k1_free(emu); + return err; + } } else { /* 5.1: Enable the additional AC97 Slots. If the emu10k1 version does not support this, it shouldn't do any harm */ - snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE); + snd_emu10k1_ptr_write(emu, AC97SLOT, 0, + AC97SLOT_CNTR|AC97SLOT_LFE); } /* initialize TRAM setup */ @@ -1916,7 +1934,7 @@ int __devinit snd_emu10k1_create(struct snd_card *card, snd_emu10k1_synth_alloc(emu, 4096); if (emu->reserved_page) emu->reserved_page->map_locked = 1; - + /* Clear silent pages and set up pointers */ memset(emu->silent_page.area, 0, PAGE_SIZE); silent_page = emu->silent_page.addr << 1; @@ -1929,19 +1947,23 @@ int __devinit snd_emu10k1_create(struct snd_card *card, emu->voices[idx].number = idx; } - if ((err = snd_emu10k1_init(emu, enable_ir, 0)) < 0) + err = snd_emu10k1_init(emu, enable_ir, 0); + if (err < 0) goto error; #ifdef CONFIG_PM - if ((err = alloc_pm_buffer(emu)) < 0) + err = alloc_pm_buffer(emu); + if (err < 0) goto error; #endif /* Initialize the effect engine */ - if ((err = snd_emu10k1_init_efx(emu)) < 0) + err = snd_emu10k1_init_efx(emu); + if (err < 0) goto error; snd_emu10k1_audio_enable(emu); - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, emu, &ops)) < 0) + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, emu, &ops); + if (err < 0) goto error; #ifdef CONFIG_PROC_FS @@ -1981,7 +2003,7 @@ static int __devinit alloc_pm_buffer(struct snd_emu10k1 *emu) if (emu->audigy) size += ARRAY_SIZE(saved_regs_audigy); emu->saved_ptr = vmalloc(4 * NUM_G * size); - if (! emu->saved_ptr) + if (!emu->saved_ptr) return -ENOMEM; if (snd_emu10k1_efx_alloc_pm_buffer(emu) < 0) return -ENOMEM; @@ -2026,7 +2048,7 @@ void snd_emu10k1_resume_init(struct snd_emu10k1 *emu) if (emu->card_capabilities->ecard) snd_emu10k1_ecard_init(emu); else if (emu->card_capabilities->emu_model) - snd_emu10k1_emu1010_init(emu); + snd_emu10k1_emu1010_init(emu); else snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE); snd_emu10k1_init(emu, emu->enable_ir, 1); diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index f34bbfb705f5..b0fb6c917c38 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1639,6 +1639,45 @@ static struct snd_kcontrol_new snd_audigy_shared_spdif __devinitdata = .put = snd_emu10k1_shared_spdif_put }; +/* workaround for too low volume on Audigy due to 16bit/24bit conversion */ + +#define snd_audigy_capture_boost_info snd_ctl_boolean_mono_info + +static int snd_audigy_capture_boost_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + + /* FIXME: better to use a cached version */ + val = snd_ac97_read(emu->ac97, AC97_REC_GAIN); + ucontrol->value.integer.value[0] = !!val; + return 0; +} + +static int snd_audigy_capture_boost_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + + if (ucontrol->value.integer.value[0]) + val = 0x0f0f; + else + val = 0; + return snd_ac97_update(emu->ac97, AC97_REC_GAIN, val); +} + +static struct snd_kcontrol_new snd_audigy_capture_boost __devinitdata = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Capture Boost", + .info = snd_audigy_capture_boost_info, + .get = snd_audigy_capture_boost_get, + .put = snd_audigy_capture_boost_put +}; + + /* */ static void snd_emu10k1_mixer_free_ac97(struct snd_ac97 *ac97) @@ -2087,5 +2126,12 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, } } + if (emu->card_capabilities->ac97_chip && emu->audigy) { + err = snd_ctl_add(card, snd_ctl_new1(&snd_audigy_capture_boost, + emu)); + if (err < 0) + return err; + } + return 0; } diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 20ee7599600b..e9c3794bbcb8 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -1953,7 +1953,7 @@ static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id) outw(inw(chip->io_port + 4) & 1, chip->io_port + 4); if (event & ESM_HWVOL_IRQ) - tasklet_hi_schedule(&chip->hwvol_tq); /* we'll do this later */ + tasklet_schedule(&chip->hwvol_tq); /* we'll do this later */ /* else ack 'em all, i imagine */ outb(0xFF, chip->io_port + 0x1A); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig new file mode 100644 index 000000000000..eb2a19b894a0 --- /dev/null +++ b/sound/pci/hda/Kconfig @@ -0,0 +1,188 @@ +menuconfig SND_HDA_INTEL + tristate "Intel HD Audio" + select SND_PCM + select SND_VMASTER + select SND_JACK if INPUT=y || INPUT=SND + help + Say Y here to include support for Intel "High Definition + Audio" (Azalia) and its compatible devices. + + This option enables the HD-audio controller. Don't forget + to choose the appropriate codec options below. + + To compile this driver as a module, choose M here: the module + will be called snd-hda-intel. + +if SND_HDA_INTEL + +config SND_HDA_HWDEP + bool "Build hwdep interface for HD-audio driver" + select SND_HWDEP + help + Say Y here to build a hwdep interface for HD-audio driver. + This interface can be used for out-of-band communication + with codecs for debugging purposes. + +config SND_HDA_RECONFIG + bool "Allow dynamic codec reconfiguration (EXPERIMENTAL)" + depends on SND_HDA_HWDEP && EXPERIMENTAL + help + Say Y here to enable the HD-audio codec re-configuration feature. + This adds the sysfs interfaces to allow user to clear the whole + codec configuration, change the codec setup, add extra verbs, + and re-configure the codec dynamically. + +config SND_HDA_INPUT_BEEP + bool "Support digital beep via input layer" + depends on INPUT=y || INPUT=SND_HDA_INTEL + help + Say Y here to build a digital beep interface for HD-audio + driver. This interface is used to generate digital beeps. + +config SND_HDA_CODEC_REALTEK + bool "Build Realtek HD-audio codec support" + default y + help + Say Y here to include Realtek HD-audio codec support in + snd-hda-intel driver, such as ALC880. + + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-realtek. + This module is automatically loaded at probing. + +config SND_HDA_CODEC_ANALOG + bool "Build Analog Device HD-audio codec support" + default y + help + Say Y here to include Analog Device HD-audio codec support in + snd-hda-intel driver, such as AD1986A. + + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-analog. + This module is automatically loaded at probing. + +config SND_HDA_CODEC_SIGMATEL + bool "Build IDT/Sigmatel HD-audio codec support" + default y + help + Say Y here to include IDT (Sigmatel) HD-audio codec support in + snd-hda-intel driver, such as STAC9200. + + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-idt. + This module is automatically loaded at probing. + +config SND_HDA_CODEC_VIA + bool "Build VIA HD-audio codec support" + default y + help + Say Y here to include VIA HD-audio codec support in + snd-hda-intel driver, such as VT1708. + + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-via. + This module is automatically loaded at probing. + +config SND_HDA_CODEC_ATIHDMI + bool "Build ATI HDMI HD-audio codec support" + default y + help + Say Y here to include ATI HDMI HD-audio codec support in + snd-hda-intel driver, such as ATI RS600 HDMI. + + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-atihdmi. + This module is automatically loaded at probing. + +config SND_HDA_CODEC_NVHDMI + bool "Build NVIDIA HDMI HD-audio codec support" + default y + help + Say Y here to include NVIDIA HDMI HD-audio codec support in + snd-hda-intel driver, such as NVIDIA MCP78 HDMI. + + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-nvhdmi. + This module is automatically loaded at probing. + +config SND_HDA_CODEC_INTELHDMI + bool "Build INTEL HDMI HD-audio codec support" + default y + help + Say Y here to include INTEL HDMI HD-audio codec support in + snd-hda-intel driver, such as Eaglelake integrated HDMI. + + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-intelhdmi. + This module is automatically loaded at probing. + +config SND_HDA_ELD + def_bool y + depends on SND_HDA_CODEC_INTELHDMI + +config SND_HDA_CODEC_CONEXANT + bool "Build Conexant HD-audio codec support" + default y + help + Say Y here to include Conexant HD-audio codec support in + snd-hda-intel driver, such as CX20549. + + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-conexant. + This module is automatically loaded at probing. + +config SND_HDA_CODEC_CMEDIA + bool "Build C-Media HD-audio codec support" + default y + help + Say Y here to include C-Media HD-audio codec support in + snd-hda-intel driver, such as CMI9880. + + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-cmedia. + This module is automatically loaded at probing. + +config SND_HDA_CODEC_SI3054 + bool "Build Silicon Labs 3054 HD-modem codec support" + default y + help + Say Y here to include Silicon Labs 3054 HD-modem codec + (and compatibles) support in snd-hda-intel driver. + + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-si3054. + This module is automatically loaded at probing. + +config SND_HDA_GENERIC + bool "Enable generic HD-audio codec parser" + default y + help + Say Y here to enable the generic HD-audio codec parser + in snd-hda-intel driver. + +config SND_HDA_POWER_SAVE + bool "Aggressive power-saving on HD-audio" + help + Say Y here to enable more aggressive power-saving mode on + HD-audio driver. The power-saving timeout can be configured + via power_save option or over sysfs on-the-fly. + +config SND_HDA_POWER_SAVE_DEFAULT + int "Default time-out for HD-audio power-save mode" + depends on SND_HDA_POWER_SAVE + default 0 + help + The default time-out value in seconds for HD-audio automatic + power-save mode. 0 means to disable the power-save mode. + +endif diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 1980c6d207e7..50f9d0967251 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,20 +1,59 @@ -snd-hda-intel-y := hda_intel.o -# since snd-hda-intel is the only driver using hda-codec, -# merge it into a single module although it was originally -# designed to be individual modules -snd-hda-intel-y += hda_codec.o -snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o -snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o -snd-hda-intel-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o -snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_REALTEK) += patch_realtek.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CMEDIA) += patch_cmedia.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ANALOG) += patch_analog.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SIGMATEL) += patch_sigmatel.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SI3054) += patch_si3054.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ATIHDMI) += patch_atihdmi.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CONEXANT) += patch_conexant.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_VIA) += patch_via.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_NVHDMI) += patch_nvhdmi.o +snd-hda-intel-objs := hda_intel.o +snd-hda-codec-y := hda_codec.o +snd-hda-codec-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o +snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o +# snd-hda-codec-$(CONFIG_SND_HDA_ELD) += hda_eld.o +snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o +snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o + +snd-hda-codec-realtek-objs := patch_realtek.o +snd-hda-codec-cmedia-objs := patch_cmedia.o +snd-hda-codec-analog-objs := patch_analog.o +snd-hda-codec-idt-objs := patch_sigmatel.o +snd-hda-codec-si3054-objs := patch_si3054.o +snd-hda-codec-atihdmi-objs := patch_atihdmi.o +snd-hda-codec-conexant-objs := patch_conexant.o +snd-hda-codec-via-objs := patch_via.o +snd-hda-codec-nvhdmi-objs := patch_nvhdmi.o +snd-hda-codec-intelhdmi-objs := patch_intelhdmi.o hda_eld.o + +# common driver +obj-$(CONFIG_SND_HDA_INTEL) := snd-hda-codec.o + +# codec drivers (note: CONFIG_SND_HDA_CODEC_XXX are booleans) +ifdef CONFIG_SND_HDA_CODEC_REALTEK +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-realtek.o +endif +ifdef CONFIG_SND_HDA_CODEC_CMEDIA +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cmedia.o +endif +ifdef CONFIG_SND_HDA_CODEC_ANALOG +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-analog.o +endif +ifdef CONFIG_SND_HDA_CODEC_SIGMATEL +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-idt.o +endif +ifdef CONFIG_SND_HDA_CODEC_SI3054 +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-si3054.o +endif +ifdef CONFIG_SND_HDA_CODEC_ATIHDMI +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-atihdmi.o +endif +ifdef CONFIG_SND_HDA_CODEC_CONEXANT +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o +endif +ifdef CONFIG_SND_HDA_CODEC_VIA +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-via.o +endif +ifdef CONFIG_SND_HDA_CODEC_NVHDMI +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-nvhdmi.o +endif +ifdef CONFIG_SND_HDA_CODEC_INTELHDMI +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-intelhdmi.o +endif + +# this must be the last entry after codec drivers; +# otherwise the codec patches won't be hooked before the PCI probe +# when built in kernel obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 3ecd7e797dee..e00421c0d8ba 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -128,6 +128,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) INIT_WORK(&beep->beep_work, &snd_hda_generate_beep); return 0; } +EXPORT_SYMBOL_HDA(snd_hda_attach_beep_device); void snd_hda_detach_beep_device(struct hda_codec *codec) { @@ -140,3 +141,4 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) kfree(beep); } } +EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index eb9164176dab..e16cf63821ae 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -31,15 +31,6 @@ #include <sound/initval.h> #include "hda_local.h" #include <sound/hda_hwdep.h> -#include "hda_patch.h" /* codec presets */ - -#ifdef CONFIG_SND_HDA_POWER_SAVE -/* define this option here to hide as static */ -static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; -module_param(power_save, int, 0644); -MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " - "(in second, 0 = disable)."); -#endif /* * vendor / preset table @@ -55,6 +46,7 @@ static struct hda_vendor_id hda_vendor_ids[] = { { 0x1002, "ATI" }, { 0x1057, "Motorola" }, { 0x1095, "Silicon Image" }, + { 0x10de, "Nvidia" }, { 0x10ec, "Realtek" }, { 0x1106, "VIA" }, { 0x111d, "IDT" }, @@ -66,40 +58,31 @@ static struct hda_vendor_id hda_vendor_ids[] = { { 0x1854, "LG" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x434d, "C-Media" }, + { 0x8086, "Intel" }, { 0x8384, "SigmaTel" }, {} /* terminator */ }; -static const struct hda_codec_preset *hda_preset_tables[] = { -#ifdef CONFIG_SND_HDA_CODEC_REALTEK - snd_hda_preset_realtek, -#endif -#ifdef CONFIG_SND_HDA_CODEC_CMEDIA - snd_hda_preset_cmedia, -#endif -#ifdef CONFIG_SND_HDA_CODEC_ANALOG - snd_hda_preset_analog, -#endif -#ifdef CONFIG_SND_HDA_CODEC_SIGMATEL - snd_hda_preset_sigmatel, -#endif -#ifdef CONFIG_SND_HDA_CODEC_SI3054 - snd_hda_preset_si3054, -#endif -#ifdef CONFIG_SND_HDA_CODEC_ATIHDMI - snd_hda_preset_atihdmi, -#endif -#ifdef CONFIG_SND_HDA_CODEC_CONEXANT - snd_hda_preset_conexant, -#endif -#ifdef CONFIG_SND_HDA_CODEC_VIA - snd_hda_preset_via, -#endif -#ifdef CONFIG_SND_HDA_CODEC_NVHDMI - snd_hda_preset_nvhdmi, -#endif - NULL -}; +static DEFINE_MUTEX(preset_mutex); +static LIST_HEAD(hda_preset_tables); + +int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset) +{ + mutex_lock(&preset_mutex); + list_add_tail(&preset->list, &hda_preset_tables); + mutex_unlock(&preset_mutex); + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_add_codec_preset); + +int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset) +{ + mutex_lock(&preset_mutex); + list_del(&preset->list); + mutex_unlock(&preset_mutex); + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset); #ifdef CONFIG_SND_HDA_POWER_SAVE static void hda_power_work(struct work_struct *work); @@ -108,6 +91,72 @@ static void hda_keep_power_on(struct hda_codec *codec); static inline void hda_keep_power_on(struct hda_codec *codec) {} #endif +const char *snd_hda_get_jack_location(u32 cfg) +{ + static char *bases[7] = { + "N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom", + }; + static unsigned char specials_idx[] = { + 0x07, 0x08, + 0x17, 0x18, 0x19, + 0x37, 0x38 + }; + static char *specials[] = { + "Rear Panel", "Drive Bar", + "Riser", "HDMI", "ATAPI", + "Mobile-In", "Mobile-Out" + }; + int i; + cfg = (cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT; + if ((cfg & 0x0f) < 7) + return bases[cfg & 0x0f]; + for (i = 0; i < ARRAY_SIZE(specials_idx); i++) { + if (cfg == specials_idx[i]) + return specials[i]; + } + return "UNKNOWN"; +} +EXPORT_SYMBOL_HDA(snd_hda_get_jack_location); + +const char *snd_hda_get_jack_connectivity(u32 cfg) +{ + static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; + + return jack_locations[(cfg >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3]; +} +EXPORT_SYMBOL_HDA(snd_hda_get_jack_connectivity); + +const char *snd_hda_get_jack_type(u32 cfg) +{ + static char *jack_types[16] = { + "Line Out", "Speaker", "HP Out", "CD", + "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", + "Line In", "Aux", "Mic", "Telephony", + "SPDIF In", "Digitial In", "Reserved", "Other" + }; + + return jack_types[(cfg & AC_DEFCFG_DEVICE) + >> AC_DEFCFG_DEVICE_SHIFT]; +} +EXPORT_SYMBOL_HDA(snd_hda_get_jack_type); + +/* + * Compose a 32bit command word to be sent to the HD-audio controller + */ +static inline unsigned int +make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, + unsigned int verb, unsigned int parm) +{ + u32 val; + + val = (u32)(codec->addr & 0x0f) << 28; + val |= (u32)direct << 27; + val |= (u32)nid << 20; + val |= verb << 8; + val |= parm; + return val; +} + /** * snd_hda_codec_read - send a command and get the response * @codec: the HDA codec @@ -124,17 +173,21 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm) { + struct hda_bus *bus = codec->bus; unsigned int res; + + res = make_codec_cmd(codec, nid, direct, verb, parm); snd_hda_power_up(codec); - mutex_lock(&codec->bus->cmd_mutex); - if (!codec->bus->ops.command(codec, nid, direct, verb, parm)) - res = codec->bus->ops.get_response(codec); + mutex_lock(&bus->cmd_mutex); + if (!bus->ops.command(bus, res)) + res = bus->ops.get_response(bus); else res = (unsigned int)-1; - mutex_unlock(&codec->bus->cmd_mutex); + mutex_unlock(&bus->cmd_mutex); snd_hda_power_down(codec); return res; } +EXPORT_SYMBOL_HDA(snd_hda_codec_read); /** * snd_hda_codec_write - send a single command without waiting for response @@ -151,14 +204,19 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm) { + struct hda_bus *bus = codec->bus; + unsigned int res; int err; + + res = make_codec_cmd(codec, nid, direct, verb, parm); snd_hda_power_up(codec); - mutex_lock(&codec->bus->cmd_mutex); - err = codec->bus->ops.command(codec, nid, direct, verb, parm); - mutex_unlock(&codec->bus->cmd_mutex); + mutex_lock(&bus->cmd_mutex); + err = bus->ops.command(bus, res); + mutex_unlock(&bus->cmd_mutex); snd_hda_power_down(codec); return err; } +EXPORT_SYMBOL_HDA(snd_hda_codec_write); /** * snd_hda_sequence_write - sequence writes @@ -173,6 +231,7 @@ void snd_hda_sequence_write(struct hda_codec *codec, const struct hda_verb *seq) for (; seq->nid; seq++) snd_hda_codec_write(codec, seq->nid, 0, seq->verb, seq->param); } +EXPORT_SYMBOL_HDA(snd_hda_sequence_write); /** * snd_hda_get_sub_nodes - get the range of sub nodes @@ -194,6 +253,7 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, *start_id = (parm >> 16) & 0x7fff; return (int)(parm & 0x7fff); } +EXPORT_SYMBOL_HDA(snd_hda_get_sub_nodes); /** * snd_hda_get_connections - get connection list @@ -282,6 +342,7 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, } return conns; } +EXPORT_SYMBOL_HDA(snd_hda_get_connections); /** @@ -316,6 +377,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) return 0; } +EXPORT_SYMBOL_HDA(snd_hda_queue_unsol_event); /* * process queued unsolicited events @@ -345,7 +407,7 @@ static void process_unsol_events(struct work_struct *work) /* * initialize unsolicited queue */ -static int __devinit init_unsol_queue(struct hda_bus *bus) +static int init_unsol_queue(struct hda_bus *bus) { struct hda_bus_unsolicited *unsol; @@ -391,9 +453,24 @@ static int snd_hda_bus_free(struct hda_bus *bus) static int snd_hda_bus_dev_free(struct snd_device *device) { struct hda_bus *bus = device->device_data; + bus->shutdown = 1; return snd_hda_bus_free(bus); } +#ifdef CONFIG_SND_HDA_HWDEP +static int snd_hda_bus_dev_register(struct snd_device *device) +{ + struct hda_bus *bus = device->device_data; + struct hda_codec *codec; + list_for_each_entry(codec, &bus->codec_list, list) { + snd_hda_hwdep_add_sysfs(codec); + } + return 0; +} +#else +#define snd_hda_bus_dev_register NULL +#endif + /** * snd_hda_bus_new - create a HDA bus * @card: the card entry @@ -402,13 +479,14 @@ static int snd_hda_bus_dev_free(struct snd_device *device) * * Returns 0 if successful, or a negative error code. */ -int __devinit snd_hda_bus_new(struct snd_card *card, +int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, struct hda_bus **busp) { struct hda_bus *bus; int err; static struct snd_device_ops dev_ops = { + .dev_register = snd_hda_bus_dev_register, .dev_free = snd_hda_bus_dev_free, }; @@ -430,6 +508,7 @@ int __devinit snd_hda_bus_new(struct snd_card *card, bus->private_data = temp->private_data; bus->pci = temp->pci; bus->modelname = temp->modelname; + bus->power_save = temp->power_save; bus->ops = temp->ops; mutex_init(&bus->cmd_mutex); @@ -444,27 +523,42 @@ int __devinit snd_hda_bus_new(struct snd_card *card, *busp = bus; return 0; } +EXPORT_SYMBOL_HDA(snd_hda_bus_new); #ifdef CONFIG_SND_HDA_GENERIC #define is_generic_config(codec) \ - (codec->bus->modelname && !strcmp(codec->bus->modelname, "generic")) + (codec->modelname && !strcmp(codec->modelname, "generic")) #else #define is_generic_config(codec) 0 #endif +#ifdef MODULE +#define HDA_MODREQ_MAX_COUNT 2 /* two request_modules()'s */ +#else +#define HDA_MODREQ_MAX_COUNT 0 /* all presets are statically linked */ +#endif + /* * find a matching codec preset */ -static const struct hda_codec_preset __devinit * +static const struct hda_codec_preset * find_codec_preset(struct hda_codec *codec) { - const struct hda_codec_preset **tbl, *preset; + struct hda_codec_preset_list *tbl; + const struct hda_codec_preset *preset; + int mod_requested = 0; if (is_generic_config(codec)) return NULL; /* use the generic parser */ - for (tbl = hda_preset_tables; *tbl; tbl++) { - for (preset = *tbl; preset->id; preset++) { + again: + mutex_lock(&preset_mutex); + list_for_each_entry(tbl, &hda_preset_tables, list) { + if (!try_module_get(tbl->owner)) { + snd_printk(KERN_ERR "hda_codec: cannot module_get\n"); + continue; + } + for (preset = tbl->preset; preset->id; preset++) { u32 mask = preset->mask; if (preset->afg && preset->afg != codec->afg) continue; @@ -474,23 +568,40 @@ find_codec_preset(struct hda_codec *codec) mask = ~0; if (preset->id == (codec->vendor_id & mask) && (!preset->rev || - preset->rev == codec->revision_id)) + preset->rev == codec->revision_id)) { + mutex_unlock(&preset_mutex); + codec->owner = tbl->owner; return preset; + } } + module_put(tbl->owner); + } + mutex_unlock(&preset_mutex); + + if (mod_requested < HDA_MODREQ_MAX_COUNT) { + char name[32]; + if (!mod_requested) + snprintf(name, sizeof(name), "snd-hda-codec-id:%08x", + codec->vendor_id); + else + snprintf(name, sizeof(name), "snd-hda-codec-id:%04x*", + (codec->vendor_id >> 16) & 0xffff); + request_module(name); + mod_requested++; + goto again; } return NULL; } /* - * snd_hda_get_codec_name - store the codec name + * get_codec_name - store the codec name */ -void snd_hda_get_codec_name(struct hda_codec *codec, - char *name, int namelen) +static int get_codec_name(struct hda_codec *codec) { const struct hda_vendor_id *c; const char *vendor = NULL; u16 vendor_id = codec->vendor_id >> 16; - char tmp[16]; + char tmp[16], name[32]; for (c = hda_vendor_ids; c->id; c++) { if (c->id == vendor_id) { @@ -503,16 +614,21 @@ void snd_hda_get_codec_name(struct hda_codec *codec, vendor = tmp; } if (codec->preset && codec->preset->name) - snprintf(name, namelen, "%s %s", vendor, codec->preset->name); + snprintf(name, sizeof(name), "%s %s", vendor, + codec->preset->name); else - snprintf(name, namelen, "%s ID %x", vendor, + snprintf(name, sizeof(name), "%s ID %x", vendor, codec->vendor_id & 0xffff); + codec->name = kstrdup(name, GFP_KERNEL); + if (!codec->name) + return -ENOMEM; + return 0; } /* * look for an AFG and MFG nodes */ -static void __devinit setup_fg_nodes(struct hda_codec *codec) +static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec) { int i, total_nodes; hda_nid_t nid; @@ -571,11 +687,15 @@ static void snd_hda_codec_free(struct hda_codec *codec) flush_scheduled_work(); #endif list_del(&codec->list); + snd_array_free(&codec->mixers); codec->bus->caddr_tbl[codec->addr] = NULL; if (codec->patch_ops.free) codec->patch_ops.free(codec); + module_put(codec->owner); free_hda_cache(&codec->amp_cache); free_hda_cache(&codec->cmd_cache); + kfree(codec->name); + kfree(codec->modelname); kfree(codec->wcaps); kfree(codec); } @@ -588,8 +708,8 @@ static void snd_hda_codec_free(struct hda_codec *codec) * * Returns 0 if successful, or a negative error code. */ -int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - struct hda_codec **codecp) +int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, + int do_init, struct hda_codec **codecp) { struct hda_codec *codec; char component[31]; @@ -617,6 +737,14 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, mutex_init(&codec->spdif_mutex); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); + snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32); + if (codec->bus->modelname) { + codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); + if (!codec->modelname) { + snd_hda_codec_free(codec); + return -ENODEV; + } + } #ifdef CONFIG_SND_HDA_POWER_SAVE INIT_DELAYED_WORK(&codec->power_work, hda_power_work); @@ -662,12 +790,44 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_SUBSYSTEM_ID, 0); } + if (bus->modelname) + codec->modelname = kstrdup(bus->modelname, GFP_KERNEL); + + if (do_init) { + err = snd_hda_codec_configure(codec); + if (err < 0) { + snd_hda_codec_free(codec); + return err; + } + } + snd_hda_codec_proc_new(codec); + + snd_hda_create_hwdep(codec); + + sprintf(component, "HDA:%08x,%08x,%08x", codec->vendor_id, + codec->subsystem_id, codec->revision_id); + snd_component_add(codec->bus->card, component); + + if (codecp) + *codecp = codec; + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_codec_new); + +int snd_hda_codec_configure(struct hda_codec *codec) +{ + int err; codec->preset = find_codec_preset(codec); + if (!codec->name) { + err = get_codec_name(codec); + if (err < 0) + return err; + } /* audio codec should override the mixer name */ - if (codec->afg || !*bus->card->mixername) - snd_hda_get_codec_name(codec, bus->card->mixername, - sizeof(bus->card->mixername)); + if (codec->afg || !*codec->bus->card->mixername) + strlcpy(codec->bus->card->mixername, codec->name, + sizeof(codec->bus->card->mixername)); if (is_generic_config(codec)) { err = snd_hda_parse_generic_codec(codec); @@ -684,25 +844,9 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, printk(KERN_ERR "hda-codec: No codec parser is available\n"); patched: - if (err < 0) { - snd_hda_codec_free(codec); - return err; - } - - if (codec->patch_ops.unsol_event) - init_unsol_queue(bus); - - snd_hda_codec_proc_new(codec); -#ifdef CONFIG_SND_HDA_HWDEP - snd_hda_create_hwdep(codec); -#endif - - sprintf(component, "HDA:%08x,%08x,%08x", codec->vendor_id, codec->subsystem_id, codec->revision_id); - snd_component_add(codec->bus->card, component); - - if (codecp) - *codecp = codec; - return 0; + if (!err && codec->patch_ops.unsol_event) + err = init_unsol_queue(codec->bus); + return err; } /** @@ -728,6 +872,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, msleep(1); snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, format); } +EXPORT_SYMBOL_HDA(snd_hda_codec_setup_stream); void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) { @@ -741,6 +886,7 @@ void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); #endif } +EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream); /* * amp access functions @@ -752,17 +898,17 @@ void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) #define INFO_AMP_VOL(ch) (1 << (1 + (ch))) /* initialize the hash table */ -static void __devinit init_hda_cache(struct hda_cache_rec *cache, +static void /*__devinit*/ init_hda_cache(struct hda_cache_rec *cache, unsigned int record_size) { memset(cache, 0, sizeof(*cache)); memset(cache->hash, 0xff, sizeof(cache->hash)); - cache->record_size = record_size; + snd_array_init(&cache->buf, record_size, 64); } static void free_hda_cache(struct hda_cache_rec *cache) { - kfree(cache->buffer); + snd_array_free(&cache->buf); } /* query the hash. allocate an entry if not found. */ @@ -774,35 +920,17 @@ static struct hda_cache_head *get_alloc_hash(struct hda_cache_rec *cache, struct hda_cache_head *info; while (cur != 0xffff) { - info = (struct hda_cache_head *)(cache->buffer + - cur * cache->record_size); + info = snd_array_elem(&cache->buf, cur); if (info->key == key) return info; cur = info->next; } /* add a new hash entry */ - if (cache->num_entries >= cache->size) { - /* reallocate the array */ - unsigned int new_size = cache->size + 64; - void *new_buffer; - new_buffer = kcalloc(new_size, cache->record_size, GFP_KERNEL); - if (!new_buffer) { - snd_printk(KERN_ERR "hda_codec: " - "can't malloc amp_info\n"); - return NULL; - } - if (cache->buffer) { - memcpy(new_buffer, cache->buffer, - cache->size * cache->record_size); - kfree(cache->buffer); - } - cache->size = new_size; - cache->buffer = new_buffer; - } - cur = cache->num_entries++; - info = (struct hda_cache_head *)(cache->buffer + - cur * cache->record_size); + info = snd_array_new(&cache->buf); + if (!info) + return NULL; + cur = snd_array_index(&cache->buf, info); info->key = key; info->val = 0; info->next = cache->hash[idx]; @@ -840,6 +968,7 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) } return info->amp_caps; } +EXPORT_SYMBOL_HDA(query_amp_caps); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps) @@ -853,6 +982,7 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, info->head.val |= INFO_AMP_CAPS; return 0; } +EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps); /* * read the current volume to info @@ -906,6 +1036,7 @@ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, return 0; return get_vol_mute(codec, info, nid, ch, direction, index); } +EXPORT_SYMBOL_HDA(snd_hda_codec_amp_read); /* * update the AMP value, mask = bit mask to set, val = the value @@ -925,6 +1056,7 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, put_vol_mute(codec, info, nid, ch, direction, idx, val); return 1; } +EXPORT_SYMBOL_HDA(snd_hda_codec_amp_update); /* * update the AMP stereo with the same mask and value @@ -938,15 +1070,16 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, idx, mask, val); return ret; } +EXPORT_SYMBOL_HDA(snd_hda_codec_amp_stereo); #ifdef SND_HDA_NEEDS_RESUME /* resume the all amp commands from the cache */ void snd_hda_codec_resume_amp(struct hda_codec *codec) { - struct hda_amp_info *buffer = codec->amp_cache.buffer; + struct hda_amp_info *buffer = codec->amp_cache.buf.list; int i; - for (i = 0; i < codec->amp_cache.size; i++, buffer++) { + for (i = 0; i < codec->amp_cache.buf.used; i++, buffer++) { u32 key = buffer->head.key; hda_nid_t nid; unsigned int idx, dir, ch; @@ -963,6 +1096,7 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec) } } } +EXPORT_SYMBOL_HDA(snd_hda_codec_resume_amp); #endif /* SND_HDA_NEEDS_RESUME */ /* volume */ @@ -990,6 +1124,7 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = caps; return 0; } +EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_info); int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1009,6 +1144,7 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, & HDA_AMP_VOLMASK; return 0; } +EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_get); int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1033,6 +1169,7 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, snd_hda_power_down(codec); return change; } +EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_put); int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv) @@ -1059,6 +1196,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -EFAULT; return 0; } +EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_tlv); /* * set (static) TLV for virtual master volume; recalculated as max 0dB @@ -1078,6 +1216,7 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, tlv[2] = -nums * step; tlv[3] = step; } +EXPORT_SYMBOL_HDA(snd_hda_set_vmaster_tlv); /* find a mixer control element with the given name */ static struct snd_kcontrol * @@ -1097,6 +1236,69 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, { return _snd_hda_find_mixer_ctl(codec, name, 0); } +EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl); + +/* Add a control element and assign to the codec */ +int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl) +{ + int err; + struct snd_kcontrol **knewp; + + err = snd_ctl_add(codec->bus->card, kctl); + if (err < 0) + return err; + knewp = snd_array_new(&codec->mixers); + if (!knewp) + return -ENOMEM; + *knewp = kctl; + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_ctl_add); + +#ifdef CONFIG_SND_HDA_RECONFIG +/* Clear all controls assigned to the given codec */ +void snd_hda_ctls_clear(struct hda_codec *codec) +{ + int i; + struct snd_kcontrol **kctls = codec->mixers.list; + for (i = 0; i < codec->mixers.used; i++) + snd_ctl_remove(codec->bus->card, kctls[i]); + snd_array_free(&codec->mixers); +} + +void snd_hda_codec_reset(struct hda_codec *codec) +{ + int i; + +#ifdef CONFIG_SND_HDA_POWER_SAVE + cancel_delayed_work(&codec->power_work); + flush_scheduled_work(); +#endif + snd_hda_ctls_clear(codec); + /* relase PCMs */ + for (i = 0; i < codec->num_pcms; i++) { + if (codec->pcm_info[i].pcm) { + snd_device_free(codec->bus->card, + codec->pcm_info[i].pcm); + clear_bit(codec->pcm_info[i].device, + codec->bus->pcm_dev_bits); + } + } + if (codec->patch_ops.free) + codec->patch_ops.free(codec); + codec->proc_widget_hook = NULL; + codec->spec = NULL; + free_hda_cache(&codec->amp_cache); + free_hda_cache(&codec->cmd_cache); + init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); + init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); + codec->num_pcms = 0; + codec->pcm_info = NULL; + codec->preset = NULL; + module_put(codec->owner); + codec->owner = NULL; +} +#endif /* CONFIG_SND_HDA_RECONFIG */ /* create a virtual master control and add slaves */ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, @@ -1115,7 +1317,7 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, kctl = snd_ctl_make_virtual_master(name, tlv); if (!kctl) return -ENOMEM; - err = snd_ctl_add(codec->bus->card, kctl); + err = snd_hda_ctl_add(codec, kctl); if (err < 0) return err; @@ -1133,6 +1335,7 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, } return 0; } +EXPORT_SYMBOL_HDA(snd_hda_add_vmaster); /* switch */ int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, @@ -1146,6 +1349,7 @@ int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = 1; return 0; } +EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_info); int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1165,6 +1369,7 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, HDA_AMP_MUTE) ? 0 : 1; return 0; } +EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_get); int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1195,6 +1400,7 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, snd_hda_power_down(codec); return change; } +EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put); /* * bound volume controls @@ -1220,6 +1426,7 @@ int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol, mutex_unlock(&codec->spdif_mutex); return err; } +EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_get); int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1243,6 +1450,7 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, mutex_unlock(&codec->spdif_mutex); return err < 0 ? err : change; } +EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_put); /* * generic bound volume/swtich controls @@ -1262,6 +1470,7 @@ int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol, mutex_unlock(&codec->spdif_mutex); return err; } +EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_info); int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1278,6 +1487,7 @@ int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol, mutex_unlock(&codec->spdif_mutex); return err; } +EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_get); int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1300,6 +1510,7 @@ int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol, mutex_unlock(&codec->spdif_mutex); return err < 0 ? err : change; } +EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_put); int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) @@ -1316,6 +1527,7 @@ int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag, mutex_unlock(&codec->spdif_mutex); return err; } +EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_tlv); struct hda_ctl_ops snd_hda_bind_vol = { .info = snd_hda_mixer_amp_volume_info, @@ -1323,6 +1535,7 @@ struct hda_ctl_ops snd_hda_bind_vol = { .put = snd_hda_mixer_amp_volume_put, .tlv = snd_hda_mixer_amp_tlv }; +EXPORT_SYMBOL_HDA(snd_hda_bind_vol); struct hda_ctl_ops snd_hda_bind_sw = { .info = snd_hda_mixer_amp_switch_info, @@ -1330,6 +1543,7 @@ struct hda_ctl_ops snd_hda_bind_sw = { .put = snd_hda_mixer_amp_switch_put, .tlv = snd_hda_mixer_amp_tlv }; +EXPORT_SYMBOL_HDA(snd_hda_bind_sw); /* * SPDIF out controls @@ -1577,9 +1791,11 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) } for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); + if (!kctl) + return -ENOMEM; kctl->id.index = idx; kctl->private_value = nid; - err = snd_ctl_add(codec->bus->card, kctl); + err = snd_hda_ctl_add(codec, kctl); if (err < 0) return err; } @@ -1589,6 +1805,7 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls); return 0; } +EXPORT_SYMBOL_HDA(snd_hda_create_spdif_out_ctls); /* * SPDIF sharing with analog output @@ -1623,9 +1840,10 @@ int snd_hda_create_spdif_share_sw(struct hda_codec *codec, if (!mout->dig_out_nid) return 0; /* ATTENTION: here mout is passed as private_data, instead of codec */ - return snd_ctl_add(codec->bus->card, + return snd_hda_ctl_add(codec, snd_ctl_new1(&spdif_share_sw, mout)); } +EXPORT_SYMBOL_HDA(snd_hda_create_spdif_share_sw); /* * SPDIF input @@ -1725,7 +1943,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); kctl->private_value = nid; - err = snd_ctl_add(codec->bus->card, kctl); + err = snd_hda_ctl_add(codec, kctl); if (err < 0) return err; } @@ -1735,6 +1953,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) AC_DIG1_ENABLE; return 0; } +EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); #ifdef SND_HDA_NEEDS_RESUME /* @@ -1761,10 +1980,14 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm) { + struct hda_bus *bus = codec->bus; + unsigned int res; int err; + + res = make_codec_cmd(codec, nid, direct, verb, parm); snd_hda_power_up(codec); - mutex_lock(&codec->bus->cmd_mutex); - err = codec->bus->ops.command(codec, nid, direct, verb, parm); + mutex_lock(&bus->cmd_mutex); + err = bus->ops.command(bus, res); if (!err) { struct hda_cache_head *c; u32 key = build_cmd_cache_key(nid, verb); @@ -1772,18 +1995,19 @@ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, if (c) c->val = parm; } - mutex_unlock(&codec->bus->cmd_mutex); + mutex_unlock(&bus->cmd_mutex); snd_hda_power_down(codec); return err; } +EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); /* resume the all commands from the cache */ void snd_hda_codec_resume_cache(struct hda_codec *codec) { - struct hda_cache_head *buffer = codec->cmd_cache.buffer; + struct hda_cache_head *buffer = codec->cmd_cache.buf.list; int i; - for (i = 0; i < codec->cmd_cache.size; i++, buffer++) { + for (i = 0; i < codec->cmd_cache.buf.used; i++, buffer++) { u32 key = buffer->key; if (!key) continue; @@ -1791,6 +2015,7 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec) get_cmd_cache_cmd(key), buffer->val); } } +EXPORT_SYMBOL_HDA(snd_hda_codec_resume_cache); /** * snd_hda_sequence_write_cache - sequence writes with caching @@ -1808,6 +2033,7 @@ void snd_hda_sequence_write_cache(struct hda_codec *codec, snd_hda_codec_write_cache(codec, seq->nid, 0, seq->verb, seq->param); } +EXPORT_SYMBOL_HDA(snd_hda_sequence_write_cache); #endif /* SND_HDA_NEEDS_RESUME */ /* @@ -1868,6 +2094,17 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, } } +#ifdef CONFIG_SND_HDA_HWDEP +/* execute additional init verbs */ +static void hda_exec_init_verbs(struct hda_codec *codec) +{ + if (codec->init_verbs.list) + snd_hda_sequence_write(codec, codec->init_verbs.list); +} +#else +static inline void hda_exec_init_verbs(struct hda_codec *codec) {} +#endif + #ifdef SND_HDA_NEEDS_RESUME /* * call suspend and power-down; used both from PM and power-save @@ -1894,6 +2131,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); + hda_exec_init_verbs(codec); if (codec->patch_ops.resume) codec->patch_ops.resume(codec); else { @@ -1914,28 +2152,37 @@ static void hda_call_codec_resume(struct hda_codec *codec) * * Returns 0 if successful, otherwise a negative error code. */ -int __devinit snd_hda_build_controls(struct hda_bus *bus) +int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus) { struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - int err = 0; - /* fake as if already powered-on */ - hda_keep_power_on(codec); - /* then fire up */ - hda_set_power_state(codec, - codec->afg ? codec->afg : codec->mfg, - AC_PWRST_D0); - /* continue to initialize... */ - if (codec->patch_ops.init) - err = codec->patch_ops.init(codec); - if (!err && codec->patch_ops.build_controls) - err = codec->patch_ops.build_controls(codec); - snd_hda_power_down(codec); + int err = snd_hda_codec_build_controls(codec); if (err < 0) return err; } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_build_controls); +int snd_hda_codec_build_controls(struct hda_codec *codec) +{ + int err = 0; + /* fake as if already powered-on */ + hda_keep_power_on(codec); + /* then fire up */ + hda_set_power_state(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_D0); + hda_exec_init_verbs(codec); + /* continue to initialize... */ + if (codec->patch_ops.init) + err = codec->patch_ops.init(codec); + if (!err && codec->patch_ops.build_controls) + err = codec->patch_ops.build_controls(codec); + snd_hda_power_down(codec); + if (err < 0) + return err; return 0; } @@ -2028,6 +2275,7 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, return val; } +EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format); /** * snd_hda_query_supported_pcm - query the supported PCM rates and formats @@ -2042,7 +2290,7 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, * * Returns 0 if successful, otherwise a negative error code. */ -int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, +static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u32 *ratesp, u64 *formatsp, unsigned int *bpsp) { int i; @@ -2207,6 +2455,7 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, return 1; } +EXPORT_SYMBOL_HDA(snd_hda_is_supported_format); /* * PCM stuff @@ -2236,8 +2485,8 @@ static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo, return 0; } -static int __devinit set_pcm_default_values(struct hda_codec *codec, - struct hda_pcm_stream *info) +static int set_pcm_default_values(struct hda_codec *codec, + struct hda_pcm_stream *info) { /* query support PCM information from the given NID */ if (info->nid && (!info->rates || !info->formats)) { @@ -2263,6 +2512,110 @@ static int __devinit set_pcm_default_values(struct hda_codec *codec, return 0; } +/* + * get the empty PCM device number to assign + */ +static int get_empty_pcm_device(struct hda_bus *bus, int type) +{ + static const char *dev_name[HDA_PCM_NTYPES] = { + "Audio", "SPDIF", "HDMI", "Modem" + }; + /* starting device index for each PCM type */ + static int dev_idx[HDA_PCM_NTYPES] = { + [HDA_PCM_TYPE_AUDIO] = 0, + [HDA_PCM_TYPE_SPDIF] = 1, + [HDA_PCM_TYPE_HDMI] = 3, + [HDA_PCM_TYPE_MODEM] = 6 + }; + /* normal audio device indices; not linear to keep compatibility */ + static int audio_idx[4] = { 0, 2, 4, 5 }; + int i, dev; + + switch (type) { + case HDA_PCM_TYPE_AUDIO: + for (i = 0; i < ARRAY_SIZE(audio_idx); i++) { + dev = audio_idx[i]; + if (!test_bit(dev, bus->pcm_dev_bits)) + break; + } + if (i >= ARRAY_SIZE(audio_idx)) { + snd_printk(KERN_WARNING "Too many audio devices\n"); + return -EAGAIN; + } + break; + case HDA_PCM_TYPE_SPDIF: + case HDA_PCM_TYPE_HDMI: + case HDA_PCM_TYPE_MODEM: + dev = dev_idx[type]; + if (test_bit(dev, bus->pcm_dev_bits)) { + snd_printk(KERN_WARNING "%s already defined\n", + dev_name[type]); + return -EAGAIN; + } + break; + default: + snd_printk(KERN_WARNING "Invalid PCM type %d\n", type); + return -EINVAL; + } + set_bit(dev, bus->pcm_dev_bits); + return dev; +} + +/* + * attach a new PCM stream + */ +static int snd_hda_attach_pcm(struct hda_codec *codec, struct hda_pcm *pcm) +{ + struct hda_bus *bus = codec->bus; + struct hda_pcm_stream *info; + int stream, err; + + if (snd_BUG_ON(!pcm->name)) + return -EINVAL; + for (stream = 0; stream < 2; stream++) { + info = &pcm->stream[stream]; + if (info->substreams) { + err = set_pcm_default_values(codec, info); + if (err < 0) + return err; + } + } + return bus->ops.attach_pcm(bus, codec, pcm); +} + +/* assign all PCMs of the given codec */ +int snd_hda_codec_build_pcms(struct hda_codec *codec) +{ + unsigned int pcm; + int err; + + if (!codec->num_pcms) { + if (!codec->patch_ops.build_pcms) + return 0; + err = codec->patch_ops.build_pcms(codec); + if (err < 0) + return err; + } + for (pcm = 0; pcm < codec->num_pcms; pcm++) { + struct hda_pcm *cpcm = &codec->pcm_info[pcm]; + int dev; + + if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams) + return 0; /* no substreams assigned */ + + if (!cpcm->pcm) { + dev = get_empty_pcm_device(codec->bus, cpcm->pcm_type); + if (dev < 0) + return 0; + cpcm->device = dev; + err = snd_hda_attach_pcm(codec, cpcm); + if (err < 0) + return err; + } + } + return 0; +} + /** * snd_hda_build_pcms - build PCM information * @bus: the BUS @@ -2294,27 +2647,13 @@ int __devinit snd_hda_build_pcms(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - unsigned int pcm, s; - int err; - if (!codec->patch_ops.build_pcms) - continue; - err = codec->patch_ops.build_pcms(codec); + int err = snd_hda_codec_build_pcms(codec); if (err < 0) return err; - for (pcm = 0; pcm < codec->num_pcms; pcm++) { - for (s = 0; s < 2; s++) { - struct hda_pcm_stream *info; - info = &codec->pcm_info[pcm].stream[s]; - if (!info->substreams) - continue; - err = set_pcm_default_values(codec, info); - if (err < 0) - return err; - } - } } return 0; } +EXPORT_SYMBOL_HDA(snd_hda_build_pcms); /** * snd_hda_check_board_config - compare the current codec with the config table @@ -2333,11 +2672,11 @@ int snd_hda_check_board_config(struct hda_codec *codec, int num_configs, const char **models, const struct snd_pci_quirk *tbl) { - if (codec->bus->modelname && models) { + if (codec->modelname && models) { int i; for (i = 0; i < num_configs; i++) { if (models[i] && - !strcmp(codec->bus->modelname, models[i])) { + !strcmp(codec->modelname, models[i])) { snd_printd(KERN_INFO "hda_codec: model '%s' is " "selected\n", models[i]); return i; @@ -2370,6 +2709,7 @@ int snd_hda_check_board_config(struct hda_codec *codec, } return -1; } +EXPORT_SYMBOL_HDA(snd_hda_check_board_config); /** * snd_hda_add_new_ctls - create controls from the array @@ -2390,7 +2730,7 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) kctl = snd_ctl_new1(knew, codec); if (!kctl) return -ENOMEM; - err = snd_ctl_add(codec->bus->card, kctl); + err = snd_hda_ctl_add(codec, kctl); if (err < 0) { if (!codec->addr) return err; @@ -2398,13 +2738,14 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) if (!kctl) return -ENOMEM; kctl->id.device = codec->addr; - err = snd_ctl_add(codec->bus->card, kctl); + err = snd_hda_ctl_add(codec, kctl); if (err < 0) return err; } } return 0; } +EXPORT_SYMBOL_HDA(snd_hda_add_new_ctls); #ifdef CONFIG_SND_HDA_POWER_SAVE static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, @@ -2414,6 +2755,7 @@ static void hda_power_work(struct work_struct *work) { struct hda_codec *codec = container_of(work, struct hda_codec, power_work.work); + struct hda_bus *bus = codec->bus; if (!codec->power_on || codec->power_count) { codec->power_transition = 0; @@ -2421,8 +2763,8 @@ static void hda_power_work(struct work_struct *work) } hda_call_codec_suspend(codec); - if (codec->bus->ops.pm_notify) - codec->bus->ops.pm_notify(codec); + if (bus->ops.pm_notify) + bus->ops.pm_notify(bus); } static void hda_keep_power_on(struct hda_codec *codec) @@ -2433,29 +2775,39 @@ static void hda_keep_power_on(struct hda_codec *codec) void snd_hda_power_up(struct hda_codec *codec) { + struct hda_bus *bus = codec->bus; + codec->power_count++; if (codec->power_on || codec->power_transition) return; codec->power_on = 1; - if (codec->bus->ops.pm_notify) - codec->bus->ops.pm_notify(codec); + if (bus->ops.pm_notify) + bus->ops.pm_notify(bus); hda_call_codec_resume(codec); cancel_delayed_work(&codec->power_work); codec->power_transition = 0; } +EXPORT_SYMBOL_HDA(snd_hda_power_up); + +#define power_save(codec) \ + ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) + +#define power_save(codec) \ + ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) void snd_hda_power_down(struct hda_codec *codec) { --codec->power_count; if (!codec->power_on || codec->power_count || codec->power_transition) return; - if (power_save) { + if (power_save(codec)) { codec->power_transition = 1; /* avoid reentrance */ schedule_delayed_work(&codec->power_work, - msecs_to_jiffies(power_save * 1000)); + msecs_to_jiffies(power_save(codec) * 1000)); } } +EXPORT_SYMBOL_HDA(snd_hda_power_down); int snd_hda_check_amp_list_power(struct hda_codec *codec, struct hda_loopback_check *check, @@ -2492,6 +2844,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, } return 0; } +EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power); #endif /* @@ -2511,6 +2864,7 @@ int snd_hda_ch_mode_info(struct hda_codec *codec, chmode[uinfo->value.enumerated.item].channels); return 0; } +EXPORT_SYMBOL_HDA(snd_hda_ch_mode_info); int snd_hda_ch_mode_get(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, @@ -2528,6 +2882,7 @@ int snd_hda_ch_mode_get(struct hda_codec *codec, } return 0; } +EXPORT_SYMBOL_HDA(snd_hda_ch_mode_get); int snd_hda_ch_mode_put(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, @@ -2548,6 +2903,7 @@ int snd_hda_ch_mode_put(struct hda_codec *codec, snd_hda_sequence_write_cache(codec, chmode[mode].sequence); return 1; } +EXPORT_SYMBOL_HDA(snd_hda_ch_mode_put); /* * input MUX helper @@ -2568,6 +2924,7 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux, strcpy(uinfo->value.enumerated.name, imux->items[index].label); return 0; } +EXPORT_SYMBOL_HDA(snd_hda_input_mux_info); int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux, @@ -2589,6 +2946,7 @@ int snd_hda_input_mux_put(struct hda_codec *codec, *cur_val = idx; return 1; } +EXPORT_SYMBOL_HDA(snd_hda_input_mux_put); /* @@ -2641,6 +2999,7 @@ int snd_hda_multi_out_dig_open(struct hda_codec *codec, mutex_unlock(&codec->spdif_mutex); return 0; } +EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_open); int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, struct hda_multi_out *mout, @@ -2653,6 +3012,7 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, mutex_unlock(&codec->spdif_mutex); return 0; } +EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_prepare); /* * release the digital out @@ -2665,6 +3025,7 @@ int snd_hda_multi_out_dig_close(struct hda_codec *codec, mutex_unlock(&codec->spdif_mutex); return 0; } +EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_close); /* * set up more restrictions for analog out @@ -2704,6 +3065,7 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, return snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); } +EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_open); /* * set up the i/o for analog out @@ -2762,6 +3124,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, } return 0; } +EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_prepare); /* * clean up the setting for analog out @@ -2788,6 +3151,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, mutex_unlock(&codec->spdif_mutex); return 0; } +EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_cleanup); /* * Helper for automatic pin configuration @@ -3073,11 +3437,13 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, return 0; } +EXPORT_SYMBOL_HDA(snd_hda_parse_pin_def_config); /* labels for input pins */ const char *auto_pin_cfg_labels[AUTO_PIN_LAST] = { "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" }; +EXPORT_SYMBOL_HDA(auto_pin_cfg_labels); #ifdef CONFIG_PM @@ -3105,11 +3471,11 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state) } return 0; } +EXPORT_SYMBOL_HDA(snd_hda_suspend); /** * snd_hda_resume - resume the codecs * @bus: the HDA bus - * @state: resume state * * Returns 0 if successful. * @@ -3126,16 +3492,79 @@ int snd_hda_resume(struct hda_bus *bus) } return 0; } -#ifdef CONFIG_SND_HDA_POWER_SAVE -int snd_hda_codecs_inuse(struct hda_bus *bus) -{ - struct hda_codec *codec; +EXPORT_SYMBOL_HDA(snd_hda_resume); +#endif /* CONFIG_PM */ - list_for_each_entry(codec, &bus->codec_list, list) { - if (snd_hda_codec_needs_resume(codec)) - return 1; +/* + * generic arrays + */ + +/* get a new element from the given array + * if it exceeds the pre-allocated array size, re-allocate the array + */ +void *snd_array_new(struct snd_array *array) +{ + if (array->used >= array->alloced) { + int num = array->alloced + array->alloc_align; + void *nlist; + if (snd_BUG_ON(num >= 4096)) + return NULL; + nlist = kcalloc(num + 1, array->elem_size, GFP_KERNEL); + if (!nlist) + return NULL; + if (array->list) { + memcpy(nlist, array->list, + array->elem_size * array->alloced); + kfree(array->list); + } + array->list = nlist; + array->alloced = num; } - return 0; + return snd_array_elem(array, array->used++); } -#endif -#endif +EXPORT_SYMBOL_HDA(snd_array_new); + +/* free the given array elements */ +void snd_array_free(struct snd_array *array) +{ + kfree(array->list); + array->used = 0; + array->alloced = 0; + array->list = NULL; +} +EXPORT_SYMBOL_HDA(snd_array_free); + +/* + * used by hda_proc.c and hda_eld.c + */ +void snd_print_pcm_rates(int pcm, char *buf, int buflen) +{ + static unsigned int rates[] = { + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200, + 96000, 176400, 192000, 384000 + }; + int i, j; + + for (i = 0, j = 0; i < ARRAY_SIZE(rates); i++) + if (pcm & (1 << i)) + j += snprintf(buf + j, buflen - j, " %d", rates[i]); + + buf[j] = '\0'; /* necessary when j == 0 */ +} +EXPORT_SYMBOL_HDA(snd_print_pcm_rates); + +void snd_print_pcm_bits(int pcm, char *buf, int buflen) +{ + static unsigned int bits[] = { 8, 16, 20, 24, 32 }; + int i, j; + + for (i = 0, j = 0; i < ARRAY_SIZE(bits); i++) + if (pcm & (AC_SUPPCM_BITS_8 << i)) + j += snprintf(buf + j, buflen - j, " %d", bits[i]); + + buf[j] = '\0'; /* necessary when j == 0 */ +} +EXPORT_SYMBOL_HDA(snd_print_pcm_bits); + +MODULE_DESCRIPTION("HDA codec core"); +MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 60468f562400..729fc7642d7f 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -520,6 +520,36 @@ enum { #define HDA_MAX_CODEC_ADDRESS 0x0f /* + * generic arrays + */ +struct snd_array { + unsigned int used; + unsigned int alloced; + unsigned int elem_size; + unsigned int alloc_align; + void *list; +}; + +void *snd_array_new(struct snd_array *array); +void snd_array_free(struct snd_array *array); +static inline void snd_array_init(struct snd_array *array, unsigned int size, + unsigned int align) +{ + array->elem_size = size; + array->alloc_align = align; +} + +static inline void *snd_array_elem(struct snd_array *array, unsigned int idx) +{ + return array->list + idx * array->elem_size; +} + +static inline unsigned int snd_array_index(struct snd_array *array, void *ptr) +{ + return (unsigned long)(ptr - array->list) / array->elem_size; +} + +/* * Structures */ @@ -536,15 +566,17 @@ typedef u16 hda_nid_t; /* bus operators */ struct hda_bus_ops { /* send a single command */ - int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct, - unsigned int verb, unsigned int parm); + int (*command)(struct hda_bus *bus, unsigned int cmd); /* get a response from the last command */ - unsigned int (*get_response)(struct hda_codec *codec); + unsigned int (*get_response)(struct hda_bus *bus); /* free the private data */ void (*private_free)(struct hda_bus *); + /* attach a PCM stream */ + int (*attach_pcm)(struct hda_bus *bus, struct hda_codec *codec, + struct hda_pcm *pcm); #ifdef CONFIG_SND_HDA_POWER_SAVE /* notify power-up/down from codec to controller */ - void (*pm_notify)(struct hda_codec *codec); + void (*pm_notify)(struct hda_bus *bus); #endif }; @@ -553,6 +585,7 @@ struct hda_bus_template { void *private_data; struct pci_dev *pci; const char *modelname; + int *power_save; struct hda_bus_ops ops; }; @@ -569,6 +602,7 @@ struct hda_bus { void *private_data; struct pci_dev *pci; const char *modelname; + int *power_save; struct hda_bus_ops ops; /* codec linked list */ @@ -581,10 +615,12 @@ struct hda_bus { /* unsolicited event queue */ struct hda_bus_unsolicited *unsol; - struct snd_info_entry *proc; + /* assigned PCMs */ + DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES); /* misc op flags */ unsigned int needs_damn_long_delay :1; + unsigned int shutdown :1; /* being unloaded */ }; /* @@ -604,6 +640,16 @@ struct hda_codec_preset { int (*patch)(struct hda_codec *codec); }; +struct hda_codec_preset_list { + const struct hda_codec_preset *preset; + struct module *owner; + struct list_head list; +}; + +/* initial hook */ +int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset); +int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset); + /* ops set by the preset patch */ struct hda_codec_ops { int (*build_controls)(struct hda_codec *codec); @@ -635,10 +681,7 @@ struct hda_amp_info { struct hda_cache_rec { u16 hash[64]; /* hash table for index */ - unsigned int num_entries; /* number of assigned entries */ - unsigned int size; /* allocated size */ - unsigned int record_size; /* record size (including header) */ - void *buffer; /* hash table entries */ + struct snd_array buf; /* record entries */ }; /* PCM callbacks */ @@ -680,7 +723,8 @@ struct hda_pcm { char *name; struct hda_pcm_stream stream[2]; unsigned int pcm_type; /* HDA_PCM_TYPE_XXX */ - int device; /* assigned device number */ + int device; /* device number to assign */ + struct snd_pcm *pcm; /* assigned PCM instance */ }; /* codec information */ @@ -699,6 +743,9 @@ struct hda_codec { /* detected preset */ const struct hda_codec_preset *preset; + struct module *owner; + const char *name; /* codec name */ + const char *modelname; /* model name for preset */ /* set by patch */ struct hda_codec_ops patch_ops; @@ -718,6 +765,8 @@ struct hda_codec { hda_nid_t start_nid; u32 *wcaps; + struct snd_array mixers; /* list of assigned mixer elements */ + struct hda_cache_rec amp_cache; /* cache for amp access */ struct hda_cache_rec cmd_cache; /* cache for other commands */ @@ -727,7 +776,11 @@ struct hda_codec { unsigned int spdif_in_enable; /* SPDIF input enable? */ hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ +#ifdef CONFIG_SND_HDA_HWDEP struct snd_hwdep *hwdep; /* assigned hwdep device */ + struct snd_array init_verbs; /* additional init verbs */ + struct snd_array hints; /* additional hints */ +#endif /* misc flags */ unsigned int spdif_status_reset :1; /* needs to toggle SPDIF for each @@ -740,6 +793,10 @@ struct hda_codec { int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ #endif + + /* codec-specific additional proc output */ + void (*proc_widget_hook)(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid); }; /* direction */ @@ -754,7 +811,7 @@ enum { int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, struct hda_bus **busp); int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - struct hda_codec **codecp); + int do_init, struct hda_codec **codecp); /* * low level functions @@ -799,11 +856,13 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec); * Mixer */ int snd_hda_build_controls(struct hda_bus *bus); +int snd_hda_codec_build_controls(struct hda_codec *codec); /* * PCM */ int snd_hda_build_pcms(struct hda_bus *bus); +int snd_hda_codec_build_pcms(struct hda_codec *codec); void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, int channel_id, int format); @@ -812,8 +871,6 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, unsigned int channels, unsigned int format, unsigned int maxbps); -int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, - u32 *ratesp, u64 *formatsp, unsigned int *bpsp); int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, unsigned int format); @@ -831,18 +888,38 @@ int snd_hda_resume(struct hda_bus *bus); #endif /* + * get widget information + */ +const char *snd_hda_get_jack_connectivity(u32 cfg); +const char *snd_hda_get_jack_type(u32 cfg); +const char *snd_hda_get_jack_location(u32 cfg); + +/* * power saving */ #ifdef CONFIG_SND_HDA_POWER_SAVE void snd_hda_power_up(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); #define snd_hda_codec_needs_resume(codec) codec->power_count -int snd_hda_codecs_inuse(struct hda_bus *bus); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} #define snd_hda_codec_needs_resume(codec) 1 -#define snd_hda_codecs_inuse(bus) 1 +#endif + +/* + * Codec modularization + */ + +/* Export symbols only for communication with codec drivers; + * When built in kernel, all HD-audio drivers are supposed to be statically + * linked to the kernel. Thus, the symbols don't have to (or shouldn't) be + * exported unless it's built as a module. + */ +#ifdef MODULE +#define EXPORT_SYMBOL_HDA(sym) EXPORT_SYMBOL_GPL(sym) +#else +#define EXPORT_SYMBOL_HDA(sym) #endif #endif /* __SOUND_HDA_CODEC_H */ diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c new file mode 100644 index 000000000000..fcad5ec31773 --- /dev/null +++ b/sound/pci/hda/hda_eld.c @@ -0,0 +1,590 @@ +/* + * Generic routines and proc interface for ELD(EDID Like Data) information + * + * Copyright(c) 2008 Intel Corporation. + * + * Authors: + * Wu Fengguang <wfg@linux.intel.com> + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <sound/core.h> +#include <asm/unaligned.h> +#include "hda_codec.h" +#include "hda_local.h" + +enum eld_versions { + ELD_VER_CEA_861D = 2, + ELD_VER_PARTIAL = 31, +}; + +enum cea_edid_versions { + CEA_EDID_VER_NONE = 0, + CEA_EDID_VER_CEA861 = 1, + CEA_EDID_VER_CEA861A = 2, + CEA_EDID_VER_CEA861BCD = 3, + CEA_EDID_VER_RESERVED = 4, +}; + +static char *cea_speaker_allocation_names[] = { + /* 0 */ "FL/FR", + /* 1 */ "LFE", + /* 2 */ "FC", + /* 3 */ "RL/RR", + /* 4 */ "RC", + /* 5 */ "FLC/FRC", + /* 6 */ "RLC/RRC", + /* 7 */ "FLW/FRW", + /* 8 */ "FLH/FRH", + /* 9 */ "TC", + /* 10 */ "FCH", +}; + +static char *eld_connection_type_names[4] = { + "HDMI", + "DisplayPort", + "2-reserved", + "3-reserved" +}; + +enum cea_audio_coding_types { + AUDIO_CODING_TYPE_REF_STREAM_HEADER = 0, + AUDIO_CODING_TYPE_LPCM = 1, + AUDIO_CODING_TYPE_AC3 = 2, + AUDIO_CODING_TYPE_MPEG1 = 3, + AUDIO_CODING_TYPE_MP3 = 4, + AUDIO_CODING_TYPE_MPEG2 = 5, + AUDIO_CODING_TYPE_AACLC = 6, + AUDIO_CODING_TYPE_DTS = 7, + AUDIO_CODING_TYPE_ATRAC = 8, + AUDIO_CODING_TYPE_SACD = 9, + AUDIO_CODING_TYPE_EAC3 = 10, + AUDIO_CODING_TYPE_DTS_HD = 11, + AUDIO_CODING_TYPE_MLP = 12, + AUDIO_CODING_TYPE_DST = 13, + AUDIO_CODING_TYPE_WMAPRO = 14, + AUDIO_CODING_TYPE_REF_CXT = 15, + /* also include valid xtypes below */ + AUDIO_CODING_TYPE_HE_AAC = 15, + AUDIO_CODING_TYPE_HE_AAC2 = 16, + AUDIO_CODING_TYPE_MPEG_SURROUND = 17, +}; + +enum cea_audio_coding_xtypes { + AUDIO_CODING_XTYPE_HE_REF_CT = 0, + AUDIO_CODING_XTYPE_HE_AAC = 1, + AUDIO_CODING_XTYPE_HE_AAC2 = 2, + AUDIO_CODING_XTYPE_MPEG_SURROUND = 3, + AUDIO_CODING_XTYPE_FIRST_RESERVED = 4, +}; + +static char *cea_audio_coding_type_names[] = { + /* 0 */ "undefined", + /* 1 */ "LPCM", + /* 2 */ "AC-3", + /* 3 */ "MPEG1", + /* 4 */ "MP3", + /* 5 */ "MPEG2", + /* 6 */ "AAC-LC", + /* 7 */ "DTS", + /* 8 */ "ATRAC", + /* 9 */ "DSD (One Bit Audio)", + /* 10 */ "E-AC-3/DD+ (Dolby Digital Plus)", + /* 11 */ "DTS-HD", + /* 12 */ "MLP (Dolby TrueHD)", + /* 13 */ "DST", + /* 14 */ "WMAPro", + /* 15 */ "HE-AAC", + /* 16 */ "HE-AACv2", + /* 17 */ "MPEG Surround", +}; + +/* + * The following two lists are shared between + * - HDMI audio InfoFrame (source to sink) + * - CEA E-EDID Extension (sink to source) + */ + +/* + * SS1:SS0 index => sample size + */ +static int cea_sample_sizes[4] = { + 0, /* 0: Refer to Stream Header */ + AC_SUPPCM_BITS_16, /* 1: 16 bits */ + AC_SUPPCM_BITS_20, /* 2: 20 bits */ + AC_SUPPCM_BITS_24, /* 3: 24 bits */ +}; + +/* + * SF2:SF1:SF0 index => sampling frequency + */ +static int cea_sampling_frequencies[8] = { + 0, /* 0: Refer to Stream Header */ + SNDRV_PCM_RATE_32000, /* 1: 32000Hz */ + SNDRV_PCM_RATE_44100, /* 2: 44100Hz */ + SNDRV_PCM_RATE_48000, /* 3: 48000Hz */ + SNDRV_PCM_RATE_88200, /* 4: 88200Hz */ + SNDRV_PCM_RATE_96000, /* 5: 96000Hz */ + SNDRV_PCM_RATE_176400, /* 6: 176400Hz */ + SNDRV_PCM_RATE_192000, /* 7: 192000Hz */ +}; + +static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid, + int byte_index) +{ + unsigned int val; + + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_HDMI_ELDD, byte_index); + +#ifdef BE_PARANOID + printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val); +#endif + + if ((val & AC_ELDD_ELD_VALID) == 0) { + snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n", + byte_index); + val = 0; + } + + return val & AC_ELDD_ELD_DATA; +} + +#define GRAB_BITS(buf, byte, lowbit, bits) \ +({ \ + BUILD_BUG_ON(lowbit > 7); \ + BUILD_BUG_ON(bits > 8); \ + BUILD_BUG_ON(bits <= 0); \ + \ + (buf[byte] >> (lowbit)) & ((1 << (bits)) - 1); \ +}) + +static void hdmi_update_short_audio_desc(struct cea_sad *a, + const unsigned char *buf) +{ + int i; + int val; + + val = GRAB_BITS(buf, 1, 0, 7); + a->rates = 0; + for (i = 0; i < 7; i++) + if (val & (1 << i)) + a->rates |= cea_sampling_frequencies[i + 1]; + + a->channels = GRAB_BITS(buf, 0, 0, 3); + a->channels++; + + a->format = GRAB_BITS(buf, 0, 3, 4); + switch (a->format) { + case AUDIO_CODING_TYPE_REF_STREAM_HEADER: + snd_printd(KERN_INFO + "HDMI: audio coding type 0 not expected\n"); + break; + + case AUDIO_CODING_TYPE_LPCM: + val = GRAB_BITS(buf, 2, 0, 3); + a->sample_bits = 0; + for (i = 0; i < 3; i++) + if (val & (1 << i)) + a->sample_bits |= cea_sample_sizes[i + 1]; + break; + + case AUDIO_CODING_TYPE_AC3: + case AUDIO_CODING_TYPE_MPEG1: + case AUDIO_CODING_TYPE_MP3: + case AUDIO_CODING_TYPE_MPEG2: + case AUDIO_CODING_TYPE_AACLC: + case AUDIO_CODING_TYPE_DTS: + case AUDIO_CODING_TYPE_ATRAC: + a->max_bitrate = GRAB_BITS(buf, 2, 0, 8); + a->max_bitrate *= 8000; + break; + + case AUDIO_CODING_TYPE_SACD: + break; + + case AUDIO_CODING_TYPE_EAC3: + break; + + case AUDIO_CODING_TYPE_DTS_HD: + break; + + case AUDIO_CODING_TYPE_MLP: + break; + + case AUDIO_CODING_TYPE_DST: + break; + + case AUDIO_CODING_TYPE_WMAPRO: + a->profile = GRAB_BITS(buf, 2, 0, 3); + break; + + case AUDIO_CODING_TYPE_REF_CXT: + a->format = GRAB_BITS(buf, 2, 3, 5); + if (a->format == AUDIO_CODING_XTYPE_HE_REF_CT || + a->format >= AUDIO_CODING_XTYPE_FIRST_RESERVED) { + snd_printd(KERN_INFO + "HDMI: audio coding xtype %d not expected\n", + a->format); + a->format = 0; + } else + a->format += AUDIO_CODING_TYPE_HE_AAC - + AUDIO_CODING_XTYPE_HE_AAC; + break; + } +} + +/* + * Be careful, ELD buf could be totally rubbish! + */ +static int hdmi_update_eld(struct hdmi_eld *e, + const unsigned char *buf, int size) +{ + int mnl; + int i; + + e->eld_ver = GRAB_BITS(buf, 0, 3, 5); + if (e->eld_ver != ELD_VER_CEA_861D && + e->eld_ver != ELD_VER_PARTIAL) { + snd_printd(KERN_INFO "HDMI: Unknown ELD version %d\n", + e->eld_ver); + goto out_fail; + } + + e->eld_size = size; + e->baseline_len = GRAB_BITS(buf, 2, 0, 8); + mnl = GRAB_BITS(buf, 4, 0, 5); + e->cea_edid_ver = GRAB_BITS(buf, 4, 5, 3); + + e->support_hdcp = GRAB_BITS(buf, 5, 0, 1); + e->support_ai = GRAB_BITS(buf, 5, 1, 1); + e->conn_type = GRAB_BITS(buf, 5, 2, 2); + e->sad_count = GRAB_BITS(buf, 5, 4, 4); + + e->aud_synch_delay = GRAB_BITS(buf, 6, 0, 8) * 2; + e->spk_alloc = GRAB_BITS(buf, 7, 0, 7); + + e->port_id = get_unaligned_le64(buf + 8); + + /* not specified, but the spec's tendency is little endian */ + e->manufacture_id = get_unaligned_le16(buf + 16); + e->product_id = get_unaligned_le16(buf + 18); + + if (mnl > ELD_MAX_MNL) { + snd_printd(KERN_INFO "HDMI: MNL is reserved value %d\n", mnl); + goto out_fail; + } else if (ELD_FIXED_BYTES + mnl > size) { + snd_printd(KERN_INFO "HDMI: out of range MNL %d\n", mnl); + goto out_fail; + } else + strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl); + + for (i = 0; i < e->sad_count; i++) { + if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) { + snd_printd(KERN_INFO "HDMI: out of range SAD %d\n", i); + goto out_fail; + } + hdmi_update_short_audio_desc(e->sad + i, + buf + ELD_FIXED_BYTES + mnl + 3 * i); + } + + return 0; + +out_fail: + e->eld_ver = 0; + return -EINVAL; +} + +static int hdmi_present_sense(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); +} + +static int hdmi_eld_valid(struct hda_codec *codec, hda_nid_t nid) +{ + int eldv; + int present; + + present = hdmi_present_sense(codec, nid); + eldv = (present & AC_PINSENSE_ELDV); + present = (present & AC_PINSENSE_PRESENCE); + +#ifdef CONFIG_SND_DEBUG_VERBOSE + printk(KERN_INFO "HDMI: sink_present = %d, eld_valid = %d\n", + !!present, !!eldv); +#endif + + return eldv && present; +} + +int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_SIZE, + AC_DIPSIZE_ELD_BUF); +} + +int snd_hdmi_get_eld(struct hdmi_eld *eld, + struct hda_codec *codec, hda_nid_t nid) +{ + int i; + int ret; + int size; + unsigned char *buf; + + if (!hdmi_eld_valid(codec, nid)) + return -ENOENT; + + size = snd_hdmi_get_eld_size(codec, nid); + if (size == 0) { + /* wfg: workaround for ASUS P5E-VM HDMI board */ + snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n"); + size = 128; + } + if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) { + snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size); + return -ERANGE; + } + + buf = kmalloc(size, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + for (i = 0; i < size; i++) + buf[i] = hdmi_get_eld_byte(codec, nid, i); + + ret = hdmi_update_eld(eld, buf, size); + + kfree(buf); + return ret; +} + +static void hdmi_show_short_audio_desc(struct cea_sad *a) +{ + char buf[SND_PRINT_RATES_ADVISED_BUFSIZE]; + char buf2[8 + SND_PRINT_BITS_ADVISED_BUFSIZE] = ", bits ="; + + if (!a->format) + return; + + snd_print_pcm_rates(a->rates, buf, sizeof(buf)); + + if (a->format == AUDIO_CODING_TYPE_LPCM) + snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8)); + else if (a->max_bitrate) + snprintf(buf2, sizeof(buf2), + ", max bitrate = %d", a->max_bitrate); + else + buf2[0] = '\0'; + + printk(KERN_INFO "HDMI: supports coding type %s:" + " channels = %d, rates =%s%s\n", + cea_audio_coding_type_names[a->format], + a->channels, + buf, + buf2); +} + +void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen) +{ + int i, j; + + for (i = 0, j = 0; i < ARRAY_SIZE(cea_speaker_allocation_names); i++) { + if (spk_alloc & (1 << i)) + j += snprintf(buf + j, buflen - j, " %s", + cea_speaker_allocation_names[i]); + } + buf[j] = '\0'; /* necessary when j == 0 */ +} + +void snd_hdmi_show_eld(struct hdmi_eld *e) +{ + int i; + + printk(KERN_INFO "HDMI: detected monitor %s at connection type %s\n", + e->monitor_name, + eld_connection_type_names[e->conn_type]); + + if (e->spk_alloc) { + char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; + snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf)); + printk(KERN_INFO "HDMI: available speakers:%s\n", buf); + } + + for (i = 0; i < e->sad_count; i++) + hdmi_show_short_audio_desc(e->sad + i); +} + +#ifdef CONFIG_PROC_FS + +static void hdmi_print_sad_info(int i, struct cea_sad *a, + struct snd_info_buffer *buffer) +{ + char buf[SND_PRINT_RATES_ADVISED_BUFSIZE]; + + snd_iprintf(buffer, "sad%d_coding_type\t[0x%x] %s\n", + i, a->format, cea_audio_coding_type_names[a->format]); + snd_iprintf(buffer, "sad%d_channels\t\t%d\n", i, a->channels); + + snd_print_pcm_rates(a->rates, buf, sizeof(buf)); + snd_iprintf(buffer, "sad%d_rates\t\t[0x%x]%s\n", i, a->rates, buf); + + if (a->format == AUDIO_CODING_TYPE_LPCM) { + snd_print_pcm_bits(a->sample_bits, buf, sizeof(buf)); + snd_iprintf(buffer, "sad%d_bits\t\t[0x%x]%s\n", + i, a->sample_bits, buf); + } + + if (a->max_bitrate) + snd_iprintf(buffer, "sad%d_max_bitrate\t%d\n", + i, a->max_bitrate); + + if (a->profile) + snd_iprintf(buffer, "sad%d_profile\t\t%d\n", i, a->profile); +} + +static void hdmi_print_eld_info(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct hdmi_eld *e = entry->private_data; + char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; + int i; + static char *eld_versoin_names[32] = { + "reserved", + "reserved", + "CEA-861D or below", + [3 ... 30] = "reserved", + [31] = "partial" + }; + static char *cea_edid_version_names[8] = { + "no CEA EDID Timing Extension block present", + "CEA-861", + "CEA-861-A", + "CEA-861-B, C or D", + [4 ... 7] = "reserved" + }; + + snd_iprintf(buffer, "monitor_name\t\t%s\n", e->monitor_name); + snd_iprintf(buffer, "connection_type\t\t%s\n", + eld_connection_type_names[e->conn_type]); + snd_iprintf(buffer, "eld_version\t\t[0x%x] %s\n", e->eld_ver, + eld_versoin_names[e->eld_ver]); + snd_iprintf(buffer, "edid_version\t\t[0x%x] %s\n", e->cea_edid_ver, + cea_edid_version_names[e->cea_edid_ver]); + snd_iprintf(buffer, "manufacture_id\t\t0x%x\n", e->manufacture_id); + snd_iprintf(buffer, "product_id\t\t0x%x\n", e->product_id); + snd_iprintf(buffer, "port_id\t\t\t0x%llx\n", (long long)e->port_id); + snd_iprintf(buffer, "support_hdcp\t\t%d\n", e->support_hdcp); + snd_iprintf(buffer, "support_ai\t\t%d\n", e->support_ai); + snd_iprintf(buffer, "audio_sync_delay\t%d\n", e->aud_synch_delay); + + snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf)); + snd_iprintf(buffer, "speakers\t\t[0x%x]%s\n", e->spk_alloc, buf); + + snd_iprintf(buffer, "sad_count\t\t%d\n", e->sad_count); + + for (i = 0; i < e->sad_count; i++) + hdmi_print_sad_info(i, e->sad + i, buffer); +} + +static void hdmi_write_eld_info(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct hdmi_eld *e = entry->private_data; + char line[64]; + char name[64]; + char *sname; + long long val; + int n; + + while (!snd_info_get_line(buffer, line, sizeof(line))) { + if (sscanf(line, "%s %llx", name, &val) != 2) + continue; + /* + * We don't allow modification to these fields: + * monitor_name manufacture_id product_id + * eld_version edid_version + */ + if (!strcmp(name, "connection_type")) + e->conn_type = val; + else if (!strcmp(name, "port_id")) + e->port_id = val; + else if (!strcmp(name, "support_hdcp")) + e->support_hdcp = val; + else if (!strcmp(name, "support_ai")) + e->support_ai = val; + else if (!strcmp(name, "audio_sync_delay")) + e->aud_synch_delay = val; + else if (!strcmp(name, "speakers")) + e->spk_alloc = val; + else if (!strcmp(name, "sad_count")) + e->sad_count = val; + else if (!strncmp(name, "sad", 3)) { + sname = name + 4; + n = name[3] - '0'; + if (name[4] >= '0' && name[4] <= '9') { + sname++; + n = 10 * n + name[4] - '0'; + } + if (n < 0 || n > 31) /* double the CEA limit */ + continue; + if (!strcmp(sname, "_coding_type")) + e->sad[n].format = val; + else if (!strcmp(sname, "_channels")) + e->sad[n].channels = val; + else if (!strcmp(sname, "_rates")) + e->sad[n].rates = val; + else if (!strcmp(sname, "_bits")) + e->sad[n].sample_bits = val; + else if (!strcmp(sname, "_max_bitrate")) + e->sad[n].max_bitrate = val; + else if (!strcmp(sname, "_profile")) + e->sad[n].profile = val; + if (n >= e->sad_count) + e->sad_count = n + 1; + } + } +} + + +int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld) +{ + char name[32]; + struct snd_info_entry *entry; + int err; + + snprintf(name, sizeof(name), "eld#%d", codec->addr); + err = snd_card_proc_new(codec->bus->card, name, &entry); + if (err < 0) + return err; + + snd_info_set_text_ops(entry, eld, hdmi_print_eld_info); + entry->c.text.write = hdmi_write_eld_info; + entry->mode |= S_IWUSR; + eld->proc_entry = entry; + + return 0; +} + +void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld) +{ + if (!codec->bus->shutdown && eld->proc_entry) { + snd_device_free(codec->bus->card, eld->proc_entry); + eld->proc_entry = NULL; + } +} + +#endif /* CONFIG_PROC_FS */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 0ca30894f7c6..65745e96dc70 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -723,7 +723,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, if (is_loopback) add_input_loopback(codec, node->nid, HDA_INPUT, index); snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index); - if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0) + err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + if (err < 0) return err; created = 1; } else if ((node->wid_caps & AC_WCAP_OUT_AMP) && @@ -732,7 +733,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, if (is_loopback) add_input_loopback(codec, node->nid, HDA_OUTPUT, 0); snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid); - if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0) + err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + if (err < 0) return err; created = 1; } @@ -745,14 +747,16 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, (node->amp_in_caps & AC_AMPCAP_NUM_STEPS)) { knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, index, HDA_INPUT); snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index); - if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0) + err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + if (err < 0) return err; created = 1; } else if ((node->wid_caps & AC_WCAP_OUT_AMP) && (node->amp_out_caps & AC_AMPCAP_NUM_STEPS)) { knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, 0, HDA_OUTPUT); snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid); - if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0) + err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + if (err < 0) return err; created = 1; } @@ -849,8 +853,8 @@ static int build_input_controls(struct hda_codec *codec) } /* create input MUX if multiple sources are available */ - if ((err = snd_ctl_add(codec->bus->card, - snd_ctl_new1(&cap_sel, codec))) < 0) + err = snd_hda_ctl_add(codec, snd_ctl_new1(&cap_sel, codec)); + if (err < 0) return err; /* no volume control? */ @@ -867,8 +871,8 @@ static int build_input_controls(struct hda_codec *codec) HDA_CODEC_VOLUME(name, adc_node->nid, spec->input_mux.items[i].index, HDA_INPUT); - if ((err = snd_ctl_add(codec->bus->card, - snd_ctl_new1(&knew, codec))) < 0) + err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + if (err < 0) return err; } @@ -1097,3 +1101,4 @@ int snd_hda_parse_generic_codec(struct hda_codec *codec) snd_hda_generic_free(codec); return err; } +EXPORT_SYMBOL(snd_hda_parse_generic_codec); diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 6e18a422d993..300ab407cf42 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -23,10 +23,12 @@ #include <linux/pci.h> #include <linux/compat.h> #include <linux/mutex.h> +#include <linux/ctype.h> #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" #include <sound/hda_hwdep.h> +#include <sound/minors.h> /* * write/read an out-of-bound verb @@ -95,7 +97,26 @@ static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file) return 0; } -int __devinit snd_hda_create_hwdep(struct hda_codec *codec) +static void clear_hwdep_elements(struct hda_codec *codec) +{ + char **head; + int i; + + /* clear init verbs */ + snd_array_free(&codec->init_verbs); + /* clear hints */ + head = codec->hints.list; + for (i = 0; i < codec->hints.used; i++, head++) + kfree(*head); + snd_array_free(&codec->hints); +} + +static void hwdep_free(struct snd_hwdep *hwdep) +{ + clear_hwdep_elements(hwdep->private_data); +} + +int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) { char hwname[16]; struct snd_hwdep *hwdep; @@ -109,6 +130,7 @@ int __devinit snd_hda_create_hwdep(struct hda_codec *codec) sprintf(hwdep->name, "HDA Codec %d", codec->addr); hwdep->iface = SNDRV_HWDEP_IFACE_HDA; hwdep->private_data = codec; + hwdep->private_free = hwdep_free; hwdep->exclusive = 1; hwdep->ops.open = hda_hwdep_open; @@ -117,5 +139,215 @@ int __devinit snd_hda_create_hwdep(struct hda_codec *codec) hwdep->ops.ioctl_compat = hda_hwdep_ioctl_compat; #endif + snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32); + snd_array_init(&codec->hints, sizeof(char *), 32); + return 0; } + +#ifdef CONFIG_SND_HDA_RECONFIG + +/* + * sysfs interface + */ + +static int clear_codec(struct hda_codec *codec) +{ + snd_hda_codec_reset(codec); + clear_hwdep_elements(codec); + return 0; +} + +static int reconfig_codec(struct hda_codec *codec) +{ + int err; + + snd_printk(KERN_INFO "hda-codec: reconfiguring\n"); + snd_hda_codec_reset(codec); + err = snd_hda_codec_configure(codec); + if (err < 0) + return err; + /* rebuild PCMs */ + err = snd_hda_codec_build_pcms(codec); + if (err < 0) + return err; + /* rebuild mixers */ + err = snd_hda_codec_build_controls(codec); + if (err < 0) + return err; + return 0; +} + +/* + * allocate a string at most len chars, and remove the trailing EOL + */ +static char *kstrndup_noeol(const char *src, size_t len) +{ + char *s = kstrndup(src, len, GFP_KERNEL); + char *p; + if (!s) + return NULL; + p = strchr(s, '\n'); + if (p) + *p = 0; + return s; +} + +#define CODEC_INFO_SHOW(type) \ +static ssize_t type##_show(struct device *dev, \ + struct device_attribute *attr, \ + char *buf) \ +{ \ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); \ + struct hda_codec *codec = hwdep->private_data; \ + return sprintf(buf, "0x%x\n", codec->type); \ +} + +#define CODEC_INFO_STR_SHOW(type) \ +static ssize_t type##_show(struct device *dev, \ + struct device_attribute *attr, \ + char *buf) \ +{ \ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); \ + struct hda_codec *codec = hwdep->private_data; \ + return sprintf(buf, "%s\n", \ + codec->type ? codec->type : ""); \ +} + +CODEC_INFO_SHOW(vendor_id); +CODEC_INFO_SHOW(subsystem_id); +CODEC_INFO_SHOW(revision_id); +CODEC_INFO_SHOW(afg); +CODEC_INFO_SHOW(mfg); +CODEC_INFO_STR_SHOW(name); +CODEC_INFO_STR_SHOW(modelname); + +#define CODEC_INFO_STORE(type) \ +static ssize_t type##_store(struct device *dev, \ + struct device_attribute *attr, \ + const char *buf, size_t count) \ +{ \ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); \ + struct hda_codec *codec = hwdep->private_data; \ + char *after; \ + codec->type = simple_strtoul(buf, &after, 0); \ + return count; \ +} + +#define CODEC_INFO_STR_STORE(type) \ +static ssize_t type##_store(struct device *dev, \ + struct device_attribute *attr, \ + const char *buf, size_t count) \ +{ \ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); \ + struct hda_codec *codec = hwdep->private_data; \ + char *s = kstrndup_noeol(buf, 64); \ + if (!s) \ + return -ENOMEM; \ + kfree(codec->type); \ + codec->type = s; \ + return count; \ +} + +CODEC_INFO_STORE(vendor_id); +CODEC_INFO_STORE(subsystem_id); +CODEC_INFO_STORE(revision_id); +CODEC_INFO_STR_STORE(name); +CODEC_INFO_STR_STORE(modelname); + +#define CODEC_ACTION_STORE(type) \ +static ssize_t type##_store(struct device *dev, \ + struct device_attribute *attr, \ + const char *buf, size_t count) \ +{ \ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); \ + struct hda_codec *codec = hwdep->private_data; \ + int err = 0; \ + if (*buf) \ + err = type##_codec(codec); \ + return err < 0 ? err : count; \ +} + +CODEC_ACTION_STORE(reconfig); +CODEC_ACTION_STORE(clear); + +static ssize_t init_verbs_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + char *p; + struct hda_verb verb, *v; + + verb.nid = simple_strtoul(buf, &p, 0); + verb.verb = simple_strtoul(p, &p, 0); + verb.param = simple_strtoul(p, &p, 0); + if (!verb.nid || !verb.verb || !verb.param) + return -EINVAL; + v = snd_array_new(&codec->init_verbs); + if (!v) + return -ENOMEM; + *v = verb; + return count; +} + +static ssize_t hints_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + char *p; + char **hint; + + if (!*buf || isspace(*buf) || *buf == '#' || *buf == '\n') + return count; + p = kstrndup_noeol(buf, 1024); + if (!p) + return -ENOMEM; + hint = snd_array_new(&codec->hints); + if (!hint) { + kfree(p); + return -ENOMEM; + } + *hint = p; + return count; +} + +#define CODEC_ATTR_RW(type) \ + __ATTR(type, 0644, type##_show, type##_store) +#define CODEC_ATTR_RO(type) \ + __ATTR_RO(type) +#define CODEC_ATTR_WO(type) \ + __ATTR(type, 0200, NULL, type##_store) + +static struct device_attribute codec_attrs[] = { + CODEC_ATTR_RW(vendor_id), + CODEC_ATTR_RW(subsystem_id), + CODEC_ATTR_RW(revision_id), + CODEC_ATTR_RO(afg), + CODEC_ATTR_RO(mfg), + CODEC_ATTR_RW(name), + CODEC_ATTR_RW(modelname), + CODEC_ATTR_WO(init_verbs), + CODEC_ATTR_WO(hints), + CODEC_ATTR_WO(reconfig), + CODEC_ATTR_WO(clear), +}; + +/* + * create sysfs files on hwdep directory + */ +int snd_hda_hwdep_add_sysfs(struct hda_codec *codec) +{ + struct snd_hwdep *hwdep = codec->hwdep; + int i; + + for (i = 0; i < ARRAY_SIZE(codec_attrs); i++) + snd_add_device_sysfs_file(SNDRV_DEVICE_TYPE_HWDEP, hwdep->card, + hwdep->device, &codec_attrs[i]); + return 0; +} + +#endif /* CONFIG_SND_HDA_RECONFIG */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 35722ec920cb..f04de115ee11 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -58,6 +58,7 @@ static char *model[SNDRV_CARDS]; static int position_fix[SNDRV_CARDS]; static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; +static int probe_only[SNDRV_CARDS]; static int single_cmd; static int enable_msi; @@ -76,6 +77,8 @@ module_param_array(bdl_pos_adj, int, NULL, 0644); MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1)."); +module_param_array(probe_only, bool, NULL, 0444); +MODULE_PARM_DESC(probe_only, "Only probing and no codec initialization."); module_param(single_cmd, bool, 0444); MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs " "(for debugging only)."); @@ -83,7 +86,10 @@ module_param(enable_msi, int, 0444); MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)"); #ifdef CONFIG_SND_HDA_POWER_SAVE -/* power_save option is defined in hda_codec.c */ +static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; +module_param(power_save, int, 0644); +MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " + "(in second, 0 = disable)."); /* reset the HD-audio controller in power save mode. * this may give more power-saving, but will take longer time to @@ -292,6 +298,8 @@ enum { /* Define VIA HD Audio Device ID*/ #define VIA_HDAC_DEVICE_ID 0x3288 +/* HD Audio class code */ +#define PCI_CLASS_MULTIMEDIA_HD_AUDIO 0x0403 /* */ @@ -392,6 +400,7 @@ struct azx { unsigned int msi :1; unsigned int irq_pending_warned :1; unsigned int via_dmapos_patch :1; /* enable DMA-position fix for VIA */ + unsigned int probing :1; /* codec probing phase */ /* for debugging */ unsigned int last_cmd; /* last issued command (to sync) */ @@ -414,6 +423,7 @@ enum { AZX_DRIVER_ULI, AZX_DRIVER_NVIDIA, AZX_DRIVER_TERA, + AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -427,6 +437,7 @@ static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_ULI] = "HDA ULI M5461", [AZX_DRIVER_NVIDIA] = "HDA NVidia", [AZX_DRIVER_TERA] = "HDA Teradici", + [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; /* @@ -527,9 +538,9 @@ static void azx_free_cmd_io(struct azx *chip) } /* send a command */ -static int azx_corb_send_cmd(struct hda_codec *codec, u32 val) +static int azx_corb_send_cmd(struct hda_bus *bus, u32 val) { - struct azx *chip = codec->bus->private_data; + struct azx *chip = bus->private_data; unsigned int wp; /* add command to corb */ @@ -577,9 +588,9 @@ static void azx_update_rirb(struct azx *chip) } /* receive a response */ -static unsigned int azx_rirb_get_response(struct hda_codec *codec) +static unsigned int azx_rirb_get_response(struct hda_bus *bus) { - struct azx *chip = codec->bus->private_data; + struct azx *chip = bus->private_data; unsigned long timeout; again: @@ -596,7 +607,7 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) } if (time_after(jiffies, timeout)) break; - if (codec->bus->needs_damn_long_delay) + if (bus->needs_damn_long_delay) msleep(2); /* temporary workaround */ else { udelay(10); @@ -624,6 +635,14 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) goto again; } + if (chip->probing) { + /* If this critical timeout happens during the codec probing + * phase, this is likely an access to a non-existing codec + * slot. Better to return an error and reset the system. + */ + return -1; + } + snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, " "switching to single_cmd mode: last cmd=0x%08x\n", chip->last_cmd); @@ -646,9 +665,9 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) */ /* send a command */ -static int azx_single_send_cmd(struct hda_codec *codec, u32 val) +static int azx_single_send_cmd(struct hda_bus *bus, u32 val) { - struct azx *chip = codec->bus->private_data; + struct azx *chip = bus->private_data; int timeout = 50; while (timeout--) { @@ -671,9 +690,9 @@ static int azx_single_send_cmd(struct hda_codec *codec, u32 val) } /* receive a response */ -static unsigned int azx_single_get_response(struct hda_codec *codec) +static unsigned int azx_single_get_response(struct hda_bus *bus) { - struct azx *chip = codec->bus->private_data; + struct azx *chip = bus->private_data; int timeout = 50; while (timeout--) { @@ -696,38 +715,29 @@ static unsigned int azx_single_get_response(struct hda_codec *codec) */ /* send a command */ -static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, - unsigned int para) +static int azx_send_cmd(struct hda_bus *bus, unsigned int val) { - struct azx *chip = codec->bus->private_data; - u32 val; - - val = (u32)(codec->addr & 0x0f) << 28; - val |= (u32)direct << 27; - val |= (u32)nid << 20; - val |= verb << 8; - val |= para; - chip->last_cmd = val; + struct azx *chip = bus->private_data; + chip->last_cmd = val; if (chip->single_cmd) - return azx_single_send_cmd(codec, val); + return azx_single_send_cmd(bus, val); else - return azx_corb_send_cmd(codec, val); + return azx_corb_send_cmd(bus, val); } /* get a response */ -static unsigned int azx_get_response(struct hda_codec *codec) +static unsigned int azx_get_response(struct hda_bus *bus) { - struct azx *chip = codec->bus->private_data; + struct azx *chip = bus->private_data; if (chip->single_cmd) - return azx_single_get_response(codec); + return azx_single_get_response(bus); else - return azx_rirb_get_response(codec); + return azx_rirb_get_response(bus); } #ifdef CONFIG_SND_HDA_POWER_SAVE -static void azx_power_notify(struct hda_codec *codec); +static void azx_power_notify(struct hda_bus *bus); #endif /* reset codec link */ @@ -1184,6 +1194,28 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) return 0; } +/* + * Probe the given codec address + */ +static int probe_codec(struct azx *chip, int addr) +{ + unsigned int cmd = (addr << 28) | (AC_NODE_ROOT << 20) | + (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; + unsigned int res; + + chip->probing = 1; + azx_send_cmd(chip->bus, cmd); + res = azx_get_response(chip->bus); + chip->probing = 0; + if (res == -1) + return -EIO; + snd_printdd("hda_intel: codec #%d probed OK\n", addr); + return 0; +} + +static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, + struct hda_pcm *cpcm); +static void azx_stop_chip(struct azx *chip); /* * Codec initialization @@ -1194,21 +1226,13 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = { [AZX_DRIVER_TERA] = 1, }; -/* number of slots to probe as default - * this can be different from azx_max_codecs[] -- e.g. some boards - * report wrongly the non-existing 4th slot availability - */ -static unsigned int azx_default_codecs[AZX_NUM_DRIVERS] __devinitdata = { - [AZX_DRIVER_ICH] = 3, - [AZX_DRIVER_ATI] = 3, -}; - static int __devinit azx_codec_create(struct azx *chip, const char *model, - unsigned int codec_probe_mask) + unsigned int codec_probe_mask, + int no_init) { struct hda_bus_template bus_temp; - int c, codecs, audio_codecs, err; - int def_slots, max_slots; + int c, codecs, err; + int max_slots; memset(&bus_temp, 0, sizeof(bus_temp)); bus_temp.private_data = chip; @@ -1216,7 +1240,9 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, bus_temp.pci = chip->pci; bus_temp.ops.command = azx_send_cmd; bus_temp.ops.get_response = azx_get_response; + bus_temp.ops.attach_pcm = azx_attach_pcm_stream; #ifdef CONFIG_SND_HDA_POWER_SAVE + bus_temp.power_save = &power_save; bus_temp.ops.pm_notify = azx_power_notify; #endif @@ -1227,33 +1253,43 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, if (chip->driver_type == AZX_DRIVER_NVIDIA) chip->bus->needs_damn_long_delay = 1; - codecs = audio_codecs = 0; + codecs = 0; max_slots = azx_max_codecs[chip->driver_type]; if (!max_slots) max_slots = AZX_MAX_CODECS; - def_slots = azx_default_codecs[chip->driver_type]; - if (!def_slots) - def_slots = max_slots; - for (c = 0; c < def_slots; c++) { + + /* First try to probe all given codec slots */ + for (c = 0; c < max_slots; c++) { + if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { + if (probe_codec(chip, c) < 0) { + /* Some BIOSen give you wrong codec addresses + * that don't exist + */ + snd_printk(KERN_WARNING + "hda_intel: Codec #%d probe error; " + "disabling it...\n", c); + chip->codec_mask &= ~(1 << c); + /* More badly, accessing to a non-existing + * codec often screws up the controller chip, + * and distrubs the further communications. + * Thus if an error occurs during probing, + * better to reset the controller chip to + * get back to the sanity state. + */ + azx_stop_chip(chip); + azx_init_chip(chip); + } + } + } + + /* Then create codec instances */ + for (c = 0; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { struct hda_codec *codec; - err = snd_hda_codec_new(chip->bus, c, &codec); + err = snd_hda_codec_new(chip->bus, c, !no_init, &codec); if (err < 0) continue; codecs++; - if (codec->afg) - audio_codecs++; - } - } - if (!audio_codecs) { - /* probe additional slots if no codec is found */ - for (; c < max_slots; c++) { - if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { - err = snd_hda_codec_new(chip->bus, c, NULL); - if (err < 0) - continue; - codecs++; - } } } if (!codecs) { @@ -1722,111 +1758,59 @@ static struct snd_pcm_ops azx_pcm_ops = { static void azx_pcm_free(struct snd_pcm *pcm) { - kfree(pcm->private_data); + struct azx_pcm *apcm = pcm->private_data; + if (apcm) { + apcm->chip->pcm[pcm->device] = NULL; + kfree(apcm); + } } -static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, - struct hda_pcm *cpcm) +static int +azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, + struct hda_pcm *cpcm) { - int err; + struct azx *chip = bus->private_data; struct snd_pcm *pcm; struct azx_pcm *apcm; + int pcm_dev = cpcm->device; + int s, err; - /* if no substreams are defined for both playback and capture, - * it's just a placeholder. ignore it. - */ - if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams) - return 0; - - if (snd_BUG_ON(!cpcm->name)) + if (pcm_dev >= AZX_MAX_PCMS) { + snd_printk(KERN_ERR SFX "Invalid PCM device number %d\n", + pcm_dev); return -EINVAL; - - err = snd_pcm_new(chip->card, cpcm->name, cpcm->device, - cpcm->stream[0].substreams, - cpcm->stream[1].substreams, + } + if (chip->pcm[pcm_dev]) { + snd_printk(KERN_ERR SFX "PCM %d already exists\n", pcm_dev); + return -EBUSY; + } + err = snd_pcm_new(chip->card, cpcm->name, pcm_dev, + cpcm->stream[SNDRV_PCM_STREAM_PLAYBACK].substreams, + cpcm->stream[SNDRV_PCM_STREAM_CAPTURE].substreams, &pcm); if (err < 0) return err; strcpy(pcm->name, cpcm->name); - apcm = kmalloc(sizeof(*apcm), GFP_KERNEL); + apcm = kzalloc(sizeof(*apcm), GFP_KERNEL); if (apcm == NULL) return -ENOMEM; apcm->chip = chip; apcm->codec = codec; - apcm->hinfo[0] = &cpcm->stream[0]; - apcm->hinfo[1] = &cpcm->stream[1]; pcm->private_data = apcm; pcm->private_free = azx_pcm_free; - if (cpcm->stream[0].substreams) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &azx_pcm_ops); - if (cpcm->stream[1].substreams) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &azx_pcm_ops); + if (cpcm->pcm_type == HDA_PCM_TYPE_MODEM) + pcm->dev_class = SNDRV_PCM_CLASS_MODEM; + chip->pcm[pcm_dev] = pcm; + cpcm->pcm = pcm; + for (s = 0; s < 2; s++) { + apcm->hinfo[s] = &cpcm->stream[s]; + if (cpcm->stream[s].substreams) + snd_pcm_set_ops(pcm, s, &azx_pcm_ops); + } + /* buffer pre-allocation */ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(chip->pci), 1024 * 64, 32 * 1024 * 1024); - chip->pcm[cpcm->device] = pcm; - return 0; -} - -static int __devinit azx_pcm_create(struct azx *chip) -{ - static const char *dev_name[HDA_PCM_NTYPES] = { - "Audio", "SPDIF", "HDMI", "Modem" - }; - /* starting device index for each PCM type */ - static int dev_idx[HDA_PCM_NTYPES] = { - [HDA_PCM_TYPE_AUDIO] = 0, - [HDA_PCM_TYPE_SPDIF] = 1, - [HDA_PCM_TYPE_HDMI] = 3, - [HDA_PCM_TYPE_MODEM] = 6 - }; - /* normal audio device indices; not linear to keep compatibility */ - static int audio_idx[4] = { 0, 2, 4, 5 }; - struct hda_codec *codec; - int c, err; - int num_devs[HDA_PCM_NTYPES]; - - err = snd_hda_build_pcms(chip->bus); - if (err < 0) - return err; - - /* create audio PCMs */ - memset(num_devs, 0, sizeof(num_devs)); - list_for_each_entry(codec, &chip->bus->codec_list, list) { - for (c = 0; c < codec->num_pcms; c++) { - struct hda_pcm *cpcm = &codec->pcm_info[c]; - int type = cpcm->pcm_type; - switch (type) { - case HDA_PCM_TYPE_AUDIO: - if (num_devs[type] >= ARRAY_SIZE(audio_idx)) { - snd_printk(KERN_WARNING - "Too many audio devices\n"); - continue; - } - cpcm->device = audio_idx[num_devs[type]]; - break; - case HDA_PCM_TYPE_SPDIF: - case HDA_PCM_TYPE_HDMI: - case HDA_PCM_TYPE_MODEM: - if (num_devs[type]) { - snd_printk(KERN_WARNING - "%s already defined\n", - dev_name[type]); - continue; - } - cpcm->device = dev_idx[type]; - break; - default: - snd_printk(KERN_WARNING - "Invalid PCM type %d\n", type); - continue; - } - num_devs[type]++; - err = create_codec_pcm(chip, codec, cpcm); - if (err < 0) - return err; - } - } return 0; } @@ -1903,13 +1887,13 @@ static void azx_stop_chip(struct azx *chip) #ifdef CONFIG_SND_HDA_POWER_SAVE /* power-up/down the controller */ -static void azx_power_notify(struct hda_codec *codec) +static void azx_power_notify(struct hda_bus *bus) { - struct azx *chip = codec->bus->private_data; + struct azx *chip = bus->private_data; struct hda_codec *c; int power_on = 0; - list_for_each_entry(c, &codec->bus->codec_list, list) { + list_for_each_entry(c, &bus->codec_list, list) { if (c->power_on) { power_on = 1; break; @@ -1926,6 +1910,18 @@ static void azx_power_notify(struct hda_codec *codec) /* * power management */ + +static int snd_hda_codecs_inuse(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + if (snd_hda_codec_needs_resume(codec)) + return 1; + } + return 0; +} + static int azx_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); @@ -1951,13 +1947,16 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) return 0; } +static int azx_resume_early(struct pci_dev *pci) +{ + return pci_restore_state(pci); +} + static int azx_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct azx *chip = card->private_data; - pci_set_power_state(pci, PCI_D0); - pci_restore_state(pci); if (pci_enable_device(pci) < 0) { printk(KERN_ERR "hda-intel: pci_enable_device failed, " "disabling device\n"); @@ -2095,6 +2094,10 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { SND_PCI_QUIRK(0x1014, 0x05b7, "Thinkpad Z60", 0x01), SND_PCI_QUIRK(0x17aa, 0x2010, "Thinkpad X/T/R60", 0x01), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X/T/R61", 0x01), + /* broken BIOS */ + SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01), + /* including bogus ALC268 in slot#2 that conflicts with ALC888 */ + SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01), {} }; @@ -2229,6 +2232,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->playback_streams = ATIHDMI_NUM_PLAYBACK; chip->capture_streams = ATIHDMI_NUM_CAPTURE; break; + case AZX_DRIVER_GENERIC: default: chip->playback_streams = ICH6_NUM_PLAYBACK; chip->capture_streams = ICH6_NUM_CAPTURE; @@ -2338,40 +2342,31 @@ static int __devinit azx_probe(struct pci_dev *pci, } err = azx_create(card, pci, dev, pci_id->driver_data, &chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; card->private_data = chip; /* create codec instances */ - err = azx_codec_create(chip, model[dev], probe_mask[dev]); - if (err < 0) { - snd_card_free(card); - return err; - } + err = azx_codec_create(chip, model[dev], probe_mask[dev], + probe_only[dev]); + if (err < 0) + goto out_free; /* create PCM streams */ - err = azx_pcm_create(chip); - if (err < 0) { - snd_card_free(card); - return err; - } + err = snd_hda_build_pcms(chip->bus); + if (err < 0) + goto out_free; /* create mixer controls */ err = azx_mixer_create(chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; snd_card_set_dev(card, &pci->dev); err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; pci_set_drvdata(pci, card); chip->running = 1; @@ -2380,6 +2375,9 @@ static int __devinit azx_probe(struct pci_dev *pci, dev++; return err; +out_free: + snd_card_free(card); + return err; } static void __devexit azx_remove(struct pci_dev *pci) @@ -2453,6 +2451,11 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA }, /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, + /* AMD Generic, PCI class code and Vendor ID for HD Audio */ + { PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_GENERIC }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); @@ -2465,6 +2468,7 @@ static struct pci_driver driver = { .remove = __devexit_p(azx_remove), #ifdef CONFIG_PM .suspend = azx_suspend, + .resume_early = azx_resume_early, .resume = azx_resume, #endif }; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 7957fefda730..6f2fe0f9fdd8 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -96,6 +96,8 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name); int snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char **slaves); +void snd_hda_codec_reset(struct hda_codec *codec); +int snd_hda_codec_configure(struct hda_codec *codec); /* amp value bits */ #define HDA_AMP_MUTE 0x80 @@ -282,6 +284,12 @@ int snd_hda_codec_proc_new(struct hda_codec *codec); static inline int snd_hda_codec_proc_new(struct hda_codec *codec) { return 0; } #endif +#define SND_PRINT_RATES_ADVISED_BUFSIZE 80 +void snd_print_pcm_rates(int pcm, char *buf, int buflen); + +#define SND_PRINT_BITS_ADVISED_BUFSIZE 16 +void snd_print_pcm_bits(int pcm, char *buf, int buflen); + /* * Misc */ @@ -364,17 +372,17 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, /* amp values */ #define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8)) #define AMP_IN_UNMUTE(idx) (0x7000 | ((idx)<<8)) -#define AMP_OUT_MUTE 0xb080 -#define AMP_OUT_UNMUTE 0xb000 -#define AMP_OUT_ZERO 0xb000 +#define AMP_OUT_MUTE 0xb080 +#define AMP_OUT_UNMUTE 0xb000 +#define AMP_OUT_ZERO 0xb000 /* pinctl values */ #define PIN_IN (AC_PINCTL_IN_EN) -#define PIN_VREFHIZ (AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ) +#define PIN_VREFHIZ (AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ) #define PIN_VREF50 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_50) -#define PIN_VREFGRD (AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD) +#define PIN_VREFGRD (AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD) #define PIN_VREF80 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_80) -#define PIN_VREF100 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_100) -#define PIN_OUT (AC_PINCTL_OUT_EN) +#define PIN_VREF100 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_100) +#define PIN_OUT (AC_PINCTL_OUT_EN) #define PIN_HP (AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN) #define PIN_HP_AMP (AC_PINCTL_HP_EN) @@ -393,10 +401,26 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); +int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl); +void snd_hda_ctls_clear(struct hda_codec *codec); + /* * hwdep interface */ +#ifdef CONFIG_SND_HDA_HWDEP int snd_hda_create_hwdep(struct hda_codec *codec); +#else +static inline int snd_hda_create_hwdep(struct hda_codec *codec) { return 0; } +#endif + +#ifdef CONFIG_SND_HDA_RECONFIG +int snd_hda_hwdep_add_sysfs(struct hda_codec *codec); +#else +static inline int snd_hda_hwdep_add_sysfs(struct hda_codec *codec) +{ + return 0; +} +#endif /* * power-management @@ -430,4 +454,66 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, #define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) #define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) +/* + * CEA Short Audio Descriptor data + */ +struct cea_sad { + int channels; + int format; /* (format == 0) indicates invalid SAD */ + int rates; + int sample_bits; /* for LPCM */ + int max_bitrate; /* for AC3...ATRAC */ + int profile; /* for WMAPRO */ +}; + +#define ELD_FIXED_BYTES 20 +#define ELD_MAX_MNL 16 +#define ELD_MAX_SAD 16 + +/* + * ELD: EDID Like Data + */ +struct hdmi_eld { + int eld_size; + int baseline_len; + int eld_ver; /* (eld_ver == 0) indicates invalid ELD */ + int cea_edid_ver; + char monitor_name[ELD_MAX_MNL + 1]; + int manufacture_id; + int product_id; + u64 port_id; + int support_hdcp; + int support_ai; + int conn_type; + int aud_synch_delay; + int spk_alloc; + int sad_count; + struct cea_sad sad[ELD_MAX_SAD]; +#ifdef CONFIG_PROC_FS + struct snd_info_entry *proc_entry; +#endif +}; + +int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid); +int snd_hdmi_get_eld(struct hdmi_eld *, struct hda_codec *, hda_nid_t); +void snd_hdmi_show_eld(struct hdmi_eld *eld); + +#ifdef CONFIG_PROC_FS +int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld); +void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld); +#else +static inline int snd_hda_eld_proc_new(struct hda_codec *codec, + struct hdmi_eld *eld) +{ + return 0; +} +static inline void snd_hda_eld_proc_free(struct hda_codec *codec, + struct hdmi_eld *eld) +{ +} +#endif + +#define SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE 80 +void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen); + #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h deleted file mode 100644 index dfbcfa88da44..000000000000 --- a/sound/pci/hda/hda_patch.h +++ /dev/null @@ -1,22 +0,0 @@ -/* - * HDA Patches - included by hda_codec.c - */ - -/* Realtek codecs */ -extern struct hda_codec_preset snd_hda_preset_realtek[]; -/* C-Media codecs */ -extern struct hda_codec_preset snd_hda_preset_cmedia[]; -/* Analog Devices codecs */ -extern struct hda_codec_preset snd_hda_preset_analog[]; -/* SigmaTel codecs */ -extern struct hda_codec_preset snd_hda_preset_sigmatel[]; -/* SiLabs 3054/3055 modem codecs */ -extern struct hda_codec_preset snd_hda_preset_si3054[]; -/* ATI HDMI codecs */ -extern struct hda_codec_preset snd_hda_preset_atihdmi[]; -/* Conexant audio codec */ -extern struct hda_codec_preset snd_hda_preset_conexant[]; -/* VIA codecs */ -extern struct hda_codec_preset snd_hda_preset_via[]; -/* NVIDIA HDMI codecs */ -extern struct hda_codec_preset snd_hda_preset_nvhdmi[]; diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index c39af986bff1..7ca66d654148 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -91,31 +91,21 @@ static void print_amp_vals(struct snd_info_buffer *buffer, static void print_pcm_rates(struct snd_info_buffer *buffer, unsigned int pcm) { - static unsigned int rates[] = { - 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200, - 96000, 176400, 192000, 384000 - }; - int i; + char buf[SND_PRINT_RATES_ADVISED_BUFSIZE]; pcm &= AC_SUPPCM_RATES; snd_iprintf(buffer, " rates [0x%x]:", pcm); - for (i = 0; i < ARRAY_SIZE(rates); i++) - if (pcm & (1 << i)) - snd_iprintf(buffer, " %d", rates[i]); - snd_iprintf(buffer, "\n"); + snd_print_pcm_rates(pcm, buf, sizeof(buf)); + snd_iprintf(buffer, "%s\n", buf); } static void print_pcm_bits(struct snd_info_buffer *buffer, unsigned int pcm) { - static unsigned int bits[] = { 8, 16, 20, 24, 32 }; - int i; + char buf[SND_PRINT_BITS_ADVISED_BUFSIZE]; - pcm = (pcm >> 16) & 0xff; - snd_iprintf(buffer, " bits [0x%x]:", pcm); - for (i = 0; i < ARRAY_SIZE(bits); i++) - if (pcm & (1 << i)) - snd_iprintf(buffer, " %d", bits[i]); - snd_iprintf(buffer, "\n"); + snd_iprintf(buffer, " bits [0x%x]:", (pcm >> 16) & 0xff); + snd_print_pcm_bits(pcm, buf, sizeof(buf)); + snd_iprintf(buffer, "%s\n", buf); } static void print_pcm_formats(struct snd_info_buffer *buffer, @@ -145,32 +135,6 @@ static void print_pcm_caps(struct snd_info_buffer *buffer, print_pcm_formats(buffer, stream); } -static const char *get_jack_location(u32 cfg) -{ - static char *bases[7] = { - "N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom", - }; - static unsigned char specials_idx[] = { - 0x07, 0x08, - 0x17, 0x18, 0x19, - 0x37, 0x38 - }; - static char *specials[] = { - "Rear Panel", "Drive Bar", - "Riser", "HDMI", "ATAPI", - "Mobile-In", "Mobile-Out" - }; - int i; - cfg = (cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT; - if ((cfg & 0x0f) < 7) - return bases[cfg & 0x0f]; - for (i = 0; i < ARRAY_SIZE(specials_idx); i++) { - if (cfg == specials_idx[i]) - return specials[i]; - } - return "UNKNOWN"; -} - static const char *get_jack_connection(u32 cfg) { static char *names[16] = { @@ -206,13 +170,6 @@ static void print_pin_caps(struct snd_info_buffer *buffer, int *supports_vref) { static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" }; - static char *jack_types[16] = { - "Line Out", "Speaker", "HP Out", "CD", - "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", - "Line In", "Aux", "Mic", "Telephony", - "SPDIF In", "Digitial In", "Reserved", "Other" - }; - static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; unsigned int caps, val; caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); @@ -274,9 +231,9 @@ static void print_pin_caps(struct snd_info_buffer *buffer, caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps, jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT], - jack_types[(caps & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT], - jack_locations[(caps >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3], - get_jack_location(caps)); + snd_hda_get_jack_type(caps), + snd_hda_get_jack_connectivity(caps), + snd_hda_get_jack_location(caps)); snd_iprintf(buffer, " Conn = %s, Color = %s\n", get_jack_connection(caps), get_jack_color(caps)); @@ -457,17 +414,6 @@ static void print_conn_list(struct snd_info_buffer *buffer, } } -static void print_realtek_coef(struct snd_info_buffer *buffer, - struct hda_codec *codec, hda_nid_t nid) -{ - int coeff = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PROC_COEF, 0); - snd_iprintf(buffer, " Processing Coefficient: 0x%02x\n", coeff); - coeff = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_COEF_INDEX, 0); - snd_iprintf(buffer, " Coefficient Index: 0x%02x\n", coeff); -} - static void print_gpio(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { @@ -500,12 +446,13 @@ static void print_gpio(struct snd_info_buffer *buffer, for (i = 0; i < max; ++i) snd_iprintf(buffer, " IO[%d]: enable=%d, dir=%d, wake=%d, " - "sticky=%d, data=%d\n", i, + "sticky=%d, data=%d, unsol=%d\n", i, (enable & (1<<i)) ? 1 : 0, (direction & (1<<i)) ? 1 : 0, (wake & (1<<i)) ? 1 : 0, (sticky & (1<<i)) ? 1 : 0, - (data & (1<<i)) ? 1 : 0); + (data & (1<<i)) ? 1 : 0, + (unsol & (1<<i)) ? 1 : 0); /* FIXME: add GPO and GPI pin information */ } @@ -513,12 +460,11 @@ static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct hda_codec *codec = entry->private_data; - char buf[32]; hda_nid_t nid; int i, nodes; - snd_hda_get_codec_name(codec, buf, sizeof(buf)); - snd_iprintf(buffer, "Codec: %s\n", buf); + snd_iprintf(buffer, "Codec: %s\n", + codec->name ? codec->name : "Not Set"); snd_iprintf(buffer, "Address: %d\n", codec->addr); snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); @@ -547,6 +493,8 @@ static void print_codec_info(struct snd_info_entry *entry, } print_gpio(buffer, codec, codec->afg); + if (codec->proc_widget_hook) + codec->proc_widget_hook(buffer, codec, codec->afg); for (i = 0; i < nodes; i++, nid++) { unsigned int wid_caps = @@ -649,9 +597,8 @@ static void print_codec_info(struct snd_info_entry *entry, if (wid_caps & AC_WCAP_PROC_WID) print_proc_caps(buffer, codec, nid); - /* NID 0x20 == Realtek Define Registers */ - if (codec->vendor_id == 0x10ec && nid == 0x20) - print_realtek_coef(buffer, codec, nid); + if (codec->proc_widget_hook) + codec->proc_widget_hook(buffer, codec, nid); } snd_hda_power_down(codec); } diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 686c77491dea..26247cfe749d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -27,7 +27,6 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" struct ad198x_spec { struct snd_kcontrol_new *mixers[5]; @@ -67,8 +66,7 @@ struct ad198x_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; - unsigned int num_kctl_alloc, num_kctl_used; - struct snd_kcontrol_new *kctl_alloc; + struct snd_array kctls; struct hda_input_mux private_imux; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; @@ -154,6 +152,8 @@ static const char *ad_slave_sws[] = { NULL }; +static void ad198x_free_kctls(struct hda_codec *codec); + static int ad198x_build_controls(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; @@ -202,6 +202,7 @@ static int ad198x_build_controls(struct hda_codec *codec) return err; } + ad198x_free_kctls(codec); /* no longer needed */ return 0; } @@ -375,16 +376,27 @@ static int ad198x_build_pcms(struct hda_codec *codec) return 0; } -static void ad198x_free(struct hda_codec *codec) +static void ad198x_free_kctls(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; - unsigned int i; - if (spec->kctl_alloc) { - for (i = 0; i < spec->num_kctl_used; i++) - kfree(spec->kctl_alloc[i].name); - kfree(spec->kctl_alloc); + if (spec->kctls.list) { + struct snd_kcontrol_new *kctl = spec->kctls.list; + int i; + for (i = 0; i < spec->kctls.used; i++) + kfree(kctl[i].name); } + snd_array_free(&spec->kctls); +} + +static void ad198x_free(struct hda_codec *codec) +{ + struct ad198x_spec *spec = codec->spec; + + if (!spec) + return; + + ad198x_free_kctls(codec); kfree(codec->spec); } @@ -629,6 +641,36 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw), HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "External Amplifier", + .info = ad198x_eapd_info, + .get = ad198x_eapd_get, + .put = ad198x_eapd_put, + .private_value = 0x1b | (1 << 8), /* port-D, inversed */ + }, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1986a_samsung_mixers[] = { + HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), + HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), @@ -917,6 +959,7 @@ enum { AD1986A_LAPTOP_EAPD, AD1986A_LAPTOP_AUTOMUTE, AD1986A_ULTRA, + AD1986A_SAMSUNG, AD1986A_MODELS }; @@ -927,6 +970,7 @@ static const char *ad1986a_models[AD1986A_MODELS] = { [AD1986A_LAPTOP_EAPD] = "laptop-eapd", [AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute", [AD1986A_ULTRA] = "ultra", + [AD1986A_SAMSUNG] = "samsung", }; static struct snd_pci_quirk ad1986a_cfg_tbl[] = { @@ -949,9 +993,9 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x144d, 0xc024, "Samsung R65", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x144d, 0xc026, "Samsung X11", AD1986A_LAPTOP_EAPD), + SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_SAMSUNG), + SND_PCI_QUIRK(0x144d, 0xc024, "Samsung R65", AD1986A_SAMSUNG), + SND_PCI_QUIRK(0x144d, 0xc026, "Samsung X11", AD1986A_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA), SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP), @@ -1033,6 +1077,17 @@ static int patch_ad1986a(struct hda_codec *codec) break; case AD1986A_LAPTOP_EAPD: spec->mixers[0] = ad1986a_laptop_eapd_mixers; + spec->num_init_verbs = 2; + spec->init_verbs[1] = ad1986a_eapd_init_verbs; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = ad1986a_laptop_dac_nids; + if (!is_jack_available(codec, 0x25)) + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1986a_laptop_eapd_capture_source; + break; + case AD1986A_SAMSUNG: + spec->mixers[0] = ad1986a_samsung_mixers; spec->num_init_verbs = 3; spec->init_verbs[1] = ad1986a_eapd_init_verbs; spec->init_verbs[2] = ad1986a_automic_verbs; @@ -2452,9 +2507,6 @@ static struct hda_amp_list ad1988_loopbacks[] = { * Automatic parse of I/O pins from the BIOS configuration */ -#define NUM_CONTROL_ALLOC 32 -#define NUM_VERB_ALLOC 32 - enum { AD_CTL_WIDGET_VOL, AD_CTL_WIDGET_MUTE, @@ -2472,27 +2524,15 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name, { struct snd_kcontrol_new *knew; - if (spec->num_kctl_used >= spec->num_kctl_alloc) { - int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC; - - knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */ - if (! knew) - return -ENOMEM; - if (spec->kctl_alloc) { - memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc); - kfree(spec->kctl_alloc); - } - spec->kctl_alloc = knew; - spec->num_kctl_alloc = num; - } - - knew = &spec->kctl_alloc[spec->num_kctl_used]; + snd_array_init(&spec->kctls, sizeof(*knew), 32); + knew = snd_array_new(&spec->kctls); + if (!knew) + return -ENOMEM; *knew = ad1988_control_templates[type]; knew->name = kstrdup(name, GFP_KERNEL); if (! knew->name) return -ENOMEM; knew->private_value = val; - spec->num_kctl_used++; return 0; } @@ -2846,8 +2886,8 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_in_pin) spec->dig_in_nid = AD1988_SPDIF_IN; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; spec->init_verbs[spec->num_init_verbs++] = ad1988_6stack_init_verbs; @@ -3861,6 +3901,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = { static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), @@ -4267,7 +4308,7 @@ static int patch_ad1882(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_analog[] = { +static struct hda_codec_preset snd_hda_preset_analog[] = { { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a }, { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a }, @@ -4285,3 +4326,26 @@ struct hda_codec_preset snd_hda_preset_analog[] = { { .id = 0x11d4989b, .name = "AD1989B", .patch = patch_ad1988 }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:11d4*"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Analog Devices HD-audio codec"); + +static struct hda_codec_preset_list analog_list = { + .preset = snd_hda_preset_analog, + .owner = THIS_MODULE, +}; + +static int __init patch_analog_init(void) +{ + return snd_hda_add_codec_preset(&analog_list); +} + +static void __exit patch_analog_exit(void) +{ + snd_hda_delete_codec_preset(&analog_list); +} + +module_init(patch_analog_init) +module_exit(patch_analog_exit) diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index ba61575983fd..233e4778bba9 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -27,7 +27,6 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" struct atihdmi_spec { struct hda_multi_out multiout; @@ -187,13 +186,40 @@ static int patch_atihdmi(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_atihdmi[] = { - { .id = 0x1002793c, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, - { .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, - { .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi }, - { .id = 0x1002aa01, .name = "ATI R6xx HDMI", .patch = patch_atihdmi }, +static struct hda_codec_preset snd_hda_preset_atihdmi[] = { + { .id = 0x1002793c, .name = "RS600 HDMI", .patch = patch_atihdmi }, + { .id = 0x10027919, .name = "RS600 HDMI", .patch = patch_atihdmi }, + { .id = 0x1002791a, .name = "RS690/780 HDMI", .patch = patch_atihdmi }, + { .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_atihdmi }, { .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_atihdmi }, - { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_atihdmi }, { .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_atihdmi }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:1002793c"); +MODULE_ALIAS("snd-hda-codec-id:10027919"); +MODULE_ALIAS("snd-hda-codec-id:1002791a"); +MODULE_ALIAS("snd-hda-codec-id:1002aa01"); +MODULE_ALIAS("snd-hda-codec-id:10951390"); +MODULE_ALIAS("snd-hda-codec-id:17e80047"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ATI HDMI HD-audio codec"); + +static struct hda_codec_preset_list atihdmi_list = { + .preset = snd_hda_preset_atihdmi, + .owner = THIS_MODULE, +}; + +static int __init patch_atihdmi_init(void) +{ + return snd_hda_add_codec_preset(&atihdmi_list); +} + +static void __exit patch_atihdmi_exit(void) +{ + snd_hda_delete_codec_preset(&atihdmi_list); +} + +module_init(patch_atihdmi_init) +module_exit(patch_atihdmi_exit) diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 6ef57fbfb6eb..f3ebe837f2d5 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -28,7 +28,6 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" #define NUM_PINS 11 @@ -736,8 +735,32 @@ static int patch_cmi9880(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_cmedia[] = { +static struct hda_codec_preset snd_hda_preset_cmedia[] = { { .id = 0x13f69880, .name = "CMI9880", .patch = patch_cmi9880 }, { .id = 0x434d4980, .name = "CMI9880", .patch = patch_cmi9880 }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:13f69880"); +MODULE_ALIAS("snd-hda-codec-id:434d4980"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("C-Media HD-audio codec"); + +static struct hda_codec_preset_list cmedia_list = { + .preset = snd_hda_preset_cmedia, + .owner = THIS_MODULE, +}; + +static int __init patch_cmedia_init(void) +{ + return snd_hda_add_codec_preset(&cmedia_list); +} + +static void __exit patch_cmedia_exit(void) +{ + snd_hda_delete_codec_preset(&cmedia_list); +} + +module_init(patch_cmedia_init) +module_exit(patch_cmedia_exit) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7c1eb23f0cec..b20e1cede00b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -27,7 +27,6 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" #define CXT_PIN_DIR_IN 0x00 #define CXT_PIN_DIR_OUT 0x01 @@ -86,8 +85,6 @@ struct conexant_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; - unsigned int num_kctl_alloc, num_kctl_used; - struct snd_kcontrol_new *kctl_alloc; struct hda_input_mux private_imux; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; @@ -344,15 +341,6 @@ static int conexant_init(struct hda_codec *codec) static void conexant_free(struct hda_codec *codec) { - struct conexant_spec *spec = codec->spec; - unsigned int i; - - if (spec->kctl_alloc) { - for (i = 0; i < spec->num_kctl_used; i++) - kfree(spec->kctl_alloc[i].name); - kfree(spec->kctl_alloc); - } - kfree(codec->spec); } @@ -1782,7 +1770,7 @@ static int patch_cxt5051(struct hda_codec *codec) /* */ -struct hda_codec_preset snd_hda_preset_conexant[] = { +static struct hda_codec_preset snd_hda_preset_conexant[] = { { .id = 0x14f15045, .name = "CX20549 (Venice)", .patch = patch_cxt5045 }, { .id = 0x14f15047, .name = "CX20551 (Waikiki)", @@ -1791,3 +1779,28 @@ struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5051 }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:14f15045"); +MODULE_ALIAS("snd-hda-codec-id:14f15047"); +MODULE_ALIAS("snd-hda-codec-id:14f15051"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Conexant HD-audio codec"); + +static struct hda_codec_preset_list conexant_list = { + .preset = snd_hda_preset_conexant, + .owner = THIS_MODULE, +}; + +static int __init patch_conexant_init(void) +{ + return snd_hda_add_codec_preset(&conexant_list); +} + +static void __exit patch_conexant_exit(void) +{ + snd_hda_delete_codec_preset(&conexant_list); +} + +module_init(patch_conexant_init) +module_exit(patch_conexant_exit) diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c new file mode 100644 index 000000000000..3564f4e4b74c --- /dev/null +++ b/sound/pci/hda/patch_intelhdmi.c @@ -0,0 +1,711 @@ +/* + * + * patch_intelhdmi.c - Patch for Intel HDMI codecs + * + * Copyright(c) 2008 Intel Corporation. All rights reserved. + * + * Authors: + * Jiang Zhe <zhe.jiang@intel.com> + * Wu Fengguang <wfg@linux.intel.com> + * + * Maintained by: + * Wu Fengguang <wfg@linux.intel.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY + * or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License + * for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software Foundation, + * Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <sound/core.h> +#include "hda_codec.h" +#include "hda_local.h" + +#define CVT_NID 0x02 /* audio converter */ +#define PIN_NID 0x03 /* HDMI output pin */ + +#define INTEL_HDMI_EVENT_TAG 0x08 + +struct intel_hdmi_spec { + struct hda_multi_out multiout; + struct hda_pcm pcm_rec; + struct hdmi_eld sink_eld; +}; + +static struct hda_verb pinout_enable_verb[] = { + {PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {} /* terminator */ +}; + +static struct hda_verb pinout_disable_verb[] = { + {PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00}, + {} +}; + +static struct hda_verb unsolicited_response_verb[] = { + {PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | + INTEL_HDMI_EVENT_TAG}, + {} +}; + +static struct hda_verb def_chan_map[] = { + {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x00}, + {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x11}, + {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x22}, + {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x33}, + {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x44}, + {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x55}, + {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x66}, + {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x77}, + {} +}; + + +struct hdmi_audio_infoframe { + u8 type; /* 0x84 */ + u8 ver; /* 0x01 */ + u8 len; /* 0x0a */ + + u8 checksum; /* PB0 */ + u8 CC02_CT47; /* CC in bits 0:2, CT in 4:7 */ + u8 SS01_SF24; + u8 CXT04; + u8 CA; + u8 LFEPBL01_LSV36_DM_INH7; + u8 reserved[5]; /* PB6 - PB10 */ +}; + +/* + * CEA speaker placement: + * + * FLH FCH FRH + * FLW FL FLC FC FRC FR FRW + * + * LFE + * TC + * + * RL RLC RC RRC RR + * + * The Left/Right Surround channel _notions_ LS/RS in SMPTE 320M corresponds to + * CEA RL/RR; The SMPTE channel _assignment_ C/LFE is swapped to CEA LFE/FC. + */ +enum cea_speaker_placement { + FL = (1 << 0), /* Front Left */ + FC = (1 << 1), /* Front Center */ + FR = (1 << 2), /* Front Right */ + FLC = (1 << 3), /* Front Left Center */ + FRC = (1 << 4), /* Front Right Center */ + RL = (1 << 5), /* Rear Left */ + RC = (1 << 6), /* Rear Center */ + RR = (1 << 7), /* Rear Right */ + RLC = (1 << 8), /* Rear Left Center */ + RRC = (1 << 9), /* Rear Right Center */ + LFE = (1 << 10), /* Low Frequency Effect */ + FLW = (1 << 11), /* Front Left Wide */ + FRW = (1 << 12), /* Front Right Wide */ + FLH = (1 << 13), /* Front Left High */ + FCH = (1 << 14), /* Front Center High */ + FRH = (1 << 15), /* Front Right High */ + TC = (1 << 16), /* Top Center */ +}; + +/* + * ELD SA bits in the CEA Speaker Allocation data block + */ +static int eld_speaker_allocation_bits[] = { + [0] = FL | FR, + [1] = LFE, + [2] = FC, + [3] = RL | RR, + [4] = RC, + [5] = FLC | FRC, + [6] = RLC | RRC, + /* the following are not defined in ELD yet */ + [7] = FLW | FRW, + [8] = FLH | FRH, + [9] = TC, + [10] = FCH, +}; + +struct cea_channel_speaker_allocation { + int ca_index; + int speakers[8]; + + /* derived values, just for convenience */ + int channels; + int spk_mask; +}; + +/* + * This is an ordered list! + * + * The preceding ones have better chances to be selected by + * hdmi_setup_channel_allocation(). + */ +static struct cea_channel_speaker_allocation channel_allocations[] = { +/* channel: 8 7 6 5 4 3 2 1 */ +{ .ca_index = 0x00, .speakers = { 0, 0, 0, 0, 0, 0, FR, FL } }, + /* 2.1 */ +{ .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } }, + /* Dolby Surround */ +{ .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } }, +{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, +{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, +{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, +{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, +{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, +{ .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x09, .speakers = { 0, 0, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, + /* 5.1 */ +{ .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } }, + /* 6.1 */ +{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } }, + /* 7.1 */ +{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } }, +{ .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } }, +{ .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } }, +{ .ca_index = 0x17, .speakers = { FRC, FLC, 0, 0, FC, LFE, FR, FL } }, +{ .ca_index = 0x18, .speakers = { FRC, FLC, 0, RC, 0, 0, FR, FL } }, +{ .ca_index = 0x19, .speakers = { FRC, FLC, 0, RC, 0, LFE, FR, FL } }, +{ .ca_index = 0x1a, .speakers = { FRC, FLC, 0, RC, FC, 0, FR, FL } }, +{ .ca_index = 0x1b, .speakers = { FRC, FLC, 0, RC, FC, LFE, FR, FL } }, +{ .ca_index = 0x1c, .speakers = { FRC, FLC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x1d, .speakers = { FRC, FLC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x1e, .speakers = { FRC, FLC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x1f, .speakers = { FRC, FLC, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x20, .speakers = { 0, FCH, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x21, .speakers = { 0, FCH, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x22, .speakers = { TC, 0, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x23, .speakers = { TC, 0, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x24, .speakers = { FRH, FLH, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x25, .speakers = { FRH, FLH, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x26, .speakers = { FRW, FLW, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x27, .speakers = { FRW, FLW, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x28, .speakers = { TC, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x29, .speakers = { TC, RC, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x2a, .speakers = { FCH, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x2b, .speakers = { FCH, RC, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x2c, .speakers = { TC, FCH, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x2d, .speakers = { TC, FCH, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x2e, .speakers = { FRH, FLH, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x2f, .speakers = { FRH, FLH, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x30, .speakers = { FRW, FLW, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } }, +}; + +/* + * HDMI routines + */ + +#ifdef BE_PARANOID +static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t nid, + int *packet_index, int *byte_index) +{ + int val; + + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_INDEX, 0); + + *packet_index = val >> 5; + *byte_index = val & 0x1f; +} +#endif + +static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t nid, + int packet_index, int byte_index) +{ + int val; + + val = (packet_index << 5) | (byte_index & 0x1f); + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); +} + +static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid, + unsigned char val) +{ + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val); +} + +static void hdmi_enable_output(struct hda_codec *codec) +{ + /* Enable Audio InfoFrame Transmission */ + hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); + snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, + AC_DIPXMIT_BEST); + /* Unmute */ + if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, PIN_NID, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + /* Enable pin out */ + snd_hda_sequence_write(codec, pinout_enable_verb); +} + +static void hdmi_disable_output(struct hda_codec *codec) +{ + snd_hda_sequence_write(codec, pinout_disable_verb); + if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, PIN_NID, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + /* + * FIXME: noises may arise when playing music after reloading the + * kernel module, until the next X restart or monitor repower. + */ +} + +static int hdmi_get_channel_count(struct hda_codec *codec) +{ + return 1 + snd_hda_codec_read(codec, CVT_NID, 0, + AC_VERB_GET_CVT_CHAN_COUNT, 0); +} + +static void hdmi_set_channel_count(struct hda_codec *codec, int chs) +{ + snd_hda_codec_write(codec, CVT_NID, 0, + AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); + + if (chs != hdmi_get_channel_count(codec)) + snd_printd(KERN_INFO "HDMI channel count: expect %d, get %d\n", + chs, hdmi_get_channel_count(codec)); +} + +static void hdmi_debug_channel_mapping(struct hda_codec *codec) +{ +#ifdef CONFIG_SND_DEBUG_VERBOSE + int i; + int slot; + + for (i = 0; i < 8; i++) { + slot = snd_hda_codec_read(codec, CVT_NID, 0, + AC_VERB_GET_HDMI_CHAN_SLOT, i); + printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", + slot >> 4, slot & 0x7); + } +#endif +} + +static void hdmi_parse_eld(struct hda_codec *codec) +{ + struct intel_hdmi_spec *spec = codec->spec; + struct hdmi_eld *eld = &spec->sink_eld; + + if (!snd_hdmi_get_eld(eld, codec, PIN_NID)) + snd_hdmi_show_eld(eld); +} + + +/* + * Audio InfoFrame routines + */ + +static void hdmi_debug_dip_size(struct hda_codec *codec) +{ +#ifdef CONFIG_SND_DEBUG_VERBOSE + int i; + int size; + + size = snd_hdmi_get_eld_size(codec, PIN_NID); + printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size); + + for (i = 0; i < 8; i++) { + size = snd_hda_codec_read(codec, PIN_NID, 0, + AC_VERB_GET_HDMI_DIP_SIZE, i); + printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size); + } +#endif +} + +static void hdmi_clear_dip_buffers(struct hda_codec *codec) +{ +#ifdef BE_PARANOID + int i, j; + int size; + int pi, bi; + for (i = 0; i < 8; i++) { + size = snd_hda_codec_read(codec, PIN_NID, 0, + AC_VERB_GET_HDMI_DIP_SIZE, i); + if (size == 0) + continue; + + hdmi_set_dip_index(codec, PIN_NID, i, 0x0); + for (j = 1; j < 1000; j++) { + hdmi_write_dip_byte(codec, PIN_NID, 0x0); + hdmi_get_dip_index(codec, PIN_NID, &pi, &bi); + if (pi != i) + snd_printd(KERN_INFO "dip index %d: %d != %d\n", + bi, pi, i); + if (bi == 0) /* byte index wrapped around */ + break; + } + snd_printd(KERN_INFO + "HDMI: DIP GP[%d] buf reported size=%d, written=%d\n", + i, size, j); + } +#endif +} + +static void hdmi_fill_audio_infoframe(struct hda_codec *codec, + struct hdmi_audio_infoframe *ai) +{ + u8 *params = (u8 *)ai; + int i; + + hdmi_debug_dip_size(codec); + hdmi_clear_dip_buffers(codec); /* be paranoid */ + + hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); + for (i = 0; i < sizeof(ai); i++) + hdmi_write_dip_byte(codec, PIN_NID, params[i]); +} + +/* + * Compute derived values in channel_allocations[]. + */ +static void init_channel_allocations(void) +{ + int i, j; + struct cea_channel_speaker_allocation *p; + + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + p = channel_allocations + i; + p->channels = 0; + p->spk_mask = 0; + for (j = 0; j < ARRAY_SIZE(p->speakers); j++) + if (p->speakers[j]) { + p->channels++; + p->spk_mask |= p->speakers[j]; + } + } +} + +/* + * The transformation takes two steps: + * + * eld->spk_alloc => (eld_speaker_allocation_bits[]) => spk_mask + * spk_mask => (channel_allocations[]) => ai->CA + * + * TODO: it could select the wrong CA from multiple candidates. +*/ +static int hdmi_setup_channel_allocation(struct hda_codec *codec, + struct hdmi_audio_infoframe *ai) +{ + struct intel_hdmi_spec *spec = codec->spec; + struct hdmi_eld *eld = &spec->sink_eld; + int i; + int spk_mask = 0; + int channels = 1 + (ai->CC02_CT47 & 0x7); + char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; + + /* + * CA defaults to 0 for basic stereo audio + */ + if (!eld->eld_ver) + return 0; + if (!eld->spk_alloc) + return 0; + if (channels <= 2) + return 0; + + /* + * expand ELD's speaker allocation mask + * + * ELD tells the speaker mask in a compact(paired) form, + * expand ELD's notions to match the ones used by Audio InfoFrame. + */ + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (eld->spk_alloc & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + /* search for the first working match in the CA table */ + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if (channels == channel_allocations[i].channels && + (spk_mask & channel_allocations[i].spk_mask) == + channel_allocations[i].spk_mask) { + ai->CA = channel_allocations[i].ca_index; + break; + } + } + + snd_print_channel_allocation(eld->spk_alloc, buf, sizeof(buf)); + snd_printdd(KERN_INFO + "HDMI: select CA 0x%x for %d-channel allocation: %s\n", + ai->CA, channels, buf); + + return ai->CA; +} + +static void hdmi_setup_channel_mapping(struct hda_codec *codec, + struct hdmi_audio_infoframe *ai) +{ + if (!ai->CA) + return; + + /* + * TODO: adjust channel mapping if necessary + * ALSA sequence is front/surr/clfe/side? + */ + + snd_hda_sequence_write(codec, def_chan_map); + hdmi_debug_channel_mapping(codec); +} + + +static void hdmi_setup_audio_infoframe(struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct hdmi_audio_infoframe ai = { + .type = 0x84, + .ver = 0x01, + .len = 0x0a, + .CC02_CT47 = substream->runtime->channels - 1, + }; + + hdmi_setup_channel_allocation(codec, &ai); + hdmi_setup_channel_mapping(codec, &ai); + + hdmi_fill_audio_infoframe(codec, &ai); +} + + +/* + * Unsolicited events + */ + +static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +{ + int pind = !!(res & AC_UNSOL_RES_PD); + int eldv = !!(res & AC_UNSOL_RES_ELDV); + + printk(KERN_INFO + "HDMI hot plug event: Presence_Detect=%d ELD_Valid=%d\n", + pind, eldv); + + if (pind && eldv) { + hdmi_parse_eld(codec); + /* TODO: do real things about ELD */ + } +} + +static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) +{ + int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; + int cp_state = !!(res & AC_UNSOL_RES_CP_STATE); + int cp_ready = !!(res & AC_UNSOL_RES_CP_READY); + + printk(KERN_INFO + "HDMI content protection event: SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", + subtag, + cp_state, + cp_ready); + + /* TODO */ + if (cp_state) + ; + if (cp_ready) + ; +} + + +static void intel_hdmi_unsol_event(struct hda_codec *codec, unsigned int res) +{ + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; + int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; + + if (tag != INTEL_HDMI_EVENT_TAG) { + snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag); + return; + } + + if (subtag == 0) + hdmi_intrinsic_event(codec, res); + else + hdmi_non_intrinsic_event(codec, res); +} + +/* + * Callbacks + */ + +static int intel_hdmi_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct intel_hdmi_spec *spec = codec->spec; + + return snd_hda_multi_out_dig_open(codec, &spec->multiout); +} + +static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct intel_hdmi_spec *spec = codec->spec; + + hdmi_disable_output(codec); + + return snd_hda_multi_out_dig_close(codec, &spec->multiout); +} + +static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct intel_hdmi_spec *spec = codec->spec; + + snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, + format, substream); + + hdmi_set_channel_count(codec, substream->runtime->channels); + + hdmi_setup_audio_infoframe(codec, substream); + + hdmi_enable_output(codec); + + return 0; +} + +static struct hda_pcm_stream intel_hdmi_pcm_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .nid = CVT_NID, /* NID to query formats and rates and setup streams */ + .ops = { + .open = intel_hdmi_playback_pcm_open, + .close = intel_hdmi_playback_pcm_close, + .prepare = intel_hdmi_playback_pcm_prepare + }, +}; + +static int intel_hdmi_build_pcms(struct hda_codec *codec) +{ + struct intel_hdmi_spec *spec = codec->spec; + struct hda_pcm *info = &spec->pcm_rec; + + codec->num_pcms = 1; + codec->pcm_info = info; + + info->name = "INTEL HDMI"; + info->pcm_type = HDA_PCM_TYPE_HDMI; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = intel_hdmi_pcm_playback; + + return 0; +} + +static int intel_hdmi_build_controls(struct hda_codec *codec) +{ + struct intel_hdmi_spec *spec = codec->spec; + int err; + + err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); + if (err < 0) + return err; + + return 0; +} + +static int intel_hdmi_init(struct hda_codec *codec) +{ + /* disable audio output as early as possible */ + hdmi_disable_output(codec); + + snd_hda_sequence_write(codec, unsolicited_response_verb); + + return 0; +} + +static void intel_hdmi_free(struct hda_codec *codec) +{ + struct intel_hdmi_spec *spec = codec->spec; + + snd_hda_eld_proc_free(codec, &spec->sink_eld); + kfree(spec); +} + +static struct hda_codec_ops intel_hdmi_patch_ops = { + .init = intel_hdmi_init, + .free = intel_hdmi_free, + .build_pcms = intel_hdmi_build_pcms, + .build_controls = intel_hdmi_build_controls, + .unsol_event = intel_hdmi_unsol_event, +}; + +static int patch_intel_hdmi(struct hda_codec *codec) +{ + struct intel_hdmi_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + spec->multiout.num_dacs = 0; /* no analog */ + spec->multiout.max_channels = 8; + spec->multiout.dig_out_nid = CVT_NID; + + codec->spec = spec; + codec->patch_ops = intel_hdmi_patch_ops; + + snd_hda_eld_proc_new(codec, &spec->sink_eld); + + init_channel_allocations(); + + return 0; +} + +static struct hda_codec_preset snd_hda_preset_intelhdmi[] = { + { .id = 0x808629fb, .name = "G45 DEVCL", .patch = patch_intel_hdmi }, + { .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi }, + { .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi }, + { .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi }, + { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, + {} /* terminator */ +}; + +MODULE_ALIAS("snd-hda-codec-id:808629fb"); +MODULE_ALIAS("snd-hda-codec-id:80862801"); +MODULE_ALIAS("snd-hda-codec-id:80862802"); +MODULE_ALIAS("snd-hda-codec-id:80862803"); +MODULE_ALIAS("snd-hda-codec-id:10951392"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Intel HDMI HD-audio codec"); + +static struct hda_codec_preset_list intel_list = { + .preset = snd_hda_preset_intelhdmi, + .owner = THIS_MODULE, +}; + +static int __init patch_intelhdmi_init(void) +{ + return snd_hda_add_codec_preset(&intel_list); +} + +static void __exit patch_intelhdmi_exit(void) +{ + snd_hda_delete_codec_preset(&intel_list); +} + +module_init(patch_intelhdmi_init) +module_exit(patch_intelhdmi_exit) diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 2eed2c8b98da..0270fda0bda5 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -158,8 +158,34 @@ static int patch_nvhdmi(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_nvhdmi[] = { - { .id = 0x10de0002, .name = "NVIDIA MCP78 HDMI", .patch = patch_nvhdmi }, - { .id = 0x10de0007, .name = "NVIDIA MCP7A HDMI", .patch = patch_nvhdmi }, +static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { + { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi }, + { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi }, + { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:10de0002"); +MODULE_ALIAS("snd-hda-codec-id:10de0007"); +MODULE_ALIAS("snd-hda-codec-id:10de0067"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Nvidia HDMI HD-audio codec"); + +static struct hda_codec_preset_list nvhdmi_list = { + .preset = snd_hda_preset_nvhdmi, + .owner = THIS_MODULE, +}; + +static int __init patch_nvhdmi_init(void) +{ + return snd_hda_add_codec_preset(&nvhdmi_list); +} + +static void __exit patch_nvhdmi_exit(void) +{ + snd_hda_delete_codec_preset(&nvhdmi_list); +} + +module_init(patch_nvhdmi_init) +module_exit(patch_nvhdmi_exit) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a378c0145125..0bd4e6bf354d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -30,7 +30,6 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" #define ALC880_FRONT_EVENT 0x01 #define ALC880_DCVOL_EVENT 0x02 @@ -114,6 +113,7 @@ enum { ALC268_3ST, ALC268_TOSHIBA, ALC268_ACER, + ALC268_ACER_DMIC, ALC268_ACER_ASPIRE_ONE, ALC268_DELL, ALC268_ZEPTO, @@ -130,6 +130,8 @@ enum { ALC269_QUANTA_FL1, ALC269_ASUS_EEEPC_P703, ALC269_ASUS_EEEPC_P901, + ALC269_FUJITSU, + ALC269_LIFEBOOK, ALC269_AUTO, ALC269_MODEL_LAST /* last tag */ }; @@ -152,6 +154,7 @@ enum { enum { ALC660VD_3ST, ALC660VD_3ST_DIG, + ALC660VD_ASUS_V1S, ALC861VD_3ST, ALC861VD_3ST_DIG, ALC861VD_6ST_DIG, @@ -212,6 +215,7 @@ enum { ALC883_TARGA_2ch_DIG, ALC883_ACER, ALC883_ACER_ASPIRE, + ALC888_ACER_ASPIRE_4930G, ALC883_MEDION, ALC883_MEDION_MD2, ALC883_LAPTOP_EAPD, @@ -225,9 +229,11 @@ enum { ALC883_MITAC, ALC883_CLEVO_M720, ALC883_FUJITSU_PI2515, + ALC888_FUJITSU_XA3530, ALC883_3ST_6ch_INTEL, ALC888_ASUS_M90V, ALC888_ASUS_EEE1601, + ALC1200_ASUS_P5Q, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -239,6 +245,7 @@ struct alc_spec { /* codec parameterization */ struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ unsigned int num_mixers; + struct snd_kcontrol_new *cap_mixer; /* capture mixer */ const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL @@ -268,6 +275,7 @@ struct alc_spec { hda_nid_t *adc_nids; hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ + unsigned char is_mix_capture; /* matrix-style capture (non-mux) */ /* capture source */ unsigned int num_mux_defs; @@ -284,8 +292,7 @@ struct alc_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; - unsigned int num_kctl_alloc, num_kctl_used; - struct snd_kcontrol_new *kctl_alloc; + struct snd_array kctls; struct hda_input_mux private_imux; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; @@ -323,6 +330,7 @@ struct alc_config_preset { struct snd_kcontrol_new *mixers[5]; /* should be identical size * with spec */ + struct snd_kcontrol_new *cap_mixer; /* capture mixer */ const struct hda_verb *init_verbs[5]; unsigned int num_dacs; hda_nid_t *dac_nids; @@ -375,14 +383,39 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; + const struct hda_input_mux *imux; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - unsigned int mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; + unsigned int mux_idx; hda_nid_t nid = spec->capsrc_nids ? spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx]; - return snd_hda_input_mux_put(codec, &spec->input_mux[mux_idx], ucontrol, - nid, &spec->cur_mux[adc_idx]); -} + mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; + imux = &spec->input_mux[mux_idx]; + + if (spec->is_mix_capture) { + /* Matrix-mixer style (e.g. ALC882) */ + unsigned int *cur_val = &spec->cur_mux[adc_idx]; + unsigned int i, idx; + + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + if (*cur_val == idx) + return 0; + for (i = 0; i < imux->num_items; i++) { + unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE; + snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, + imux->items[i].index, + HDA_AMP_MUTE, v); + } + *cur_val = idx; + return 1; + } else { + /* MUX style (e.g. ALC880) */ + return snd_hda_input_mux_put(codec, imux, ucontrol, nid, + &spec->cur_mux[adc_idx]); + } +} /* * channel mode setting @@ -717,6 +750,43 @@ static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol, #endif /* CONFIG_SND_DEBUG */ /* + */ +static void add_mixer(struct alc_spec *spec, struct snd_kcontrol_new *mix) +{ + if (snd_BUG_ON(spec->num_mixers >= ARRAY_SIZE(spec->mixers))) + return; + spec->mixers[spec->num_mixers++] = mix; +} + +static void add_verb(struct alc_spec *spec, const struct hda_verb *verb) +{ + if (snd_BUG_ON(spec->num_init_verbs >= ARRAY_SIZE(spec->init_verbs))) + return; + spec->init_verbs[spec->num_init_verbs++] = verb; +} + +#ifdef CONFIG_PROC_FS +/* + * hook for proc + */ +static void print_realtek_coef(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int coeff; + + if (nid != 0x20) + return; + coeff = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PROC_COEF, 0); + snd_iprintf(buffer, " Processing Coefficient: 0x%02x\n", coeff); + coeff = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_COEF_INDEX, 0); + snd_iprintf(buffer, " Coefficient Index: 0x%02x\n", coeff); +} +#else +#define print_realtek_coef NULL +#endif + +/* * set up from the preset table */ static void setup_preset(struct alc_spec *spec, @@ -725,11 +795,11 @@ static void setup_preset(struct alc_spec *spec, int i; for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++) - spec->mixers[spec->num_mixers++] = preset->mixers[i]; + add_mixer(spec, preset->mixers[i]); + spec->cap_mixer = preset->cap_mixer; for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i]; i++) - spec->init_verbs[spec->num_init_verbs++] = - preset->init_verbs[i]; + add_verb(spec, preset->init_verbs[i]); spec->channel_mode = preset->channel_mode; spec->num_channel_mode = preset->num_channel_mode; @@ -1107,6 +1177,226 @@ static void alc_fix_pincfg(struct hda_codec *codec, } /* + * ALC888 + */ + +/* + * 2ch mode + */ +static struct hda_verb alc888_4ST_ch2_intel_init[] = { +/* Mic-in jack as mic in */ + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, +/* Line-in jack as Line in */ + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, +/* Line-Out as Front */ + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } /* end */ +}; + +/* + * 4ch mode + */ +static struct hda_verb alc888_4ST_ch4_intel_init[] = { +/* Mic-in jack as mic in */ + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, +/* Line-in jack as Surround */ + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, +/* Line-Out as Front */ + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } /* end */ +}; + +/* + * 6ch mode + */ +static struct hda_verb alc888_4ST_ch6_intel_init[] = { +/* Mic-in jack as CLFE */ + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, +/* Line-in jack as Surround */ + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, +/* Line-Out as CLFE (workaround because Mic-in is not loud enough) */ + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, + { } /* end */ +}; + +/* + * 8ch mode + */ +static struct hda_verb alc888_4ST_ch8_intel_init[] = { +/* Mic-in jack as CLFE */ + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, +/* Line-in jack as Surround */ + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, +/* Line-Out as Side */ + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, + { } /* end */ +}; + +static struct hda_channel_mode alc888_4ST_8ch_intel_modes[4] = { + { 2, alc888_4ST_ch2_intel_init }, + { 4, alc888_4ST_ch4_intel_init }, + { 6, alc888_4ST_ch6_intel_init }, + { 8, alc888_4ST_ch8_intel_init }, +}; + +/* + * ALC888 Fujitsu Siemens Amillo xa3530 + */ + +static struct hda_verb alc888_fujitsu_xa3530_verbs[] = { +/* Front Mic: set to PIN_IN (empty by default) */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, +/* Connect Internal HP to Front */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect Bass HP to Front */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect Line-Out side jack (SPDIF) to Side */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, +/* Connect Mic jack to CLFE */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Connect Line-in jack to Surround */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect HP out jack to Front */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Enable unsolicited event for HP jack and Line-out jack */ + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {} +}; + +static void alc888_fujitsu_xa3530_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned int bits; + /* Line out presence */ + present = snd_hda_codec_read(codec, 0x17, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + /* HP out presence */ + present = present || snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? HDA_AMP_MUTE : 0; + /* Toggle internal speakers muting */ + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + /* Toggle internal bass muting */ + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc888_fujitsu_xa3530_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if (res >> 26 == ALC880_HP_EVENT) + alc888_fujitsu_xa3530_automute(codec); +} + + +/* + * ALC888 Acer Aspire 4930G model + */ + +static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { +/* Front Mic: set to PIN_IN (empty by default) */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, +/* Unselect Front Mic by default in input mixer 3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, +/* Enable unsolicited event for HP jack */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, +/* Connect Internal HP to front */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect HP out to front */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } +}; + +static struct hda_input_mux alc888_2_capture_sources[2] = { + /* Front mic only available on one ADC */ + { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Front Mic", 0xb }, + }, + }, + { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, + } +}; + +static struct snd_kcontrol_new alc888_base_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, + HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + { } /* end */ +}; + +static void alc888_acer_aspire_4930g_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned int bits; + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc888_acer_aspire_4930g_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if (res >> 26 == ALC880_HP_EVENT) + alc888_acer_aspire_4930g_automute(codec); +} + +/* * ALC880 3-stack model * * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e) @@ -1205,49 +1495,126 @@ static struct snd_kcontrol_new alc880_three_stack_mixer[] = { }; /* capture mixer elements */ -static struct snd_kcontrol_new alc880_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 3, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, - { } /* end */ -}; +static int alc_cap_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + int err; -/* capture mixer elements (in case NID 0x07 not available) */ -static struct snd_kcontrol_new alc880_capture_alt_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, - { } /* end */ -}; + mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0, + HDA_INPUT); + err = snd_hda_mixer_amp_volume_info(kcontrol, uinfo); + mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */ + return err; +} + +static int alc_cap_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + int err; + mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0, + HDA_INPUT); + err = snd_hda_mixer_amp_tlv(kcontrol, op_flag, size, tlv); + mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */ + return err; +} + +typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + +static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, + getput_call_t func) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + int err; + + mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx], + 3, 0, HDA_INPUT); + err = func(kcontrol, ucontrol); + mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */ + return err; +} + +static int alc_cap_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return alc_cap_getput_caller(kcontrol, ucontrol, + snd_hda_mixer_amp_volume_get); +} + +static int alc_cap_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return alc_cap_getput_caller(kcontrol, ucontrol, + snd_hda_mixer_amp_volume_put); +} + +/* capture mixer elements */ +#define alc_cap_sw_info snd_ctl_boolean_stereo_info + +static int alc_cap_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return alc_cap_getput_caller(kcontrol, ucontrol, + snd_hda_mixer_amp_switch_get); +} + +static int alc_cap_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return alc_cap_getput_caller(kcontrol, ucontrol, + snd_hda_mixer_amp_switch_put); +} + +#define DEFINE_CAPMIX(num) \ +static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Capture Switch", \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .count = num, \ + .info = alc_cap_sw_info, \ + .get = alc_cap_sw_get, \ + .put = alc_cap_sw_put, \ + }, \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Capture Volume", \ + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK), \ + .count = num, \ + .info = alc_cap_vol_info, \ + .get = alc_cap_vol_get, \ + .put = alc_cap_vol_put, \ + .tlv = { .c = alc_cap_vol_tlv }, \ + }, \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + /* .name = "Capture Source", */ \ + .name = "Input Source", \ + .count = num, \ + .info = alc_mux_enum_info, \ + .get = alc_mux_enum_get, \ + .put = alc_mux_enum_put, \ + }, \ + { } /* end */ \ +} + +/* up to three ADCs */ +DEFINE_CAPMIX(1); +DEFINE_CAPMIX(2); +DEFINE_CAPMIX(3); /* @@ -1533,18 +1900,6 @@ static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0B, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, { } /* end */ }; @@ -1619,6 +1974,7 @@ static const char *alc_slave_vols[] = { "Speaker Playback Volume", "Mono Playback Volume", "Line-Out Playback Volume", + "PCM Playback Volume", NULL, }; @@ -1638,6 +1994,9 @@ static const char *alc_slave_sws[] = { /* * build control elements */ + +static void alc_free_kctls(struct hda_codec *codec); + static int alc_build_controls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1649,7 +2008,11 @@ static int alc_build_controls(struct hda_codec *codec) if (err < 0) return err; } - + if (spec->cap_mixer) { + err = snd_hda_add_new_ctls(codec, spec->cap_mixer); + if (err < 0) + return err; + } if (spec->multiout.dig_out_nid) { err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); @@ -1684,6 +2047,7 @@ static int alc_build_controls(struct hda_codec *codec) return err; } + alc_free_kctls(codec); /* no longer needed */ return 0; } @@ -2774,19 +3138,27 @@ static int alc_build_pcms(struct hda_codec *codec) return 0; } +static void alc_free_kctls(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (spec->kctls.list) { + struct snd_kcontrol_new *kctl = spec->kctls.list; + int i; + for (i = 0; i < spec->kctls.used; i++) + kfree(kctl[i].name); + } + snd_array_free(&spec->kctls); +} + static void alc_free(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int i; if (!spec) return; - if (spec->kctl_alloc) { - for (i = 0; i < spec->num_kctl_used; i++) - kfree(spec->kctl_alloc[i].name); - kfree(spec->kctl_alloc); - } + alc_free_kctls(codec); kfree(spec); codec->spec = NULL; /* to be sure */ } @@ -3268,6 +3640,8 @@ static struct alc_config_preset alc880_presets[] = { alc880_gpio2_init_verbs }, .num_dacs = ARRAY_SIZE(alc880_dac_nids), .dac_nids = alc880_dac_nids, + .adc_nids = alc880_adc_nids_alt, /* FIXME: correct? */ + .num_adc_nids = 1, /* single ADC */ .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), .channel_mode = alc880_2_jack_modes, @@ -3532,9 +3906,6 @@ static struct alc_config_preset alc880_presets[] = { * Automatic parse of I/O pins from the BIOS configuration */ -#define NUM_CONTROL_ALLOC 32 -#define NUM_VERB_ALLOC 32 - enum { ALC_CTL_WIDGET_VOL, ALC_CTL_WIDGET_MUTE, @@ -3552,29 +3923,15 @@ static int add_control(struct alc_spec *spec, int type, const char *name, { struct snd_kcontrol_new *knew; - if (spec->num_kctl_used >= spec->num_kctl_alloc) { - int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC; - - /* array + terminator */ - knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); - if (!knew) - return -ENOMEM; - if (spec->kctl_alloc) { - memcpy(knew, spec->kctl_alloc, - sizeof(*knew) * spec->num_kctl_alloc); - kfree(spec->kctl_alloc); - } - spec->kctl_alloc = knew; - spec->num_kctl_alloc = num; - } - - knew = &spec->kctl_alloc[spec->num_kctl_used]; + snd_array_init(&spec->kctls, sizeof(*knew), 32); + knew = snd_array_new(&spec->kctls); + if (!knew) + return -ENOMEM; *knew = alc880_control_templates[type]; knew->name = kstrdup(name, GFP_KERNEL); if (!knew->name) return -ENOMEM; knew->private_value = val; - spec->num_kctl_used++; return 0; } @@ -3898,10 +4255,10 @@ static int alc880_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_in_pin) spec->dig_in_nid = ALC880_DIGIN_NID; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + add_mixer(spec, spec->kctls.list); - spec->init_verbs[spec->num_init_verbs++] = alc880_volume_init_verbs; + add_verb(spec, alc880_volume_init_verbs); spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; @@ -3925,6 +4282,17 @@ static void alc880_auto_init(struct hda_codec *codec) * OK, here we have finally the patch for ALC880 */ +static void set_capture_mixer(struct alc_spec *spec) +{ + static struct snd_kcontrol_new *caps[3] = { + alc_capture_mixer1, + alc_capture_mixer2, + alc_capture_mixer3, + }; + if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) + spec->cap_mixer = caps[spec->num_adc_nids - 1]; +} + static int patch_alc880(struct hda_codec *codec) { struct alc_spec *spec; @@ -3980,16 +4348,12 @@ static int patch_alc880(struct hda_codec *codec) if (wcap != AC_WID_AUD_IN) { spec->adc_nids = alc880_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids_alt); - spec->mixers[spec->num_mixers] = - alc880_capture_alt_mixer; - spec->num_mixers++; } else { spec->adc_nids = alc880_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids); - spec->mixers[spec->num_mixers] = alc880_capture_mixer; - spec->num_mixers++; } } + set_capture_mixer(spec); spec->vmaster_nid = 0x0c; @@ -4000,6 +4364,7 @@ static int patch_alc880(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc880_loopbacks; #endif + codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -4024,11 +4389,6 @@ static hda_nid_t alc260_adc_nids_alt[1] = { 0x05, }; -static hda_nid_t alc260_hp_adc_nids[2] = { - /* ADC1, 0 */ - 0x05, 0x04 -}; - /* NIDs used when simultaneous access to both ADCs makes sense. Note that * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC. */ @@ -4157,13 +4517,13 @@ static void alc260_hp_master_update(struct hda_codec *codec, struct alc_spec *spec = codec->spec; unsigned int val = spec->master_sw ? PIN_HP : 0; /* change HP and line-out pins */ - snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + snd_hda_codec_write(codec, hp, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + snd_hda_codec_write(codec, line, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val); /* mono (speaker) depending on the HP jack sense */ val = (val && !spec->jack_present) ? PIN_OUT : 0; - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + snd_hda_codec_write(codec, mono, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val); } @@ -4242,7 +4602,7 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { .info = snd_ctl_boolean_mono_info, .get = alc260_hp_master_sw_get, .put = alc260_hp_master_sw_put, - .private_value = (0x10 << 16) | (0x15 << 8) | 0x11 + .private_value = (0x15 << 16) | (0x10 << 8) | 0x11 }, HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT), @@ -4295,7 +4655,7 @@ static void alc260_hp_3013_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0); spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; - alc260_hp_master_update(codec, 0x10, 0x15, 0x11); + alc260_hp_master_update(codec, 0x15, 0x10, 0x11); } static void alc260_hp_3013_unsol_event(struct hda_codec *codec, @@ -4427,45 +4787,6 @@ static struct snd_kcontrol_new alc260_replacer_672v_mixer[] = { { } /* end */ }; -/* capture mixer elements */ -static struct snd_kcontrol_new alc260_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x04, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x05, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x05, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, - { } /* end */ -}; - -static struct snd_kcontrol_new alc260_capture_alt_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x05, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x05, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, - { } /* end */ -}; - /* * initialization verbs */ @@ -5282,7 +5603,6 @@ static struct hda_verb alc260_volume_init_verbs[] = { static int alc260_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int wcap; int err; static hda_nid_t alc260_ignore[] = { 0x17, 0 }; @@ -5293,7 +5613,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) err = alc260_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; - if (!spec->kctl_alloc) + if (!spec->kctls.list) return 0; /* can't find valid BIOS pin config */ err = alc260_auto_create_analog_input_ctls(spec, &spec->autocfg); if (err < 0) @@ -5303,28 +5623,14 @@ static int alc260_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = ALC260_DIGOUT_NID; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + add_mixer(spec, spec->kctls.list); - spec->init_verbs[spec->num_init_verbs++] = alc260_volume_init_verbs; + add_verb(spec, alc260_volume_init_verbs); spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; - /* check whether NID 0x04 is valid */ - wcap = get_wcaps(codec, 0x04); - wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */ - if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) { - spec->adc_nids = alc260_adc_nids_alt; - spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt); - spec->mixers[spec->num_mixers] = alc260_capture_alt_mixer; - } else { - spec->adc_nids = alc260_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids); - spec->mixers[spec->num_mixers] = alc260_capture_mixer; - } - spec->num_mixers++; - store_pin_configs(codec); return 1; } @@ -5394,12 +5700,11 @@ static struct alc_config_preset alc260_presets[] = { [ALC260_BASIC] = { .mixers = { alc260_base_output_mixer, alc260_input_mixer, - alc260_pc_beep_mixer, - alc260_capture_mixer }, + alc260_pc_beep_mixer }, .init_verbs = { alc260_init_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_adc_nids), + .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), .adc_nids = alc260_adc_nids, .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, @@ -5407,14 +5712,13 @@ static struct alc_config_preset alc260_presets[] = { }, [ALC260_HP] = { .mixers = { alc260_hp_output_mixer, - alc260_input_mixer, - alc260_capture_alt_mixer }, + alc260_input_mixer }, .init_verbs = { alc260_init_verbs, alc260_hp_unsol_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids), - .adc_nids = alc260_hp_adc_nids, + .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt), + .adc_nids = alc260_adc_nids_alt, .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_capture_source, @@ -5423,14 +5727,13 @@ static struct alc_config_preset alc260_presets[] = { }, [ALC260_HP_DC7600] = { .mixers = { alc260_hp_dc7600_mixer, - alc260_input_mixer, - alc260_capture_alt_mixer }, + alc260_input_mixer }, .init_verbs = { alc260_init_verbs, alc260_hp_dc7600_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids), - .adc_nids = alc260_hp_adc_nids, + .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt), + .adc_nids = alc260_adc_nids_alt, .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_capture_source, @@ -5439,14 +5742,13 @@ static struct alc_config_preset alc260_presets[] = { }, [ALC260_HP_3013] = { .mixers = { alc260_hp_3013_mixer, - alc260_input_mixer, - alc260_capture_alt_mixer }, + alc260_input_mixer }, .init_verbs = { alc260_hp_3013_init_verbs, alc260_hp_3013_unsol_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids), - .adc_nids = alc260_hp_adc_nids, + .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt), + .adc_nids = alc260_adc_nids_alt, .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_capture_source, @@ -5454,8 +5756,7 @@ static struct alc_config_preset alc260_presets[] = { .init_hook = alc260_hp_3013_automute, }, [ALC260_FUJITSU_S702X] = { - .mixers = { alc260_fujitsu_mixer, - alc260_capture_mixer }, + .mixers = { alc260_fujitsu_mixer }, .init_verbs = { alc260_fujitsu_init_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, @@ -5467,8 +5768,7 @@ static struct alc_config_preset alc260_presets[] = { .input_mux = alc260_fujitsu_capture_sources, }, [ALC260_ACER] = { - .mixers = { alc260_acer_mixer, - alc260_capture_mixer }, + .mixers = { alc260_acer_mixer }, .init_verbs = { alc260_acer_init_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, @@ -5480,8 +5780,7 @@ static struct alc_config_preset alc260_presets[] = { .input_mux = alc260_acer_capture_sources, }, [ALC260_WILL] = { - .mixers = { alc260_will_mixer, - alc260_capture_mixer }, + .mixers = { alc260_will_mixer }, .init_verbs = { alc260_init_verbs, alc260_will_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, @@ -5493,8 +5792,7 @@ static struct alc_config_preset alc260_presets[] = { .input_mux = &alc260_capture_source, }, [ALC260_REPLACER_672V] = { - .mixers = { alc260_replacer_672v_mixer, - alc260_capture_mixer }, + .mixers = { alc260_replacer_672v_mixer }, .init_verbs = { alc260_init_verbs, alc260_replacer_672v_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, @@ -5509,8 +5807,7 @@ static struct alc_config_preset alc260_presets[] = { }, #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = { - .mixers = { alc260_test_mixer, - alc260_capture_mixer }, + .mixers = { alc260_test_mixer }, .init_verbs = { alc260_test_init_verbs }, .num_dacs = ARRAY_SIZE(alc260_test_dac_nids), .dac_nids = alc260_test_dac_nids, @@ -5569,6 +5866,21 @@ static int patch_alc260(struct hda_codec *codec) spec->stream_digital_playback = &alc260_pcm_digital_playback; spec->stream_digital_capture = &alc260_pcm_digital_capture; + if (!spec->adc_nids && spec->input_mux) { + /* check whether NID 0x04 is valid */ + unsigned int wcap = get_wcaps(codec, 0x04); + wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + /* get type */ + if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) { + spec->adc_nids = alc260_adc_nids_alt; + spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt); + } else { + spec->adc_nids = alc260_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids); + } + } + set_capture_mixer(spec); + spec->vmaster_nid = 0x08; codec->patch_ops = alc_patch_ops; @@ -5578,6 +5890,7 @@ static int patch_alc260(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc260_loopbacks; #endif + codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -5625,36 +5938,6 @@ static struct hda_input_mux alc882_capture_source = { { "CD", 0x4 }, }, }; -#define alc882_mux_enum_info alc_mux_enum_info -#define alc882_mux_enum_get alc_mux_enum_get - -static int alc882_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - hda_nid_t nid = spec->capsrc_nids ? - spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx]; - unsigned int *cur_val = &spec->cur_mux[adc_idx]; - unsigned int i, idx; - - idx = ucontrol->value.enumerated.item[0]; - if (idx >= imux->num_items) - idx = imux->num_items - 1; - if (*cur_val == idx) - return 0; - for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, - imux->items[i].index, - HDA_AMP_MUTE, v); - } - *cur_val = idx; - return 1; -} - /* * 2ch mode */ @@ -6337,49 +6620,6 @@ static struct hda_verb alc882_auto_init_verbs[] = { { } }; -/* capture mixer elements */ -static struct snd_kcontrol_new alc882_capture_alt_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, - { } /* end */ -}; - -static struct snd_kcontrol_new alc882_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 3, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, - { } /* end */ -}; - #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc882_loopbacks alc880_loopbacks #endif @@ -6508,8 +6748,7 @@ static struct alc_config_preset alc882_presets[] = { .init_hook = alc885_imac24_init_hook, }, [ALC882_TARGA] = { - .mixers = { alc882_targa_mixer, alc882_chmode_mixer, - alc882_capture_mixer }, + .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, .init_verbs = { alc882_init_verbs, alc882_targa_verbs}, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, @@ -6525,8 +6764,7 @@ static struct alc_config_preset alc882_presets[] = { .init_hook = alc882_targa_automute, }, [ALC882_ASUS_A7J] = { - .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer, - alc882_capture_mixer }, + .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer }, .init_verbs = { alc882_init_verbs, alc882_asus_a7j_verbs}, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, @@ -6831,6 +7069,7 @@ static int patch_alc882(struct hda_codec *codec) spec->stream_digital_playback = &alc882_pcm_digital_playback; spec->stream_digital_capture = &alc882_pcm_digital_capture; + spec->is_mix_capture = 1; /* matrix-style capture */ if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -6840,17 +7079,13 @@ static int patch_alc882(struct hda_codec *codec) spec->adc_nids = alc882_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids_alt); spec->capsrc_nids = alc882_capsrc_nids_alt; - spec->mixers[spec->num_mixers] = - alc882_capture_alt_mixer; - spec->num_mixers++; } else { spec->adc_nids = alc882_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids); spec->capsrc_nids = alc882_capsrc_nids; - spec->mixers[spec->num_mixers] = alc882_capture_mixer; - spec->num_mixers++; } } + set_capture_mixer(spec); spec->vmaster_nid = 0x0c; @@ -6861,6 +7096,7 @@ static int patch_alc882(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc882_loopbacks; #endif + codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -6879,6 +7115,8 @@ static int patch_alc882(struct hda_codec *codec) #define ALC883_DIGOUT_NID 0x06 #define ALC883_DIGIN_NID 0x0a +#define ALC1200_DIGOUT_NID 0x10 + static hda_nid_t alc883_dac_nids[4] = { /* front, rear, clfe, rear_surr */ 0x02, 0x03, 0x04, 0x05 @@ -6889,8 +7127,20 @@ static hda_nid_t alc883_adc_nids[2] = { 0x08, 0x09, }; +static hda_nid_t alc883_adc_nids_alt[1] = { + /* ADC1 */ + 0x08, +}; + +static hda_nid_t alc883_adc_nids_rev[2] = { + /* ADC2-1 */ + 0x09, 0x08 +}; + static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 }; +static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 }; + /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -6957,11 +7207,6 @@ static struct hda_input_mux alc883_asus_eee1601_capture_source = { }, }; -#define alc883_mux_enum_info alc_mux_enum_info -#define alc883_mux_enum_get alc_mux_enum_get -/* ALC883 has the ALC882-type input selection */ -#define alc883_mux_enum_put alc882_mux_enum_put - /* * 2ch mode */ @@ -7115,19 +7360,6 @@ static struct snd_kcontrol_new alc883_base_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7145,19 +7377,6 @@ static struct snd_kcontrol_new alc883_mitac_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7172,19 +7391,6 @@ static struct snd_kcontrol_new alc883_clevo_m720_mixer[] = { HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7199,19 +7405,6 @@ static struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = { HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7231,19 +7424,6 @@ static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7269,17 +7449,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7306,19 +7475,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7344,18 +7500,6 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7376,19 +7520,6 @@ static struct snd_kcontrol_new alc883_tagra_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7404,19 +7535,6 @@ static struct snd_kcontrol_new alc883_tagra_2ch_mixer[] = { HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7429,17 +7547,6 @@ static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7453,19 +7560,6 @@ static struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("iMic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("iMic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7479,19 +7573,6 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7504,19 +7585,6 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7544,19 +7612,6 @@ static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, { } /* end */ }; @@ -7587,6 +7642,10 @@ static struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = { HDA_BIND_VOL("Capture Volume", &alc883_bind_cap_vol), HDA_BIND_SW("Capture Switch", &alc883_bind_cap_switch), { @@ -7594,9 +7653,9 @@ static struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = { /* .name = "Capture Source", */ .name = "Input Source", .count = 1, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, }, { } /* end */ }; @@ -8251,27 +8310,6 @@ static struct hda_verb alc883_auto_init_verbs[] = { { } }; -/* capture mixer elements */ -static struct snd_kcontrol_new alc883_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, - { } /* end */ -}; - static struct hda_verb alc888_asus_m90v_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -8394,6 +8432,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_TARGA_2ch_DIG] = "targa-2ch-dig", [ALC883_ACER] = "acer", [ALC883_ACER_ASPIRE] = "acer-aspire", + [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", [ALC883_LAPTOP_EAPD] = "laptop-eapd", @@ -8407,7 +8446,9 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_MITAC] = "mitac", [ALC883_CLEVO_M720] = "clevo-m720", [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515", + [ALC888_FUJITSU_XA3530] = "fujitsu-xa3530", [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", + [ALC1200_ASUS_P5Q] = "asus-p5q", [ALC883_AUTO] = "auto", }; @@ -8418,6 +8459,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G", + ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */ SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), @@ -8426,6 +8469,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), + SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601), SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), @@ -8452,6 +8496,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7260, "MSI 7260", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), @@ -8463,6 +8508,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515), + SND_PCI_QUIRK(0x1734, 0x113d, "Fujitsu AMILO Xa3530", + ALC888_FUJITSU_XA3530), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763), @@ -8553,6 +8600,8 @@ static struct alc_config_preset alc883_presets[] = { .init_verbs = { alc883_init_verbs, alc883_tagra_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, + .adc_nids = alc883_adc_nids_alt, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt), .dig_out_nid = ALC883_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, @@ -8586,6 +8635,26 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc883_acer_aspire_unsol_event, .init_hook = alc883_acer_aspire_automute, }, + [ALC888_ACER_ASPIRE_4930G] = { + .mixers = { alc888_base_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, + alc888_acer_aspire_4930g_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), + .adc_nids = alc883_adc_nids_rev, + .capsrc_nids = alc883_capsrc_nids_rev, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), + .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, + .num_mux_defs = + ARRAY_SIZE(alc888_2_capture_sources), + .input_mux = alc888_2_capture_sources, + .unsol_event = alc888_acer_aspire_4930g_unsol_event, + .init_hook = alc888_acer_aspire_4930g_automute, + }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer, alc883_chmode_mixer }, @@ -8593,6 +8662,8 @@ static struct alc_config_preset alc883_presets[] = { alc883_medion_eapd_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, + .adc_nids = alc883_adc_nids_alt, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt), .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, @@ -8635,6 +8706,8 @@ static struct alc_config_preset alc883_presets[] = { .init_verbs = { alc883_init_verbs, alc883_lenovo_101e_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, + .adc_nids = alc883_adc_nids_alt, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt), .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_lenovo_101e_capture_source, @@ -8725,14 +8798,30 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc883_2ch_fujitsu_pi2515_unsol_event, .init_hook = alc883_2ch_fujitsu_pi2515_automute, }, + [ALC888_FUJITSU_XA3530] = { + .mixers = { alc888_base_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, + alc888_fujitsu_xa3530_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), + .adc_nids = alc883_adc_nids_rev, + .capsrc_nids = alc883_capsrc_nids_rev, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc888_4ST_8ch_intel_modes), + .channel_mode = alc888_4ST_8ch_intel_modes, + .num_mux_defs = + ARRAY_SIZE(alc888_2_capture_sources), + .input_mux = alc888_2_capture_sources, + .unsol_event = alc888_fujitsu_xa3530_unsol_event, + .init_hook = alc888_fujitsu_xa3530_automute, + }, [ALC888_LENOVO_SKY] = { .mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc888_lenovo_sky_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, .need_dac_fix = 1, @@ -8756,6 +8845,7 @@ static struct alc_config_preset alc883_presets[] = { }, [ALC888_ASUS_EEE1601] = { .mixers = { alc883_asus_eee1601_mixer }, + .cap_mixer = alc883_asus_eee1601_cap_mixer, .init_verbs = { alc883_init_verbs, alc888_asus_eee1601_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, @@ -8768,6 +8858,17 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc883_eee1601_unsol_event, .init_hook = alc883_eee1601_inithook, }, + [ALC1200_ASUS_P5Q] = { + .mixers = { alc883_base_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC1200_DIGOUT_NID, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), + .channel_mode = alc883_sixstack_modes, + .input_mux = &alc883_capture_source, + }, }; @@ -8862,8 +8963,6 @@ static int alc883_parse_auto_config(struct hda_codec *codec) /* hack - override the init verbs */ spec->init_verbs[0] = alc883_auto_init_verbs; - spec->mixers[spec->num_mixers] = alc883_capture_mixer; - spec->num_mixers++; return 1; /* config found */ } @@ -8946,9 +9045,15 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_digital_playback = &alc883_pcm_digital_playback; spec->stream_digital_capture = &alc883_pcm_digital_capture; - spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); - spec->adc_nids = alc883_adc_nids; - spec->capsrc_nids = alc883_capsrc_nids; + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc883_capsrc_nids; + spec->is_mix_capture = 1; /* matrix-style capture */ + if (!spec->cap_mixer) + set_capture_mixer(spec); spec->vmaster_nid = 0x0c; @@ -8960,6 +9065,7 @@ static int patch_alc883(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc883_loopbacks; #endif + codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -9439,20 +9545,6 @@ static struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = { HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, { } /* end */ }; @@ -9969,7 +10061,7 @@ static int alc262_ultra_mux_enum_put(struct snd_kcontrol *kcontrol, struct alc_spec *spec = codec->spec; int ret; - ret = alc882_mux_enum_put(kcontrol, ucontrol); + ret = alc_mux_enum_put(kcontrol, ucontrol); if (!ret) return 0; /* reprogram the HP pin as mic or HP according to the input source */ @@ -9986,8 +10078,8 @@ static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, .put = alc262_ultra_mux_enum_put, }, { } /* end */ @@ -10380,10 +10472,10 @@ static int alc262_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_in_pin) spec->dig_in_nid = ALC262_DIGIN_NID; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + add_mixer(spec, spec->kctls.list); - spec->init_verbs[spec->num_init_verbs++] = alc262_volume_init_verbs; + add_verb(spec, alc262_volume_init_verbs); spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; @@ -10466,6 +10558,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x104d, 0x9033, "Sony VAIO VGN-SR19XN", + ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", ALC262_TOSHIBA_RX1), SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), @@ -10624,7 +10718,8 @@ static struct alc_config_preset alc262_presets[] = { .init_hook = alc262_hippo_automute, }, [ALC262_ULTRA] = { - .mixers = { alc262_ultra_mixer, alc262_ultra_capture_mixer }, + .mixers = { alc262_ultra_mixer }, + .cap_mixer = alc262_ultra_capture_mixer, .init_verbs = { alc262_ultra_verbs }, .num_dacs = ARRAY_SIZE(alc262_dac_nids), .dac_nids = alc262_dac_nids, @@ -10750,6 +10845,7 @@ static int patch_alc262(struct hda_codec *codec) spec->stream_digital_playback = &alc262_pcm_digital_playback; spec->stream_digital_capture = &alc262_pcm_digital_capture; + spec->is_mix_capture = 1; if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -10760,17 +10856,14 @@ static int patch_alc262(struct hda_codec *codec) spec->adc_nids = alc262_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids_alt); spec->capsrc_nids = alc262_capsrc_nids_alt; - spec->mixers[spec->num_mixers] = - alc262_capture_alt_mixer; - spec->num_mixers++; } else { spec->adc_nids = alc262_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids); spec->capsrc_nids = alc262_capsrc_nids; - spec->mixers[spec->num_mixers] = alc262_capture_mixer; - spec->num_mixers++; } } + if (!spec->cap_mixer) + set_capture_mixer(spec); spec->vmaster_nid = 0x0c; @@ -10781,6 +10874,7 @@ static int patch_alc262(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc262_loopbacks; #endif + codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -10942,6 +11036,22 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { { } }; +static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { + /* output mixer control */ + HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc268_acer_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT), + { } +}; + static struct hda_verb alc268_acer_aspire_one_verbs[] = { {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -11218,10 +11328,6 @@ static struct hda_verb alc268_volume_init_verbs[] = { { } }; -#define alc268_mux_enum_info alc_mux_enum_info -#define alc268_mux_enum_get alc_mux_enum_get -#define alc268_mux_enum_put alc_mux_enum_put - static struct snd_kcontrol_new alc268_capture_alt_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), @@ -11233,9 +11339,9 @@ static struct snd_kcontrol_new alc268_capture_alt_mixer[] = { /* .name = "Capture Source", */ .name = "Input Source", .count = 1, - .info = alc268_mux_enum_info, - .get = alc268_mux_enum_get, - .put = alc268_mux_enum_put, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, }, { } /* end */ }; @@ -11253,9 +11359,9 @@ static struct snd_kcontrol_new alc268_capture_mixer[] = { /* .name = "Capture Source", */ .name = "Input Source", .count = 2, - .info = alc268_mux_enum_info, - .get = alc268_mux_enum_get, - .put = alc268_mux_enum_put, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, }, { } /* end */ }; @@ -11274,6 +11380,15 @@ static struct hda_input_mux alc268_acer_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, + { "Internal Mic", 0x1 }, + { "Line", 0x2 }, + }, +}; + +static struct hda_input_mux alc268_acer_dmic_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, { "Internal Mic", 0x6 }, { "Line", 0x2 }, }, @@ -11512,13 +11627,13 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = ALC268_DIGOUT_NID; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + add_mixer(spec, spec->kctls.list); if (spec->autocfg.speaker_pins[0] != 0x1d) - spec->mixers[spec->num_mixers++] = alc268_beep_mixer; + add_mixer(spec, alc268_beep_mixer); - spec->init_verbs[spec->num_init_verbs++] = alc268_volume_init_verbs; + add_verb(spec, alc268_volume_init_verbs); spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; @@ -11554,6 +11669,7 @@ static const char *alc268_models[ALC268_MODEL_LAST] = { [ALC268_3ST] = "3stack", [ALC268_TOSHIBA] = "toshiba", [ALC268_ACER] = "acer", + [ALC268_ACER_DMIC] = "acer-dmic", [ALC268_ACER_ASPIRE_ONE] = "acer-aspire", [ALC268_DELL] = "dell", [ALC268_ZEPTO] = "zepto", @@ -11649,6 +11765,23 @@ static struct alc_config_preset alc268_presets[] = { .unsol_event = alc268_acer_unsol_event, .init_hook = alc268_acer_init_hook, }, + [ALC268_ACER_DMIC] = { + .mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer, + alc268_beep_mixer }, + .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, + alc268_acer_verbs }, + .num_dacs = ARRAY_SIZE(alc268_dac_nids), + .dac_nids = alc268_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), + .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, + .hp_nid = 0x02, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .input_mux = &alc268_acer_dmic_capture_source, + .unsol_event = alc268_acer_unsol_event, + .init_hook = alc268_acer_init_hook, + }, [ALC268_ACER_ASPIRE_ONE] = { .mixers = { alc268_acer_aspire_one_mixer, alc268_capture_alt_mixer }, @@ -11787,15 +11920,11 @@ static int patch_alc268(struct hda_codec *codec) if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) { spec->adc_nids = alc268_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt); - spec->mixers[spec->num_mixers] = - alc268_capture_alt_mixer; - spec->num_mixers++; + add_mixer(spec, alc268_capture_alt_mixer); } else { spec->adc_nids = alc268_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids); - spec->mixers[spec->num_mixers] = - alc268_capture_mixer; - spec->num_mixers++; + add_mixer(spec, alc268_capture_mixer); } spec->capsrc_nids = alc268_capsrc_nids; /* set default input source */ @@ -11811,6 +11940,8 @@ static int patch_alc268(struct hda_codec *codec) if (board_config == ALC268_AUTO) spec->init_hook = alc268_auto_init; + codec->proc_widget_hook = print_realtek_coef; + return 0; } @@ -11893,6 +12024,31 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { { } }; +static struct snd_kcontrol_new alc269_lifebook_mixer[] = { + /* output mixer control */ + HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc268_acer_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Boost", 0x1b, 0, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT), + { } +}; + /* bind volumes of both NID 0x0c and 0x0d */ static struct hda_bind_ctls alc269_epc_bind_vol = { .ops = &snd_hda_bind_vol, @@ -11911,28 +12067,18 @@ static struct snd_kcontrol_new alc269_eeepc_mixer[] = { }; /* capture mixer elements */ -static struct snd_kcontrol_new alc269_capture_mixer[] = { +static struct snd_kcontrol_new alc269_epc_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), { } /* end */ }; -/* capture mixer elements */ -static struct snd_kcontrol_new alc269_epc_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), +/* FSC amilo */ +static struct snd_kcontrol_new alc269_fujitsu_mixer[] = { + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_BIND_VOL("PCM Playback Volume", &alc269_epc_bind_vol), { } /* end */ }; @@ -11953,6 +12099,20 @@ static struct hda_verb alc269_quanta_fl1_verbs[] = { { } }; +static struct hda_verb alc269_lifebook_verbs[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + { } +}; + /* toggle speaker-output according to the hp-jack state */ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec) { @@ -11978,6 +12138,37 @@ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec) AC_VERB_SET_PROC_COEF, 0x480); } +/* toggle speaker-output according to the hp-jacks state */ +static void alc269_lifebook_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + /* Check laptop headphone socket */ + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + /* Check port replicator headphone socket */ + present |= snd_hda_codec_read(codec, 0x1a, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + bits = present ? AMP_IN_MUTE(0) : 0; + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, + AMP_IN_MUTE(0), bits); + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, + AMP_IN_MUTE(0), bits); + + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_COEF_INDEX, 0x0c); + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x680); + + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_COEF_INDEX, 0x0c); + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x480); +} + static void alc269_quanta_fl1_mic_automute(struct hda_codec *codec) { unsigned int present; @@ -11988,6 +12179,29 @@ static void alc269_quanta_fl1_mic_automute(struct hda_codec *codec) AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x1); } +static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec) +{ + unsigned int present_laptop; + unsigned int present_dock; + + present_laptop = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + present_dock = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + /* Laptop mic port overrides dock mic port, design decision */ + if (present_dock) + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, 0x3); + if (present_laptop) + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, 0x0); + if (!present_dock && !present_laptop) + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, 0x1); +} + static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec, unsigned int res) { @@ -11997,12 +12211,27 @@ static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec, alc269_quanta_fl1_mic_automute(codec); } +static void alc269_lifebook_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc269_lifebook_speaker_automute(codec); + if ((res >> 26) == ALC880_MIC_EVENT) + alc269_lifebook_mic_autoswitch(codec); +} + static void alc269_quanta_fl1_init_hook(struct hda_codec *codec) { alc269_quanta_fl1_speaker_automute(codec); alc269_quanta_fl1_mic_automute(codec); } +static void alc269_lifebook_init_hook(struct hda_codec *codec) +{ + alc269_lifebook_speaker_automute(codec); + alc269_lifebook_mic_autoswitch(codec); +} + static struct hda_verb alc269_eeepc_dmic_init_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x23, AC_VERB_SET_CONNECT_SEL, 0x05}, @@ -12303,17 +12532,17 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = ALC269_DIGOUT_NID; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + add_mixer(spec, spec->kctls.list); /* create a beep mixer control if the pin 0x1d isn't assigned */ for (i = 0; i < ARRAY_SIZE(spec->autocfg.input_pins); i++) if (spec->autocfg.input_pins[i] == 0x1d) break; if (i >= ARRAY_SIZE(spec->autocfg.input_pins)) - spec->mixers[spec->num_mixers++] = alc269_beep_mixer; + add_mixer(spec, alc269_beep_mixer); - spec->init_verbs[spec->num_init_verbs++] = alc269_init_verbs; + add_verb(spec, alc269_init_verbs); spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; /* set default input source */ @@ -12325,8 +12554,8 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - spec->mixers[spec->num_mixers] = alc269_capture_mixer; - spec->num_mixers++; + if (!spec->cap_mixer) + set_capture_mixer(spec); store_pin_configs(codec); return 1; @@ -12355,7 +12584,9 @@ static const char *alc269_models[ALC269_MODEL_LAST] = { [ALC269_BASIC] = "basic", [ALC269_QUANTA_FL1] = "quanta", [ALC269_ASUS_EEEPC_P703] = "eeepc-p703", - [ALC269_ASUS_EEEPC_P901] = "eeepc-p901" + [ALC269_ASUS_EEEPC_P901] = "eeepc-p901", + [ALC269_FUJITSU] = "fujitsu", + [ALC269_LIFEBOOK] = "lifebook" }; static struct snd_pci_quirk alc269_cfg_tbl[] = { @@ -12366,12 +12597,14 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = { ALC269_ASUS_EEEPC_P901), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101", ALC269_ASUS_EEEPC_P901), + SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), + SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK), {} }; static struct alc_config_preset alc269_presets[] = { [ALC269_BASIC] = { - .mixers = { alc269_base_mixer, alc269_capture_mixer }, + .mixers = { alc269_base_mixer }, .init_verbs = { alc269_init_verbs }, .num_dacs = ARRAY_SIZE(alc269_dac_nids), .dac_nids = alc269_dac_nids, @@ -12393,7 +12626,8 @@ static struct alc_config_preset alc269_presets[] = { .init_hook = alc269_quanta_fl1_init_hook, }, [ALC269_ASUS_EEEPC_P703] = { - .mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer }, + .mixers = { alc269_eeepc_mixer }, + .cap_mixer = alc269_epc_capture_mixer, .init_verbs = { alc269_init_verbs, alc269_eeepc_amic_init_verbs }, .num_dacs = ARRAY_SIZE(alc269_dac_nids), @@ -12406,7 +12640,8 @@ static struct alc_config_preset alc269_presets[] = { .init_hook = alc269_eeepc_amic_inithook, }, [ALC269_ASUS_EEEPC_P901] = { - .mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer}, + .mixers = { alc269_eeepc_mixer }, + .cap_mixer = alc269_epc_capture_mixer, .init_verbs = { alc269_init_verbs, alc269_eeepc_dmic_init_verbs }, .num_dacs = ARRAY_SIZE(alc269_dac_nids), @@ -12418,6 +12653,32 @@ static struct alc_config_preset alc269_presets[] = { .unsol_event = alc269_eeepc_dmic_unsol_event, .init_hook = alc269_eeepc_dmic_inithook, }, + [ALC269_FUJITSU] = { + .mixers = { alc269_fujitsu_mixer, alc269_beep_mixer }, + .cap_mixer = alc269_epc_capture_mixer, + .init_verbs = { alc269_init_verbs, + alc269_eeepc_dmic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .input_mux = &alc269_eeepc_dmic_capture_source, + .unsol_event = alc269_eeepc_dmic_unsol_event, + .init_hook = alc269_eeepc_dmic_inithook, + }, + [ALC269_LIFEBOOK] = { + .mixers = { alc269_lifebook_mixer }, + .init_verbs = { alc269_init_verbs, alc269_lifebook_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .input_mux = &alc269_capture_source, + .unsol_event = alc269_lifebook_unsol_event, + .init_hook = alc269_lifebook_init_hook, + }, }; static int patch_alc269(struct hda_codec *codec) @@ -12472,6 +12733,8 @@ static int patch_alc269(struct hda_codec *codec) spec->adc_nids = alc269_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); spec->capsrc_nids = alc269_capsrc_nids; + if (!spec->cap_mixer) + set_capture_mixer(spec); codec->patch_ops = alc_patch_ops; if (board_config == ALC269_AUTO) @@ -12480,6 +12743,7 @@ static int patch_alc269(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc269_loopbacks; #endif + codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -12612,17 +12876,6 @@ static struct snd_kcontrol_new alc861_base_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT), - /* Capture mixer control */ - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, { } /* end */ }; @@ -12646,17 +12899,6 @@ static struct snd_kcontrol_new alc861_3ST_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT), - /* Capture mixer control */ - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -12674,18 +12916,6 @@ static struct snd_kcontrol_new alc861_toshiba_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT), - /*Capture mixer control */ - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, - { } /* end */ }; @@ -12709,17 +12939,6 @@ static struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT), - /* Capture mixer control */ - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -12751,17 +12970,6 @@ static struct snd_kcontrol_new alc861_asus_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_OUTPUT), - /* Capture mixer control */ - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -13293,25 +13501,6 @@ static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec, return 0; } -static struct snd_kcontrol_new alc861_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, - { } /* end */ -}; - static void alc861_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int dac_idx) @@ -13402,18 +13591,17 @@ static int alc861_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = ALC861_DIGOUT_NID; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + add_mixer(spec, spec->kctls.list); - spec->init_verbs[spec->num_init_verbs++] = alc861_auto_init_verbs; + add_verb(spec, alc861_auto_init_verbs); spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; spec->adc_nids = alc861_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); - spec->mixers[spec->num_mixers] = alc861_capture_mixer; - spec->num_mixers++; + set_capture_mixer(spec); store_pin_configs(codec); return 1; @@ -13644,6 +13832,7 @@ static int patch_alc861(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc861_loopbacks; #endif + codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -13709,11 +13898,6 @@ static struct hda_input_mux alc861vd_hp_capture_source = { }, }; -#define alc861vd_mux_enum_info alc_mux_enum_info -#define alc861vd_mux_enum_get alc_mux_enum_get -/* ALC861VD has the ALC882-type input selection (but has only one ADC) */ -#define alc861vd_mux_enum_put alc882_mux_enum_put - /* * 2ch mode */ @@ -13759,25 +13943,6 @@ static struct snd_kcontrol_new alc861vd_chmode_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc861vd_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc861vd_mux_enum_info, - .get = alc861vd_mux_enum_get, - .put = alc861vd_mux_enum_put, - }, - { } /* end */ -}; - /* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b */ @@ -14169,6 +14334,7 @@ static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int re static const char *alc861vd_models[ALC861VD_MODEL_LAST] = { [ALC660VD_3ST] = "3stack-660", [ALC660VD_3ST_DIG] = "3stack-660-digout", + [ALC660VD_ASUS_V1S] = "asus-v1s", [ALC861VD_3ST] = "3stack", [ALC861VD_3ST_DIG] = "3stack-digout", [ALC861VD_6ST_DIG] = "6stack-digout", @@ -14183,7 +14349,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP), SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), - SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC861VD_LENOVO), + SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO), @@ -14290,6 +14456,21 @@ static struct alc_config_preset alc861vd_presets[] = { .unsol_event = alc861vd_dallas_unsol_event, .init_hook = alc861vd_dallas_automute, }, + [ALC660VD_ASUS_V1S] = { + .mixers = { alc861vd_lenovo_mixer }, + .init_verbs = { alc861vd_volume_init_verbs, + alc861vd_3stack_init_verbs, + alc861vd_eapd_verbs, + alc861vd_lenovo_unsol_verbs }, + .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), + .dac_nids = alc660vd_dac_nids, + .dig_out_nid = ALC861VD_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), + .channel_mode = alc861vd_3stack_2ch_modes, + .input_mux = &alc861vd_capture_source, + .unsol_event = alc861vd_lenovo_unsol_event, + .init_hook = alc861vd_lenovo_automute, + }, }; /* @@ -14514,11 +14695,10 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = ALC861VD_DIGOUT_NID; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + add_mixer(spec, spec->kctls.list); - spec->init_verbs[spec->num_init_verbs++] - = alc861vd_volume_init_verbs; + add_verb(spec, alc861vd_volume_init_verbs); spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; @@ -14585,7 +14765,7 @@ static int patch_alc861vd(struct hda_codec *codec) spec->stream_name_analog = "ALC660-VD Analog"; spec->stream_name_digital = "ALC660-VD Digital"; /* always turn on EAPD */ - spec->init_verbs[spec->num_init_verbs++] = alc660vd_eapd_verbs; + add_verb(spec, alc660vd_eapd_verbs); } else { spec->stream_name_analog = "ALC861VD Analog"; spec->stream_name_digital = "ALC861VD Digital"; @@ -14600,9 +14780,9 @@ static int patch_alc861vd(struct hda_codec *codec) spec->adc_nids = alc861vd_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids); spec->capsrc_nids = alc861vd_capsrc_nids; + spec->is_mix_capture = 1; - spec->mixers[spec->num_mixers] = alc861vd_capture_mixer; - spec->num_mixers++; + set_capture_mixer(spec); spec->vmaster_nid = 0x02; @@ -14614,6 +14794,7 @@ static int patch_alc861vd(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc861vd_loopbacks; #endif + codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -14689,10 +14870,6 @@ static struct hda_input_mux alc663_m51va_capture_source = { }, }; -#define alc662_mux_enum_info alc_mux_enum_info -#define alc662_mux_enum_get alc_mux_enum_get -#define alc662_mux_enum_put alc882_mux_enum_put - /* * 2ch mode */ @@ -15278,25 +15455,6 @@ static struct hda_verb alc662_ecs_init_verbs[] = { {} }; -/* capture mixer elements */ -static struct snd_kcontrol_new alc662_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, - { } /* end */ -}; - static struct snd_kcontrol_new alc662_auto_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), @@ -15868,7 +16026,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { static struct alc_config_preset alc662_presets[] = { [ALC662_3ST_2ch_DIG] = { - .mixers = { alc662_3ST_2ch_mixer, alc662_capture_mixer }, + .mixers = { alc662_3ST_2ch_mixer }, .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -15879,8 +16037,7 @@ static struct alc_config_preset alc662_presets[] = { .input_mux = &alc662_capture_source, }, [ALC662_3ST_6ch_DIG] = { - .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer, - alc662_capture_mixer }, + .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer }, .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -15892,8 +16049,7 @@ static struct alc_config_preset alc662_presets[] = { .input_mux = &alc662_capture_source, }, [ALC662_3ST_6ch] = { - .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer, - alc662_capture_mixer }, + .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer }, .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -15903,8 +16059,7 @@ static struct alc_config_preset alc662_presets[] = { .input_mux = &alc662_capture_source, }, [ALC662_5ST_DIG] = { - .mixers = { alc662_base_mixer, alc662_chmode_mixer, - alc662_capture_mixer }, + .mixers = { alc662_base_mixer, alc662_chmode_mixer }, .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -15915,7 +16070,7 @@ static struct alc_config_preset alc662_presets[] = { .input_mux = &alc662_capture_source, }, [ALC662_LENOVO_101E] = { - .mixers = { alc662_lenovo_101e_mixer, alc662_capture_mixer }, + .mixers = { alc662_lenovo_101e_mixer }, .init_verbs = { alc662_init_verbs, alc662_sue_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -15926,7 +16081,7 @@ static struct alc_config_preset alc662_presets[] = { .init_hook = alc662_lenovo_101e_all_automute, }, [ALC662_ASUS_EEEPC_P701] = { - .mixers = { alc662_eeepc_p701_mixer, alc662_capture_mixer }, + .mixers = { alc662_eeepc_p701_mixer }, .init_verbs = { alc662_init_verbs, alc662_eeepc_sue_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), @@ -15938,7 +16093,7 @@ static struct alc_config_preset alc662_presets[] = { .init_hook = alc662_eeepc_inithook, }, [ALC662_ASUS_EEEPC_EP20] = { - .mixers = { alc662_eeepc_ep20_mixer, alc662_capture_mixer, + .mixers = { alc662_eeepc_ep20_mixer, alc662_chmode_mixer }, .init_verbs = { alc662_init_verbs, alc662_eeepc_ep20_sue_init_verbs }, @@ -15951,7 +16106,7 @@ static struct alc_config_preset alc662_presets[] = { .init_hook = alc662_eeepc_ep20_inithook, }, [ALC662_ECS] = { - .mixers = { alc662_ecs_mixer, alc662_capture_mixer }, + .mixers = { alc662_ecs_mixer }, .init_verbs = { alc662_init_verbs, alc662_ecs_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), @@ -15963,7 +16118,7 @@ static struct alc_config_preset alc662_presets[] = { .init_hook = alc662_eeepc_inithook, }, [ALC663_ASUS_M51VA] = { - .mixers = { alc663_m51va_mixer, alc662_capture_mixer}, + .mixers = { alc663_m51va_mixer }, .init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -15975,7 +16130,7 @@ static struct alc_config_preset alc662_presets[] = { .init_hook = alc663_m51va_inithook, }, [ALC663_ASUS_G71V] = { - .mixers = { alc663_g71v_mixer, alc662_capture_mixer}, + .mixers = { alc663_g71v_mixer }, .init_verbs = { alc662_init_verbs, alc663_g71v_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -15987,7 +16142,7 @@ static struct alc_config_preset alc662_presets[] = { .init_hook = alc663_g71v_inithook, }, [ALC663_ASUS_H13] = { - .mixers = { alc663_m51va_mixer, alc662_capture_mixer}, + .mixers = { alc663_m51va_mixer }, .init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -15998,7 +16153,7 @@ static struct alc_config_preset alc662_presets[] = { .init_hook = alc663_m51va_inithook, }, [ALC663_ASUS_G50V] = { - .mixers = { alc663_g50v_mixer, alc662_capture_mixer}, + .mixers = { alc663_g50v_mixer }, .init_verbs = { alc662_init_verbs, alc663_g50v_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -16010,7 +16165,8 @@ static struct alc_config_preset alc662_presets[] = { .init_hook = alc663_g50v_inithook, }, [ALC663_ASUS_MODE1] = { - .mixers = { alc663_m51va_mixer, alc662_auto_capture_mixer }, + .mixers = { alc663_m51va_mixer }, + .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, alc663_21jd_amic_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), @@ -16024,7 +16180,8 @@ static struct alc_config_preset alc662_presets[] = { .init_hook = alc663_mode1_inithook, }, [ALC662_ASUS_MODE2] = { - .mixers = { alc662_1bjd_mixer, alc662_auto_capture_mixer }, + .mixers = { alc662_1bjd_mixer }, + .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, alc662_1bjd_amic_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), @@ -16037,7 +16194,8 @@ static struct alc_config_preset alc662_presets[] = { .init_hook = alc662_mode2_inithook, }, [ALC663_ASUS_MODE3] = { - .mixers = { alc663_two_hp_m1_mixer, alc662_auto_capture_mixer }, + .mixers = { alc663_two_hp_m1_mixer }, + .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, alc663_two_hp_amic_m1_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), @@ -16051,8 +16209,8 @@ static struct alc_config_preset alc662_presets[] = { .init_hook = alc663_mode3_inithook, }, [ALC663_ASUS_MODE4] = { - .mixers = { alc663_asus_21jd_clfe_mixer, - alc662_auto_capture_mixer}, + .mixers = { alc663_asus_21jd_clfe_mixer }, + .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, alc663_21jd_amic_init_verbs}, .num_dacs = ARRAY_SIZE(alc662_dac_nids), @@ -16066,8 +16224,8 @@ static struct alc_config_preset alc662_presets[] = { .init_hook = alc663_mode4_inithook, }, [ALC663_ASUS_MODE5] = { - .mixers = { alc663_asus_15jd_clfe_mixer, - alc662_auto_capture_mixer }, + .mixers = { alc663_asus_15jd_clfe_mixer }, + .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, alc663_15jd_amic_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), @@ -16081,7 +16239,8 @@ static struct alc_config_preset alc662_presets[] = { .init_hook = alc663_mode5_inithook, }, [ALC663_ASUS_MODE6] = { - .mixers = { alc663_two_hp_m2_mixer, alc662_auto_capture_mixer }, + .mixers = { alc663_two_hp_m2_mixer }, + .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, alc663_two_hp_amic_m2_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), @@ -16342,24 +16501,20 @@ static int alc662_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + add_mixer(spec, spec->kctls.list); spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; - spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs; + add_verb(spec, alc662_auto_init_verbs); if (codec->vendor_id == 0x10ec0663) - spec->init_verbs[spec->num_init_verbs++] = - alc663_auto_init_verbs; + add_verb(spec, alc663_auto_init_verbs); err = alc_auto_add_mic_boost(codec); if (err < 0) return err; - spec->mixers[spec->num_mixers] = alc662_capture_mixer; - spec->num_mixers++; - store_pin_configs(codec); return 1; } @@ -16435,6 +16590,10 @@ static int patch_alc662(struct hda_codec *codec) spec->adc_nids = alc662_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); spec->capsrc_nids = alc662_capsrc_nids; + spec->is_mix_capture = 1; + + if (!spec->cap_mixer) + set_capture_mixer(spec); spec->vmaster_nid = 0x02; @@ -16445,6 +16604,7 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc662_loopbacks; #endif + codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -16452,7 +16612,7 @@ static int patch_alc662(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_realtek[] = { +static struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, @@ -16484,3 +16644,26 @@ struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:10ec*"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Realtek HD-audio codec"); + +static struct hda_codec_preset_list realtek_list = { + .preset = snd_hda_preset_realtek, + .owner = THIS_MODULE, +}; + +static int __init patch_realtek_init(void) +{ + return snd_hda_add_codec_preset(&realtek_list); +} + +static void __exit patch_realtek_exit(void) +{ + snd_hda_delete_codec_preset(&realtek_list); +} + +module_init(patch_realtek_init) +module_exit(patch_realtek_exit) diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 9332b63e406c..43b436c5d01b 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -28,7 +28,6 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" /* si3054 verbs */ #define SI3054_VERB_READ_NODE 0x900 @@ -283,7 +282,7 @@ static int patch_si3054(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_si3054[] = { +static struct hda_codec_preset snd_hda_preset_si3054[] = { { .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x11c13026, .name = "Si3054", .patch = patch_si3054 }, @@ -301,3 +300,35 @@ struct hda_codec_preset snd_hda_preset_si3054[] = { {} }; +MODULE_ALIAS("snd-hda-codec-id:163c3055"); +MODULE_ALIAS("snd-hda-codec-id:163c3155"); +MODULE_ALIAS("snd-hda-codec-id:11c13026"); +MODULE_ALIAS("snd-hda-codec-id:11c13055"); +MODULE_ALIAS("snd-hda-codec-id:11c13155"); +MODULE_ALIAS("snd-hda-codec-id:10573055"); +MODULE_ALIAS("snd-hda-codec-id:10573057"); +MODULE_ALIAS("snd-hda-codec-id:10573155"); +MODULE_ALIAS("snd-hda-codec-id:11063288"); +MODULE_ALIAS("snd-hda-codec-id:15433155"); +MODULE_ALIAS("snd-hda-codec-id:18540018"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Si3054 HD-audio modem codec"); + +static struct hda_codec_preset_list si3054_list = { + .preset = snd_hda_preset_si3054, + .owner = THIS_MODULE, +}; + +static int __init patch_si3054_init(void) +{ + return snd_hda_add_codec_preset(&si3054_list); +} + +static void __exit patch_si3054_exit(void) +{ + snd_hda_delete_codec_preset(&si3054_list); +} + +module_init(patch_si3054_init) +module_exit(patch_si3054_exit) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b77f330d2650..35b83dc6e19e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -30,17 +30,17 @@ #include <linux/pci.h> #include <sound/core.h> #include <sound/asoundef.h> +#include <sound/jack.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" #include "hda_beep.h" -#define NUM_CONTROL_ALLOC 32 - -#define STAC_VREF_EVENT 0x00 -#define STAC_INSERT_EVENT 0x10 -#define STAC_PWR_EVENT 0x20 -#define STAC_HP_EVENT 0x30 +enum { + STAC_VREF_EVENT = 1, + STAC_INSERT_EVENT, + STAC_PWR_EVENT, + STAC_HP_EVENT, +}; enum { STAC_REF, @@ -137,6 +137,19 @@ enum { STAC_927X_MODELS }; +struct sigmatel_event { + hda_nid_t nid; + unsigned char type; + unsigned char tag; + int data; +}; + +struct sigmatel_jack { + hda_nid_t nid; + int type; + struct snd_jack *jack; +}; + struct sigmatel_spec { struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; @@ -144,8 +157,6 @@ struct sigmatel_spec { int board_config; unsigned int eapd_switch: 1; unsigned int surr_switch: 1; - unsigned int line_switch: 1; - unsigned int mic_switch: 1; unsigned int alt_switch: 1; unsigned int hp_detect: 1; unsigned int spdif_mute: 1; @@ -170,12 +181,20 @@ struct sigmatel_spec { hda_nid_t *pwr_nids; hda_nid_t *dac_list; + /* jack detection */ + struct snd_array jacks; + + /* events */ + struct snd_array events; + /* playback */ struct hda_input_mux *mono_mux; struct hda_input_mux *amp_mux; unsigned int cur_mmux; struct hda_multi_out multiout; hda_nid_t dac_nids[5]; + hda_nid_t hp_dacs[5]; + hda_nid_t speaker_dacs[5]; /* capture */ hda_nid_t *adc_nids; @@ -199,7 +218,6 @@ struct sigmatel_spec { hda_nid_t *pin_nids; unsigned int num_pins; unsigned int *pin_configs; - unsigned int *bios_pin_configs; /* codec specific stuff */ struct hda_verb *init; @@ -220,15 +238,16 @@ struct sigmatel_spec { /* i/o switches */ unsigned int io_switch[2]; unsigned int clfe_swap; - unsigned int hp_switch; /* NID of HP as line-out */ + hda_nid_t line_switch; /* shared line-in for input and output */ + hda_nid_t mic_switch; /* shared mic-in for input and output */ + hda_nid_t hp_switch; /* NID of HP as line-out */ unsigned int aloopback; struct hda_pcm pcm_rec[2]; /* PCM information */ /* dynamic controls and input_mux */ struct auto_pin_cfg autocfg; - unsigned int num_kctl_alloc, num_kctl_used; - struct snd_kcontrol_new *kctl_alloc; + struct snd_array kctls; struct hda_input_mux private_dimux; struct hda_input_mux private_imux; struct hda_input_mux private_smux; @@ -272,9 +291,6 @@ static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = { }; #define STAC92HD73_DAC_COUNT 5 -static hda_nid_t stac92hd73xx_dac_nids[STAC92HD73_DAC_COUNT] = { - 0x15, 0x16, 0x17, 0x18, 0x19, -}; static hda_nid_t stac92hd73xx_mux_nids[4] = { 0x28, 0x29, 0x2a, 0x2b, @@ -293,11 +309,7 @@ static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = { 0x11, 0x12, 0 }; -#define STAC92HD81_DAC_COUNT 2 #define STAC92HD83_DAC_COUNT 3 -static hda_nid_t stac92hd83xxx_dac_nids[STAC92HD73_DAC_COUNT] = { - 0x13, 0x14, 0x22, -}; static hda_nid_t stac92hd83xxx_dmux_nids[2] = { 0x17, 0x18, @@ -339,10 +351,6 @@ static hda_nid_t stac92hd71bxx_smux_nids[2] = { 0x24, 0x25, }; -static hda_nid_t stac92hd71bxx_dac_nids[1] = { - 0x10, /*0x11, */ -}; - #define STAC92HD71BXX_NUM_DMICS 2 static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = { 0x18, 0x19, 0 @@ -574,12 +582,12 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, else nid = codec->slave_dig_outs[smux_idx - 1]; if (spec->cur_smux[smux_idx] == smux->num_items - 1) - val = AMP_OUT_MUTE; + val = HDA_AMP_MUTE; else - val = AMP_OUT_UNMUTE; + val = 0; /* un/mute SPDIF out */ - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, val); + snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, val); } return 0; } @@ -744,10 +752,6 @@ static struct hda_verb stac9200_eapd_init[] = { static struct hda_verb stac92hd73xx_6ch_core_init[] = { /* set master volume and direct control */ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* setup audio connections */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00}, - { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02}, /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, @@ -766,10 +770,6 @@ static struct hda_verb dell_eq_core_init[] = { /* set master volume to max value without distortion * and direct control */ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, - /* setup audio connections */ - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, - { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x02}, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x01}, /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, @@ -783,10 +783,6 @@ static struct hda_verb dell_eq_core_init[] = { static struct hda_verb dell_m6_core_init[] = { { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* setup audio connections */ - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, - { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, @@ -801,13 +797,6 @@ static struct hda_verb dell_m6_core_init[] = { static struct hda_verb stac92hd73xx_8ch_core_init[] = { /* set master volume and direct control */ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* setup audio connections */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00}, - { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* connect hp ports to dac3 */ - { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x03}, - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x03}, /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, @@ -825,15 +814,8 @@ static struct hda_verb stac92hd73xx_8ch_core_init[] = { static struct hda_verb stac92hd73xx_10ch_core_init[] = { /* set master volume and direct control */ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* setup audio connections */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, - { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02 }, /* dac3 is connected to import3 mux */ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb07f}, - /* connect hp ports to dac4 */ - { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x04}, - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x04}, /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, @@ -865,8 +847,6 @@ static struct hda_verb stac92hd83xxx_core_init[] = { static struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* connect headphone jack to dac1 */ - { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -886,8 +866,6 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* connect headphone jack to dac1 */ - { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* unmute right and left channels for nodes 0x0a, 0xd */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -1087,21 +1065,21 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0, HDA_INPUT), - HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0, HDA_INPUT), + HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x2, HDA_INPUT), /* - HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x1, HDA_INPUT), */ { } /* end */ }; @@ -1240,9 +1218,14 @@ static const char *slave_sws[] = { NULL }; +static void stac92xx_free_kctls(struct hda_codec *codec); +static int stac92xx_add_jack(struct hda_codec *codec, hda_nid_t nid, int type); + static int stac92xx_build_controls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid; int err; int i; @@ -1257,7 +1240,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) } if (spec->num_dmuxes > 0) { stac_dmux_mixer.count = spec->num_dmuxes; - err = snd_ctl_add(codec->bus->card, + err = snd_hda_ctl_add(codec, snd_ctl_new1(&stac_dmux_mixer, codec)); if (err < 0) return err; @@ -1273,7 +1256,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) spec->spdif_mute = 1; } stac_smux_mixer.count = spec->num_smuxes; - err = snd_ctl_add(codec->bus->card, + err = snd_hda_ctl_add(codec, snd_ctl_new1(&stac_smux_mixer, codec)); if (err < 0) return err; @@ -1312,6 +1295,37 @@ static int stac92xx_build_controls(struct hda_codec *codec) return err; } + stac92xx_free_kctls(codec); /* no longer needed */ + + /* create jack input elements */ + if (spec->hp_detect) { + for (i = 0; i < cfg->hp_outs; i++) { + int type = SND_JACK_HEADPHONE; + nid = cfg->hp_pins[i]; + /* jack detection */ + if (cfg->hp_outs == i) + type |= SND_JACK_LINEOUT; + err = stac92xx_add_jack(codec, nid, type); + if (err < 0) + return err; + } + } + for (i = 0; i < cfg->line_outs; i++) { + err = stac92xx_add_jack(codec, cfg->line_out_pins[i], + SND_JACK_LINEOUT); + if (err < 0) + return err; + } + for (i = 0; i < AUTO_PIN_LAST; i++) { + nid = cfg->input_pins[i]; + if (nid) { + err = stac92xx_add_jack(codec, nid, + SND_JACK_MICROPHONE); + if (err < 0) + return err; + } + } + return 0; } @@ -1720,6 +1734,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP dv5", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4, "HP dv7", STAC_HP_M4), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc, + "HP dv7", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a, "unknown HP", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, @@ -2203,12 +2219,11 @@ static int stac92xx_save_bios_config_regs(struct hda_codec *codec) int i; struct sigmatel_spec *spec = codec->spec; - if (! spec->bios_pin_configs) { - spec->bios_pin_configs = kcalloc(spec->num_pins, - sizeof(*spec->bios_pin_configs), GFP_KERNEL); - if (! spec->bios_pin_configs) - return -ENOMEM; - } + kfree(spec->pin_configs); + spec->pin_configs = kcalloc(spec->num_pins, sizeof(*spec->pin_configs), + GFP_KERNEL); + if (!spec->pin_configs) + return -ENOMEM; for (i = 0; i < spec->num_pins; i++) { hda_nid_t nid = spec->pin_nids[i]; @@ -2218,7 +2233,7 @@ static int stac92xx_save_bios_config_regs(struct hda_codec *codec) AC_VERB_GET_CONFIG_DEFAULT, 0x00); snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n", nid, pin_cfg); - spec->bios_pin_configs[i] = pin_cfg; + spec->pin_configs[i] = pin_cfg; } return 0; @@ -2260,6 +2275,39 @@ static void stac92xx_set_config_regs(struct hda_codec *codec) spec->pin_configs[i]); } +static int stac_save_pin_cfgs(struct hda_codec *codec, unsigned int *pins) +{ + struct sigmatel_spec *spec = codec->spec; + + if (!pins) + return stac92xx_save_bios_config_regs(codec); + + kfree(spec->pin_configs); + spec->pin_configs = kmemdup(pins, + spec->num_pins * sizeof(*pins), + GFP_KERNEL); + if (!spec->pin_configs) + return -ENOMEM; + + stac92xx_set_config_regs(codec); + return 0; +} + +static void stac_change_pin_config(struct hda_codec *codec, hda_nid_t nid, + unsigned int cfg) +{ + struct sigmatel_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_pins; i++) { + if (spec->pin_nids[i] == nid) { + spec->pin_configs[i] = cfg; + stac92xx_set_config_reg(codec, nid, cfg); + break; + } + } +} + /* * Analog playback callbacks */ @@ -2337,7 +2385,7 @@ static int stac92xx_capture_pcm_prepare(struct hda_pcm_stream *hinfo, if (spec->powerdown_adcs) { msleep(40); - snd_hda_codec_write_cache(codec, nid, 0, + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); } snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); @@ -2353,7 +2401,7 @@ static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, snd_hda_codec_cleanup_stream(codec, nid); if (spec->powerdown_adcs) - snd_hda_codec_write_cache(codec, nid, 0, + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); return 0; } @@ -2485,6 +2533,9 @@ static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol, return 0; } +static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid, + unsigned char type); + static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -2497,7 +2548,7 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, /* check to be sure that the ports are upto date with * switch changes */ - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + stac_issue_unsol_event(codec, nid, STAC_HP_EVENT); return 1; } @@ -2537,7 +2588,7 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ * appropriately according to the pin direction */ if (spec->hp_detect) - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + stac_issue_unsol_event(codec, nid, STAC_HP_EVENT); return 1; } @@ -2632,28 +2683,16 @@ static int stac92xx_add_control_temp(struct sigmatel_spec *spec, { struct snd_kcontrol_new *knew; - if (spec->num_kctl_used >= spec->num_kctl_alloc) { - int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC; - - knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */ - if (! knew) - return -ENOMEM; - if (spec->kctl_alloc) { - memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc); - kfree(spec->kctl_alloc); - } - spec->kctl_alloc = knew; - spec->num_kctl_alloc = num; - } - - knew = &spec->kctl_alloc[spec->num_kctl_used]; + snd_array_init(&spec->kctls, sizeof(*knew), 32); + knew = snd_array_new(&spec->kctls); + if (!knew) + return -ENOMEM; *knew = *ktemp; knew->index = idx; knew->name = kstrdup(name, GFP_KERNEL); if (!knew->name) return -ENOMEM; knew->private_value = val; - spec->num_kctl_used++; return 0; } @@ -2674,70 +2713,53 @@ static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type, return stac92xx_add_control_idx(spec, type, 0, name, val); } -/* flag inputs as additional dynamic lineouts */ -static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cfg *cfg) +/* check whether the line-input can be used as line-out */ +static hda_nid_t check_line_out_switch(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - unsigned int wcaps, wtype; - int i, num_dacs = 0; - - /* use the wcaps cache to count all DACs available for line-outs */ - for (i = 0; i < codec->num_nodes; i++) { - wcaps = codec->wcaps[i]; - wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid; + unsigned int pincap; - if (wtype == AC_WID_AUD_OUT && !(wcaps & AC_WCAP_DIGITAL)) - num_dacs++; - } + if (cfg->line_out_type != AUTO_PIN_LINE_OUT) + return 0; + nid = cfg->input_pins[AUTO_PIN_LINE]; + pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + if (pincap & AC_PINCAP_OUT) + return nid; + return 0; +} - snd_printdd("%s: total dac count=%d\n", __func__, num_dacs); - - switch (cfg->line_outs) { - case 3: - /* add line-in as side */ - if (cfg->input_pins[AUTO_PIN_LINE] && num_dacs > 3) { - cfg->line_out_pins[cfg->line_outs] = - cfg->input_pins[AUTO_PIN_LINE]; - spec->line_switch = 1; - cfg->line_outs++; - } - break; - case 2: - /* add line-in as clfe and mic as side */ - if (cfg->input_pins[AUTO_PIN_LINE] && num_dacs > 2) { - cfg->line_out_pins[cfg->line_outs] = - cfg->input_pins[AUTO_PIN_LINE]; - spec->line_switch = 1; - cfg->line_outs++; - } - if (cfg->input_pins[AUTO_PIN_MIC] && num_dacs > 3) { - cfg->line_out_pins[cfg->line_outs] = - cfg->input_pins[AUTO_PIN_MIC]; - spec->mic_switch = 1; - cfg->line_outs++; - } - break; - case 1: - /* add line-in as surr and mic as clfe */ - if (cfg->input_pins[AUTO_PIN_LINE] && num_dacs > 1) { - cfg->line_out_pins[cfg->line_outs] = - cfg->input_pins[AUTO_PIN_LINE]; - spec->line_switch = 1; - cfg->line_outs++; - } - if (cfg->input_pins[AUTO_PIN_MIC] && num_dacs > 2) { - cfg->line_out_pins[cfg->line_outs] = - cfg->input_pins[AUTO_PIN_MIC]; - spec->mic_switch = 1; - cfg->line_outs++; +/* check whether the mic-input can be used as line-out */ +static hda_nid_t check_mic_out_switch(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int def_conf, pincap; + unsigned int mic_pin; + + if (cfg->line_out_type != AUTO_PIN_LINE_OUT) + return 0; + mic_pin = AUTO_PIN_MIC; + for (;;) { + hda_nid_t nid = cfg->input_pins[mic_pin]; + def_conf = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + /* some laptops have an internal analog microphone + * which can't be used as a output */ + if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { + pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + if (pincap & AC_PINCAP_OUT) + return nid; } - break; + if (mic_pin == AUTO_PIN_MIC) + mic_pin = AUTO_PIN_FRONT_MIC; + else + break; } - return 0; } - static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) { int i; @@ -2750,6 +2772,52 @@ static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) return 0; } +static int check_all_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) +{ + int i; + if (is_in_dac_nids(spec, nid)) + return 1; + for (i = 0; i < spec->autocfg.hp_outs; i++) + if (spec->hp_dacs[i] == nid) + return 1; + for (i = 0; i < spec->autocfg.speaker_outs; i++) + if (spec->speaker_dacs[i] == nid) + return 1; + return 0; +} + +static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + int j, conn_len; + hda_nid_t conn[HDA_MAX_CONNECTIONS]; + unsigned int wcaps, wtype; + + conn_len = snd_hda_get_connections(codec, nid, conn, + HDA_MAX_CONNECTIONS); + for (j = 0; j < conn_len; j++) { + wcaps = snd_hda_param_read(codec, conn[j], + AC_PAR_AUDIO_WIDGET_CAP); + wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + /* we check only analog outputs */ + if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL)) + continue; + /* if this route has a free DAC, assign it */ + if (!check_all_dac_nids(spec, conn[j])) { + if (conn_len > 1) { + /* select this DAC in the pin's input mux */ + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, j); + } + return conn[j]; + } + } + return 0; +} + +static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid); +static int add_spec_extra_dacs(struct sigmatel_spec *spec, hda_nid_t nid); + /* * Fill in the dac_nids table from the parsed pin configuration * This function only works when every pin in line_out_pins[] @@ -2757,31 +2825,17 @@ static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) * codecs are not connected directly to a DAC, such as the 9200 * and 9202/925x. For those, dac_nids[] must be hard-coded. */ -static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, - struct auto_pin_cfg *cfg) +static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - int i, j, conn_len = 0; - hda_nid_t nid, conn[HDA_MAX_CONNECTIONS]; - unsigned int wcaps, wtype; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + hda_nid_t nid, dac; for (i = 0; i < cfg->line_outs; i++) { nid = cfg->line_out_pins[i]; - conn_len = snd_hda_get_connections(codec, nid, conn, - HDA_MAX_CONNECTIONS); - for (j = 0; j < conn_len; j++) { - wcaps = snd_hda_param_read(codec, conn[j], - AC_PAR_AUDIO_WIDGET_CAP); - wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wtype != AC_WID_AUD_OUT || - (wcaps & AC_WCAP_DIGITAL)) - continue; - /* conn[j] is a DAC routed to this line-out */ - if (!is_in_dac_nids(spec, conn[j])) - break; - } - - if (j == conn_len) { + dac = get_unassigned_dac(codec, nid); + if (!dac) { if (spec->multiout.num_dacs > 0) { /* we have already working output pins, * so let's drop the broken ones again @@ -2795,24 +2849,64 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, __func__, nid); return -ENODEV; } + add_spec_dacs(spec, dac); + } - spec->multiout.dac_nids[i] = conn[j]; - spec->multiout.num_dacs++; - if (conn_len > 1) { - /* select this DAC in the pin's input mux */ - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_CONNECT_SEL, j); + /* add line-in as output */ + nid = check_line_out_switch(codec); + if (nid) { + dac = get_unassigned_dac(codec, nid); + if (dac) { + snd_printdd("STAC: Add line-in 0x%x as output %d\n", + nid, cfg->line_outs); + cfg->line_out_pins[cfg->line_outs] = nid; + cfg->line_outs++; + spec->line_switch = nid; + add_spec_dacs(spec, dac); + } + } + /* add mic as output */ + nid = check_mic_out_switch(codec); + if (nid) { + dac = get_unassigned_dac(codec, nid); + if (dac) { + snd_printdd("STAC: Add mic-in 0x%x as output %d\n", + nid, cfg->line_outs); + cfg->line_out_pins[cfg->line_outs] = nid; + cfg->line_outs++; + spec->mic_switch = nid; + add_spec_dacs(spec, dac); + } + } + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + dac = get_unassigned_dac(codec, nid); + if (dac) { + if (!spec->multiout.hp_nid) + spec->multiout.hp_nid = dac; + else + add_spec_extra_dacs(spec, dac); } + spec->hp_dacs[i] = dac; + } + + for (i = 0; i < cfg->speaker_outs; i++) { + nid = cfg->speaker_pins[i]; + dac = get_unassigned_dac(codec, nid); + if (dac) + add_spec_extra_dacs(spec, dac); + spec->speaker_dacs[i] = dac; } - snd_printd("dac_nids=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", + snd_printd("stac92xx: dac_nids=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", spec->multiout.num_dacs, spec->multiout.dac_nids[0], spec->multiout.dac_nids[1], spec->multiout.dac_nids[2], spec->multiout.dac_nids[3], spec->multiout.dac_nids[4]); + return 0; } @@ -2837,9 +2931,7 @@ static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_ static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) { - if (!spec->multiout.hp_nid) - spec->multiout.hp_nid = nid; - else if (spec->multiout.num_dacs > 4) { + if (spec->multiout.num_dacs > 4) { printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid); return 1; } else { @@ -2849,35 +2941,47 @@ static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) return 0; } -static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) +static int add_spec_extra_dacs(struct sigmatel_spec *spec, hda_nid_t nid) { - if (is_in_dac_nids(spec, nid)) - return 1; + int i; + for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) { + if (!spec->multiout.extra_out_nid[i]) { + spec->multiout.extra_out_nid[i] = nid; + return 0; + } + } + printk(KERN_WARNING "stac92xx: No space for extra DAC 0x%x\n", nid); + return 1; +} + +static int is_unique_dac(struct sigmatel_spec *spec, hda_nid_t nid) +{ + int i; + + if (spec->autocfg.line_outs != 1) + return 0; if (spec->multiout.hp_nid == nid) - return 1; - return 0; + return 0; + for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) + if (spec->multiout.extra_out_nid[i] == nid) + return 0; + return 1; } /* add playback controls from the parsed DAC table */ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { + struct sigmatel_spec *spec = codec->spec; static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; hda_nid_t nid = 0; int i, err; + unsigned int wid_caps; - struct sigmatel_spec *spec = codec->spec; - unsigned int wid_caps, pincap; - - - for (i = 0; i < cfg->line_outs && i < spec->multiout.num_dacs; i++) { - if (!spec->multiout.dac_nids[i]) - continue; - + for (i = 0; i < cfg->line_outs && spec->multiout.dac_nids[i]; i++) { nid = spec->multiout.dac_nids[i]; - if (i == 2) { /* Center/LFE */ err = create_controls(spec, "Center", nid, 1); @@ -2899,16 +3003,24 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } } else { - err = create_controls(spec, chname[i], nid, 3); + const char *name = chname[i]; + /* if it's a single DAC, assign a better name */ + if (!i && is_unique_dac(spec, nid)) { + switch (cfg->line_out_type) { + case AUTO_PIN_HP_OUT: + name = "Headphone"; + break; + case AUTO_PIN_SPEAKER_OUT: + name = "Speaker"; + break; + } + } + err = create_controls(spec, name, nid, 3); if (err < 0) return err; } } - if ((spec->multiout.num_dacs - cfg->line_outs) > 0 && - cfg->hp_outs == 1 && !spec->multiout.hp_nid) - spec->multiout.hp_nid = nid; - if (cfg->hp_outs > 1 && cfg->line_out_type == AUTO_PIN_LINE_OUT) { err = stac92xx_add_control(spec, STAC_CTL_WIDGET_HP_SWITCH, @@ -2919,45 +3031,19 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } if (spec->line_switch) { - nid = cfg->input_pins[AUTO_PIN_LINE]; - pincap = snd_hda_param_read(codec, nid, - AC_PAR_PIN_CAP); - if (pincap & AC_PINCAP_OUT) { - err = stac92xx_add_control(spec, - STAC_CTL_WIDGET_IO_SWITCH, - "Line In as Output Switch", nid << 8); - if (err < 0) - return err; - } + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, + "Line In as Output Switch", + spec->line_switch << 8); + if (err < 0) + return err; } if (spec->mic_switch) { - unsigned int def_conf; - unsigned int mic_pin = AUTO_PIN_MIC; -again: - nid = cfg->input_pins[mic_pin]; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); - /* some laptops have an internal analog microphone - * which can't be used as a output */ - if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { - pincap = snd_hda_param_read(codec, nid, - AC_PAR_PIN_CAP); - if (pincap & AC_PINCAP_OUT) { - err = stac92xx_add_control(spec, - STAC_CTL_WIDGET_IO_SWITCH, - "Mic as Output Switch", (nid << 8) | 1); - nid = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONNECT_LIST, 0) & 0xff; - if (!check_in_dac_nids(spec, nid)) - add_spec_dacs(spec, nid); - if (err < 0) - return err; - } - } else if (mic_pin == AUTO_PIN_MIC) { - mic_pin = AUTO_PIN_FRONT_MIC; - goto again; - } + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, + "Mic as Output Switch", + (spec->mic_switch << 8) | 1); + if (err < 0) + return err; } return 0; @@ -2969,55 +3055,39 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, { struct sigmatel_spec *spec = codec->spec; hda_nid_t nid; - int i, old_num_dacs, err; + int i, err, nums; - old_num_dacs = spec->multiout.num_dacs; + nums = 0; for (i = 0; i < cfg->hp_outs; i++) { + static const char *pfxs[] = { + "Headphone", "Headphone2", "Headphone3", + }; unsigned int wid_caps = get_wcaps(codec, cfg->hp_pins[i]); if (wid_caps & AC_WCAP_UNSOL_CAP) spec->hp_detect = 1; - nid = snd_hda_codec_read(codec, cfg->hp_pins[i], 0, - AC_VERB_GET_CONNECT_LIST, 0) & 0xff; - if (check_in_dac_nids(spec, nid)) - nid = 0; - if (! nid) + if (nums >= ARRAY_SIZE(pfxs)) continue; - add_spec_dacs(spec, nid); - } - for (i = 0; i < cfg->speaker_outs; i++) { - nid = snd_hda_codec_read(codec, cfg->speaker_pins[i], 0, - AC_VERB_GET_CONNECT_LIST, 0) & 0xff; - if (check_in_dac_nids(spec, nid)) - nid = 0; - if (! nid) + nid = spec->hp_dacs[i]; + if (!nid) continue; - add_spec_dacs(spec, nid); - } - for (i = 0; i < cfg->line_outs; i++) { - nid = snd_hda_codec_read(codec, cfg->line_out_pins[i], 0, - AC_VERB_GET_CONNECT_LIST, 0) & 0xff; - if (check_in_dac_nids(spec, nid)) - nid = 0; - if (! nid) - continue; - add_spec_dacs(spec, nid); + err = create_controls(spec, pfxs[nums++], nid, 3); + if (err < 0) + return err; } - for (i = old_num_dacs; i < spec->multiout.num_dacs; i++) { + nums = 0; + for (i = 0; i < cfg->speaker_outs; i++) { static const char *pfxs[] = { "Speaker", "External Speaker", "Speaker2", }; - err = create_controls(spec, pfxs[i - old_num_dacs], - spec->multiout.dac_nids[i], 3); - if (err < 0) - return err; - } - if (spec->multiout.hp_nid) { - err = create_controls(spec, "Headphone", - spec->multiout.hp_nid, 3); + if (nums >= ARRAY_SIZE(pfxs)) + continue; + nid = spec->speaker_dacs[i]; + if (!nid) + continue; + err = create_controls(spec, pfxs[nums++], nid, 3); if (err < 0) return err; } - return 0; } @@ -3355,7 +3425,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out { struct sigmatel_spec *spec = codec->spec; int err; - int hp_speaker_swap = 0; if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, @@ -3373,13 +3442,15 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out * speaker_outs so that the following routines can handle * HP pins as primary outputs. */ + snd_printdd("stac92xx: Enabling multi-HPs workaround\n"); memcpy(spec->autocfg.speaker_pins, spec->autocfg.line_out_pins, sizeof(spec->autocfg.line_out_pins)); spec->autocfg.speaker_outs = spec->autocfg.line_outs; memcpy(spec->autocfg.line_out_pins, spec->autocfg.hp_pins, sizeof(spec->autocfg.hp_pins)); spec->autocfg.line_outs = spec->autocfg.hp_outs; - hp_speaker_swap = 1; + spec->autocfg.line_out_type = AUTO_PIN_HP_OUT; + spec->autocfg.hp_outs = 0; } if (spec->autocfg.mono_out_pin) { int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) & @@ -3431,11 +3502,11 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out AC_PINCTL_OUT_EN); } - if ((err = stac92xx_add_dyn_out_pins(codec, &spec->autocfg)) < 0) - return err; - if (spec->multiout.num_dacs == 0) - if ((err = stac92xx_auto_fill_dac_nids(codec, &spec->autocfg)) < 0) + if (!spec->multiout.num_dacs) { + err = stac92xx_auto_fill_dac_nids(codec); + if (err < 0) return err; + } err = stac92xx_auto_create_multi_out_ctls(codec, &spec->autocfg); @@ -3473,19 +3544,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out } #endif - if (hp_speaker_swap == 1) { - /* Restore the hp_outs and line_outs */ - memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins, - sizeof(spec->autocfg.line_out_pins)); - spec->autocfg.hp_outs = spec->autocfg.line_outs; - memcpy(spec->autocfg.line_out_pins, spec->autocfg.speaker_pins, - sizeof(spec->autocfg.speaker_pins)); - spec->autocfg.line_outs = spec->autocfg.speaker_outs; - memset(spec->autocfg.speaker_pins, 0, - sizeof(spec->autocfg.speaker_pins)); - spec->autocfg.speaker_outs = 0; - } - err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg); if (err < 0) @@ -3530,11 +3588,12 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (dig_in && spec->autocfg.dig_in_pin) spec->dig_in_nid = dig_in; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; spec->input_mux = &spec->private_imux; - spec->dinput_mux = &spec->private_dimux; + if (!spec->dinput_mux) + spec->dinput_mux = &spec->private_dimux; spec->sinput_mux = &spec->private_smux; spec->mono_mux = &spec->private_mono_mux; spec->amp_mux = &spec->private_amp_mux; @@ -3638,8 +3697,8 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_in_pin) spec->dig_in_nid = 0x04; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; spec->input_mux = &spec->private_imux; spec->dinput_mux = &spec->private_dimux; @@ -3683,13 +3742,101 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */ } +static int stac92xx_add_jack(struct hda_codec *codec, + hda_nid_t nid, int type) +{ +#ifdef CONFIG_SND_JACK + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_jack *jack; + int def_conf = snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + int connectivity = get_defcfg_connect(def_conf); + char name[32]; + + if (connectivity && connectivity != AC_JACK_PORT_FIXED) + return 0; + + snd_array_init(&spec->jacks, sizeof(*jack), 32); + jack = snd_array_new(&spec->jacks); + if (!jack) + return -ENOMEM; + jack->nid = nid; + jack->type = type; + + sprintf(name, "%s at %s %s Jack", + snd_hda_get_jack_type(def_conf), + snd_hda_get_jack_connectivity(def_conf), + snd_hda_get_jack_location(def_conf)); + + return snd_jack_new(codec->bus->card, name, type, &jack->jack); +#else + return 0; +#endif +} + +static int stac_add_event(struct sigmatel_spec *spec, hda_nid_t nid, + unsigned char type, int data) +{ + struct sigmatel_event *event; + + snd_array_init(&spec->events, sizeof(*event), 32); + event = snd_array_new(&spec->events); + if (!event) + return -ENOMEM; + event->nid = nid; + event->type = type; + event->tag = spec->events.used; + event->data = data; + + return event->tag; +} + +static struct sigmatel_event *stac_get_event(struct hda_codec *codec, + hda_nid_t nid, unsigned char type) +{ + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_event *event = spec->events.list; + int i; + + for (i = 0; i < spec->events.used; i++, event++) { + if (event->nid == nid && event->type == type) + return event; + } + return NULL; +} + +static struct sigmatel_event *stac_get_event_from_tag(struct hda_codec *codec, + unsigned char tag) +{ + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_event *event = spec->events.list; + int i; + + for (i = 0; i < spec->events.used; i++, event++) { + if (event->tag == tag) + return event; + } + return NULL; +} + static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, - unsigned int event) + unsigned int type) { - if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | event)); + struct sigmatel_event *event; + int tag; + + if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)) + return; + event = stac_get_event(codec, nid, type); + if (event) + tag = event->tag; + else + tag = stac_add_event(codec->spec, nid, type, 0); + if (tag < 0) + return; + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | tag); } static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) @@ -3709,9 +3856,8 @@ static void stac92xx_power_down(struct hda_codec *codec) /* power down inactive DACs */ hda_nid_t *dac; for (dac = spec->dac_list; *dac; dac++) - if (!is_in_dac_nids(spec, *dac) && - spec->multiout.hp_nid != *dac) - snd_hda_codec_write_cache(codec, *dac, 0, + if (!check_all_dac_nids(spec, *dac)) + snd_hda_codec_write(codec, *dac, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); } @@ -3730,7 +3876,7 @@ static int stac92xx_init(struct hda_codec *codec) /* power down adcs initially */ if (spec->powerdown_adcs) for (i = 0; i < spec->num_adcs; i++) - snd_hda_codec_write_cache(codec, + snd_hda_codec_write(codec, spec->adc_nids[i], 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); @@ -3746,37 +3892,51 @@ static int stac92xx_init(struct hda_codec *codec) /* set up pins */ if (spec->hp_detect) { /* Enable unsolicited responses on the HP widget */ - for (i = 0; i < cfg->hp_outs; i++) - enable_pin_detect(codec, cfg->hp_pins[i], - STAC_HP_EVENT); + for (i = 0; i < cfg->hp_outs; i++) { + hda_nid_t nid = cfg->hp_pins[i]; + enable_pin_detect(codec, nid, STAC_HP_EVENT); + } /* force to enable the first line-out; the others are set up * in unsol_event */ stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0], - AC_PINCTL_OUT_EN); - stac92xx_auto_init_hp_out(codec); + AC_PINCTL_OUT_EN); /* fake event to set up pins */ - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0], + STAC_HP_EVENT); } else { stac92xx_auto_init_multi_out(codec); stac92xx_auto_init_hp_out(codec); + for (i = 0; i < cfg->hp_outs; i++) + stac_toggle_power_map(codec, cfg->hp_pins[i], 1); } for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = cfg->input_pins[i]; if (nid) { - unsigned int pinctl; + unsigned int pinctl, conf; if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC) { /* for mic pins, force to initialize */ pinctl = stac92xx_get_vref(codec, nid); + pinctl |= AC_PINCTL_IN_EN; + stac92xx_auto_set_pinctl(codec, nid, pinctl); } else { pinctl = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); /* if PINCTL already set then skip */ - if (pinctl & AC_PINCTL_IN_EN) - continue; + if (!(pinctl & AC_PINCTL_IN_EN)) { + pinctl |= AC_PINCTL_IN_EN; + stac92xx_auto_set_pinctl(codec, nid, + pinctl); + } + } + conf = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + if (get_defcfg_connect(conf) != AC_JACK_PORT_FIXED) { + enable_pin_detect(codec, nid, + STAC_INSERT_EVENT); + stac_issue_unsol_event(codec, nid, + STAC_INSERT_EVENT); } - pinctl |= AC_PINCTL_IN_EN; - stac92xx_auto_set_pinctl(codec, nid, pinctl); } } for (i = 0; i < spec->num_dmics; i++) @@ -3791,9 +3951,14 @@ static int stac92xx_init(struct hda_codec *codec) for (i = 0; i < spec->num_pwrs; i++) { hda_nid_t nid = spec->pwr_nids[i]; int pinctl, def_conf; - int event = STAC_PWR_EVENT; - if (is_nid_hp_pin(cfg, nid) && spec->hp_detect) + /* power on when no jack detection is available */ + if (!spec->hp_detect) { + stac_toggle_power_map(codec, nid, 1); + continue; + } + + if (is_nid_hp_pin(cfg, nid)) continue; /* already has an unsol event */ pinctl = snd_hda_codec_read(codec, nid, 0, @@ -3802,8 +3967,10 @@ static int stac92xx_init(struct hda_codec *codec) * any attempts on powering down a input port cause the * referenced VREF to act quirky. */ - if (pinctl & AC_PINCTL_IN_EN) + if (pinctl & AC_PINCTL_IN_EN) { + stac_toggle_power_map(codec, nid, 1); continue; + } def_conf = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); def_conf = get_defcfg_connect(def_conf); @@ -3814,30 +3981,54 @@ static int stac92xx_init(struct hda_codec *codec) stac_toggle_power_map(codec, nid, 1); continue; } - enable_pin_detect(codec, spec->pwr_nids[i], event | i); - codec->patch_ops.unsol_event(codec, (event | i) << 26); + if (!stac_get_event(codec, nid, STAC_INSERT_EVENT)) { + enable_pin_detect(codec, nid, STAC_PWR_EVENT); + stac_issue_unsol_event(codec, nid, STAC_PWR_EVENT); + } } if (spec->dac_list) stac92xx_power_down(codec); return 0; } +static void stac92xx_free_jacks(struct hda_codec *codec) +{ +#ifdef CONFIG_SND_JACK + /* free jack instances manually when clearing/reconfiguring */ + struct sigmatel_spec *spec = codec->spec; + if (!codec->bus->shutdown && spec->jacks.list) { + struct sigmatel_jack *jacks = spec->jacks.list; + int i; + for (i = 0; i < spec->jacks.used; i++) + snd_device_free(codec->bus->card, &jacks[i].jack); + } + snd_array_free(&spec->jacks); +#endif +} + +static void stac92xx_free_kctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + if (spec->kctls.list) { + struct snd_kcontrol_new *kctl = spec->kctls.list; + int i; + for (i = 0; i < spec->kctls.used; i++) + kfree(kctl[i].name); + } + snd_array_free(&spec->kctls); +} + static void stac92xx_free(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - int i; if (! spec) return; - if (spec->kctl_alloc) { - for (i = 0; i < spec->num_kctl_used; i++) - kfree(spec->kctl_alloc[i].name); - kfree(spec->kctl_alloc); - } - - if (spec->bios_pin_configs) - kfree(spec->bios_pin_configs); + kfree(spec->pin_configs); + stac92xx_free_jacks(codec); + snd_array_free(&spec->events); kfree(spec); snd_hda_detach_beep_device(codec); @@ -3856,11 +4047,7 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, * "xxx as Output" mixer switch */ struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - if ((nid == cfg->input_pins[AUTO_PIN_LINE] && - spec->line_switch) || - (nid == cfg->input_pins[AUTO_PIN_MIC] && - spec->mic_switch)) + if (nid == spec->line_switch || nid == spec->mic_switch) return; } @@ -3884,20 +4071,13 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, pin_ctl & ~flag); } -static int get_hp_pin_presence(struct hda_codec *codec, hda_nid_t nid) +static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) { if (!nid) return 0; if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0x00) - & (1 << 31)) { - unsigned int pinctl; - pinctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - if (pinctl & AC_PINCTL_IN_EN) - return 0; /* mic- or line-input */ - else - return 1; /* HP-output */ - } + & (1 << 31)) + return 1; return 0; } @@ -3909,11 +4089,9 @@ static int no_hp_sensing(struct sigmatel_spec *spec, int i) struct auto_pin_cfg *cfg = &spec->autocfg; /* ignore sensing of shared line and mic jacks */ - if (spec->line_switch && - cfg->hp_pins[i] == cfg->input_pins[AUTO_PIN_LINE]) + if (cfg->hp_pins[i] == spec->line_switch) return 1; - if (spec->mic_switch && - cfg->hp_pins[i] == cfg->input_pins[AUTO_PIN_MIC]) + if (cfg->hp_pins[i] == spec->mic_switch) return 1; /* ignore if the pin is set as line-out */ if (cfg->hp_pins[i] == spec->hp_switch) @@ -3921,7 +4099,7 @@ static int no_hp_sensing(struct sigmatel_spec *spec, int i) return 0; } -static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) +static void stac92xx_hp_detect(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; @@ -3937,7 +4115,14 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) break; if (no_hp_sensing(spec, i)) continue; - presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); + presence = get_pin_presence(codec, cfg->hp_pins[i]); + if (presence) { + unsigned int pinctl; + pinctl = snd_hda_codec_read(codec, cfg->hp_pins[i], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + if (pinctl & AC_PINCTL_IN_EN) + presence = 0; /* mic- or line-input */ + } } if (presence) { @@ -4014,50 +4199,145 @@ static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, static void stac92xx_pin_sense(struct hda_codec *codec, hda_nid_t nid) { - stac_toggle_power_map(codec, nid, get_hp_pin_presence(codec, nid)); + stac_toggle_power_map(codec, nid, get_pin_presence(codec, nid)); +} + +static void stac92xx_report_jack(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_jack *jacks = spec->jacks.list; + + if (jacks) { + int i; + for (i = 0; i < spec->jacks.used; i++) { + if (jacks->nid == nid) { + unsigned int pin_ctl = + snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_PIN_WIDGET_CONTROL, + 0x00); + int type = jacks->type; + if (type == (SND_JACK_LINEOUT + | SND_JACK_HEADPHONE)) + type = (pin_ctl & AC_PINCTL_HP_EN) + ? SND_JACK_HEADPHONE : SND_JACK_LINEOUT; + snd_jack_report(jacks->jack, + get_pin_presence(codec, nid) + ? type : 0); + } + jacks++; + } + } +} + +static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid, + unsigned char type) +{ + struct sigmatel_event *event = stac_get_event(codec, nid, type); + if (!event) + return; + codec->patch_ops.unsol_event(codec, (unsigned)event->tag << 26); } static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) { struct sigmatel_spec *spec = codec->spec; - int idx = res >> 26 & 0x0f; + struct sigmatel_event *event; + int tag, data; - switch ((res >> 26) & 0x70) { + tag = (res >> 26) & 0x7f; + event = stac_get_event_from_tag(codec, tag); + if (!event) + return; + + switch (event->type) { case STAC_HP_EVENT: - stac92xx_hp_detect(codec, res); + stac92xx_hp_detect(codec); /* fallthru */ + case STAC_INSERT_EVENT: case STAC_PWR_EVENT: if (spec->num_pwrs > 0) - stac92xx_pin_sense(codec, idx); + stac92xx_pin_sense(codec, event->nid); + stac92xx_report_jack(codec, event->nid); break; - case STAC_VREF_EVENT: { - int data = snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_DATA, 0); + case STAC_VREF_EVENT: + data = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0); /* toggle VREF state based on GPIOx status */ snd_hda_codec_write(codec, codec->afg, 0, 0x7e0, - !!(data & (1 << idx))); + !!(data & (1 << event->data))); break; - } } } +#ifdef CONFIG_PROC_FS +static void stac92hd_proc_hook(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + if (nid == codec->afg) + snd_iprintf(buffer, "Power-Map: 0x%02x\n", + snd_hda_codec_read(codec, nid, 0, 0x0fec, 0x0)); +} + +static void analog_loop_proc_hook(struct snd_info_buffer *buffer, + struct hda_codec *codec, + unsigned int verb) +{ + snd_iprintf(buffer, "Analog Loopback: 0x%02x\n", + snd_hda_codec_read(codec, codec->afg, 0, verb, 0)); +} + +/* stac92hd71bxx, stac92hd73xx */ +static void stac92hd7x_proc_hook(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + stac92hd_proc_hook(buffer, codec, nid); + if (nid == codec->afg) + analog_loop_proc_hook(buffer, codec, 0xfa0); +} + +static void stac9205_proc_hook(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + if (nid == codec->afg) + analog_loop_proc_hook(buffer, codec, 0xfe0); +} + +static void stac927x_proc_hook(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + if (nid == codec->afg) + analog_loop_proc_hook(buffer, codec, 0xfeb); +} +#else +#define stac92hd_proc_hook NULL +#define stac92hd7x_proc_hook NULL +#define stac9205_proc_hook NULL +#define stac927x_proc_hook NULL +#endif + #ifdef SND_HDA_NEEDS_RESUME static int stac92xx_resume(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; stac92xx_set_config_regs(codec); - snd_hda_sequence_write(codec, spec->init); - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data); + stac92xx_init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); - /* power down inactive DACs */ - if (spec->dac_list) - stac92xx_power_down(codec); - /* invoke unsolicited event to reset the HP state */ + /* fake event to set up pins again to override cached values */ if (spec->hp_detect) - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0], + STAC_HP_EVENT); + return 0; +} + +static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) +{ + struct sigmatel_spec *spec = codec->spec; + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data & + ~spec->eapd_mask); return 0; } #endif @@ -4069,6 +4349,7 @@ static struct hda_codec_ops stac92xx_patch_ops = { .free = stac92xx_free, .unsol_event = stac92xx_unsol_event, #ifdef SND_HDA_NEEDS_RESUME + .suspend = stac92xx_suspend, .resume = stac92xx_resume, #endif }; @@ -4091,14 +4372,12 @@ static int patch_stac9200(struct hda_codec *codec) if (spec->board_config < 0) { snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); - if (err < 0) { - stac92xx_free(codec); - return err; - } - spec->pin_configs = spec->bios_pin_configs; - } else { - spec->pin_configs = stac9200_brd_tbl[spec->board_config]; - stac92xx_set_config_regs(codec); + } else + err = stac_save_pin_cfgs(codec, + stac9200_brd_tbl[spec->board_config]); + if (err < 0) { + stac92xx_free(codec); + return err; } spec->multiout.max_channels = 2; @@ -4154,14 +4433,12 @@ static int patch_stac925x(struct hda_codec *codec) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x," "using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); - if (err < 0) { - stac92xx_free(codec); - return err; - } - spec->pin_configs = spec->bios_pin_configs; - } else if (stac925x_brd_tbl[spec->board_config] != NULL){ - spec->pin_configs = stac925x_brd_tbl[spec->board_config]; - stac92xx_set_config_regs(codec); + } else + err = stac_save_pin_cfgs(codec, + stac925x_brd_tbl[spec->board_config]); + if (err < 0) { + stac92xx_free(codec); + return err; } spec->multiout.max_channels = 2; @@ -4225,6 +4502,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) struct sigmatel_spec *spec; hda_nid_t conn[STAC92HD73_DAC_COUNT + 2]; int err = 0; + int num_dacs; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4243,43 +4521,37 @@ again: snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD73XX, using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); - if (err < 0) { - stac92xx_free(codec); - return err; - } - spec->pin_configs = spec->bios_pin_configs; - } else { - spec->pin_configs = stac92hd73xx_brd_tbl[spec->board_config]; - stac92xx_set_config_regs(codec); + } else + err = stac_save_pin_cfgs(codec, + stac92hd73xx_brd_tbl[spec->board_config]); + if (err < 0) { + stac92xx_free(codec); + return err; } - spec->multiout.num_dacs = snd_hda_get_connections(codec, 0x0a, + num_dacs = snd_hda_get_connections(codec, 0x0a, conn, STAC92HD73_DAC_COUNT + 2) - 1; - if (spec->multiout.num_dacs < 0) { + if (num_dacs < 3 || num_dacs > 5) { printk(KERN_WARNING "hda_codec: Could not determine " "number of channels defaulting to DAC count\n"); - spec->multiout.num_dacs = STAC92HD73_DAC_COUNT; + num_dacs = STAC92HD73_DAC_COUNT; } - - switch (spec->multiout.num_dacs) { + switch (num_dacs) { case 0x3: /* 6 Channel */ - spec->multiout.hp_nid = 0x17; spec->mixer = stac92hd73xx_6ch_mixer; spec->init = stac92hd73xx_6ch_core_init; break; case 0x4: /* 8 Channel */ - spec->multiout.hp_nid = 0x18; spec->mixer = stac92hd73xx_8ch_mixer; spec->init = stac92hd73xx_8ch_core_init; break; case 0x5: /* 10 Channel */ - spec->multiout.hp_nid = 0x19; spec->mixer = stac92hd73xx_10ch_mixer; spec->init = stac92hd73xx_10ch_core_init; - }; + } + spec->multiout.dac_nids = spec->dac_nids; - spec->multiout.dac_nids = stac92hd73xx_dac_nids; spec->aloopback_mask = 0x01; spec->aloopback_shift = 8; @@ -4310,9 +4582,8 @@ again: spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP]; spec->eapd_switch = 0; spec->num_amps = 1; - spec->multiout.hp_nid = 0; /* dual HPs */ - if (!spec->init) + if (spec->board_config != STAC_DELL_EQ) spec->init = dell_m6_core_init; switch (spec->board_config) { case STAC_DELL_M6_AMIC: /* Analog Mics */ @@ -4370,6 +4641,8 @@ again: codec->patch_ops = stac92xx_patch_ops; + codec->proc_widget_hook = stac92hd7x_proc_hook; + return 0; } @@ -4401,17 +4674,15 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->pwr_nids = stac92hd83xxx_pwr_nids; spec->pwr_mapping = stac92hd83xxx_pwr_mapping; spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); - spec->multiout.dac_nids = stac92hd83xxx_dac_nids; + spec->multiout.dac_nids = spec->dac_nids; spec->init = stac92hd83xxx_core_init; switch (codec->vendor_id) { case 0x111d7605: - spec->multiout.num_dacs = STAC92HD81_DAC_COUNT; break; default: spec->num_pwrs--; spec->init++; /* switch to config #2 */ - spec->multiout.num_dacs = STAC92HD83_DAC_COUNT; } spec->mixer = stac92hd83xxx_mixer; @@ -4430,14 +4701,12 @@ again: snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD83XXX, using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); - if (err < 0) { - stac92xx_free(codec); - return err; - } - spec->pin_configs = spec->bios_pin_configs; - } else { - spec->pin_configs = stac92hd83xxx_brd_tbl[spec->board_config]; - stac92xx_set_config_regs(codec); + } else + err = stac_save_pin_cfgs(codec, + stac92hd83xxx_brd_tbl[spec->board_config]); + if (err < 0) { + stac92xx_free(codec); + return err; } err = stac92xx_parse_auto_config(codec, 0x1d, 0); @@ -4458,56 +4727,10 @@ again: codec->patch_ops = stac92xx_patch_ops; - return 0; -} + codec->proc_widget_hook = stac92hd_proc_hook; -#ifdef SND_HDA_NEEDS_RESUME -static void stac92hd71xx_set_power_state(struct hda_codec *codec, int pwr) -{ - struct sigmatel_spec *spec = codec->spec; - int i; - snd_hda_codec_write_cache(codec, codec->afg, 0, - AC_VERB_SET_POWER_STATE, pwr); - - msleep(1); - for (i = 0; i < spec->num_adcs; i++) { - snd_hda_codec_write_cache(codec, - spec->adc_nids[i], 0, - AC_VERB_SET_POWER_STATE, pwr); - } -}; - -static int stac92hd71xx_resume(struct hda_codec *codec) -{ - stac92hd71xx_set_power_state(codec, AC_PWRST_D0); - return stac92xx_resume(codec); -} - -static int stac92hd71xx_suspend(struct hda_codec *codec, pm_message_t state) -{ - struct sigmatel_spec *spec = codec->spec; - - stac92hd71xx_set_power_state(codec, AC_PWRST_D3); - if (spec->eapd_mask) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data & - ~spec->eapd_mask); return 0; -}; - -#endif - -static struct hda_codec_ops stac92hd71bxx_patch_ops = { - .build_controls = stac92xx_build_controls, - .build_pcms = stac92xx_build_pcms, - .init = stac92xx_init, - .free = stac92xx_free, - .unsol_event = stac92xx_unsol_event, -#ifdef SND_HDA_NEEDS_RESUME - .resume = stac92hd71xx_resume, - .suspend = stac92hd71xx_suspend, -#endif -}; +} static struct hda_input_mux stac92hd71bxx_dmux = { .num_items = 4, @@ -4544,14 +4767,12 @@ again: snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD71BXX, using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); - if (err < 0) { - stac92xx_free(codec); - return err; - } - spec->pin_configs = spec->bios_pin_configs; - } else { - spec->pin_configs = stac92hd71bxx_brd_tbl[spec->board_config]; - stac92xx_set_config_regs(codec); + } else + err = stac_save_pin_cfgs(codec, + stac92hd71bxx_brd_tbl[spec->board_config]); + if (err < 0) { + stac92xx_free(codec); + return err; } if (spec->board_config > STAC_92HD71BXX_REF) { @@ -4574,21 +4795,21 @@ again: switch (spec->board_config) { case STAC_HP_M4: /* Enable VREF power saving on GPIO1 detect */ + err = stac_add_event(spec, codec->afg, + STAC_VREF_EVENT, 0x02); + if (err < 0) + return err; snd_hda_codec_write_cache(codec, codec->afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02); snd_hda_codec_write_cache(codec, codec->afg, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | STAC_VREF_EVENT | 0x01)); + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | err); spec->gpio_mask |= 0x02; break; } if ((codec->revision_id & 0xf) == 0 || - (codec->revision_id & 0xf) == 1) { -#ifdef SND_HDA_NEEDS_RESUME - codec->patch_ops = stac92hd71bxx_patch_ops; -#endif + (codec->revision_id & 0xf) == 1) spec->stream_delay = 40; /* 40 milliseconds */ - } /* no output amps */ spec->num_pwrs = 0; @@ -4597,15 +4818,11 @@ again: /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; - stac92xx_set_config_reg(codec, 0xf, 0x40f000f0); + stac_change_pin_config(codec, 0xf, 0x40f000f0); break; case 0x111d7603: /* 6 Port with Analog Mixer */ - if ((codec->revision_id & 0xf) == 1) { -#ifdef SND_HDA_NEEDS_RESUME - codec->patch_ops = stac92hd71bxx_patch_ops; -#endif + if ((codec->revision_id & 0xf) == 1) spec->stream_delay = 40; /* 40 milliseconds */ - } /* no output amps */ spec->num_pwrs = 0; @@ -4635,7 +4852,7 @@ again: switch (spec->board_config) { case STAC_HP_M4: /* enable internal microphone */ - stac92xx_set_config_reg(codec, 0x0e, 0x01813040); + stac_change_pin_config(codec, 0x0e, 0x01813040); stac92xx_auto_set_pinctl(codec, 0x0e, AC_PINCTL_IN_EN | AC_PINCTL_VREF_80); /* fallthru */ @@ -4656,9 +4873,7 @@ again: spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); }; - spec->multiout.num_dacs = 1; - spec->multiout.hp_nid = 0x11; - spec->multiout.dac_nids = stac92hd71bxx_dac_nids; + spec->multiout.dac_nids = spec->dac_nids; if (spec->dinput_mux) spec->private_dimux.num_items += spec->num_dmics - @@ -4680,6 +4895,8 @@ again: return err; } + codec->proc_widget_hook = stac92hd7x_proc_hook; + return 0; }; @@ -4741,14 +4958,12 @@ static int patch_stac922x(struct hda_codec *codec) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, " "using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); - if (err < 0) { - stac92xx_free(codec); - return err; - } - spec->pin_configs = spec->bios_pin_configs; - } else if (stac922x_brd_tbl[spec->board_config] != NULL) { - spec->pin_configs = stac922x_brd_tbl[spec->board_config]; - stac92xx_set_config_regs(codec); + } else + err = stac_save_pin_cfgs(codec, + stac922x_brd_tbl[spec->board_config]); + if (err < 0) { + stac92xx_free(codec); + return err; } spec->adc_nids = stac922x_adc_nids; @@ -4811,14 +5026,12 @@ static int patch_stac927x(struct hda_codec *codec) snd_printdd(KERN_INFO "hda_codec: Unknown model for" "STAC927x, using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); - if (err < 0) { - stac92xx_free(codec); - return err; - } - spec->pin_configs = spec->bios_pin_configs; - } else { - spec->pin_configs = stac927x_brd_tbl[spec->board_config]; - stac92xx_set_config_regs(codec); + } else + err = stac_save_pin_cfgs(codec, + stac927x_brd_tbl[spec->board_config]); + if (err < 0) { + stac92xx_free(codec); + return err; } spec->digbeep_nid = 0x23; @@ -4848,15 +5061,15 @@ static int patch_stac927x(struct hda_codec *codec) case 0x10280209: case 0x1028022e: /* correct the device field to SPDIF out */ - stac92xx_set_config_reg(codec, 0x21, 0x01442070); + stac_change_pin_config(codec, 0x21, 0x01442070); break; }; /* configure the analog microphone on some laptops */ - stac92xx_set_config_reg(codec, 0x0c, 0x90a79130); + stac_change_pin_config(codec, 0x0c, 0x90a79130); /* correct the front output jack as a hp out */ - stac92xx_set_config_reg(codec, 0x0f, 0x0227011f); + stac_change_pin_config(codec, 0x0f, 0x0227011f); /* correct the front input jack as a mic */ - stac92xx_set_config_reg(codec, 0x0e, 0x02a79130); + stac_change_pin_config(codec, 0x0e, 0x02a79130); /* fallthru */ case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ @@ -4904,6 +5117,8 @@ static int patch_stac927x(struct hda_codec *codec) codec->patch_ops = stac92xx_patch_ops; + codec->proc_widget_hook = stac927x_proc_hook; + /* * !!FIXME!! * The STAC927x seem to require fairly long delays for certain @@ -4942,14 +5157,12 @@ static int patch_stac9205(struct hda_codec *codec) if (spec->board_config < 0) { snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9205, using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); - if (err < 0) { - stac92xx_free(codec); - return err; - } - spec->pin_configs = spec->bios_pin_configs; - } else { - spec->pin_configs = stac9205_brd_tbl[spec->board_config]; - stac92xx_set_config_regs(codec); + } else + err = stac_save_pin_cfgs(codec, + stac9205_brd_tbl[spec->board_config]); + if (err < 0) { + stac92xx_free(codec); + return err; } spec->digbeep_nid = 0x23; @@ -4976,15 +5189,18 @@ static int patch_stac9205(struct hda_codec *codec) switch (spec->board_config){ case STAC_9205_DELL_M43: /* Enable SPDIF in/out */ - stac92xx_set_config_reg(codec, 0x1f, 0x01441030); - stac92xx_set_config_reg(codec, 0x20, 0x1c410030); + stac_change_pin_config(codec, 0x1f, 0x01441030); + stac_change_pin_config(codec, 0x20, 0x1c410030); /* Enable unsol response for GPIO4/Dock HP connection */ + err = stac_add_event(spec, codec->afg, STAC_VREF_EVENT, 0x01); + if (err < 0) + return err; snd_hda_codec_write_cache(codec, codec->afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x10); snd_hda_codec_write_cache(codec, codec->afg, 0, AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | STAC_HP_EVENT)); + AC_USRSP_EN | err); spec->gpio_dir = 0x0b; spec->eapd_mask = 0x01; @@ -5022,6 +5238,8 @@ static int patch_stac9205(struct hda_codec *codec) codec->patch_ops = stac92xx_patch_ops; + codec->proc_widget_hook = stac9205_proc_hook; + return 0; } @@ -5078,29 +5296,11 @@ static struct hda_verb vaio_ar_init[] = { {} }; -/* bind volumes of both NID 0x02 and 0x05 */ -static struct hda_bind_ctls vaio_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -/* bind volumes of both NID 0x02 and 0x05 */ -static struct hda_bind_ctls vaio_bind_master_sw = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT), - 0, - }, -}; - static struct snd_kcontrol_new vaio_mixer[] = { - HDA_BIND_VOL("Master Playback Volume", &vaio_bind_master_vol), - HDA_BIND_SW("Master Playback Switch", &vaio_bind_master_sw), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT), /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), @@ -5116,8 +5316,10 @@ static struct snd_kcontrol_new vaio_mixer[] = { }; static struct snd_kcontrol_new vaio_ar_mixer[] = { - HDA_BIND_VOL("Master Playback Volume", &vaio_bind_master_vol), - HDA_BIND_SW("Master Playback Switch", &vaio_bind_master_sw), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT), /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), @@ -5158,7 +5360,7 @@ static int stac9872_vaio_init(struct hda_codec *codec) static void stac9872_vaio_hp_detect(struct hda_codec *codec, unsigned int res) { - if (get_hp_pin_presence(codec, 0x0a)) { + if (get_pin_presence(codec, 0x0a)) { stac92xx_reset_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN); stac92xx_set_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN); } else { @@ -5269,7 +5471,7 @@ static int patch_stac9872(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_sigmatel[] = { +static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847690, .name = "STAC9200", .patch = patch_stac9200 }, { .id = 0x83847882, .name = "STAC9220 A1", .patch = patch_stac922x }, { .id = 0x83847680, .name = "STAC9221 A1", .patch = patch_stac922x }, @@ -5333,3 +5535,27 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:8384*"); +MODULE_ALIAS("snd-hda-codec-id:111d*"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("IDT/Sigmatel HD-audio codec"); + +static struct hda_codec_preset_list sigmatel_list = { + .preset = snd_hda_preset_sigmatel, + .owner = THIS_MODULE, +}; + +static int __init patch_sigmatel_init(void) +{ + return snd_hda_add_codec_preset(&sigmatel_list); +} + +static void __exit patch_sigmatel_exit(void) +{ + snd_hda_delete_codec_preset(&sigmatel_list); +} + +module_init(patch_sigmatel_init) +module_exit(patch_sigmatel_exit) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 63e4871e5d8f..c761394cbe84 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -47,15 +47,11 @@ #include <sound/asoundef.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" /* amp values */ #define AMP_VAL_IDX_SHIFT 19 #define AMP_VAL_IDX_MASK (0x0f<<19) -#define NUM_CONTROL_ALLOC 32 -#define NUM_VERB_ALLOC 32 - /* Pin Widget NID */ #define VT1708_HP_NID 0x13 #define VT1708_DIGOUT_NID 0x14 @@ -145,8 +141,6 @@ enum { AUTO_SEQ_SIDE }; -#define get_amp_nid(kc) ((kc)->private_value & 0xffff) - /* Some VT1708S based boards gets the micboost setting wrong, so we have * to apply some brute-force and re-write the TLV's by software. */ static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag, @@ -227,8 +221,7 @@ struct via_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; - unsigned int num_kctl_alloc, num_kctl_used; - struct snd_kcontrol_new *kctl_alloc; + struct snd_array kctls; struct hda_input_mux private_imux[2]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; @@ -272,33 +265,31 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, { struct snd_kcontrol_new *knew; - if (spec->num_kctl_used >= spec->num_kctl_alloc) { - int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC; - - /* array + terminator */ - knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); - if (!knew) - return -ENOMEM; - if (spec->kctl_alloc) { - memcpy(knew, spec->kctl_alloc, - sizeof(*knew) * spec->num_kctl_alloc); - kfree(spec->kctl_alloc); - } - spec->kctl_alloc = knew; - spec->num_kctl_alloc = num; - } - - knew = &spec->kctl_alloc[spec->num_kctl_used]; + snd_array_init(&spec->kctls, sizeof(*knew), 32); + knew = snd_array_new(&spec->kctls); + if (!knew) + return -ENOMEM; *knew = vt1708_control_templates[type]; knew->name = kstrdup(name, GFP_KERNEL); - if (!knew->name) return -ENOMEM; knew->private_value = val; - spec->num_kctl_used++; return 0; } +static void via_free_kctls(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + + if (spec->kctls.list) { + struct snd_kcontrol_new *kctl = spec->kctls.list; + int i; + for (i = 0; i < spec->kctls.used; i++) + kfree(kctl[i].name); + } + snd_array_free(&spec->kctls); +} + /* create input playback/capture controls for the given pin */ static int via_new_analog_input(struct via_spec *spec, hda_nid_t pin, const char *ctlname, int idx, int mix_nid) @@ -896,6 +887,7 @@ static int via_build_controls(struct hda_codec *codec) if (err < 0) return err; } + via_free_kctls(codec); /* no longer needed */ return 0; } @@ -941,17 +933,11 @@ static int via_build_pcms(struct hda_codec *codec) static void via_free(struct hda_codec *codec) { struct via_spec *spec = codec->spec; - unsigned int i; if (!spec) return; - if (spec->kctl_alloc) { - for (i = 0; i < spec->num_kctl_used; i++) - kfree(spec->kctl_alloc[i].name); - kfree(spec->kctl_alloc); - } - + via_free_kctls(codec); kfree(codec->spec); } @@ -1373,8 +1359,8 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1708_DIGIN_NID; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; spec->init_verbs[spec->num_iverbs++] = vt1708_volume_init_verbs; @@ -1846,8 +1832,8 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1709_DIGIN_NID; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; spec->input_mux = &spec->private_imux[0]; @@ -2390,8 +2376,8 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1708B_DIGIN_NID; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; spec->input_mux = &spec->private_imux[0]; @@ -2855,8 +2841,8 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) spec->extra_dig_out_nid = 0x15; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; spec->input_mux = &spec->private_imux[0]; @@ -3174,8 +3160,8 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) spec->extra_dig_out_nid = 0x1B; - if (spec->kctl_alloc) - spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; spec->input_mux = &spec->private_imux[0]; @@ -3262,74 +3248,97 @@ static int patch_vt1702(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_via[] = { - { .id = 0x11061708, .name = "VIA VT1708", .patch = patch_vt1708}, - { .id = 0x11061709, .name = "VIA VT1708", .patch = patch_vt1708}, - { .id = 0x1106170A, .name = "VIA VT1708", .patch = patch_vt1708}, - { .id = 0x1106170B, .name = "VIA VT1708", .patch = patch_vt1708}, - { .id = 0x1106E710, .name = "VIA VT1709 10-Ch", +static struct hda_codec_preset snd_hda_preset_via[] = { + { .id = 0x11061708, .name = "VT1708", .patch = patch_vt1708}, + { .id = 0x11061709, .name = "VT1708", .patch = patch_vt1708}, + { .id = 0x1106170a, .name = "VT1708", .patch = patch_vt1708}, + { .id = 0x1106170b, .name = "VT1708", .patch = patch_vt1708}, + { .id = 0x1106e710, .name = "VT1709 10-Ch", .patch = patch_vt1709_10ch}, - { .id = 0x1106E711, .name = "VIA VT1709 10-Ch", + { .id = 0x1106e711, .name = "VT1709 10-Ch", .patch = patch_vt1709_10ch}, - { .id = 0x1106E712, .name = "VIA VT1709 10-Ch", + { .id = 0x1106e712, .name = "VT1709 10-Ch", .patch = patch_vt1709_10ch}, - { .id = 0x1106E713, .name = "VIA VT1709 10-Ch", + { .id = 0x1106e713, .name = "VT1709 10-Ch", .patch = patch_vt1709_10ch}, - { .id = 0x1106E714, .name = "VIA VT1709 6-Ch", + { .id = 0x1106e714, .name = "VT1709 6-Ch", .patch = patch_vt1709_6ch}, - { .id = 0x1106E715, .name = "VIA VT1709 6-Ch", + { .id = 0x1106e715, .name = "VT1709 6-Ch", .patch = patch_vt1709_6ch}, - { .id = 0x1106E716, .name = "VIA VT1709 6-Ch", + { .id = 0x1106e716, .name = "VT1709 6-Ch", .patch = patch_vt1709_6ch}, - { .id = 0x1106E717, .name = "VIA VT1709 6-Ch", + { .id = 0x1106e717, .name = "VT1709 6-Ch", .patch = patch_vt1709_6ch}, - { .id = 0x1106E720, .name = "VIA VT1708B 8-Ch", + { .id = 0x1106e720, .name = "VT1708B 8-Ch", .patch = patch_vt1708B_8ch}, - { .id = 0x1106E721, .name = "VIA VT1708B 8-Ch", + { .id = 0x1106e721, .name = "VT1708B 8-Ch", .patch = patch_vt1708B_8ch}, - { .id = 0x1106E722, .name = "VIA VT1708B 8-Ch", + { .id = 0x1106e722, .name = "VT1708B 8-Ch", .patch = patch_vt1708B_8ch}, - { .id = 0x1106E723, .name = "VIA VT1708B 8-Ch", + { .id = 0x1106e723, .name = "VT1708B 8-Ch", .patch = patch_vt1708B_8ch}, - { .id = 0x1106E724, .name = "VIA VT1708B 4-Ch", + { .id = 0x1106e724, .name = "VT1708B 4-Ch", .patch = patch_vt1708B_4ch}, - { .id = 0x1106E725, .name = "VIA VT1708B 4-Ch", + { .id = 0x1106e725, .name = "VT1708B 4-Ch", .patch = patch_vt1708B_4ch}, - { .id = 0x1106E726, .name = "VIA VT1708B 4-Ch", + { .id = 0x1106e726, .name = "VT1708B 4-Ch", .patch = patch_vt1708B_4ch}, - { .id = 0x1106E727, .name = "VIA VT1708B 4-Ch", + { .id = 0x1106e727, .name = "VT1708B 4-Ch", .patch = patch_vt1708B_4ch}, - { .id = 0x11060397, .name = "VIA VT1708S", + { .id = 0x11060397, .name = "VT1708S", .patch = patch_vt1708S}, - { .id = 0x11061397, .name = "VIA VT1708S", + { .id = 0x11061397, .name = "VT1708S", .patch = patch_vt1708S}, - { .id = 0x11062397, .name = "VIA VT1708S", + { .id = 0x11062397, .name = "VT1708S", .patch = patch_vt1708S}, - { .id = 0x11063397, .name = "VIA VT1708S", + { .id = 0x11063397, .name = "VT1708S", .patch = patch_vt1708S}, - { .id = 0x11064397, .name = "VIA VT1708S", + { .id = 0x11064397, .name = "VT1708S", .patch = patch_vt1708S}, - { .id = 0x11065397, .name = "VIA VT1708S", + { .id = 0x11065397, .name = "VT1708S", .patch = patch_vt1708S}, - { .id = 0x11066397, .name = "VIA VT1708S", + { .id = 0x11066397, .name = "VT1708S", .patch = patch_vt1708S}, - { .id = 0x11067397, .name = "VIA VT1708S", + { .id = 0x11067397, .name = "VT1708S", .patch = patch_vt1708S}, - { .id = 0x11060398, .name = "VIA VT1702", + { .id = 0x11060398, .name = "VT1702", .patch = patch_vt1702}, - { .id = 0x11061398, .name = "VIA VT1702", + { .id = 0x11061398, .name = "VT1702", .patch = patch_vt1702}, - { .id = 0x11062398, .name = "VIA VT1702", + { .id = 0x11062398, .name = "VT1702", .patch = patch_vt1702}, - { .id = 0x11063398, .name = "VIA VT1702", + { .id = 0x11063398, .name = "VT1702", .patch = patch_vt1702}, - { .id = 0x11064398, .name = "VIA VT1702", + { .id = 0x11064398, .name = "VT1702", .patch = patch_vt1702}, - { .id = 0x11065398, .name = "VIA VT1702", + { .id = 0x11065398, .name = "VT1702", .patch = patch_vt1702}, - { .id = 0x11066398, .name = "VIA VT1702", + { .id = 0x11066398, .name = "VT1702", .patch = patch_vt1702}, - { .id = 0x11067398, .name = "VIA VT1702", + { .id = 0x11067398, .name = "VT1702", .patch = patch_vt1702}, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:1106*"); + +static struct hda_codec_preset_list via_list = { + .preset = snd_hda_preset_via, + .owner = THIS_MODULE, +}; + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("VIA HD-audio codec"); + +static int __init patch_via_init(void) +{ + return snd_hda_add_codec_preset(&via_list); +} + +static void __exit patch_via_exit(void) +{ + snd_hda_delete_codec_preset(&via_list); +} + +module_init(patch_via_init) +module_exit(patch_via_exit) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 1b3f11702713..0dfa0540ce2c 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -382,23 +382,25 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) unsigned char status_mask = VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX | VT1724_IRQ_MTPCM; int handled = 0; -#ifdef CONFIG_SND_DEBUG int timeout = 0; -#endif while (1) { status = inb(ICEREG1724(ice, IRQSTAT)); status &= status_mask; if (status == 0) break; -#ifdef CONFIG_SND_DEBUG if (++timeout > 10) { - printk(KERN_ERR - "ice1724: Too long irq loop, status = 0x%x\n", - status); + status = inb(ICEREG1724(ice, IRQSTAT)); + printk(KERN_ERR "ice1724: Too long irq loop, " + "status = 0x%x\n", status); + if (status & VT1724_IRQ_MPU_TX) { + printk(KERN_ERR "ice1724: Disabling MPU_TX\n"); + outb(inb(ICEREG1724(ice, IRQMASK)) | + VT1724_IRQ_MPU_TX, + ICEREG1724(ice, IRQMASK)); + } break; } -#endif handled = 1; if (status & VT1724_IRQ_MPU_TX) { spin_lock(&ice->reg_lock); @@ -2351,7 +2353,6 @@ static int __devinit snd_vt1724_create(struct snd_card *card, { struct snd_ice1712 *ice; int err; - unsigned char mask; static struct snd_device_ops ops = { .dev_free = snd_vt1724_dev_free, }; @@ -2412,9 +2413,9 @@ static int __devinit snd_vt1724_create(struct snd_card *card, return -EIO; } - /* unmask used interrupts */ - mask = VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX; - outb(mask, ICEREG1724(ice, IRQMASK)); + /* MPU_RX and TX irq masks are cleared later dynamically */ + outb(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX , ICEREG1724(ice, IRQMASK)); + /* don't handle FIFO overrun/underruns (just yet), * since they cause machine lockups */ diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 9ff3f9e34404..59bbaf8f3e5b 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -1670,7 +1670,7 @@ static irqreturn_t snd_m3_interrupt(int irq, void *dev_id) return IRQ_NONE; if (status & HV_INT_PENDING) - tasklet_hi_schedule(&chip->hwvol_tq); + tasklet_schedule(&chip->hwvol_tq); /* * ack an assp int if its running diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index ae7601f353a7..f23a73577c22 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1010,7 +1010,7 @@ static int __devinit snd_mixart_create(struct mixart_mgr *mgr, struct snd_card * .dev_free = snd_mixart_chip_dev_free, }; - mgr->chip[idx] = chip = kzalloc(sizeof(*chip), GFP_KERNEL); + chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (! chip) { snd_printk(KERN_ERR "cannot allocate chip\n"); return -ENOMEM; @@ -1025,6 +1025,7 @@ static int __devinit snd_mixart_create(struct mixart_mgr *mgr, struct snd_card * return err; } + mgr->chip[idx] = chip; snd_card_set_dev(card, &mgr->pci->dev); return 0; @@ -1377,6 +1378,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, sprintf(card->longname, "%s [PCM #%d]", mgr->longname, i); if ((err = snd_mixart_create(mgr, card, i)) < 0) { + snd_card_free(card); snd_mixart_free(mgr); return err; } diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index b9a06c279397..d3350f383966 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -550,7 +550,7 @@ irqreturn_t snd_mixart_interrupt(int irq, void *dev_id) mgr->msg_fifo[mgr->msg_fifo_writeptr] = msg; mgr->msg_fifo_writeptr++; mgr->msg_fifo_writeptr %= MSG_FIFO_SIZE; - tasklet_hi_schedule(&mgr->msg_taskq); + tasklet_schedule(&mgr->msg_taskq); } spin_unlock(&mgr->msg_lock); break; diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index b60f6212745a..de999c6d6dd3 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -61,6 +61,7 @@ MODULE_PARM_DESC(enable, "enable card"); enum { MODEL_CMEDIA_REF, /* C-Media's reference design */ MODEL_MERIDIAN, /* AuzenTech X-Meridian */ + MODEL_HALO, /* HT-Omega Claro halo */ }; static struct pci_device_id oxygen_ids[] __devinitdata = { @@ -74,6 +75,7 @@ static struct pci_device_id oxygen_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN }, { OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CMEDIA_REF }, + { OXYGEN_PCI_SUBID(0x7284, 0x9781), .driver_data = MODEL_HALO }, { } }; MODULE_DEVICE_TABLE(pci, oxygen_ids); @@ -301,6 +303,8 @@ static int generic_probe(struct oxygen *chip, unsigned long driver_data) PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF; + } + if (driver_data == MODEL_MERIDIAN || driver_data == MODEL_HALO) { chip->model.misc_flags = OXYGEN_MISC_MIDI; chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT; } diff --git a/sound/pci/pcxhr/Makefile b/sound/pci/pcxhr/Makefile index 10473c05918d..b06128e918ca 100644 --- a/sound/pci/pcxhr/Makefile +++ b/sound/pci/pcxhr/Makefile @@ -1,2 +1,2 @@ -snd-pcxhr-objs := pcxhr.o pcxhr_hwdep.o pcxhr_mixer.o pcxhr_core.o +snd-pcxhr-objs := pcxhr.o pcxhr_hwdep.o pcxhr_mixer.o pcxhr_core.o pcxhr_mix22.o obj-$(CONFIG_SND_PCXHR) += snd-pcxhr.o diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 73de6e989b3d..27cf2c28d113 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -40,18 +40,20 @@ #include "pcxhr_mixer.h" #include "pcxhr_hwdep.h" #include "pcxhr_core.h" +#include "pcxhr_mix22.h" #define DRIVER_NAME "pcxhr" -MODULE_AUTHOR("Markus Bollinger <bollinger@digigram.com>"); +MODULE_AUTHOR("Markus Bollinger <bollinger@digigram.com>, " + "Marc Titinger <titinger@digigram.com>"); MODULE_DESCRIPTION("Digigram " DRIVER_NAME " " PCXHR_DRIVER_VERSION_STRING); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Digigram," DRIVER_NAME "}}"); -static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ -static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int mono[SNDRV_CARDS]; /* capture in mono only */ +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ +static int mono[SNDRV_CARDS]; /* capture mono only */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Digigram " DRIVER_NAME " soundcard"); @@ -67,18 +69,58 @@ enum { PCI_ID_PCX882HR, PCI_ID_VX881HR, PCI_ID_PCX881HR, + PCI_ID_VX882E, + PCI_ID_PCX882E, + PCI_ID_VX881E, + PCI_ID_PCX881E, + PCI_ID_VX1222HR, PCI_ID_PCX1222HR, + PCI_ID_VX1221HR, PCI_ID_PCX1221HR, + PCI_ID_VX1222E, + PCI_ID_PCX1222E, + PCI_ID_VX1221E, + PCI_ID_PCX1221E, + PCI_ID_VX222HR, + PCI_ID_VX222E, + PCI_ID_PCX22HR, + PCI_ID_PCX22E, + PCI_ID_VX222HRMIC, + PCI_ID_VX222E_MIC, + PCI_ID_PCX924HR, + PCI_ID_PCX924E, + PCI_ID_PCX924HRMIC, + PCI_ID_PCX924E_MIC, PCI_ID_LAST }; static struct pci_device_id pcxhr_ids[] = { - { 0x10b5, 0x9656, 0x1369, 0xb001, 0, 0, PCI_ID_VX882HR, }, /* VX882HR */ - { 0x10b5, 0x9656, 0x1369, 0xb101, 0, 0, PCI_ID_PCX882HR, }, /* PCX882HR */ - { 0x10b5, 0x9656, 0x1369, 0xb201, 0, 0, PCI_ID_VX881HR, }, /* VX881HR */ - { 0x10b5, 0x9656, 0x1369, 0xb301, 0, 0, PCI_ID_PCX881HR, }, /* PCX881HR */ - { 0x10b5, 0x9656, 0x1369, 0xb501, 0, 0, PCI_ID_PCX1222HR, }, /* PCX1222HR */ - { 0x10b5, 0x9656, 0x1369, 0xb701, 0, 0, PCI_ID_PCX1221HR, }, /* PCX1221HR */ + { 0x10b5, 0x9656, 0x1369, 0xb001, 0, 0, PCI_ID_VX882HR, }, + { 0x10b5, 0x9656, 0x1369, 0xb101, 0, 0, PCI_ID_PCX882HR, }, + { 0x10b5, 0x9656, 0x1369, 0xb201, 0, 0, PCI_ID_VX881HR, }, + { 0x10b5, 0x9656, 0x1369, 0xb301, 0, 0, PCI_ID_PCX881HR, }, + { 0x10b5, 0x9056, 0x1369, 0xb021, 0, 0, PCI_ID_VX882E, }, + { 0x10b5, 0x9056, 0x1369, 0xb121, 0, 0, PCI_ID_PCX882E, }, + { 0x10b5, 0x9056, 0x1369, 0xb221, 0, 0, PCI_ID_VX881E, }, + { 0x10b5, 0x9056, 0x1369, 0xb321, 0, 0, PCI_ID_PCX881E, }, + { 0x10b5, 0x9656, 0x1369, 0xb401, 0, 0, PCI_ID_VX1222HR, }, + { 0x10b5, 0x9656, 0x1369, 0xb501, 0, 0, PCI_ID_PCX1222HR, }, + { 0x10b5, 0x9656, 0x1369, 0xb601, 0, 0, PCI_ID_VX1221HR, }, + { 0x10b5, 0x9656, 0x1369, 0xb701, 0, 0, PCI_ID_PCX1221HR, }, + { 0x10b5, 0x9056, 0x1369, 0xb421, 0, 0, PCI_ID_VX1222E, }, + { 0x10b5, 0x9056, 0x1369, 0xb521, 0, 0, PCI_ID_PCX1222E, }, + { 0x10b5, 0x9056, 0x1369, 0xb621, 0, 0, PCI_ID_VX1221E, }, + { 0x10b5, 0x9056, 0x1369, 0xb721, 0, 0, PCI_ID_PCX1221E, }, + { 0x10b5, 0x9056, 0x1369, 0xba01, 0, 0, PCI_ID_VX222HR, }, + { 0x10b5, 0x9056, 0x1369, 0xba21, 0, 0, PCI_ID_VX222E, }, + { 0x10b5, 0x9056, 0x1369, 0xbd01, 0, 0, PCI_ID_PCX22HR, }, + { 0x10b5, 0x9056, 0x1369, 0xbd21, 0, 0, PCI_ID_PCX22E, }, + { 0x10b5, 0x9056, 0x1369, 0xbc01, 0, 0, PCI_ID_VX222HRMIC, }, + { 0x10b5, 0x9056, 0x1369, 0xbc21, 0, 0, PCI_ID_VX222E_MIC, }, + { 0x10b5, 0x9056, 0x1369, 0xbb01, 0, 0, PCI_ID_PCX924HR, }, + { 0x10b5, 0x9056, 0x1369, 0xbb21, 0, 0, PCI_ID_PCX924E, }, + { 0x10b5, 0x9056, 0x1369, 0xbf01, 0, 0, PCI_ID_PCX924HRMIC, }, + { 0x10b5, 0x9056, 0x1369, 0xbf21, 0, 0, PCI_ID_PCX924E_MIC, }, { 0, } }; @@ -88,27 +130,55 @@ struct board_parameters { char* board_name; short playback_chips; short capture_chips; + short fw_file_set; short firmware_num; }; static struct board_parameters pcxhr_board_params[] = { -[PCI_ID_VX882HR] = { "VX882HR", 4, 4, 41, }, -[PCI_ID_PCX882HR] = { "PCX882HR", 4, 4, 41, }, -[PCI_ID_VX881HR] = { "VX881HR", 4, 4, 41, }, -[PCI_ID_PCX881HR] = { "PCX881HR", 4, 4, 41, }, -[PCI_ID_PCX1222HR] = { "PCX1222HR", 6, 1, 42, }, -[PCI_ID_PCX1221HR] = { "PCX1221HR", 6, 1, 42, }, +[PCI_ID_VX882HR] = { "VX882HR", 4, 4, 0, 41 }, +[PCI_ID_PCX882HR] = { "PCX882HR", 4, 4, 0, 41 }, +[PCI_ID_VX881HR] = { "VX881HR", 4, 4, 0, 41 }, +[PCI_ID_PCX881HR] = { "PCX881HR", 4, 4, 0, 41 }, +[PCI_ID_VX882E] = { "VX882e", 4, 4, 1, 41 }, +[PCI_ID_PCX882E] = { "PCX882e", 4, 4, 1, 41 }, +[PCI_ID_VX881E] = { "VX881e", 4, 4, 1, 41 }, +[PCI_ID_PCX881E] = { "PCX881e", 4, 4, 1, 41 }, +[PCI_ID_VX1222HR] = { "VX1222HR", 6, 1, 2, 42 }, +[PCI_ID_PCX1222HR] = { "PCX1222HR", 6, 1, 2, 42 }, +[PCI_ID_VX1221HR] = { "VX1221HR", 6, 1, 2, 42 }, +[PCI_ID_PCX1221HR] = { "PCX1221HR", 6, 1, 2, 42 }, +[PCI_ID_VX1222E] = { "VX1222e", 6, 1, 3, 42 }, +[PCI_ID_PCX1222E] = { "PCX1222e", 6, 1, 3, 42 }, +[PCI_ID_VX1221E] = { "VX1221e", 6, 1, 3, 42 }, +[PCI_ID_PCX1221E] = { "PCX1221e", 6, 1, 3, 42 }, +[PCI_ID_VX222HR] = { "VX222HR", 1, 1, 4, 44 }, +[PCI_ID_VX222E] = { "VX222e", 1, 1, 4, 44 }, +[PCI_ID_PCX22HR] = { "PCX22HR", 1, 0, 4, 44 }, +[PCI_ID_PCX22E] = { "PCX22e", 1, 0, 4, 44 }, +[PCI_ID_VX222HRMIC] = { "VX222HR-Mic", 1, 1, 5, 44 }, +[PCI_ID_VX222E_MIC] = { "VX222e-Mic", 1, 1, 5, 44 }, +[PCI_ID_PCX924HR] = { "PCX924HR", 1, 1, 5, 44 }, +[PCI_ID_PCX924E] = { "PCX924e", 1, 1, 5, 44 }, +[PCI_ID_PCX924HRMIC] = { "PCX924HR-Mic", 1, 1, 5, 44 }, +[PCI_ID_PCX924E_MIC] = { "PCX924e-Mic", 1, 1, 5, 44 }, }; +/* boards without hw AES1 and SRC onboard are all using fw_file_set==4 */ +/* VX222HR, VX222e, PCX22HR and PCX22e */ +#define PCXHR_BOARD_HAS_AES1(x) (x->fw_file_set != 4) +/* some boards do not support 192kHz on digital AES input plugs */ +#define PCXHR_BOARD_AESIN_NO_192K(x) ((x->capture_chips == 0) || \ + (x->fw_file_set == 0) || \ + (x->fw_file_set == 2)) static int pcxhr_pll_freq_register(unsigned int freq, unsigned int* pllreg, unsigned int* realfreq) { unsigned int reg; - if (freq < 6900 || freq > 110250) + if (freq < 6900 || freq > 110000) return -EINVAL; - reg = (28224000 * 10) / freq; - reg = (reg + 5) / 10; + reg = (28224000 * 2) / freq; + reg = (reg - 1) / 2; if (reg < 0x200) *pllreg = reg + 0x800; else if (reg < 0x400) @@ -121,7 +191,7 @@ static int pcxhr_pll_freq_register(unsigned int freq, unsigned int* pllreg, reg &= ~3; } if (realfreq) - *realfreq = ((28224000 * 10) / reg + 5) / 10; + *realfreq = (28224000 / (reg + 1)); return 0; } @@ -151,11 +221,6 @@ static int pcxhr_pll_freq_register(unsigned int freq, unsigned int* pllreg, #define PCXHR_FREQ_AES_3 0x03 #define PCXHR_FREQ_AES_4 0x0d -#define PCXHR_MODIFY_CLOCK_S_BIT 0x04 - -#define PCXHR_IRQ_TIMER_FREQ 92000 -#define PCXHR_IRQ_TIMER_PERIOD 48 - static int pcxhr_get_clock_reg(struct pcxhr_mgr *mgr, unsigned int rate, unsigned int *reg, unsigned int *freq) { @@ -196,19 +261,32 @@ static int pcxhr_get_clock_reg(struct pcxhr_mgr *mgr, unsigned int rate, err = pcxhr_send_msg(mgr, &rmh); if (err < 0) { snd_printk(KERN_ERR - "error CMD_ACCESS_IO_WRITE for PLL register : %x!\n", - err ); + "error CMD_ACCESS_IO_WRITE " + "for PLL register : %x!\n", err); return err; } } break; - case PCXHR_CLOCK_TYPE_WORD_CLOCK : val = PCXHR_FREQ_WORD_CLOCK; break; - case PCXHR_CLOCK_TYPE_AES_SYNC : val = PCXHR_FREQ_SYNC_AES; break; - case PCXHR_CLOCK_TYPE_AES_1 : val = PCXHR_FREQ_AES_1; break; - case PCXHR_CLOCK_TYPE_AES_2 : val = PCXHR_FREQ_AES_2; break; - case PCXHR_CLOCK_TYPE_AES_3 : val = PCXHR_FREQ_AES_3; break; - case PCXHR_CLOCK_TYPE_AES_4 : val = PCXHR_FREQ_AES_4; break; - default : return -EINVAL; + case PCXHR_CLOCK_TYPE_WORD_CLOCK: + val = PCXHR_FREQ_WORD_CLOCK; + break; + case PCXHR_CLOCK_TYPE_AES_SYNC: + val = PCXHR_FREQ_SYNC_AES; + break; + case PCXHR_CLOCK_TYPE_AES_1: + val = PCXHR_FREQ_AES_1; + break; + case PCXHR_CLOCK_TYPE_AES_2: + val = PCXHR_FREQ_AES_2; + break; + case PCXHR_CLOCK_TYPE_AES_3: + val = PCXHR_FREQ_AES_3; + break; + case PCXHR_CLOCK_TYPE_AES_4: + val = PCXHR_FREQ_AES_4; + break; + default: + return -EINVAL; } *reg = val; *freq = realfreq; @@ -216,14 +294,13 @@ static int pcxhr_get_clock_reg(struct pcxhr_mgr *mgr, unsigned int rate, } -int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate) +static int pcxhr_sub_set_clock(struct pcxhr_mgr *mgr, + unsigned int rate, + int *changed) { unsigned int val, realfreq, speed; struct pcxhr_rmh rmh; - int err, changed; - - if (rate == 0) - return 0; /* nothing to do */ + int err; err = pcxhr_get_clock_reg(mgr, rate, &val, &realfreq); if (err) @@ -237,13 +314,17 @@ int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate) else speed = 2; /* quad speed */ if (mgr->codec_speed != speed) { - pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* mute outputs */ + pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* mute outputs */ rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT; + if (DSP_EXT_CMD_SET(mgr)) { + rmh.cmd[1] = 1; + rmh.cmd_len = 2; + } err = pcxhr_send_msg(mgr, &rmh); if (err) return err; - pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* set speed ratio */ + pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* set speed ratio */ rmh.cmd[0] |= IO_NUM_SPEED_RATIO; rmh.cmd[1] = speed; rmh.cmd_len = 2; @@ -253,25 +334,57 @@ int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate) } /* set the new frequency */ snd_printdd("clock register : set %x\n", val); - err = pcxhr_write_io_num_reg_cont(mgr, PCXHR_FREQ_REG_MASK, val, &changed); + err = pcxhr_write_io_num_reg_cont(mgr, PCXHR_FREQ_REG_MASK, + val, changed); if (err) return err; + mgr->sample_rate_real = realfreq; mgr->cur_clock_type = mgr->use_clock_type; /* unmute after codec speed modes */ if (mgr->codec_speed != speed) { - pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); /* unmute outputs */ + pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); /* unmute outputs */ rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT; + if (DSP_EXT_CMD_SET(mgr)) { + rmh.cmd[1] = 1; + rmh.cmd_len = 2; + } err = pcxhr_send_msg(mgr, &rmh); if (err) return err; - mgr->codec_speed = speed; /* save new codec speed */ + mgr->codec_speed = speed; /* save new codec speed */ } + snd_printdd("pcxhr_sub_set_clock to %dHz (realfreq=%d)\n", + rate, realfreq); + return 0; +} + +#define PCXHR_MODIFY_CLOCK_S_BIT 0x04 + +#define PCXHR_IRQ_TIMER_FREQ 92000 +#define PCXHR_IRQ_TIMER_PERIOD 48 + +int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate) +{ + struct pcxhr_rmh rmh; + int err, changed; + + if (rate == 0) + return 0; /* nothing to do */ + + if (mgr->is_hr_stereo) + err = hr222_sub_set_clock(mgr, rate, &changed); + else + err = pcxhr_sub_set_clock(mgr, rate, &changed); + + if (err) + return err; + if (changed) { pcxhr_init_rmh(&rmh, CMD_MODIFY_CLOCK); - rmh.cmd[0] |= PCXHR_MODIFY_CLOCK_S_BIT; /* resync fifos */ + rmh.cmd[0] |= PCXHR_MODIFY_CLOCK_S_BIT; /* resync fifos */ if (rate < PCXHR_IRQ_TIMER_FREQ) rmh.cmd[1] = PCXHR_IRQ_TIMER_PERIOD; else @@ -282,26 +395,39 @@ int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate) if (err) return err; } - snd_printdd("pcxhr_set_clock to %dHz (realfreq=%d)\n", rate, realfreq); return 0; } -int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_type, - int *sample_rate) +static int pcxhr_sub_get_external_clock(struct pcxhr_mgr *mgr, + enum pcxhr_clock_type clock_type, + int *sample_rate) { struct pcxhr_rmh rmh; unsigned char reg; int err, rate; switch (clock_type) { - case PCXHR_CLOCK_TYPE_WORD_CLOCK : reg = REG_STATUS_WORD_CLOCK; break; - case PCXHR_CLOCK_TYPE_AES_SYNC : reg = REG_STATUS_AES_SYNC; break; - case PCXHR_CLOCK_TYPE_AES_1 : reg = REG_STATUS_AES_1; break; - case PCXHR_CLOCK_TYPE_AES_2 : reg = REG_STATUS_AES_2; break; - case PCXHR_CLOCK_TYPE_AES_3 : reg = REG_STATUS_AES_3; break; - case PCXHR_CLOCK_TYPE_AES_4 : reg = REG_STATUS_AES_4; break; - default : return -EINVAL; + case PCXHR_CLOCK_TYPE_WORD_CLOCK: + reg = REG_STATUS_WORD_CLOCK; + break; + case PCXHR_CLOCK_TYPE_AES_SYNC: + reg = REG_STATUS_AES_SYNC; + break; + case PCXHR_CLOCK_TYPE_AES_1: + reg = REG_STATUS_AES_1; + break; + case PCXHR_CLOCK_TYPE_AES_2: + reg = REG_STATUS_AES_2; + break; + case PCXHR_CLOCK_TYPE_AES_3: + reg = REG_STATUS_AES_3; + break; + case PCXHR_CLOCK_TYPE_AES_4: + reg = REG_STATUS_AES_4; + break; + default: + return -EINVAL; } pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); rmh.cmd_len = 2; @@ -311,7 +437,7 @@ int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_ err = pcxhr_send_msg(mgr, &rmh); if (err) return err; - udelay(100); /* wait minimum 2 sample_frames at 32kHz ! */ + udelay(100); /* wait minimum 2 sample_frames at 32kHz ! */ mgr->last_reg_stat = reg; } rmh.cmd[1] = REG_STATUS_CURRENT; @@ -336,6 +462,18 @@ int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_ } +int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, + enum pcxhr_clock_type clock_type, + int *sample_rate) +{ + if (mgr->is_hr_stereo) + return hr222_get_external_clock(mgr, clock_type, + sample_rate); + else + return pcxhr_sub_get_external_clock(mgr, clock_type, + sample_rate); +} + /* * start or stop playback/capture substream */ @@ -350,7 +488,8 @@ static int pcxhr_set_stream_state(struct pcxhr_stream *stream) start = 1; else { if (stream->status != PCXHR_STREAM_STATUS_SCHEDULE_STOP) { - snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state CANNOT be stopped\n"); + snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state " + "CANNOT be stopped\n"); return -EINVAL; } start = 0; @@ -359,11 +498,12 @@ static int pcxhr_set_stream_state(struct pcxhr_stream *stream) return -EINVAL; stream->timer_abs_periods = 0; - stream->timer_period_frag = 0; /* reset theoretical stream pos */ + stream->timer_period_frag = 0; /* reset theoretical stream pos */ stream->timer_buf_periods = 0; stream->timer_is_synced = 0; - stream_mask = stream->pipe->is_capture ? 1 : 1<<stream->substream->number; + stream_mask = + stream->pipe->is_capture ? 1 : 1<<stream->substream->number; pcxhr_init_rmh(&rmh, start ? CMD_START_STREAM : CMD_STOP_STREAM); pcxhr_set_pipe_cmd_params(&rmh, stream->pipe->is_capture, @@ -373,8 +513,10 @@ static int pcxhr_set_stream_state(struct pcxhr_stream *stream) err = pcxhr_send_msg(chip->mgr, &rmh); if (err) - snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state err=%x;\n", err); - stream->status = start ? PCXHR_STREAM_STATUS_STARTED : PCXHR_STREAM_STATUS_STOPPED; + snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state err=%x;\n", + err); + stream->status = + start ? PCXHR_STREAM_STATUS_STARTED : PCXHR_STREAM_STATUS_STOPPED; return err; } @@ -399,13 +541,15 @@ static int pcxhr_set_format(struct pcxhr_stream *stream) header = HEADER_FMT_BASE_LIN; break; case SNDRV_PCM_FORMAT_S16_LE: - header = HEADER_FMT_BASE_LIN | HEADER_FMT_16BITS | HEADER_FMT_INTEL; + header = HEADER_FMT_BASE_LIN | + HEADER_FMT_16BITS | HEADER_FMT_INTEL; break; case SNDRV_PCM_FORMAT_S16_BE: header = HEADER_FMT_BASE_LIN | HEADER_FMT_16BITS; break; case SNDRV_PCM_FORMAT_S24_3LE: - header = HEADER_FMT_BASE_LIN | HEADER_FMT_24BITS | HEADER_FMT_INTEL; + header = HEADER_FMT_BASE_LIN | + HEADER_FMT_24BITS | HEADER_FMT_INTEL; break; case SNDRV_PCM_FORMAT_S24_3BE: header = HEADER_FMT_BASE_LIN | HEADER_FMT_24BITS; @@ -414,7 +558,8 @@ static int pcxhr_set_format(struct pcxhr_stream *stream) header = HEADER_FMT_BASE_FLOAT | HEADER_FMT_INTEL; break; default: - snd_printk(KERN_ERR "error pcxhr_set_format() : unknown format\n"); + snd_printk(KERN_ERR + "error pcxhr_set_format() : unknown format\n"); return -EINVAL; } chip = snd_pcm_substream_chip(stream->substream); @@ -432,14 +577,31 @@ static int pcxhr_set_format(struct pcxhr_stream *stream) is_capture = stream->pipe->is_capture; stream_num = is_capture ? 0 : stream->substream->number; - pcxhr_init_rmh(&rmh, is_capture ? CMD_FORMAT_STREAM_IN : CMD_FORMAT_STREAM_OUT); - pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio, stream_num, 0); - if (is_capture) - rmh.cmd[0] |= 1<<12; + pcxhr_init_rmh(&rmh, is_capture ? + CMD_FORMAT_STREAM_IN : CMD_FORMAT_STREAM_OUT); + pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio, + stream_num, 0); + if (is_capture) { + /* bug with old dsp versions: */ + /* bit 12 also sets the format of the playback stream */ + if (DSP_EXT_CMD_SET(chip->mgr)) + rmh.cmd[0] |= 1<<10; + else + rmh.cmd[0] |= 1<<12; + } rmh.cmd[1] = 0; - rmh.cmd[2] = header >> 8; - rmh.cmd[3] = (header & 0xff) << 16; - rmh.cmd_len = 4; + rmh.cmd_len = 2; + if (DSP_EXT_CMD_SET(chip->mgr)) { + /* add channels and set bit 19 if channels>2 */ + rmh.cmd[1] = stream->channels; + if (!is_capture) { + /* playback : add channel mask to command */ + rmh.cmd[2] = (stream->channels == 1) ? 0x01 : 0x03; + rmh.cmd_len = 3; + } + } + rmh.cmd[rmh.cmd_len++] = header >> 8; + rmh.cmd[rmh.cmd_len++] = (header & 0xff) << 16; err = pcxhr_send_msg(chip->mgr, &rmh); if (err) snd_printk(KERN_ERR "ERROR pcxhr_set_format err=%x;\n", err); @@ -456,30 +618,38 @@ static int pcxhr_update_r_buffer(struct pcxhr_stream *stream) is_capture = (subs->stream == SNDRV_PCM_STREAM_CAPTURE); stream_num = is_capture ? 0 : subs->number; - snd_printdd("pcxhr_update_r_buffer(pcm%c%d) : addr(%p) bytes(%zx) subs(%d)\n", + snd_printdd("pcxhr_update_r_buffer(pcm%c%d) : " + "addr(%p) bytes(%zx) subs(%d)\n", is_capture ? 'c' : 'p', chip->chip_idx, (void *)(long)subs->runtime->dma_addr, subs->runtime->dma_bytes, subs->number); pcxhr_init_rmh(&rmh, CMD_UPDATE_R_BUFFERS); - pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio, stream_num, 0); + pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio, + stream_num, 0); /* max buffer size is 2 MByte */ snd_BUG_ON(subs->runtime->dma_bytes >= 0x200000); - rmh.cmd[1] = subs->runtime->dma_bytes * 8; /* size in bits */ - rmh.cmd[2] = subs->runtime->dma_addr >> 24; /* most significant byte */ - rmh.cmd[2] |= 1<<19; /* this is a circular buffer */ - rmh.cmd[3] = subs->runtime->dma_addr & MASK_DSP_WORD; /* least 3 significant bytes */ + /* size in bits */ + rmh.cmd[1] = subs->runtime->dma_bytes * 8; + /* most significant byte */ + rmh.cmd[2] = subs->runtime->dma_addr >> 24; + /* this is a circular buffer */ + rmh.cmd[2] |= 1<<19; + /* least 3 significant bytes */ + rmh.cmd[3] = subs->runtime->dma_addr & MASK_DSP_WORD; rmh.cmd_len = 4; err = pcxhr_send_msg(chip->mgr, &rmh); if (err) - snd_printk(KERN_ERR "ERROR CMD_UPDATE_R_BUFFERS err=%x;\n", err); + snd_printk(KERN_ERR + "ERROR CMD_UPDATE_R_BUFFERS err=%x;\n", err); return err; } #if 0 -static int pcxhr_pipe_sample_count(struct pcxhr_stream *stream, snd_pcm_uframes_t *sample_count) +static int pcxhr_pipe_sample_count(struct pcxhr_stream *stream, + snd_pcm_uframes_t *sample_count) { struct pcxhr_rmh rmh; int err; @@ -533,8 +703,8 @@ static void pcxhr_trigger_tasklet(unsigned long arg) for (j = 0; j < chip->nb_streams_play; j++) { if (pcxhr_stream_scheduled_get_pipe(&chip->playback_stream[j], &pipe)) { playback_mask |= (1 << pipe->first_audio); - break; /* add only once, as all playback streams of - * one chip use the same pipe + break; /* add only once, as all playback + * streams of one chip use the same pipe */ } } @@ -545,19 +715,21 @@ static void pcxhr_trigger_tasklet(unsigned long arg) return; } - snd_printdd("pcxhr_trigger_tasklet : playback_mask=%x capture_mask=%x\n", + snd_printdd("pcxhr_trigger_tasklet : " + "playback_mask=%x capture_mask=%x\n", playback_mask, capture_mask); /* synchronous stop of all the pipes concerned */ err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 0); if (err) { mutex_unlock(&mgr->setup_mutex); - snd_printk(KERN_ERR "pcxhr_trigger_tasklet : error stop pipes (P%x C%x)\n", + snd_printk(KERN_ERR "pcxhr_trigger_tasklet : " + "error stop pipes (P%x C%x)\n", playback_mask, capture_mask); return; } - /* unfortunately the dsp lost format and buffer info with the stop pipe */ + /* the dsp lost format and buffer info with the stop pipe */ for (i = 0; i < mgr->num_cards; i++) { struct pcxhr_stream *stream; chip = mgr->chip[i]; @@ -596,12 +768,15 @@ static void pcxhr_trigger_tasklet(unsigned long arg) err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 1); if (err) { mutex_unlock(&mgr->setup_mutex); - snd_printk(KERN_ERR "pcxhr_trigger_tasklet : error start pipes (P%x C%x)\n", + snd_printk(KERN_ERR "pcxhr_trigger_tasklet : " + "error start pipes (P%x C%x)\n", playback_mask, capture_mask); return; } - /* put the streams into the running state now (increment pointer by interrupt) */ + /* put the streams into the running state now + * (increment pointer by interrupt) + */ spin_lock_irqsave(&mgr->lock, flags); for ( i =0; i < mgr->num_cards; i++) { struct pcxhr_stream *stream; @@ -615,7 +790,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg) stream = &chip->playback_stream[j]; if (stream->status == PCXHR_STREAM_STATUS_STARTED) { /* playback will already have advanced ! */ - stream->timer_period_frag += PCXHR_GRANULARITY; + stream->timer_period_frag += mgr->granularity; stream->status = PCXHR_STREAM_STATUS_RUNNING; } } @@ -653,7 +828,7 @@ static int pcxhr_trigger(struct snd_pcm_substream *subs, int cmd) PCXHR_STREAM_STATUS_SCHEDULE_RUN; snd_pcm_trigger_done(s, subs); } - tasklet_hi_schedule(&chip->mgr->trigger_taskq); + tasklet_schedule(&chip->mgr->trigger_taskq); } else { stream = subs->runtime->private_data; snd_printdd("Only one Substream %c %d\n", @@ -697,12 +872,14 @@ static int pcxhr_hardware_timer(struct pcxhr_mgr *mgr, int start) pcxhr_init_rmh(&rmh, CMD_SET_TIMER_INTERRUPT); if (start) { - mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID; /* last dsp time invalid */ - rmh.cmd[0] |= PCXHR_GRANULARITY; + /* last dsp time invalid */ + mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID; + rmh.cmd[0] |= mgr->granularity; } err = pcxhr_send_msg(mgr, &rmh); if (err < 0) - snd_printk(KERN_ERR "error pcxhr_hardware_timer err(%x)\n", err); + snd_printk(KERN_ERR "error pcxhr_hardware_timer err(%x)\n", + err); return err; } @@ -713,38 +890,16 @@ static int pcxhr_prepare(struct snd_pcm_substream *subs) { struct snd_pcxhr *chip = snd_pcm_substream_chip(subs); struct pcxhr_mgr *mgr = chip->mgr; - /* - struct pcxhr_stream *stream = (pcxhr_stream_t*)subs->runtime->private_data; - */ int err = 0; snd_printdd("pcxhr_prepare : period_size(%lx) periods(%x) buffer_size(%lx)\n", subs->runtime->period_size, subs->runtime->periods, subs->runtime->buffer_size); - /* - if(subs->runtime->period_size <= PCXHR_GRANULARITY) { - snd_printk(KERN_ERR "pcxhr_prepare : error period_size too small (%x)\n", - (unsigned int)subs->runtime->period_size); - return -EINVAL; - } - */ - mutex_lock(&mgr->setup_mutex); do { - /* if the stream was stopped before, format and buffer were reset */ - /* - if(stream->status == PCXHR_STREAM_STATUS_STOPPED) { - err = pcxhr_set_format(stream); - if(err) break; - err = pcxhr_update_r_buffer(stream); - if(err) break; - } - */ - /* only the first stream can choose the sample rate */ - /* the further opened streams will be limited to its frequency (see open) */ /* set the clock only once (first stream) */ if (mgr->sample_rate != subs->runtime->rate) { err = pcxhr_set_clock(mgr, subs->runtime->rate); @@ -787,22 +942,9 @@ static int pcxhr_hw_params(struct snd_pcm_substream *subs, stream->channels = channels; stream->format = format; - /* set the format to the board */ - /* - err = pcxhr_set_format(stream); - if(err) { - mutex_unlock(&mgr->setup_mutex); - return err; - } - */ /* allocate buffer */ err = snd_pcm_lib_malloc_pages(subs, params_buffer_bytes(hw)); - /* - if (err > 0) { - err = pcxhr_update_r_buffer(stream); - } - */ mutex_unlock(&mgr->setup_mutex); return err; @@ -820,14 +962,18 @@ static int pcxhr_hw_free(struct snd_pcm_substream *subs) */ static struct snd_pcm_hardware pcxhr_caps = { - .info = ( SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START | - 0 /*SNDRV_PCM_INFO_PAUSE*/), - .formats = ( SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | - SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | - SNDRV_PCM_FMTBIT_FLOAT_LE ), - .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000, + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_3BE | + SNDRV_PCM_FMTBIT_FLOAT_LE), + .rates = (SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_192000), .rate_min = 8000, .rate_max = 192000, .channels_min = 1, @@ -847,6 +993,7 @@ static int pcxhr_open(struct snd_pcm_substream *subs) struct pcxhr_mgr *mgr = chip->mgr; struct snd_pcm_runtime *runtime = subs->runtime; struct pcxhr_stream *stream; + int err; mutex_lock(&mgr->setup_mutex); @@ -874,6 +1021,18 @@ static int pcxhr_open(struct snd_pcm_substream *subs) return -EBUSY; } + /* float format support is in some cases buggy on stereo cards */ + if (mgr->is_hr_stereo) + runtime->hw.formats &= ~SNDRV_PCM_FMTBIT_FLOAT_LE; + + /* buffer-size should better be multiple of period-size */ + err = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) { + mutex_unlock(&mgr->setup_mutex); + return err; + } + /* if a sample rate is already used or fixed by external clock, * the stream cannot change */ @@ -889,7 +1048,8 @@ static int pcxhr_open(struct snd_pcm_substream *subs) mutex_unlock(&mgr->setup_mutex); return -EBUSY; } - runtime->hw.rate_min = runtime->hw.rate_max = external_rate; + runtime->hw.rate_min = external_rate; + runtime->hw.rate_max = external_rate; } } @@ -899,9 +1059,11 @@ static int pcxhr_open(struct snd_pcm_substream *subs) runtime->private_data = stream; - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 4); - + /* better get a divisor of granularity values (96 or 192) */ + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32); + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32); snd_pcm_set_sync(subs); mgr->ref_count_rate++; @@ -919,11 +1081,12 @@ static int pcxhr_close(struct snd_pcm_substream *subs) mutex_lock(&mgr->setup_mutex); - snd_printdd("pcxhr_close chip%d subs%d\n", chip->chip_idx, subs->number); + snd_printdd("pcxhr_close chip%d subs%d\n", + chip->chip_idx, subs->number); /* sample rate released */ if (--mgr->ref_count_rate == 0) { - mgr->sample_rate = 0; /* the sample rate is no more locked */ + mgr->sample_rate = 0; /* the sample rate is no more locked */ pcxhr_hardware_timer(mgr, 0); /* stop the DSP-timer */ } @@ -1016,7 +1179,8 @@ static int pcxhr_chip_dev_free(struct snd_device *device) /* */ -static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card, int idx) +static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, + struct snd_card *card, int idx) { int err; struct snd_pcxhr *chip; @@ -1024,7 +1188,7 @@ static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card, .dev_free = pcxhr_chip_dev_free, }; - mgr->chip[idx] = chip = kzalloc(sizeof(*chip), GFP_KERNEL); + chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (! chip) { snd_printk(KERN_ERR "cannot allocate chip\n"); return -ENOMEM; @@ -1040,7 +1204,7 @@ static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card, if (idx < mgr->capture_chips) { if (mgr->mono_capture) - chip->nb_streams_capt = 2; /* 2 mono streams (left+right) */ + chip->nb_streams_capt = 2; /* 2 mono streams */ else chip->nb_streams_capt = 1; /* or 1 stereo stream */ } @@ -1050,13 +1214,15 @@ static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card, return err; } + mgr->chip[idx] = chip; snd_card_set_dev(card, &mgr->pci->dev); return 0; } /* proc interface */ -static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) +static void pcxhr_proc_info(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) { struct snd_pcxhr *chip = entry->private_data; struct pcxhr_mgr *mgr = chip->mgr; @@ -1069,8 +1235,10 @@ static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer short ver_maj = (mgr->dsp_version >> 16) & 0xff; short ver_min = (mgr->dsp_version >> 8) & 0xff; short ver_build = mgr->dsp_version & 0xff; - snd_iprintf(buffer, "module version %s\n", PCXHR_DRIVER_VERSION_STRING); - snd_iprintf(buffer, "dsp version %d.%d.%d\n", ver_maj, ver_min, ver_build); + snd_iprintf(buffer, "module version %s\n", + PCXHR_DRIVER_VERSION_STRING); + snd_iprintf(buffer, "dsp version %d.%d.%d\n", + ver_maj, ver_min, ver_build); if (mgr->board_has_analog) snd_iprintf(buffer, "analog io available\n"); else @@ -1084,18 +1252,22 @@ static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer if (ref > 0) { if (mgr->sample_rate_real != 0 && mgr->sample_rate_real != 48000) { - ref = (ref * 48000) / mgr->sample_rate_real; - if (mgr->sample_rate_real >= PCXHR_IRQ_TIMER_FREQ) + ref = (ref * 48000) / + mgr->sample_rate_real; + if (mgr->sample_rate_real >= + PCXHR_IRQ_TIMER_FREQ) ref *= 2; } cur = 100 - (100 * cur) / ref; snd_iprintf(buffer, "cpu load %d%%\n", cur); - snd_iprintf(buffer, "buffer pool %d/%d kWords\n", + snd_iprintf(buffer, "buffer pool %d/%d\n", rmh.stat[2], rmh.stat[3]); } } - snd_iprintf(buffer, "dma granularity : %d\n", PCXHR_GRANULARITY); - snd_iprintf(buffer, "dsp time errors : %d\n", mgr->dsp_time_err); + snd_iprintf(buffer, "dma granularity : %d\n", + mgr->granularity); + snd_iprintf(buffer, "dsp time errors : %d\n", + mgr->dsp_time_err); snd_iprintf(buffer, "dsp async pipe xrun errors : %d\n", mgr->async_err_pipe_xrun); snd_iprintf(buffer, "dsp async stream xrun errors : %d\n", @@ -1110,33 +1282,52 @@ static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer rmh.cmd_idx = CMD_LAST_INDEX; if( ! pcxhr_send_msg(mgr, &rmh) ) { int i; + if (rmh.stat_len > 8) + rmh.stat_len = 8; for (i = 0; i < rmh.stat_len; i++) - snd_iprintf(buffer, "debug[%02d] = %06x\n", i, rmh.stat[i]); + snd_iprintf(buffer, "debug[%02d] = %06x\n", + i, rmh.stat[i]); } } else snd_iprintf(buffer, "no firmware loaded\n"); snd_iprintf(buffer, "\n"); } -static void pcxhr_proc_sync(struct snd_info_entry *entry, struct snd_info_buffer *buffer) +static void pcxhr_proc_sync(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) { struct snd_pcxhr *chip = entry->private_data; struct pcxhr_mgr *mgr = chip->mgr; - static char *texts[7] = { - "Internal", "Word", "AES Sync", "AES 1", "AES 2", "AES 3", "AES 4" + static const char *textsHR22[3] = { + "Internal", "AES Sync", "AES 1" + }; + static const char *textsPCXHR[7] = { + "Internal", "Word", "AES Sync", + "AES 1", "AES 2", "AES 3", "AES 4" }; + const char **texts; + int max_clock; + if (mgr->is_hr_stereo) { + texts = textsHR22; + max_clock = HR22_CLOCK_TYPE_MAX; + } else { + texts = textsPCXHR; + max_clock = PCXHR_CLOCK_TYPE_MAX; + } snd_iprintf(buffer, "\n%s\n", mgr->longname); - snd_iprintf(buffer, "Current Sample Clock\t: %s\n", texts[mgr->cur_clock_type]); - snd_iprintf(buffer, "Current Sample Rate\t= %d\n", mgr->sample_rate_real); - + snd_iprintf(buffer, "Current Sample Clock\t: %s\n", + texts[mgr->cur_clock_type]); + snd_iprintf(buffer, "Current Sample Rate\t= %d\n", + mgr->sample_rate_real); /* commands available when embedded DSP is running */ if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) { int i, err, sample_rate; - for (i = PCXHR_CLOCK_TYPE_WORD_CLOCK; i< (3 + mgr->capture_chips); i++) { + for (i = 1; i <= max_clock; i++) { err = pcxhr_get_external_clock(mgr, i, &sample_rate); if (err) break; - snd_iprintf(buffer, "%s Clock\t\t= %d\n", texts[i], sample_rate); + snd_iprintf(buffer, "%s Clock\t\t= %d\n", + texts[i], sample_rate); } } else snd_iprintf(buffer, "no firmware loaded\n"); @@ -1194,7 +1385,8 @@ static int pcxhr_free(struct pcxhr_mgr *mgr) /* * probe function - creates the card manager */ -static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) +static int __devinit pcxhr_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) { static int dev; struct pcxhr_mgr *mgr; @@ -1217,7 +1409,8 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id /* check if we can restrict PCI DMA transfers to 32 bits */ if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) { - snd_printk(KERN_ERR "architecture does not support 32bit PCI busmaster DMA\n"); + snd_printk(KERN_ERR "architecture does not support " + "32bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; } @@ -1234,11 +1427,25 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id pci_disable_device(pci); return -ENODEV; } - card_name = pcxhr_board_params[pci_id->driver_data].board_name; - mgr->playback_chips = pcxhr_board_params[pci_id->driver_data].playback_chips; - mgr->capture_chips = pcxhr_board_params[pci_id->driver_data].capture_chips; - mgr->firmware_num = pcxhr_board_params[pci_id->driver_data].firmware_num; + card_name = + pcxhr_board_params[pci_id->driver_data].board_name; + mgr->playback_chips = + pcxhr_board_params[pci_id->driver_data].playback_chips; + mgr->capture_chips = + pcxhr_board_params[pci_id->driver_data].capture_chips; + mgr->fw_file_set = + pcxhr_board_params[pci_id->driver_data].fw_file_set; + mgr->firmware_num = + pcxhr_board_params[pci_id->driver_data].firmware_num; mgr->mono_capture = mono[dev]; + mgr->is_hr_stereo = (mgr->playback_chips == 1); + mgr->board_has_aes1 = PCXHR_BOARD_HAS_AES1(mgr); + mgr->board_aes_in_192k = !PCXHR_BOARD_AESIN_NO_192K(mgr); + + if (mgr->is_hr_stereo) + mgr->granularity = PCXHR_GRANULARITY_HR22; + else + mgr->granularity = PCXHR_GRANULARITY; /* resource assignment */ if ((err = pci_request_regions(pci, card_name)) < 0) { @@ -1261,7 +1468,8 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id mgr->irq = pci->irq; sprintf(mgr->shortname, "Digigram %s", card_name); - sprintf(mgr->longname, "%s at 0x%lx & 0x%lx, 0x%lx irq %i", mgr->shortname, + sprintf(mgr->longname, "%s at 0x%lx & 0x%lx, 0x%lx irq %i", + mgr->shortname, mgr->port[0], mgr->port[1], mgr->port[2], mgr->irq); /* ISR spinlock */ @@ -1272,10 +1480,14 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id mutex_init(&mgr->setup_mutex); /* init taslket */ - tasklet_init(&mgr->msg_taskq, pcxhr_msg_tasklet, (unsigned long) mgr); - tasklet_init(&mgr->trigger_taskq, pcxhr_trigger_tasklet, (unsigned long) mgr); + tasklet_init(&mgr->msg_taskq, pcxhr_msg_tasklet, + (unsigned long) mgr); + tasklet_init(&mgr->trigger_taskq, pcxhr_trigger_tasklet, + (unsigned long) mgr); + mgr->prmh = kmalloc(sizeof(*mgr->prmh) + - sizeof(u32) * (PCXHR_SIZE_MAX_LONG_STATUS - PCXHR_SIZE_MAX_STATUS), + sizeof(u32) * (PCXHR_SIZE_MAX_LONG_STATUS - + PCXHR_SIZE_MAX_STATUS), GFP_KERNEL); if (! mgr->prmh) { pcxhr_free(mgr); @@ -1296,7 +1508,8 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id else idx = index[dev] + i; - snprintf(tmpid, sizeof(tmpid), "%s-%d", id[dev] ? id[dev] : card_name, i); + snprintf(tmpid, sizeof(tmpid), "%s-%d", + id[dev] ? id[dev] : card_name, i); card = snd_card_new(idx, tmpid, THIS_MODULE, 0); if (! card) { @@ -1310,6 +1523,7 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id sprintf(card->longname, "%s [PCM #%d]", mgr->longname, i); if ((err = pcxhr_create(mgr, card, i)) < 0) { + snd_card_free(card); pcxhr_free(mgr); return err; } diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h index 652064787a55..84131a916c92 100644 --- a/sound/pci/pcxhr/pcxhr.h +++ b/sound/pci/pcxhr/pcxhr.h @@ -27,15 +27,18 @@ #include <linux/mutex.h> #include <sound/pcm.h> -#define PCXHR_DRIVER_VERSION 0x000804 /* 0.8.4 */ -#define PCXHR_DRIVER_VERSION_STRING "0.8.4" /* 0.8.4 */ +#define PCXHR_DRIVER_VERSION 0x000905 /* 0.9.5 */ +#define PCXHR_DRIVER_VERSION_STRING "0.9.5" /* 0.9.5 */ -#define PCXHR_MAX_CARDS 6 -#define PCXHR_PLAYBACK_STREAMS 4 +#define PCXHR_MAX_CARDS 6 +#define PCXHR_PLAYBACK_STREAMS 4 -#define PCXHR_GRANULARITY 96 /* transfer granularity (should be min 96 and multiple of 48) */ -#define PCXHR_GRANULARITY_MIN 96 /* transfer granularity of pipes and the dsp time (MBOX4) */ +#define PCXHR_GRANULARITY 96 /* min 96 and multiple of 48 */ +/* transfer granularity of pipes and the dsp time (MBOX4) */ +#define PCXHR_GRANULARITY_MIN 96 +/* TODO : granularity could be 64 or 128 */ +#define PCXHR_GRANULARITY_HR22 192 /* granularity for stereo cards */ struct snd_pcxhr; struct pcxhr_mgr; @@ -51,6 +54,11 @@ enum pcxhr_clock_type { PCXHR_CLOCK_TYPE_AES_2, PCXHR_CLOCK_TYPE_AES_3, PCXHR_CLOCK_TYPE_AES_4, + PCXHR_CLOCK_TYPE_MAX = PCXHR_CLOCK_TYPE_AES_4, + HR22_CLOCK_TYPE_INTERNAL = PCXHR_CLOCK_TYPE_INTERNAL, + HR22_CLOCK_TYPE_AES_SYNC, + HR22_CLOCK_TYPE_AES_1, + HR22_CLOCK_TYPE_MAX = HR22_CLOCK_TYPE_AES_1, }; struct pcxhr_mgr { @@ -61,6 +69,8 @@ struct pcxhr_mgr { int irq; + int granularity; + /* card access with 1 mem bar and 2 io bar's */ unsigned long port[3]; @@ -83,11 +93,16 @@ struct pcxhr_mgr { /* hardware interface */ unsigned int dsp_loaded; /* bit flags of loaded dsp indices */ unsigned int dsp_version; /* read from embedded once firmware is loaded */ - int board_has_analog; /* if 0 the board is digital only */ - int mono_capture; /* if 1 the board does mono capture */ - int playback_chips; /* 4 or 6 */ - int capture_chips; /* 4 or 1 */ - int firmware_num; /* 41 or 42 */ + int playback_chips; + int capture_chips; + int fw_file_set; + int firmware_num; + int is_hr_stereo:1; + int board_has_aes1:1; /* if 1 board has AES1 plug and SRC */ + int board_has_analog:1; /* if 0 the board is digital only */ + int board_has_mic:1; /* if 1 the board has microphone input */ + int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */ + int mono_capture:1; /* if 1 the board does mono capture */ struct snd_dma_buffer hostport; @@ -106,6 +121,9 @@ struct pcxhr_mgr { int async_err_stream_xrun; int async_err_pipe_xrun; int async_err_other_last; + + unsigned char xlx_cfg; /* copy of PCXHR_XLX_CFG register */ + unsigned char xlx_selmic; /* copy of PCXHR_XLX_SELMIC register */ }; @@ -155,24 +173,30 @@ struct snd_pcxhr { struct snd_pcm *pcm; /* PCM */ - struct pcxhr_pipe playback_pipe; /* 1 stereo pipe only */ - struct pcxhr_pipe capture_pipe[2]; /* 1 stereo pipe or 2 mono pipes */ + struct pcxhr_pipe playback_pipe; /* 1 stereo pipe only */ + struct pcxhr_pipe capture_pipe[2]; /* 1 stereo or 2 mono pipes */ struct pcxhr_stream playback_stream[PCXHR_PLAYBACK_STREAMS]; - struct pcxhr_stream capture_stream[2]; /* 1 stereo stream or 2 mono streams */ + struct pcxhr_stream capture_stream[2]; /* 1 stereo or 2 mono streams */ int nb_streams_play; int nb_streams_capt; - int analog_playback_active[2]; /* Mixer : Master Playback active (!mute) */ - int analog_playback_volume[2]; /* Mixer : Master Playback Volume */ - int analog_capture_volume[2]; /* Mixer : Master Capture Volume */ - int digital_playback_active[PCXHR_PLAYBACK_STREAMS][2]; /* Mixer : Digital Playback Active [streams][stereo]*/ - int digital_playback_volume[PCXHR_PLAYBACK_STREAMS][2]; /* Mixer : Digital Playback Volume [streams][stereo]*/ - int digital_capture_volume[2]; /* Mixer : Digital Capture Volume [stereo] */ - int monitoring_active[2]; /* Mixer : Monitoring Active */ - int monitoring_volume[2]; /* Mixer : Monitoring Volume */ - int audio_capture_source; /* Mixer : Audio Capture Source */ - unsigned char aes_bits[5]; /* Mixer : IEC958_AES bits */ + int analog_playback_active[2]; /* Mixer : Master Playback !mute */ + int analog_playback_volume[2]; /* Mixer : Master Playback Volume */ + int analog_capture_volume[2]; /* Mixer : Master Capture Volume */ + int digital_playback_active[PCXHR_PLAYBACK_STREAMS][2]; + int digital_playback_volume[PCXHR_PLAYBACK_STREAMS][2]; + int digital_capture_volume[2]; /* Mixer : Digital Capture Volume */ + int monitoring_active[2]; /* Mixer : Monitoring Active */ + int monitoring_volume[2]; /* Mixer : Monitoring Volume */ + int audio_capture_source; /* Mixer : Audio Capture Source */ + int mic_volume; /* used by cards with MIC only */ + int mic_boost; /* used by cards with MIC only */ + int mic_active; /* used by cards with MIC only */ + int analog_capture_active; /* used by cards with MIC only */ + int phantom_power; /* used by cards with MIC only */ + + unsigned char aes_bits[5]; /* Mixer : IEC958_AES bits */ }; struct pcxhr_hostport @@ -184,6 +208,8 @@ struct pcxhr_hostport /* exported */ int pcxhr_create_pcm(struct snd_pcxhr *chip); int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate); -int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_type, int *sample_rate); +int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, + enum pcxhr_clock_type clock_type, + int *sample_rate); #endif /* __SOUND_PCXHR_H */ diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 7143259cfe34..833e7180ad2d 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -132,13 +132,15 @@ static int pcxhr_check_reg_bit(struct pcxhr_mgr *mgr, unsigned int reg, *read = PCXHR_INPB(mgr, reg); if ((*read & mask) == bit) { if (i > 100) - snd_printdd("ATTENTION! check_reg(%x) loopcount=%d\n", + snd_printdd("ATTENTION! check_reg(%x) " + "loopcount=%d\n", reg, i); return 0; } i++; } while (time_after_eq(end_time, jiffies)); - snd_printk(KERN_ERR "pcxhr_check_reg_bit: timeout, reg=%x, mask=0x%x, val=0x%x\n", + snd_printk(KERN_ERR + "pcxhr_check_reg_bit: timeout, reg=%x, mask=0x%x, val=%x\n", reg, mask, *read); return -EIO; } @@ -159,18 +161,22 @@ static int pcxhr_check_reg_bit(struct pcxhr_mgr *mgr, unsigned int reg, #define PCXHR_IT_TEST_XILINX (0x0000003C | PCXHR_MASK_IT_HF1 | \ PCXHR_MASK_IT_MANAGE_HF5) #define PCXHR_IT_DOWNLOAD_BOOT (0x0000000C | PCXHR_MASK_IT_HF1 | \ - PCXHR_MASK_IT_MANAGE_HF5 | PCXHR_MASK_IT_WAIT) + PCXHR_MASK_IT_MANAGE_HF5 | \ + PCXHR_MASK_IT_WAIT) #define PCXHR_IT_RESET_BOARD_FUNC (0x0000000C | PCXHR_MASK_IT_HF0 | \ - PCXHR_MASK_IT_MANAGE_HF5 | PCXHR_MASK_IT_WAIT_EXTRA) + PCXHR_MASK_IT_MANAGE_HF5 | \ + PCXHR_MASK_IT_WAIT_EXTRA) #define PCXHR_IT_DOWNLOAD_DSP (0x0000000C | \ - PCXHR_MASK_IT_MANAGE_HF5 | PCXHR_MASK_IT_WAIT) + PCXHR_MASK_IT_MANAGE_HF5 | \ + PCXHR_MASK_IT_WAIT) #define PCXHR_IT_DEBUG (0x0000005A | PCXHR_MASK_IT_NO_HF0_HF1) #define PCXHR_IT_RESET_SEMAPHORE (0x0000005C | PCXHR_MASK_IT_NO_HF0_HF1) #define PCXHR_IT_MESSAGE (0x00000074 | PCXHR_MASK_IT_NO_HF0_HF1) #define PCXHR_IT_RESET_CHK (0x00000076 | PCXHR_MASK_IT_NO_HF0_HF1) #define PCXHR_IT_UPDATE_RBUFFER (0x00000078 | PCXHR_MASK_IT_NO_HF0_HF1) -static int pcxhr_send_it_dsp(struct pcxhr_mgr *mgr, unsigned int itdsp, int atomic) +static int pcxhr_send_it_dsp(struct pcxhr_mgr *mgr, + unsigned int itdsp, int atomic) { int err; unsigned char reg; @@ -178,17 +184,21 @@ static int pcxhr_send_it_dsp(struct pcxhr_mgr *mgr, unsigned int itdsp, int atom if (itdsp & PCXHR_MASK_IT_MANAGE_HF5) { /* clear hf5 bit */ PCXHR_OUTPL(mgr, PCXHR_PLX_MBOX0, - PCXHR_INPL(mgr, PCXHR_PLX_MBOX0) & ~PCXHR_MBOX0_HF5); + PCXHR_INPL(mgr, PCXHR_PLX_MBOX0) & + ~PCXHR_MBOX0_HF5); } if ((itdsp & PCXHR_MASK_IT_NO_HF0_HF1) == 0) { - reg = PCXHR_ICR_HI08_RREQ | PCXHR_ICR_HI08_TREQ | PCXHR_ICR_HI08_HDRQ; + reg = (PCXHR_ICR_HI08_RREQ | + PCXHR_ICR_HI08_TREQ | + PCXHR_ICR_HI08_HDRQ); if (itdsp & PCXHR_MASK_IT_HF0) reg |= PCXHR_ICR_HI08_HF0; if (itdsp & PCXHR_MASK_IT_HF1) reg |= PCXHR_ICR_HI08_HF1; PCXHR_OUTPB(mgr, PCXHR_DSP_ICR, reg); } - reg = (unsigned char)(((itdsp & PCXHR_MASK_EXTRA_INFO) >> 1) | PCXHR_CVR_HI08_HC); + reg = (unsigned char)(((itdsp & PCXHR_MASK_EXTRA_INFO) >> 1) | + PCXHR_CVR_HI08_HC); PCXHR_OUTPB(mgr, PCXHR_DSP_CVR, reg); if (itdsp & PCXHR_MASK_IT_WAIT) { if (atomic) @@ -211,10 +221,14 @@ static int pcxhr_send_it_dsp(struct pcxhr_mgr *mgr, unsigned int itdsp, int atom } if (itdsp & PCXHR_MASK_IT_MANAGE_HF5) { /* wait for hf5 bit */ - err = pcxhr_check_reg_bit(mgr, PCXHR_PLX_MBOX0, PCXHR_MBOX0_HF5, - PCXHR_MBOX0_HF5, PCXHR_TIMEOUT_DSP, ®); + err = pcxhr_check_reg_bit(mgr, PCXHR_PLX_MBOX0, + PCXHR_MBOX0_HF5, + PCXHR_MBOX0_HF5, + PCXHR_TIMEOUT_DSP, + ®); if (err) { - snd_printk(KERN_ERR "pcxhr_send_it_dsp : TIMEOUT HF5\n"); + snd_printk(KERN_ERR + "pcxhr_send_it_dsp : TIMEOUT HF5\n"); return err; } } @@ -263,7 +277,8 @@ void pcxhr_enable_dsp(struct pcxhr_mgr *mgr) /* * load the xilinx image */ -int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilinx, int second) +int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, + const struct firmware *xilinx, int second) { unsigned int i; unsigned int chipsc; @@ -274,7 +289,9 @@ int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilin /* test first xilinx */ chipsc = PCXHR_INPL(mgr, PCXHR_PLX_CHIPSC); /* REV01 cards do not support the PCXHR_CHIPSC_GPI_USERI bit anymore */ - /* this bit will always be 1; no possibility to test presence of first xilinx */ + /* this bit will always be 1; + * no possibility to test presence of first xilinx + */ if(second) { if ((chipsc & PCXHR_CHIPSC_GPI_USERI) == 0) { snd_printk(KERN_ERR "error loading first xilinx\n"); @@ -290,7 +307,8 @@ int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilin data = *image; mask = 0x80; while (mask) { - chipsc &= ~(PCXHR_CHIPSC_DATA_CLK | PCXHR_CHIPSC_DATA_IN); + chipsc &= ~(PCXHR_CHIPSC_DATA_CLK | + PCXHR_CHIPSC_DATA_IN); if (data & mask) chipsc |= PCXHR_CHIPSC_DATA_IN; PCXHR_OUTPL(mgr, PCXHR_PLX_CHIPSC, chipsc); @@ -330,15 +348,20 @@ static int pcxhr_download_dsp(struct pcxhr_mgr *mgr, const struct firmware *dsp) data = dsp->data + i; if (i == 0) { /* test data header consistency */ - len = (unsigned int)((data[0]<<16) + (data[1]<<8) + data[2]); - if (len && dsp->size != (len + 2) * 3) + len = (unsigned int)((data[0]<<16) + + (data[1]<<8) + + data[2]); + if (len && (dsp->size != (len + 2) * 3)) return -EINVAL; } /* wait DSP ready for new transfer */ - err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_TRDY, - PCXHR_ISR_HI08_TRDY, PCXHR_TIMEOUT_DSP, &dummy); + err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, + PCXHR_ISR_HI08_TRDY, + PCXHR_ISR_HI08_TRDY, + PCXHR_TIMEOUT_DSP, &dummy); if (err) { - snd_printk(KERN_ERR "dsp loading error at position %d\n", i); + snd_printk(KERN_ERR + "dsp loading error at position %d\n", i); return err; } /* send host data */ @@ -357,7 +380,8 @@ static int pcxhr_download_dsp(struct pcxhr_mgr *mgr, const struct firmware *dsp) /* * load the eeprom image */ -int pcxhr_load_eeprom_binary(struct pcxhr_mgr *mgr, const struct firmware *eeprom) +int pcxhr_load_eeprom_binary(struct pcxhr_mgr *mgr, + const struct firmware *eeprom) { int err; unsigned char reg; @@ -365,7 +389,9 @@ int pcxhr_load_eeprom_binary(struct pcxhr_mgr *mgr, const struct firmware *eepro /* init value of the ICR register */ reg = PCXHR_ICR_HI08_RREQ | PCXHR_ICR_HI08_TREQ | PCXHR_ICR_HI08_HDRQ; if (PCXHR_INPL(mgr, PCXHR_PLX_MBOX0) & PCXHR_MBOX0_BOOT_HERE) { - /* no need to load the eeprom binary, but init the HI08 interface */ + /* no need to load the eeprom binary, + * but init the HI08 interface + */ PCXHR_OUTPB(mgr, PCXHR_DSP_ICR, reg | PCXHR_ICR_HI08_INIT); msleep(PCXHR_WAIT_DEFAULT); PCXHR_OUTPB(mgr, PCXHR_DSP_ICR, reg); @@ -429,8 +455,10 @@ int pcxhr_load_dsp_binary(struct pcxhr_mgr *mgr, const struct firmware *dsp) if (err) return err; /* wait for chk bit */ - return pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_CHK, - PCXHR_ISR_HI08_CHK, PCXHR_TIMEOUT_DSP, &dummy); + return pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, + PCXHR_ISR_HI08_CHK, + PCXHR_ISR_HI08_CHK, + PCXHR_TIMEOUT_DSP, &dummy); } @@ -443,8 +471,8 @@ struct pcxhr_cmd_info { /* RMH status type */ enum { RMH_SSIZE_FIXED = 0, /* status size fix (st_length = 0..x) */ - RMH_SSIZE_ARG = 1, /* status size given in the LSB byte (used with st_length = 1) */ - RMH_SSIZE_MASK = 2, /* status size given in bitmask (used with st_length = 1) */ + RMH_SSIZE_ARG = 1, /* status size given in the LSB byte */ + RMH_SSIZE_MASK = 2, /* status size given in bitmask */ }; /* @@ -474,7 +502,7 @@ static struct pcxhr_cmd_info pcxhr_dsp_cmds[] = { [CMD_UPDATE_R_BUFFERS] = { 0x840000, 0, RMH_SSIZE_FIXED }, [CMD_FORMAT_STREAM_OUT] = { 0x860000, 0, RMH_SSIZE_FIXED }, [CMD_FORMAT_STREAM_IN] = { 0x870000, 0, RMH_SSIZE_FIXED }, -[CMD_STREAM_SAMPLE_COUNT] = { 0x902000, 2, RMH_SSIZE_FIXED }, /* stat_len = nb_streams * 2 */ +[CMD_STREAM_SAMPLE_COUNT] = { 0x902000, 2, RMH_SSIZE_FIXED }, [CMD_AUDIO_LEVEL_ADJUST] = { 0xc22000, 0, RMH_SSIZE_FIXED }, }; @@ -524,10 +552,13 @@ static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) for (i = 0; i < rmh->stat_len; i++) { /* wait for receiver full */ - err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_RXDF, - PCXHR_ISR_HI08_RXDF, PCXHR_TIMEOUT_DSP, ®); + err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, + PCXHR_ISR_HI08_RXDF, + PCXHR_ISR_HI08_RXDF, + PCXHR_TIMEOUT_DSP, ®); if (err) { - snd_printk(KERN_ERR "ERROR RMH stat: ISR:RXDF=1 (ISR = %x; i=%d )\n", + snd_printk(KERN_ERR "ERROR RMH stat: " + "ISR:RXDF=1 (ISR = %x; i=%d )\n", reg, i); return err; } @@ -537,10 +568,10 @@ static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) data |= PCXHR_INPB(mgr, PCXHR_DSP_TXL); /* need to update rmh->stat_len on the fly ?? */ - if (i==0) { + if (!i) { if (rmh->dsp_stat != RMH_SSIZE_FIXED) { if (rmh->dsp_stat == RMH_SSIZE_ARG) { - rmh->stat_len = (u16)(data & 0x0000ff) + 1; + rmh->stat_len = (data & 0x0000ff) + 1; data &= 0xffff00; } else { /* rmh->dsp_stat == RMH_SSIZE_MASK */ @@ -562,7 +593,8 @@ static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) rmh->stat[i] = data; } if (rmh->stat_len > max_stat_len) { - snd_printdd("PCXHR : rmh->stat_len=%x too big\n", rmh->stat_len); + snd_printdd("PCXHR : rmh->stat_len=%x too big\n", + rmh->stat_len); rmh->stat_len = max_stat_len; } return 0; @@ -605,7 +637,8 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) data &= 0xff7fff; /* MASK_1_WORD_COMMAND */ #ifdef CONFIG_SND_DEBUG_VERBOSE if (rmh->cmd_idx < CMD_LAST_INDEX) - snd_printdd("MSG cmd[0]=%x (%s)\n", data, cmd_names[rmh->cmd_idx]); + snd_printdd("MSG cmd[0]=%x (%s)\n", + data, cmd_names[rmh->cmd_idx]); #endif err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_TRDY, @@ -619,8 +652,10 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) if (rmh->cmd_len > 1) { /* send length */ data = rmh->cmd_len - 1; - err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_TRDY, - PCXHR_ISR_HI08_TRDY, PCXHR_TIMEOUT_DSP, ®); + err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, + PCXHR_ISR_HI08_TRDY, + PCXHR_ISR_HI08_TRDY, + PCXHR_TIMEOUT_DSP, ®); if (err) return err; PCXHR_OUTPB(mgr, PCXHR_DSP_TXH, (data>>16)&0xFF); @@ -653,8 +688,10 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) /* test status ISR */ if (reg & PCXHR_ISR_HI08_ERR) { /* ERROR, wait for receiver full */ - err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_RXDF, - PCXHR_ISR_HI08_RXDF, PCXHR_TIMEOUT_DSP, ®); + err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, + PCXHR_ISR_HI08_RXDF, + PCXHR_ISR_HI08_RXDF, + PCXHR_TIMEOUT_DSP, ®); if (err) { snd_printk(KERN_ERR "ERROR RMH: ISR:RXDF=1 (ISR = %x)\n", reg); return err; @@ -663,7 +700,8 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) data = PCXHR_INPB(mgr, PCXHR_DSP_TXH) << 16; data |= PCXHR_INPB(mgr, PCXHR_DSP_TXM) << 8; data |= PCXHR_INPB(mgr, PCXHR_DSP_TXL); - snd_printk(KERN_ERR "ERROR RMH(%d): 0x%x\n", rmh->cmd_idx, data); + snd_printk(KERN_ERR "ERROR RMH(%d): 0x%x\n", + rmh->cmd_idx, data); err = -EINVAL; } else { /* read the response data */ @@ -732,8 +770,9 @@ int pcxhr_send_msg(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) static inline int pcxhr_pipes_running(struct pcxhr_mgr *mgr) { int start_mask = PCXHR_INPL(mgr, PCXHR_PLX_MBOX2); - /* least segnificant 12 bits are the pipe states for the playback audios */ - /* next 12 bits are the pipe states for the capture audios + /* least segnificant 12 bits are the pipe states + * for the playback audios + * next 12 bits are the pipe states for the capture audios * (PCXHR_PIPE_STATE_CAPTURE_OFFSET) */ start_mask &= 0xffffff; @@ -744,7 +783,8 @@ static inline int pcxhr_pipes_running(struct pcxhr_mgr *mgr) #define PCXHR_PIPE_STATE_CAPTURE_OFFSET 12 #define MAX_WAIT_FOR_DSP 20 -static int pcxhr_prepair_pipe_start(struct pcxhr_mgr *mgr, int audio_mask, int *retry) +static int pcxhr_prepair_pipe_start(struct pcxhr_mgr *mgr, + int audio_mask, int *retry) { struct pcxhr_rmh rmh; int err; @@ -760,17 +800,20 @@ static int pcxhr_prepair_pipe_start(struct pcxhr_mgr *mgr, int audio_mask, int * } else { /* can start capture pipe */ pcxhr_set_pipe_cmd_params(&rmh, 1, audio - - PCXHR_PIPE_STATE_CAPTURE_OFFSET, - 0, 0); + PCXHR_PIPE_STATE_CAPTURE_OFFSET, + 0, 0); } err = pcxhr_send_msg(mgr, &rmh); if (err) { snd_printk(KERN_ERR - "error pipe start (CMD_CAN_START_PIPE) err=%x!\n", + "error pipe start " + "(CMD_CAN_START_PIPE) err=%x!\n", err); return err; } - /* if the pipe couldn't be prepaired for start, retry it later */ + /* if the pipe couldn't be prepaired for start, + * retry it later + */ if (rmh.stat[0] == 0) *retry |= (1<<audio); } @@ -795,14 +838,14 @@ static int pcxhr_stop_pipes(struct pcxhr_mgr *mgr, int audio_mask) } else { /* stop capture pipe */ pcxhr_set_pipe_cmd_params(&rmh, 1, audio - - PCXHR_PIPE_STATE_CAPTURE_OFFSET, - 0, 0); + PCXHR_PIPE_STATE_CAPTURE_OFFSET, + 0, 0); } err = pcxhr_send_msg(mgr, &rmh); if (err) { snd_printk(KERN_ERR - "error pipe stop (CMD_STOP_PIPE) err=%x!\n", - err); + "error pipe stop " + "(CMD_STOP_PIPE) err=%x!\n", err); return err; } } @@ -822,15 +865,16 @@ static int pcxhr_toggle_pipes(struct pcxhr_mgr *mgr, int audio_mask) if (audio_mask & 1) { pcxhr_init_rmh(&rmh, CMD_CONF_PIPE); if (audio < PCXHR_PIPE_STATE_CAPTURE_OFFSET) - pcxhr_set_pipe_cmd_params(&rmh, 0, 0, 0, 1 << audio); + pcxhr_set_pipe_cmd_params(&rmh, 0, 0, 0, + 1 << audio); else pcxhr_set_pipe_cmd_params(&rmh, 1, 0, 0, 1 << (audio - PCXHR_PIPE_STATE_CAPTURE_OFFSET)); err = pcxhr_send_msg(mgr, &rmh); if (err) { snd_printk(KERN_ERR - "error pipe start (CMD_CONF_PIPE) err=%x!\n", - err); + "error pipe start " + "(CMD_CONF_PIPE) err=%x!\n", err); return err; } } @@ -841,7 +885,9 @@ static int pcxhr_toggle_pipes(struct pcxhr_mgr *mgr, int audio_mask) pcxhr_init_rmh(&rmh, CMD_SEND_IRQA); err = pcxhr_send_msg(mgr, &rmh); if (err) { - snd_printk(KERN_ERR "error pipe start (CMD_SEND_IRQA) err=%x!\n", err ); + snd_printk(KERN_ERR + "error pipe start (CMD_SEND_IRQA) err=%x!\n", + err); return err; } return 0; @@ -849,7 +895,8 @@ static int pcxhr_toggle_pipes(struct pcxhr_mgr *mgr, int audio_mask) -int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_mask, int start) +int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, + int capture_mask, int start) { int state, i, err; int audio_mask; @@ -858,21 +905,23 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m struct timeval my_tv1, my_tv2; do_gettimeofday(&my_tv1); #endif - audio_mask = (playback_mask | (capture_mask << PCXHR_PIPE_STATE_CAPTURE_OFFSET)); + audio_mask = (playback_mask | + (capture_mask << PCXHR_PIPE_STATE_CAPTURE_OFFSET)); /* current pipe state (playback + record) */ state = pcxhr_pipes_running(mgr); snd_printdd("pcxhr_set_pipe_state %s (mask %x current %x)\n", start ? "START" : "STOP", audio_mask, state); if (start) { - audio_mask &= ~state; /* start only pipes that are not yet started */ + /* start only pipes that are not yet started */ + audio_mask &= ~state; state = audio_mask; for (i = 0; i < MAX_WAIT_FOR_DSP; i++) { err = pcxhr_prepair_pipe_start(mgr, state, &state); if (err) return err; if (state == 0) - break; /* success, all pipes prepaired for start */ - mdelay(1); /* otherwise wait 1 millisecond and retry */ + break; /* success, all pipes prepaired */ + mdelay(1); /* wait 1 millisecond and retry */ } } else { audio_mask &= state; /* stop only pipes that are started */ @@ -891,7 +940,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m if ((state & audio_mask) == (start ? audio_mask : 0)) break; if (++i >= MAX_WAIT_FOR_DSP * 100) { - snd_printk(KERN_ERR "error pipe start/stop (ED_NO_RESPONSE_AT_IRQA)\n"); + snd_printk(KERN_ERR "error pipe start/stop\n"); return -EBUSY; } udelay(10); /* wait 10 microseconds */ @@ -918,7 +967,8 @@ int pcxhr_write_io_num_reg_cont(struct pcxhr_mgr *mgr, unsigned int mask, spin_lock_irqsave(&mgr->msg_lock, flags); if ((mgr->io_num_reg_cont & mask) == value) { - snd_printdd("IO_NUM_REG_CONT mask %x already is set to %x\n", mask, value); + snd_printdd("IO_NUM_REG_CONT mask %x already is set to %x\n", + mask, value); if (changed) *changed = 0; spin_unlock_irqrestore(&mgr->msg_lock, flags); @@ -971,7 +1021,8 @@ static int pcxhr_handle_async_err(struct pcxhr_mgr *mgr, u32 err, err = ((err >> 12) & 0xfff); if (!err) return 0; - snd_printdd("CMD_ASYNC : Error %s %s Pipe %d err=%x\n", err_src_name[err_src], + snd_printdd("CMD_ASYNC : Error %s %s Pipe %d err=%x\n", + err_src_name[err_src], is_capture ? "Record" : "Play", pipe, err); if (err == 0xe01) mgr->async_err_stream_xrun++; @@ -996,6 +1047,13 @@ void pcxhr_msg_tasklet(unsigned long arg) snd_printdd("TASKLET : PCXHR_IRQ_TIME_CODE event occured\n"); if (mgr->src_it_dsp & PCXHR_IRQ_NOTIFY) snd_printdd("TASKLET : PCXHR_IRQ_NOTIFY event occured\n"); + if (mgr->src_it_dsp & (PCXHR_IRQ_FREQ_CHANGE | PCXHR_IRQ_TIME_CODE)) { + /* clear events FREQ_CHANGE and TIME_CODE */ + pcxhr_init_rmh(prmh, CMD_TEST_IT); + err = pcxhr_send_msg(mgr, prmh); + snd_printdd("CMD_TEST_IT : err=%x, stat=%x\n", + err, prmh->stat[0]); + } if (mgr->src_it_dsp & PCXHR_IRQ_ASYNC) { snd_printdd("TASKLET : PCXHR_IRQ_ASYNC event occured\n"); @@ -1005,18 +1063,22 @@ void pcxhr_msg_tasklet(unsigned long arg) prmh->stat_len = PCXHR_SIZE_MAX_LONG_STATUS; err = pcxhr_send_msg(mgr, prmh); if (err) - snd_printk(KERN_ERR "ERROR pcxhr_msg_tasklet=%x;\n", err); + snd_printk(KERN_ERR "ERROR pcxhr_msg_tasklet=%x;\n", + err); i = 1; while (i < prmh->stat_len) { - int nb_audio = (prmh->stat[i] >> FIELD_SIZE) & MASK_FIRST_FIELD; - int nb_stream = (prmh->stat[i] >> (2*FIELD_SIZE)) & MASK_FIRST_FIELD; + int nb_audio = ((prmh->stat[i] >> FIELD_SIZE) & + MASK_FIRST_FIELD); + int nb_stream = ((prmh->stat[i] >> (2*FIELD_SIZE)) & + MASK_FIRST_FIELD); int pipe = prmh->stat[i] & MASK_FIRST_FIELD; int is_capture = prmh->stat[i] & 0x400000; u32 err2; if (prmh->stat[i] & 0x800000) { /* if BIT_END */ snd_printdd("TASKLET : End%sPipe %d\n", - is_capture ? "Record" : "Play", pipe); + is_capture ? "Record" : "Play", + pipe); } i++; err2 = prmh->stat[i] ? prmh->stat[i] : prmh->stat[i+1]; @@ -1062,7 +1124,7 @@ static u_int64_t pcxhr_stream_read_position(struct pcxhr_mgr *mgr, pcxhr_init_rmh(&rmh, CMD_STREAM_SAMPLE_COUNT); pcxhr_set_pipe_cmd_params(&rmh, stream->pipe->is_capture, stream->pipe->first_audio, 0, stream_mask); - /* rmh.stat_len = 2; */ /* 2 resp data for each stream of the pipe */ + /* rmh.stat_len = 2; */ /* 2 resp data for each stream of the pipe */ err = pcxhr_send_msg(mgr, &rmh); if (err) @@ -1072,18 +1134,21 @@ static u_int64_t pcxhr_stream_read_position(struct pcxhr_mgr *mgr, hw_sample_count += (u_int64_t)rmh.stat[1]; snd_printdd("stream %c%d : abs samples real(%ld) timer(%ld)\n", - stream->pipe->is_capture ? 'C':'P', stream->substream->number, + stream->pipe->is_capture ? 'C' : 'P', + stream->substream->number, (long unsigned int)hw_sample_count, (long unsigned int)(stream->timer_abs_periods + - stream->timer_period_frag + PCXHR_GRANULARITY)); - + stream->timer_period_frag + + mgr->granularity)); return hw_sample_count; } static void pcxhr_update_timer_pos(struct pcxhr_mgr *mgr, - struct pcxhr_stream *stream, int samples_to_add) + struct pcxhr_stream *stream, + int samples_to_add) { - if (stream->substream && (stream->status == PCXHR_STREAM_STATUS_RUNNING)) { + if (stream->substream && + (stream->status == PCXHR_STREAM_STATUS_RUNNING)) { u_int64_t new_sample_count; int elapsed = 0; int hardware_read = 0; @@ -1092,20 +1157,22 @@ static void pcxhr_update_timer_pos(struct pcxhr_mgr *mgr, if (samples_to_add < 0) { stream->timer_is_synced = 0; /* add default if no hardware_read possible */ - samples_to_add = PCXHR_GRANULARITY; + samples_to_add = mgr->granularity; } if (!stream->timer_is_synced) { - if (stream->timer_abs_periods != 0 || - stream->timer_period_frag + PCXHR_GRANULARITY >= - runtime->period_size) { - new_sample_count = pcxhr_stream_read_position(mgr, stream); + if ((stream->timer_abs_periods != 0) || + ((stream->timer_period_frag + samples_to_add) >= + runtime->period_size)) { + new_sample_count = + pcxhr_stream_read_position(mgr, stream); hardware_read = 1; - if (new_sample_count >= PCXHR_GRANULARITY_MIN) { - /* sub security offset because of jitter and - * finer granularity of dsp time (MBOX4) + if (new_sample_count >= mgr->granularity) { + /* sub security offset because of + * jitter and finer granularity of + * dsp time (MBOX4) */ - new_sample_count -= PCXHR_GRANULARITY_MIN; + new_sample_count -= mgr->granularity; stream->timer_is_synced = 1; } } @@ -1128,12 +1195,15 @@ static void pcxhr_update_timer_pos(struct pcxhr_mgr *mgr, stream->timer_buf_periods = 0; stream->timer_abs_periods = new_elapse_pos; } - if (new_sample_count >= stream->timer_abs_periods) - stream->timer_period_frag = (u_int32_t)(new_sample_count - - stream->timer_abs_periods); - else - snd_printk(KERN_ERR "ERROR new_sample_count too small ??? %lx\n", + if (new_sample_count >= stream->timer_abs_periods) { + stream->timer_period_frag = + (u_int32_t)(new_sample_count - + stream->timer_abs_periods); + } else { + snd_printk(KERN_ERR + "ERROR new_sample_count too small ??? %ld\n", (long unsigned int)new_sample_count); + } if (elapsed) { spin_unlock(&mgr->lock); @@ -1143,7 +1213,6 @@ static void pcxhr_update_timer_pos(struct pcxhr_mgr *mgr, } } - irqreturn_t pcxhr_interrupt(int irq, void *dev_id) { struct pcxhr_mgr *mgr = dev_id; @@ -1156,7 +1225,8 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) reg = PCXHR_INPL(mgr, PCXHR_PLX_IRQCS); if (! (reg & PCXHR_IRQCS_ACTIVE_PCIDB)) { spin_unlock(&mgr->lock); - return IRQ_NONE; /* this device did not cause the interrupt */ + /* this device did not cause the interrupt */ + return IRQ_NONE; } /* clear interrupt */ @@ -1167,10 +1237,12 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) if (reg & PCXHR_IRQ_TIMER) { int timer_toggle = reg & PCXHR_IRQ_TIMER; /* is a 24 bit counter */ - int dsp_time_new = PCXHR_INPL(mgr, PCXHR_PLX_MBOX4) & PCXHR_DSP_TIME_MASK; + int dsp_time_new = + PCXHR_INPL(mgr, PCXHR_PLX_MBOX4) & PCXHR_DSP_TIME_MASK; int dsp_time_diff = dsp_time_new - mgr->dsp_time_last; - if (dsp_time_diff < 0 && mgr->dsp_time_last != PCXHR_DSP_TIME_INVALID) { + if ((dsp_time_diff < 0) && + (mgr->dsp_time_last != PCXHR_DSP_TIME_INVALID)) { snd_printdd("ERROR DSP TIME old(%d) new(%d) -> " "resynchronize all streams\n", mgr->dsp_time_last, dsp_time_new); @@ -1178,42 +1250,51 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) } #ifdef CONFIG_SND_DEBUG_VERBOSE if (dsp_time_diff == 0) - snd_printdd("ERROR DSP TIME NO DIFF time(%d)\n", dsp_time_new); - else if (dsp_time_diff >= (2*PCXHR_GRANULARITY)) + snd_printdd("ERROR DSP TIME NO DIFF time(%d)\n", + dsp_time_new); + else if (dsp_time_diff >= (2*mgr->granularity)) snd_printdd("ERROR DSP TIME TOO BIG old(%d) add(%d)\n", - mgr->dsp_time_last, dsp_time_new - mgr->dsp_time_last); + mgr->dsp_time_last, + dsp_time_new - mgr->dsp_time_last); + else if (dsp_time_diff % mgr->granularity) + snd_printdd("ERROR DSP TIME increased by %d\n", + dsp_time_diff); #endif mgr->dsp_time_last = dsp_time_new; - if (timer_toggle == mgr->timer_toggle) + if (timer_toggle == mgr->timer_toggle) { snd_printdd("ERROR TIMER TOGGLE\n"); + mgr->dsp_time_err++; + } mgr->timer_toggle = timer_toggle; reg &= ~PCXHR_IRQ_TIMER; for (i = 0; i < mgr->num_cards; i++) { chip = mgr->chip[i]; for (j = 0; j < chip->nb_streams_capt; j++) - pcxhr_update_timer_pos(mgr, &chip->capture_stream[j], - dsp_time_diff); + pcxhr_update_timer_pos(mgr, + &chip->capture_stream[j], + dsp_time_diff); } for (i = 0; i < mgr->num_cards; i++) { chip = mgr->chip[i]; for (j = 0; j < chip->nb_streams_play; j++) - pcxhr_update_timer_pos(mgr, &chip->playback_stream[j], - dsp_time_diff); + pcxhr_update_timer_pos(mgr, + &chip->playback_stream[j], + dsp_time_diff); } } /* other irq's handled in the tasklet */ if (reg & PCXHR_IRQ_MASK) { - - /* as we didn't request any notifications, some kind of xrun error - * will probably occured - */ - /* better resynchronize all streams next interrupt : */ - mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID; - + if (reg & PCXHR_IRQ_ASYNC) { + /* as we didn't request any async notifications, + * some kind of xrun error will probably occured + */ + /* better resynchronize all streams next interrupt : */ + mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID; + } mgr->src_it_dsp = reg; - tasklet_hi_schedule(&mgr->msg_taskq); + tasklet_schedule(&mgr->msg_taskq); } #ifdef CONFIG_SND_DEBUG_VERBOSE if (reg & PCXHR_FATAL_DSP_ERR) diff --git a/sound/pci/pcxhr/pcxhr_core.h b/sound/pci/pcxhr/pcxhr_core.h index d9a4ab609875..bbbd66d13a64 100644 --- a/sound/pci/pcxhr/pcxhr_core.h +++ b/sound/pci/pcxhr/pcxhr_core.h @@ -65,7 +65,7 @@ enum { CMD_RESYNC_AUDIO_INPUTS, /* cmd_len = 1 stat_len = 0 */ CMD_GET_DSP_RESOURCES, /* cmd_len = 1 stat_len = 4 */ CMD_SET_TIMER_INTERRUPT, /* cmd_len = 1 stat_len = 0 */ - CMD_RES_PIPE, /* cmd_len = 2 stat_len = 0 */ + CMD_RES_PIPE, /* cmd_len >=2 stat_len = 0 */ CMD_FREE_PIPE, /* cmd_len = 1 stat_len = 0 */ CMD_CONF_PIPE, /* cmd_len = 2 stat_len = 0 */ CMD_STOP_PIPE, /* cmd_len = 1 stat_len = 0 */ @@ -96,6 +96,8 @@ void pcxhr_init_rmh(struct pcxhr_rmh *rmh, int cmd); void pcxhr_set_pipe_cmd_params(struct pcxhr_rmh* rmh, int capture, unsigned int param1, unsigned int param2, unsigned int param3); +#define DSP_EXT_CMD_SET(x) (x->dsp_version > 0x012800) + /* send the rmh */ @@ -110,6 +112,7 @@ int pcxhr_send_msg(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh); #define IO_NUM_REG_STATUS 5 #define IO_NUM_REG_CUER 10 #define IO_NUM_UER_CHIP_REG 11 +#define IO_NUM_REG_CONFIG_SRC 12 #define IO_NUM_REG_OUT_ANA_LEVEL 20 #define IO_NUM_REG_IN_ANA_LEVEL 21 diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index 96640d9c227d..592743a298b0 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -31,6 +31,7 @@ #include "pcxhr_mixer.h" #include "pcxhr_hwdep.h" #include "pcxhr_core.h" +#include "pcxhr_mix22.h" #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) @@ -40,10 +41,10 @@ #endif +static int pcxhr_sub_init(struct pcxhr_mgr *mgr); /* * get basic information and init pcxhr card */ - static int pcxhr_init_board(struct pcxhr_mgr *mgr) { int err; @@ -68,7 +69,7 @@ static int pcxhr_init_board(struct pcxhr_mgr *mgr) if ((rmh.stat[0] & MASK_FIRST_FIELD) != mgr->playback_chips * 2) return -EINVAL; /* test 8 or 2 phys in */ - if (((rmh.stat[0] >> (2 * FIELD_SIZE)) & MASK_FIRST_FIELD) != + if (((rmh.stat[0] >> (2 * FIELD_SIZE)) & MASK_FIRST_FIELD) < mgr->capture_chips * 2) return -EINVAL; /* test max nb substream per board */ @@ -77,20 +78,34 @@ static int pcxhr_init_board(struct pcxhr_mgr *mgr) /* test max nb substream per pipe */ if (((rmh.stat[1] >> 7) & 0x5F) < PCXHR_PLAYBACK_STREAMS) return -EINVAL; + snd_printdd("supported formats : playback=%x capture=%x\n", + rmh.stat[2], rmh.stat[3]); pcxhr_init_rmh(&rmh, CMD_VERSION); /* firmware num for DSP */ rmh.cmd[0] |= mgr->firmware_num; /* transfer granularity in samples (should be multiple of 48) */ - rmh.cmd[1] = (1<<23) + PCXHR_GRANULARITY; + rmh.cmd[1] = (1<<23) + mgr->granularity; rmh.cmd_len = 2; err = pcxhr_send_msg(mgr, &rmh); if (err) return err; - snd_printdd("PCXHR DSP version is %d.%d.%d\n", - (rmh.stat[0]>>16)&0xff, (rmh.stat[0]>>8)&0xff, rmh.stat[0]&0xff); + snd_printdd("PCXHR DSP version is %d.%d.%d\n", (rmh.stat[0]>>16)&0xff, + (rmh.stat[0]>>8)&0xff, rmh.stat[0]&0xff); mgr->dsp_version = rmh.stat[0]; + if (mgr->is_hr_stereo) + err = hr222_sub_init(mgr); + else + err = pcxhr_sub_init(mgr); + return err; +} + +static int pcxhr_sub_init(struct pcxhr_mgr *mgr) +{ + int err; + struct pcxhr_rmh rmh; + /* get options */ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); rmh.cmd[0] |= IO_NUM_REG_STATUS; @@ -100,20 +115,22 @@ static int pcxhr_init_board(struct pcxhr_mgr *mgr) if (err) return err; - if ((rmh.stat[1] & REG_STATUS_OPT_DAUGHTER_MASK) == REG_STATUS_OPT_ANALOG_BOARD) - mgr->board_has_analog = 1; /* analog addon board available */ - else - /* analog addon board not available -> no support for instance */ - return -EINVAL; + if ((rmh.stat[1] & REG_STATUS_OPT_DAUGHTER_MASK) == + REG_STATUS_OPT_ANALOG_BOARD) + mgr->board_has_analog = 1; /* analog addon board found */ /* unmute inputs */ err = pcxhr_write_io_num_reg_cont(mgr, REG_CONT_UNMUTE_INPUTS, REG_CONT_UNMUTE_INPUTS, NULL); if (err) return err; - /* unmute outputs */ - pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); /* a write to IO_NUM_REG_MUTE_OUT mutes! */ + /* unmute outputs (a write to IO_NUM_REG_MUTE_OUT mutes!) */ + pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT; + if (DSP_EXT_CMD_SET(mgr)) { + rmh.cmd[1] = 1; /* unmute digital plugs */ + rmh.cmd_len = 2; + } err = pcxhr_send_msg(mgr, &rmh); return err; } @@ -124,19 +141,25 @@ void pcxhr_reset_board(struct pcxhr_mgr *mgr) if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) { /* mute outputs */ + if (!mgr->is_hr_stereo) { /* a read to IO_NUM_REG_MUTE_OUT register unmutes! */ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT; pcxhr_send_msg(mgr, &rmh); /* mute inputs */ - pcxhr_write_io_num_reg_cont(mgr, REG_CONT_UNMUTE_INPUTS, 0, NULL); + pcxhr_write_io_num_reg_cont(mgr, REG_CONT_UNMUTE_INPUTS, + 0, NULL); + } + /* stereo cards mute with reset of dsp */ } /* reset pcxhr dsp */ - if (mgr->dsp_loaded & ( 1 << PCXHR_FIRMWARE_DSP_EPRM_INDEX)) + if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_EPRM_INDEX)) pcxhr_reset_dsp(mgr); /* reset second xilinx */ - if (mgr->dsp_loaded & ( 1 << PCXHR_FIRMWARE_XLX_COM_INDEX)) + if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_XLX_COM_INDEX)) { pcxhr_reset_xilinx_com(mgr); + mgr->dsp_loaded = 1; + } return; } @@ -144,8 +167,9 @@ void pcxhr_reset_board(struct pcxhr_mgr *mgr) /* * allocate a playback/capture pipe (pcmp0/pcmc0) */ -static int pcxhr_dsp_allocate_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pipe, - int is_capture, int pin) +static int pcxhr_dsp_allocate_pipe(struct pcxhr_mgr *mgr, + struct pcxhr_pipe *pipe, + int is_capture, int pin) { int stream_count, audio_count; int err; @@ -161,15 +185,23 @@ static int pcxhr_dsp_allocate_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pi stream_count = PCXHR_PLAYBACK_STREAMS; audio_count = 2; /* always stereo */ } - snd_printdd("snd_add_ref_pipe pin(%d) pcm%c0\n", pin, is_capture ? 'c' : 'p'); + snd_printdd("snd_add_ref_pipe pin(%d) pcm%c0\n", + pin, is_capture ? 'c' : 'p'); pipe->is_capture = is_capture; pipe->first_audio = pin; /* define pipe (P_PCM_ONLY_MASK (0x020000) is not necessary) */ pcxhr_init_rmh(&rmh, CMD_RES_PIPE); - pcxhr_set_pipe_cmd_params(&rmh, is_capture, pin, audio_count, stream_count); + pcxhr_set_pipe_cmd_params(&rmh, is_capture, pin, + audio_count, stream_count); + rmh.cmd[1] |= 0x020000; /* add P_PCM_ONLY_MASK */ + if (DSP_EXT_CMD_SET(mgr)) { + /* add channel mask to command */ + rmh.cmd[rmh.cmd_len++] = (audio_count == 1) ? 0x01 : 0x03; + } err = pcxhr_send_msg(mgr, &rmh); if (err < 0) { - snd_printk(KERN_ERR "error pipe allocation (CMD_RES_PIPE) err=%x!\n", err ); + snd_printk(KERN_ERR "error pipe allocation " + "(CMD_RES_PIPE) err=%x!\n", err); return err; } pipe->status = PCXHR_PIPE_DEFINED; @@ -199,10 +231,12 @@ static int pcxhr_dsp_free_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pipe) snd_printk(KERN_ERR "error stopping pipe!\n"); /* release the pipe */ pcxhr_init_rmh(&rmh, CMD_FREE_PIPE); - pcxhr_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->first_audio, 0, 0); + pcxhr_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->first_audio, + 0, 0); err = pcxhr_send_msg(mgr, &rmh); if (err < 0) - snd_printk(KERN_ERR "error pipe release (CMD_FREE_PIPE) err(%x)\n", err); + snd_printk(KERN_ERR "error pipe release " + "(CMD_FREE_PIPE) err(%x)\n", err); pipe->status = PCXHR_PIPE_UNDEFINED; return err; } @@ -248,15 +282,16 @@ static int pcxhr_start_pipes(struct pcxhr_mgr *mgr) for (i = 0; i < mgr->num_cards; i++) { chip = mgr->chip[i]; if (chip->nb_streams_play) - playback_mask |= (1 << chip->playback_pipe.first_audio); + playback_mask |= 1 << chip->playback_pipe.first_audio; for (j = 0; j < chip->nb_streams_capt; j++) - capture_mask |= (1 << chip->capture_pipe[j].first_audio); + capture_mask |= 1 << chip->capture_pipe[j].first_audio; } return pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 1); } -static int pcxhr_dsp_load(struct pcxhr_mgr *mgr, int index, const struct firmware *dsp) +static int pcxhr_dsp_load(struct pcxhr_mgr *mgr, int index, + const struct firmware *dsp) { int err, card_index; @@ -330,22 +365,33 @@ static int pcxhr_dsp_load(struct pcxhr_mgr *mgr, int index, const struct firmwar int pcxhr_setup_firmware(struct pcxhr_mgr *mgr) { - static char *fw_files[5] = { - "xi_1_882.dat", - "xc_1_882.dat", - "e321_512.e56", - "b321_512.b56", - "d321_512.d56" + static char *fw_files[][5] = { + [0] = { "xlxint.dat", "xlxc882hr.dat", + "dspe882.e56", "dspb882hr.b56", "dspd882.d56" }, + [1] = { "xlxint.dat", "xlxc882e.dat", + "dspe882.e56", "dspb882e.b56", "dspd882.d56" }, + [2] = { "xlxint.dat", "xlxc1222hr.dat", + "dspe882.e56", "dspb1222hr.b56", "dspd1222.d56" }, + [3] = { "xlxint.dat", "xlxc1222e.dat", + "dspe882.e56", "dspb1222e.b56", "dspd1222.d56" }, + [4] = { NULL, "xlxc222.dat", + "dspe924.e56", "dspb924.b56", "dspd222.d56" }, + [5] = { NULL, "xlxc924.dat", + "dspe924.e56", "dspb924.b56", "dspd222.d56" }, }; char path[32]; const struct firmware *fw_entry; int i, err; + int fw_set = mgr->fw_file_set; - for (i = 0; i < ARRAY_SIZE(fw_files); i++) { - sprintf(path, "pcxhr/%s", fw_files[i]); + for (i = 0; i < 5; i++) { + if (!fw_files[fw_set][i]) + continue; + sprintf(path, "pcxhr/%s", fw_files[fw_set][i]); if (request_firmware(&fw_entry, path, &mgr->pci->dev)) { - snd_printk(KERN_ERR "pcxhr: can't load firmware %s\n", path); + snd_printk(KERN_ERR "pcxhr: can't load firmware %s\n", + path); return -ENOENT; } /* fake hwdep dsp record */ @@ -358,11 +404,26 @@ int pcxhr_setup_firmware(struct pcxhr_mgr *mgr) return 0; } -MODULE_FIRMWARE("pcxhr/xi_1_882.dat"); -MODULE_FIRMWARE("pcxhr/xc_1_882.dat"); -MODULE_FIRMWARE("pcxhr/e321_512.e56"); -MODULE_FIRMWARE("pcxhr/b321_512.b56"); -MODULE_FIRMWARE("pcxhr/d321_512.d56"); +MODULE_FIRMWARE("pcxhr/xlxint.dat"); +MODULE_FIRMWARE("pcxhr/xlxc882hr.dat"); +MODULE_FIRMWARE("pcxhr/xlxc882e.dat"); +MODULE_FIRMWARE("pcxhr/dspe882.e56"); +MODULE_FIRMWARE("pcxhr/dspb882hr.b56"); +MODULE_FIRMWARE("pcxhr/dspb882e.b56"); +MODULE_FIRMWARE("pcxhr/dspd882.d56"); + +MODULE_FIRMWARE("pcxhr/xlxc1222hr.dat"); +MODULE_FIRMWARE("pcxhr/xlxc1222e.dat"); +MODULE_FIRMWARE("pcxhr/dspb1222hr.b56"); +MODULE_FIRMWARE("pcxhr/dspb1222e.b56"); +MODULE_FIRMWARE("pcxhr/dspd1222.d56"); + +MODULE_FIRMWARE("pcxhr/xlxc222.dat"); +MODULE_FIRMWARE("pcxhr/xlxc924.dat"); +MODULE_FIRMWARE("pcxhr/dspe924.e56"); +MODULE_FIRMWARE("pcxhr/dspb924.b56"); +MODULE_FIRMWARE("pcxhr/dspd222.d56"); + #else /* old style firmware loading */ @@ -373,7 +434,8 @@ MODULE_FIRMWARE("pcxhr/d321_512.d56"); static int pcxhr_hwdep_dsp_status(struct snd_hwdep *hw, struct snd_hwdep_dsp_status *info) { - strcpy(info->id, "pcxhr"); + struct pcxhr_mgr *mgr = hw->private_data; + sprintf(info->id, "pcxhr%d", mgr->fw_file_set); info->num_dsps = PCXHR_FIRMWARE_FILES_MAX_INDEX; if (hw->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) @@ -393,8 +455,8 @@ static int pcxhr_hwdep_dsp_load(struct snd_hwdep *hw, fw.size = dsp->length; fw.data = vmalloc(fw.size); if (! fw.data) { - snd_printk(KERN_ERR "pcxhr: cannot allocate dsp image (%lu bytes)\n", - (unsigned long)fw.size); + snd_printk(KERN_ERR "pcxhr: cannot allocate dsp image " + "(%lu bytes)\n", (unsigned long)fw.size); return -ENOMEM; } if (copy_from_user((void *)fw.data, dsp->image, dsp->length)) { @@ -424,8 +486,11 @@ int pcxhr_setup_firmware(struct pcxhr_mgr *mgr) int err; struct snd_hwdep *hw; - /* only create hwdep interface for first cardX (see "index" module parameter)*/ - if ((err = snd_hwdep_new(mgr->chip[0]->card, PCXHR_HWDEP_ID, 0, &hw)) < 0) + /* only create hwdep interface for first cardX + * (see "index" module parameter) + */ + err = snd_hwdep_new(mgr->chip[0]->card, PCXHR_HWDEP_ID, 0, &hw); + if (err < 0) return err; hw->iface = SNDRV_HWDEP_IFACE_PCXHR; @@ -435,10 +500,13 @@ int pcxhr_setup_firmware(struct pcxhr_mgr *mgr) hw->ops.dsp_status = pcxhr_hwdep_dsp_status; hw->ops.dsp_load = pcxhr_hwdep_dsp_load; hw->exclusive = 1; + /* stereo cards don't need fw_file_0 -> dsp_loaded = 1 */ + hw->dsp_loaded = mgr->is_hr_stereo ? 1 : 0; mgr->dsp_loaded = 0; sprintf(hw->name, PCXHR_HWDEP_ID); - if ((err = snd_card_register(mgr->chip[0]->card)) < 0) + err = snd_card_register(mgr->chip[0]->card); + if (err < 0) return err; return 0; } diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c new file mode 100644 index 000000000000..ff019126b672 --- /dev/null +++ b/sound/pci/pcxhr/pcxhr_mix22.c @@ -0,0 +1,820 @@ +/* + * Driver for Digigram pcxhr compatible soundcards + * + * mixer interface for stereo cards + * + * Copyright (c) 2004 by Digigram <alsa@digigram.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/delay.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/tlv.h> +#include <sound/asoundef.h> +#include "pcxhr.h" +#include "pcxhr_core.h" +#include "pcxhr_mix22.h" + + +/* registers used on the DSP and Xilinx (port 2) : HR stereo cards only */ +#define PCXHR_DSP_RESET 0x20 +#define PCXHR_XLX_CFG 0x24 +#define PCXHR_XLX_RUER 0x28 +#define PCXHR_XLX_DATA 0x2C +#define PCXHR_XLX_STATUS 0x30 +#define PCXHR_XLX_LOFREQ 0x34 +#define PCXHR_XLX_HIFREQ 0x38 +#define PCXHR_XLX_CSUER 0x3C +#define PCXHR_XLX_SELMIC 0x40 + +#define PCXHR_DSP 2 + +/* byte access only ! */ +#define PCXHR_INPB(mgr, x) inb((mgr)->port[PCXHR_DSP] + (x)) +#define PCXHR_OUTPB(mgr, x, data) outb((data), (mgr)->port[PCXHR_DSP] + (x)) + + +/* values for PCHR_DSP_RESET register */ +#define PCXHR_DSP_RESET_DSP 0x01 +#define PCXHR_DSP_RESET_MUTE 0x02 +#define PCXHR_DSP_RESET_CODEC 0x08 + +/* values for PCHR_XLX_CFG register */ +#define PCXHR_CFG_SYNCDSP_MASK 0x80 +#define PCXHR_CFG_DEPENDENCY_MASK 0x60 +#define PCXHR_CFG_INDEPENDANT_SEL 0x00 +#define PCXHR_CFG_MASTER_SEL 0x40 +#define PCXHR_CFG_SLAVE_SEL 0x20 +#define PCXHR_CFG_DATA_UER1_SEL_MASK 0x10 /* 0 (UER0), 1(UER1) */ +#define PCXHR_CFG_DATAIN_SEL_MASK 0x08 /* 0 (ana), 1 (UER) */ +#define PCXHR_CFG_SRC_MASK 0x04 /* 0 (Bypass), 1 (SRC Actif) */ +#define PCXHR_CFG_CLOCK_UER1_SEL_MASK 0x02 /* 0 (UER0), 1(UER1) */ +#define PCXHR_CFG_CLOCKIN_SEL_MASK 0x01 /* 0 (internal), 1 (AES/EBU) */ + +/* values for PCHR_XLX_DATA register */ +#define PCXHR_DATA_CODEC 0x80 +#define AKM_POWER_CONTROL_CMD 0xA007 +#define AKM_RESET_ON_CMD 0xA100 +#define AKM_RESET_OFF_CMD 0xA103 +#define AKM_CLOCK_INF_55K_CMD 0xA240 +#define AKM_CLOCK_SUP_55K_CMD 0xA24D +#define AKM_MUTE_CMD 0xA38D +#define AKM_UNMUTE_CMD 0xA30D +#define AKM_LEFT_LEVEL_CMD 0xA600 +#define AKM_RIGHT_LEVEL_CMD 0xA700 + +/* values for PCHR_XLX_STATUS register - READ */ +#define PCXHR_STAT_SRC_LOCK 0x01 +#define PCXHR_STAT_LEVEL_IN 0x02 +#define PCXHR_STAT_MIC_CAPS 0x10 +/* values for PCHR_XLX_STATUS register - WRITE */ +#define PCXHR_STAT_FREQ_SYNC_MASK 0x01 +#define PCXHR_STAT_FREQ_UER1_MASK 0x02 +#define PCXHR_STAT_FREQ_SAVE_MASK 0x80 + +/* values for PCHR_XLX_CSUER register */ +#define PCXHR_SUER1_BIT_U_READ_MASK 0x80 +#define PCXHR_SUER1_BIT_C_READ_MASK 0x40 +#define PCXHR_SUER1_DATA_PRESENT_MASK 0x20 +#define PCXHR_SUER1_CLOCK_PRESENT_MASK 0x10 +#define PCXHR_SUER_BIT_U_READ_MASK 0x08 +#define PCXHR_SUER_BIT_C_READ_MASK 0x04 +#define PCXHR_SUER_DATA_PRESENT_MASK 0x02 +#define PCXHR_SUER_CLOCK_PRESENT_MASK 0x01 + +#define PCXHR_SUER_BIT_U_WRITE_MASK 0x02 +#define PCXHR_SUER_BIT_C_WRITE_MASK 0x01 + +/* values for PCXHR_XLX_SELMIC register - WRITE */ +#define PCXHR_SELMIC_PREAMPLI_OFFSET 2 +#define PCXHR_SELMIC_PREAMPLI_MASK 0x0C +#define PCXHR_SELMIC_PHANTOM_ALIM 0x80 + + +static const unsigned char g_hr222_p_level[] = { + 0x00, /* [000] -49.5 dB: AKM[000] = -1.#INF dB (mute) */ + 0x01, /* [001] -49.0 dB: AKM[001] = -48.131 dB (diff=0.86920 dB) */ + 0x01, /* [002] -48.5 dB: AKM[001] = -48.131 dB (diff=0.36920 dB) */ + 0x01, /* [003] -48.0 dB: AKM[001] = -48.131 dB (diff=0.13080 dB) */ + 0x01, /* [004] -47.5 dB: AKM[001] = -48.131 dB (diff=0.63080 dB) */ + 0x01, /* [005] -46.5 dB: AKM[001] = -48.131 dB (diff=1.63080 dB) */ + 0x01, /* [006] -47.0 dB: AKM[001] = -48.131 dB (diff=1.13080 dB) */ + 0x01, /* [007] -46.0 dB: AKM[001] = -48.131 dB (diff=2.13080 dB) */ + 0x01, /* [008] -45.5 dB: AKM[001] = -48.131 dB (diff=2.63080 dB) */ + 0x02, /* [009] -45.0 dB: AKM[002] = -42.110 dB (diff=2.88980 dB) */ + 0x02, /* [010] -44.5 dB: AKM[002] = -42.110 dB (diff=2.38980 dB) */ + 0x02, /* [011] -44.0 dB: AKM[002] = -42.110 dB (diff=1.88980 dB) */ + 0x02, /* [012] -43.5 dB: AKM[002] = -42.110 dB (diff=1.38980 dB) */ + 0x02, /* [013] -43.0 dB: AKM[002] = -42.110 dB (diff=0.88980 dB) */ + 0x02, /* [014] -42.5 dB: AKM[002] = -42.110 dB (diff=0.38980 dB) */ + 0x02, /* [015] -42.0 dB: AKM[002] = -42.110 dB (diff=0.11020 dB) */ + 0x02, /* [016] -41.5 dB: AKM[002] = -42.110 dB (diff=0.61020 dB) */ + 0x02, /* [017] -41.0 dB: AKM[002] = -42.110 dB (diff=1.11020 dB) */ + 0x02, /* [018] -40.5 dB: AKM[002] = -42.110 dB (diff=1.61020 dB) */ + 0x03, /* [019] -40.0 dB: AKM[003] = -38.588 dB (diff=1.41162 dB) */ + 0x03, /* [020] -39.5 dB: AKM[003] = -38.588 dB (diff=0.91162 dB) */ + 0x03, /* [021] -39.0 dB: AKM[003] = -38.588 dB (diff=0.41162 dB) */ + 0x03, /* [022] -38.5 dB: AKM[003] = -38.588 dB (diff=0.08838 dB) */ + 0x03, /* [023] -38.0 dB: AKM[003] = -38.588 dB (diff=0.58838 dB) */ + 0x03, /* [024] -37.5 dB: AKM[003] = -38.588 dB (diff=1.08838 dB) */ + 0x04, /* [025] -37.0 dB: AKM[004] = -36.090 dB (diff=0.91040 dB) */ + 0x04, /* [026] -36.5 dB: AKM[004] = -36.090 dB (diff=0.41040 dB) */ + 0x04, /* [027] -36.0 dB: AKM[004] = -36.090 dB (diff=0.08960 dB) */ + 0x04, /* [028] -35.5 dB: AKM[004] = -36.090 dB (diff=0.58960 dB) */ + 0x05, /* [029] -35.0 dB: AKM[005] = -34.151 dB (diff=0.84860 dB) */ + 0x05, /* [030] -34.5 dB: AKM[005] = -34.151 dB (diff=0.34860 dB) */ + 0x05, /* [031] -34.0 dB: AKM[005] = -34.151 dB (diff=0.15140 dB) */ + 0x05, /* [032] -33.5 dB: AKM[005] = -34.151 dB (diff=0.65140 dB) */ + 0x06, /* [033] -33.0 dB: AKM[006] = -32.568 dB (diff=0.43222 dB) */ + 0x06, /* [034] -32.5 dB: AKM[006] = -32.568 dB (diff=0.06778 dB) */ + 0x06, /* [035] -32.0 dB: AKM[006] = -32.568 dB (diff=0.56778 dB) */ + 0x07, /* [036] -31.5 dB: AKM[007] = -31.229 dB (diff=0.27116 dB) */ + 0x07, /* [037] -31.0 dB: AKM[007] = -31.229 dB (diff=0.22884 dB) */ + 0x08, /* [038] -30.5 dB: AKM[008] = -30.069 dB (diff=0.43100 dB) */ + 0x08, /* [039] -30.0 dB: AKM[008] = -30.069 dB (diff=0.06900 dB) */ + 0x09, /* [040] -29.5 dB: AKM[009] = -29.046 dB (diff=0.45405 dB) */ + 0x09, /* [041] -29.0 dB: AKM[009] = -29.046 dB (diff=0.04595 dB) */ + 0x0a, /* [042] -28.5 dB: AKM[010] = -28.131 dB (diff=0.36920 dB) */ + 0x0a, /* [043] -28.0 dB: AKM[010] = -28.131 dB (diff=0.13080 dB) */ + 0x0b, /* [044] -27.5 dB: AKM[011] = -27.303 dB (diff=0.19705 dB) */ + 0x0b, /* [045] -27.0 dB: AKM[011] = -27.303 dB (diff=0.30295 dB) */ + 0x0c, /* [046] -26.5 dB: AKM[012] = -26.547 dB (diff=0.04718 dB) */ + 0x0d, /* [047] -26.0 dB: AKM[013] = -25.852 dB (diff=0.14806 dB) */ + 0x0e, /* [048] -25.5 dB: AKM[014] = -25.208 dB (diff=0.29176 dB) */ + 0x0e, /* [049] -25.0 dB: AKM[014] = -25.208 dB (diff=0.20824 dB) */ + 0x0f, /* [050] -24.5 dB: AKM[015] = -24.609 dB (diff=0.10898 dB) */ + 0x10, /* [051] -24.0 dB: AKM[016] = -24.048 dB (diff=0.04840 dB) */ + 0x11, /* [052] -23.5 dB: AKM[017] = -23.522 dB (diff=0.02183 dB) */ + 0x12, /* [053] -23.0 dB: AKM[018] = -23.025 dB (diff=0.02535 dB) */ + 0x13, /* [054] -22.5 dB: AKM[019] = -22.556 dB (diff=0.05573 dB) */ + 0x14, /* [055] -22.0 dB: AKM[020] = -22.110 dB (diff=0.11020 dB) */ + 0x15, /* [056] -21.5 dB: AKM[021] = -21.686 dB (diff=0.18642 dB) */ + 0x17, /* [057] -21.0 dB: AKM[023] = -20.896 dB (diff=0.10375 dB) */ + 0x18, /* [058] -20.5 dB: AKM[024] = -20.527 dB (diff=0.02658 dB) */ + 0x1a, /* [059] -20.0 dB: AKM[026] = -19.831 dB (diff=0.16866 dB) */ + 0x1b, /* [060] -19.5 dB: AKM[027] = -19.504 dB (diff=0.00353 dB) */ + 0x1d, /* [061] -19.0 dB: AKM[029] = -18.883 dB (diff=0.11716 dB) */ + 0x1e, /* [062] -18.5 dB: AKM[030] = -18.588 dB (diff=0.08838 dB) */ + 0x20, /* [063] -18.0 dB: AKM[032] = -18.028 dB (diff=0.02780 dB) */ + 0x22, /* [064] -17.5 dB: AKM[034] = -17.501 dB (diff=0.00123 dB) */ + 0x24, /* [065] -17.0 dB: AKM[036] = -17.005 dB (diff=0.00475 dB) */ + 0x26, /* [066] -16.5 dB: AKM[038] = -16.535 dB (diff=0.03513 dB) */ + 0x28, /* [067] -16.0 dB: AKM[040] = -16.090 dB (diff=0.08960 dB) */ + 0x2b, /* [068] -15.5 dB: AKM[043] = -15.461 dB (diff=0.03857 dB) */ + 0x2d, /* [069] -15.0 dB: AKM[045] = -15.067 dB (diff=0.06655 dB) */ + 0x30, /* [070] -14.5 dB: AKM[048] = -14.506 dB (diff=0.00598 dB) */ + 0x33, /* [071] -14.0 dB: AKM[051] = -13.979 dB (diff=0.02060 dB) */ + 0x36, /* [072] -13.5 dB: AKM[054] = -13.483 dB (diff=0.01707 dB) */ + 0x39, /* [073] -13.0 dB: AKM[057] = -13.013 dB (diff=0.01331 dB) */ + 0x3c, /* [074] -12.5 dB: AKM[060] = -12.568 dB (diff=0.06778 dB) */ + 0x40, /* [075] -12.0 dB: AKM[064] = -12.007 dB (diff=0.00720 dB) */ + 0x44, /* [076] -11.5 dB: AKM[068] = -11.481 dB (diff=0.01937 dB) */ + 0x48, /* [077] -11.0 dB: AKM[072] = -10.984 dB (diff=0.01585 dB) */ + 0x4c, /* [078] -10.5 dB: AKM[076] = -10.515 dB (diff=0.01453 dB) */ + 0x51, /* [079] -10.0 dB: AKM[081] = -9.961 dB (diff=0.03890 dB) */ + 0x55, /* [080] -9.5 dB: AKM[085] = -9.542 dB (diff=0.04243 dB) */ + 0x5a, /* [081] -9.0 dB: AKM[090] = -9.046 dB (diff=0.04595 dB) */ + 0x60, /* [082] -8.5 dB: AKM[096] = -8.485 dB (diff=0.01462 dB) */ + 0x66, /* [083] -8.0 dB: AKM[102] = -7.959 dB (diff=0.04120 dB) */ + 0x6c, /* [084] -7.5 dB: AKM[108] = -7.462 dB (diff=0.03767 dB) */ + 0x72, /* [085] -7.0 dB: AKM[114] = -6.993 dB (diff=0.00729 dB) */ + 0x79, /* [086] -6.5 dB: AKM[121] = -6.475 dB (diff=0.02490 dB) */ + 0x80, /* [087] -6.0 dB: AKM[128] = -5.987 dB (diff=0.01340 dB) */ + 0x87, /* [088] -5.5 dB: AKM[135] = -5.524 dB (diff=0.02413 dB) */ + 0x8f, /* [089] -5.0 dB: AKM[143] = -5.024 dB (diff=0.02408 dB) */ + 0x98, /* [090] -4.5 dB: AKM[152] = -4.494 dB (diff=0.00607 dB) */ + 0xa1, /* [091] -4.0 dB: AKM[161] = -3.994 dB (diff=0.00571 dB) */ + 0xaa, /* [092] -3.5 dB: AKM[170] = -3.522 dB (diff=0.02183 dB) */ + 0xb5, /* [093] -3.0 dB: AKM[181] = -2.977 dB (diff=0.02277 dB) */ + 0xbf, /* [094] -2.5 dB: AKM[191] = -2.510 dB (diff=0.01014 dB) */ + 0xcb, /* [095] -2.0 dB: AKM[203] = -1.981 dB (diff=0.01912 dB) */ + 0xd7, /* [096] -1.5 dB: AKM[215] = -1.482 dB (diff=0.01797 dB) */ + 0xe3, /* [097] -1.0 dB: AKM[227] = -1.010 dB (diff=0.01029 dB) */ + 0xf1, /* [098] -0.5 dB: AKM[241] = -0.490 dB (diff=0.00954 dB) */ + 0xff, /* [099] +0.0 dB: AKM[255] = +0.000 dB (diff=0.00000 dB) */ +}; + + +static void hr222_config_akm(struct pcxhr_mgr *mgr, unsigned short data) +{ + unsigned short mask = 0x8000; + /* activate access to codec registers */ + PCXHR_INPB(mgr, PCXHR_XLX_HIFREQ); + + while (mask) { + PCXHR_OUTPB(mgr, PCXHR_XLX_DATA, + data & mask ? PCXHR_DATA_CODEC : 0); + mask >>= 1; + } + /* termiate access to codec registers */ + PCXHR_INPB(mgr, PCXHR_XLX_RUER); +} + + +static int hr222_set_hw_playback_level(struct pcxhr_mgr *mgr, + int idx, int level) +{ + unsigned short cmd; + if (idx > 1 || + level < 0 || + level >= ARRAY_SIZE(g_hr222_p_level)) + return -EINVAL; + + if (idx == 0) + cmd = AKM_LEFT_LEVEL_CMD; + else + cmd = AKM_RIGHT_LEVEL_CMD; + + /* conversion from PmBoardCodedLevel to AKM nonlinear programming */ + cmd += g_hr222_p_level[level]; + + hr222_config_akm(mgr, cmd); + return 0; +} + + +static int hr222_set_hw_capture_level(struct pcxhr_mgr *mgr, + int level_l, int level_r, int level_mic) +{ + /* program all input levels at the same time */ + unsigned int data; + int i; + + if (!mgr->capture_chips) + return -EINVAL; /* no PCX22 */ + + data = ((level_mic & 0xff) << 24); /* micro is mono, but apply */ + data |= ((level_mic & 0xff) << 16); /* level on both channels */ + data |= ((level_r & 0xff) << 8); /* line input right channel */ + data |= (level_l & 0xff); /* line input left channel */ + + PCXHR_INPB(mgr, PCXHR_XLX_DATA); /* activate input codec */ + /* send 32 bits (4 x 8 bits) */ + for (i = 0; i < 32; i++, data <<= 1) { + PCXHR_OUTPB(mgr, PCXHR_XLX_DATA, + (data & 0x80000000) ? PCXHR_DATA_CODEC : 0); + } + PCXHR_INPB(mgr, PCXHR_XLX_RUER); /* close input level codec */ + return 0; +} + +static void hr222_micro_boost(struct pcxhr_mgr *mgr, int level); + +int hr222_sub_init(struct pcxhr_mgr *mgr) +{ + unsigned char reg; + + mgr->board_has_analog = 1; /* analog always available */ + mgr->xlx_cfg = PCXHR_CFG_SYNCDSP_MASK; + + reg = PCXHR_INPB(mgr, PCXHR_XLX_STATUS); + if (reg & PCXHR_STAT_MIC_CAPS) + mgr->board_has_mic = 1; /* microphone available */ + snd_printdd("MIC input available = %d\n", mgr->board_has_mic); + + /* reset codec */ + PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, + PCXHR_DSP_RESET_DSP); + msleep(5); + PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, + PCXHR_DSP_RESET_DSP | + PCXHR_DSP_RESET_MUTE | + PCXHR_DSP_RESET_CODEC); + msleep(5); + + /* config AKM */ + hr222_config_akm(mgr, AKM_POWER_CONTROL_CMD); + hr222_config_akm(mgr, AKM_CLOCK_INF_55K_CMD); + hr222_config_akm(mgr, AKM_UNMUTE_CMD); + hr222_config_akm(mgr, AKM_RESET_OFF_CMD); + + /* init micro boost */ + hr222_micro_boost(mgr, 0); + + return 0; +} + + +/* calc PLL register */ +/* TODO : there is a very similar fct in pcxhr.c */ +static int hr222_pll_freq_register(unsigned int freq, + unsigned int *pllreg, + unsigned int *realfreq) +{ + unsigned int reg; + + if (freq < 6900 || freq > 219000) + return -EINVAL; + reg = (28224000 * 2) / freq; + reg = (reg - 1) / 2; + if (reg < 0x100) + *pllreg = reg + 0xC00; + else if (reg < 0x200) + *pllreg = reg + 0x800; + else if (reg < 0x400) + *pllreg = reg & 0x1ff; + else if (reg < 0x800) { + *pllreg = ((reg >> 1) & 0x1ff) + 0x200; + reg &= ~1; + } else { + *pllreg = ((reg >> 2) & 0x1ff) + 0x400; + reg &= ~3; + } + if (realfreq) + *realfreq = (28224000 / (reg + 1)); + return 0; +} + +int hr222_sub_set_clock(struct pcxhr_mgr *mgr, + unsigned int rate, + int *changed) +{ + unsigned int speed, pllreg = 0; + int err; + unsigned realfreq = rate; + + switch (mgr->use_clock_type) { + case HR22_CLOCK_TYPE_INTERNAL: + err = hr222_pll_freq_register(rate, &pllreg, &realfreq); + if (err) + return err; + + mgr->xlx_cfg &= ~(PCXHR_CFG_CLOCKIN_SEL_MASK | + PCXHR_CFG_CLOCK_UER1_SEL_MASK); + break; + case HR22_CLOCK_TYPE_AES_SYNC: + mgr->xlx_cfg |= PCXHR_CFG_CLOCKIN_SEL_MASK; + mgr->xlx_cfg &= ~PCXHR_CFG_CLOCK_UER1_SEL_MASK; + break; + case HR22_CLOCK_TYPE_AES_1: + if (!mgr->board_has_aes1) + return -EINVAL; + + mgr->xlx_cfg |= (PCXHR_CFG_CLOCKIN_SEL_MASK | + PCXHR_CFG_CLOCK_UER1_SEL_MASK); + break; + default: + return -EINVAL; + } + hr222_config_akm(mgr, AKM_MUTE_CMD); + + if (mgr->use_clock_type == HR22_CLOCK_TYPE_INTERNAL) { + PCXHR_OUTPB(mgr, PCXHR_XLX_HIFREQ, pllreg >> 8); + PCXHR_OUTPB(mgr, PCXHR_XLX_LOFREQ, pllreg & 0xff); + } + + /* set clock source */ + PCXHR_OUTPB(mgr, PCXHR_XLX_CFG, mgr->xlx_cfg); + + /* codec speed modes */ + speed = rate < 55000 ? 0 : 1; + if (mgr->codec_speed != speed) { + mgr->codec_speed = speed; + if (speed == 0) + hr222_config_akm(mgr, AKM_CLOCK_INF_55K_CMD); + else + hr222_config_akm(mgr, AKM_CLOCK_SUP_55K_CMD); + } + + mgr->sample_rate_real = realfreq; + mgr->cur_clock_type = mgr->use_clock_type; + + if (changed) + *changed = 1; + + hr222_config_akm(mgr, AKM_UNMUTE_CMD); + + snd_printdd("set_clock to %dHz (realfreq=%d pllreg=%x)\n", + rate, realfreq, pllreg); + return 0; +} + +int hr222_get_external_clock(struct pcxhr_mgr *mgr, + enum pcxhr_clock_type clock_type, + int *sample_rate) +{ + int rate, calc_rate = 0; + unsigned int ticks; + unsigned char mask, reg; + + if (clock_type == HR22_CLOCK_TYPE_AES_SYNC) { + + mask = (PCXHR_SUER_CLOCK_PRESENT_MASK | + PCXHR_SUER_DATA_PRESENT_MASK); + reg = PCXHR_STAT_FREQ_SYNC_MASK; + + } else if (clock_type == HR22_CLOCK_TYPE_AES_1 && mgr->board_has_aes1) { + + mask = (PCXHR_SUER1_CLOCK_PRESENT_MASK | + PCXHR_SUER1_DATA_PRESENT_MASK); + reg = PCXHR_STAT_FREQ_UER1_MASK; + + } else { + snd_printdd("get_external_clock : type %d not supported\n", + clock_type); + return -EINVAL; /* other clocks not supported */ + } + + if ((PCXHR_INPB(mgr, PCXHR_XLX_CSUER) & mask) != mask) { + snd_printdd("get_external_clock(%d) = 0 Hz\n", clock_type); + *sample_rate = 0; + return 0; /* no external clock locked */ + } + + PCXHR_OUTPB(mgr, PCXHR_XLX_STATUS, reg); /* calculate freq */ + + /* save the measured clock frequency */ + reg |= PCXHR_STAT_FREQ_SAVE_MASK; + + if (mgr->last_reg_stat != reg) { + udelay(500); /* wait min 2 cycles of lowest freq (8000) */ + mgr->last_reg_stat = reg; + } + + PCXHR_OUTPB(mgr, PCXHR_XLX_STATUS, reg); /* save */ + + /* get the frequency */ + ticks = (unsigned int)PCXHR_INPB(mgr, PCXHR_XLX_CFG); + ticks = (ticks & 0x03) << 8; + ticks |= (unsigned int)PCXHR_INPB(mgr, PCXHR_DSP_RESET); + + if (ticks != 0) + calc_rate = 28224000 / ticks; + /* rounding */ + if (calc_rate > 184200) + rate = 192000; + else if (calc_rate > 152200) + rate = 176400; + else if (calc_rate > 112000) + rate = 128000; + else if (calc_rate > 92100) + rate = 96000; + else if (calc_rate > 76100) + rate = 88200; + else if (calc_rate > 56000) + rate = 64000; + else if (calc_rate > 46050) + rate = 48000; + else if (calc_rate > 38050) + rate = 44100; + else if (calc_rate > 28000) + rate = 32000; + else if (calc_rate > 23025) + rate = 24000; + else if (calc_rate > 19025) + rate = 22050; + else if (calc_rate > 14000) + rate = 16000; + else if (calc_rate > 11512) + rate = 12000; + else if (calc_rate > 9512) + rate = 11025; + else if (calc_rate > 7000) + rate = 8000; + else + rate = 0; + + snd_printdd("External clock is at %d Hz (measured %d Hz)\n", + rate, calc_rate); + *sample_rate = rate; + return 0; +} + + +int hr222_update_analog_audio_level(struct snd_pcxhr *chip, + int is_capture, int channel) +{ + snd_printdd("hr222_update_analog_audio_level(%s chan=%d)\n", + is_capture ? "capture" : "playback", channel); + if (is_capture) { + int level_l, level_r, level_mic; + /* we have to update all levels */ + if (chip->analog_capture_active) { + level_l = chip->analog_capture_volume[0]; + level_r = chip->analog_capture_volume[1]; + } else { + level_l = HR222_LINE_CAPTURE_LEVEL_MIN; + level_r = HR222_LINE_CAPTURE_LEVEL_MIN; + } + if (chip->mic_active) + level_mic = chip->mic_volume; + else + level_mic = HR222_MICRO_CAPTURE_LEVEL_MIN; + return hr222_set_hw_capture_level(chip->mgr, + level_l, level_r, level_mic); + } else { + int vol; + if (chip->analog_playback_active[channel]) + vol = chip->analog_playback_volume[channel]; + else + vol = HR222_LINE_PLAYBACK_LEVEL_MIN; + return hr222_set_hw_playback_level(chip->mgr, channel, vol); + } +} + + +/*texts[5] = {"Line", "Digital", "Digi+SRC", "Mic", "Line+Mic"}*/ +#define SOURCE_LINE 0 +#define SOURCE_DIGITAL 1 +#define SOURCE_DIGISRC 2 +#define SOURCE_MIC 3 +#define SOURCE_LINEMIC 4 + +int hr222_set_audio_source(struct snd_pcxhr *chip) +{ + int digital = 0; + /* default analog source */ + chip->mgr->xlx_cfg &= ~(PCXHR_CFG_SRC_MASK | + PCXHR_CFG_DATAIN_SEL_MASK | + PCXHR_CFG_DATA_UER1_SEL_MASK); + + if (chip->audio_capture_source == SOURCE_DIGISRC) { + chip->mgr->xlx_cfg |= PCXHR_CFG_SRC_MASK; + digital = 1; + } else { + if (chip->audio_capture_source == SOURCE_DIGITAL) + digital = 1; + } + if (digital) { + chip->mgr->xlx_cfg |= PCXHR_CFG_DATAIN_SEL_MASK; + if (chip->mgr->board_has_aes1) { + /* get data from the AES1 plug */ + chip->mgr->xlx_cfg |= PCXHR_CFG_DATA_UER1_SEL_MASK; + } + /* chip->mic_active = 0; */ + /* chip->analog_capture_active = 0; */ + } else { + int update_lvl = 0; + chip->analog_capture_active = 0; + chip->mic_active = 0; + if (chip->audio_capture_source == SOURCE_LINE || + chip->audio_capture_source == SOURCE_LINEMIC) { + if (chip->analog_capture_active == 0) + update_lvl = 1; + chip->analog_capture_active = 1; + } + if (chip->audio_capture_source == SOURCE_MIC || + chip->audio_capture_source == SOURCE_LINEMIC) { + if (chip->mic_active == 0) + update_lvl = 1; + chip->mic_active = 1; + } + if (update_lvl) { + /* capture: update all 3 mutes/unmutes with one call */ + hr222_update_analog_audio_level(chip, 1, 0); + } + } + /* set the source infos (max 3 bits modified) */ + PCXHR_OUTPB(chip->mgr, PCXHR_XLX_CFG, chip->mgr->xlx_cfg); + return 0; +} + + +int hr222_iec958_capture_byte(struct snd_pcxhr *chip, + int aes_idx, unsigned char *aes_bits) +{ + unsigned char idx = (unsigned char)(aes_idx * 8); + unsigned char temp = 0; + unsigned char mask = chip->mgr->board_has_aes1 ? + PCXHR_SUER1_BIT_C_READ_MASK : PCXHR_SUER_BIT_C_READ_MASK; + int i; + for (i = 0; i < 8; i++) { + PCXHR_OUTPB(chip->mgr, PCXHR_XLX_RUER, idx++); /* idx < 192 */ + temp <<= 1; + if (PCXHR_INPB(chip->mgr, PCXHR_XLX_CSUER) & mask) + temp |= 1; + } + snd_printdd("read iec958 AES %d byte %d = 0x%x\n", + chip->chip_idx, aes_idx, temp); + *aes_bits = temp; + return 0; +} + + +int hr222_iec958_update_byte(struct snd_pcxhr *chip, + int aes_idx, unsigned char aes_bits) +{ + int i; + unsigned char new_bits = aes_bits; + unsigned char old_bits = chip->aes_bits[aes_idx]; + unsigned char idx = (unsigned char)(aes_idx * 8); + for (i = 0; i < 8; i++) { + if ((old_bits & 0x01) != (new_bits & 0x01)) { + /* idx < 192 */ + PCXHR_OUTPB(chip->mgr, PCXHR_XLX_RUER, idx); + /* write C and U bit */ + PCXHR_OUTPB(chip->mgr, PCXHR_XLX_CSUER, new_bits&0x01 ? + PCXHR_SUER_BIT_C_WRITE_MASK : 0); + } + idx++; + old_bits >>= 1; + new_bits >>= 1; + } + chip->aes_bits[aes_idx] = aes_bits; + return 0; +} + +static void hr222_micro_boost(struct pcxhr_mgr *mgr, int level) +{ + unsigned char boost_mask; + boost_mask = (unsigned char) (level << PCXHR_SELMIC_PREAMPLI_OFFSET); + if (boost_mask & (~PCXHR_SELMIC_PREAMPLI_MASK)) + return; /* only values form 0 to 3 accepted */ + + mgr->xlx_selmic &= ~PCXHR_SELMIC_PREAMPLI_MASK; + mgr->xlx_selmic |= boost_mask; + + PCXHR_OUTPB(mgr, PCXHR_XLX_SELMIC, mgr->xlx_selmic); + + snd_printdd("hr222_micro_boost : set %x\n", boost_mask); +} + +static void hr222_phantom_power(struct pcxhr_mgr *mgr, int power) +{ + if (power) + mgr->xlx_selmic |= PCXHR_SELMIC_PHANTOM_ALIM; + else + mgr->xlx_selmic &= ~PCXHR_SELMIC_PHANTOM_ALIM; + + PCXHR_OUTPB(mgr, PCXHR_XLX_SELMIC, mgr->xlx_selmic); + + snd_printdd("hr222_phantom_power : set %d\n", power); +} + + +/* mic level */ +static const DECLARE_TLV_DB_SCALE(db_scale_mic_hr222, -9850, 50, 650); + +static int hr222_mic_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = HR222_MICRO_CAPTURE_LEVEL_MIN; /* -98 dB */ + /* gains from 9 dB to 31.5 dB not recommended; use micboost instead */ + uinfo->value.integer.max = HR222_MICRO_CAPTURE_LEVEL_MAX; /* +7 dB */ + return 0; +} + +static int hr222_mic_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + mutex_lock(&chip->mgr->mixer_mutex); + ucontrol->value.integer.value[0] = chip->mic_volume; + mutex_unlock(&chip->mgr->mixer_mutex); + return 0; +} + +static int hr222_mic_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + mutex_lock(&chip->mgr->mixer_mutex); + if (chip->mic_volume != ucontrol->value.integer.value[0]) { + changed = 1; + chip->mic_volume = ucontrol->value.integer.value[0]; + hr222_update_analog_audio_level(chip, 1, 0); + } + mutex_unlock(&chip->mgr->mixer_mutex); + return changed; +} + +static struct snd_kcontrol_new hr222_control_mic_level = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Mic Capture Volume", + .info = hr222_mic_vol_info, + .get = hr222_mic_vol_get, + .put = hr222_mic_vol_put, + .tlv = { .p = db_scale_mic_hr222 }, +}; + + +/* mic boost level */ +static const DECLARE_TLV_DB_SCALE(db_scale_micboost_hr222, 0, 1800, 5400); + +static int hr222_mic_boost_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; /* 0 dB */ + uinfo->value.integer.max = 3; /* 54 dB */ + return 0; +} + +static int hr222_mic_boost_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + mutex_lock(&chip->mgr->mixer_mutex); + ucontrol->value.integer.value[0] = chip->mic_boost; + mutex_unlock(&chip->mgr->mixer_mutex); + return 0; +} + +static int hr222_mic_boost_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + mutex_lock(&chip->mgr->mixer_mutex); + if (chip->mic_boost != ucontrol->value.integer.value[0]) { + changed = 1; + chip->mic_boost = ucontrol->value.integer.value[0]; + hr222_micro_boost(chip->mgr, chip->mic_boost); + } + mutex_unlock(&chip->mgr->mixer_mutex); + return changed; +} + +static struct snd_kcontrol_new hr222_control_mic_boost = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "MicBoost Capture Volume", + .info = hr222_mic_boost_info, + .get = hr222_mic_boost_get, + .put = hr222_mic_boost_put, + .tlv = { .p = db_scale_micboost_hr222 }, +}; + + +/******************* Phantom power switch *******************/ +#define hr222_phantom_power_info snd_ctl_boolean_mono_info + +static int hr222_phantom_power_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + mutex_lock(&chip->mgr->mixer_mutex); + ucontrol->value.integer.value[0] = chip->phantom_power; + mutex_unlock(&chip->mgr->mixer_mutex); + return 0; +} + +static int hr222_phantom_power_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + int power, changed = 0; + + mutex_lock(&chip->mgr->mixer_mutex); + power = !!ucontrol->value.integer.value[0]; + if (chip->phantom_power != power) { + hr222_phantom_power(chip->mgr, power); + chip->phantom_power = power; + changed = 1; + } + mutex_unlock(&chip->mgr->mixer_mutex); + return changed; +} + +static struct snd_kcontrol_new hr222_phantom_power_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Phantom Power Switch", + .info = hr222_phantom_power_info, + .get = hr222_phantom_power_get, + .put = hr222_phantom_power_put, +}; + + +int hr222_add_mic_controls(struct snd_pcxhr *chip) +{ + int err; + if (!chip->mgr->board_has_mic) + return 0; + + /* controls */ + err = snd_ctl_add(chip->card, snd_ctl_new1(&hr222_control_mic_level, + chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, snd_ctl_new1(&hr222_control_mic_boost, + chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, snd_ctl_new1(&hr222_phantom_power_switch, + chip)); + return err; +} diff --git a/sound/pci/pcxhr/pcxhr_mix22.h b/sound/pci/pcxhr/pcxhr_mix22.h new file mode 100644 index 000000000000..6b318b2f0100 --- /dev/null +++ b/sound/pci/pcxhr/pcxhr_mix22.h @@ -0,0 +1,56 @@ +/* + * Driver for Digigram pcxhr compatible soundcards + * + * low level interface with interrupt ans message handling + * + * Copyright (c) 2004 by Digigram <alsa@digigram.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __SOUND_PCXHR_MIX22_H +#define __SOUND_PCXHR_MIX22_H + +struct pcxhr_mgr; + +int hr222_sub_init(struct pcxhr_mgr *mgr); +int hr222_sub_set_clock(struct pcxhr_mgr *mgr, unsigned int rate, + int *changed); +int hr222_get_external_clock(struct pcxhr_mgr *mgr, + enum pcxhr_clock_type clock_type, + int *sample_rate); + +#define HR222_LINE_PLAYBACK_LEVEL_MIN 0 /* -25.5 dB */ +#define HR222_LINE_PLAYBACK_ZERO_LEVEL 51 /* 0.0 dB */ +#define HR222_LINE_PLAYBACK_LEVEL_MAX 99 /* +24.0 dB */ + +#define HR222_LINE_CAPTURE_LEVEL_MIN 0 /* -111.5 dB */ +#define HR222_LINE_CAPTURE_ZERO_LEVEL 223 /* 0.0 dB */ +#define HR222_LINE_CAPTURE_LEVEL_MAX 255 /* +16 dB */ +#define HR222_MICRO_CAPTURE_LEVEL_MIN 0 /* -98.5 dB */ +#define HR222_MICRO_CAPTURE_LEVEL_MAX 210 /* +6.5 dB */ + +int hr222_update_analog_audio_level(struct snd_pcxhr *chip, + int is_capture, + int channel); +int hr222_set_audio_source(struct snd_pcxhr *chip); +int hr222_iec958_capture_byte(struct snd_pcxhr *chip, int aes_idx, + unsigned char *aes_bits); +int hr222_iec958_update_byte(struct snd_pcxhr *chip, int aes_idx, + unsigned char aes_bits); + +int hr222_add_mic_controls(struct snd_pcxhr *chip); + +#endif /* __SOUND_PCXHR_MIX22_H */ diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c index aabc7bc5321e..2436e374586f 100644 --- a/sound/pci/pcxhr/pcxhr_mixer.c +++ b/sound/pci/pcxhr/pcxhr_mixer.c @@ -33,20 +33,24 @@ #include <sound/tlv.h> #include <sound/asoundef.h> #include "pcxhr_mixer.h" +#include "pcxhr_mix22.h" +#define PCXHR_LINE_CAPTURE_LEVEL_MIN 0 /* -112.0 dB */ +#define PCXHR_LINE_CAPTURE_LEVEL_MAX 255 /* +15.5 dB */ +#define PCXHR_LINE_CAPTURE_ZERO_LEVEL 224 /* 0.0 dB ( 0 dBu -> 0 dBFS ) */ -#define PCXHR_ANALOG_CAPTURE_LEVEL_MIN 0 /* -96.0 dB */ -#define PCXHR_ANALOG_CAPTURE_LEVEL_MAX 255 /* +31.5 dB */ -#define PCXHR_ANALOG_CAPTURE_ZERO_LEVEL 224 /* +16.0 dB ( +31.5 dB - fix level +15.5 dB ) */ +#define PCXHR_LINE_PLAYBACK_LEVEL_MIN 0 /* -104.0 dB */ +#define PCXHR_LINE_PLAYBACK_LEVEL_MAX 128 /* +24.0 dB */ +#define PCXHR_LINE_PLAYBACK_ZERO_LEVEL 104 /* 0.0 dB ( 0 dBFS -> 0 dBu ) */ -#define PCXHR_ANALOG_PLAYBACK_LEVEL_MIN 0 /* -128.0 dB */ -#define PCXHR_ANALOG_PLAYBACK_LEVEL_MAX 128 /* 0.0 dB */ -#define PCXHR_ANALOG_PLAYBACK_ZERO_LEVEL 104 /* -24.0 dB ( 0.0 dB - fix level +24.0 dB ) */ - -static const DECLARE_TLV_DB_SCALE(db_scale_analog_capture, -9600, 50, 3150); +static const DECLARE_TLV_DB_SCALE(db_scale_analog_capture, -11200, 50, 1550); static const DECLARE_TLV_DB_SCALE(db_scale_analog_playback, -10400, 100, 2400); -static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, int is_capture, int channel) +static const DECLARE_TLV_DB_SCALE(db_scale_a_hr222_capture, -11150, 50, 1600); +static const DECLARE_TLV_DB_SCALE(db_scale_a_hr222_playback, -2550, 50, 2400); + +static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, + int is_capture, int channel) { int err, vol; struct pcxhr_rmh rmh; @@ -60,15 +64,17 @@ static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, int is_captur if (chip->analog_playback_active[channel]) vol = chip->analog_playback_volume[channel]; else - vol = PCXHR_ANALOG_PLAYBACK_LEVEL_MIN; - rmh.cmd[2] = PCXHR_ANALOG_PLAYBACK_LEVEL_MAX - vol; /* playback analog levels are inversed */ + vol = PCXHR_LINE_PLAYBACK_LEVEL_MIN; + /* playback analog levels are inversed */ + rmh.cmd[2] = PCXHR_LINE_PLAYBACK_LEVEL_MAX - vol; } rmh.cmd[1] = 1 << ((2 * chip->chip_idx) + channel); /* audio mask */ rmh.cmd_len = 3; err = pcxhr_send_msg(chip->mgr, &rmh); if (err < 0) { - snd_printk(KERN_DEBUG "error update_analog_audio_level card(%d) " - "is_capture(%d) err(%x)\n", chip->chip_idx, is_capture, err); + snd_printk(KERN_DEBUG "error update_analog_audio_level card(%d)" + " is_capture(%d) err(%x)\n", + chip->chip_idx, is_capture, err); return -EINVAL; } return 0; @@ -80,14 +86,34 @@ static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, int is_captur static int pcxhr_analog_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; if (kcontrol->private_value == 0) { /* playback */ - uinfo->value.integer.min = PCXHR_ANALOG_PLAYBACK_LEVEL_MIN; /* -128 dB */ - uinfo->value.integer.max = PCXHR_ANALOG_PLAYBACK_LEVEL_MAX; /* 0 dB */ + if (chip->mgr->is_hr_stereo) { + uinfo->value.integer.min = + HR222_LINE_PLAYBACK_LEVEL_MIN; /* -25 dB */ + uinfo->value.integer.max = + HR222_LINE_PLAYBACK_LEVEL_MAX; /* +24 dB */ + } else { + uinfo->value.integer.min = + PCXHR_LINE_PLAYBACK_LEVEL_MIN; /*-104 dB */ + uinfo->value.integer.max = + PCXHR_LINE_PLAYBACK_LEVEL_MAX; /* +24 dB */ + } } else { /* capture */ - uinfo->value.integer.min = PCXHR_ANALOG_CAPTURE_LEVEL_MIN; /* -96 dB */ - uinfo->value.integer.max = PCXHR_ANALOG_CAPTURE_LEVEL_MAX; /* 31.5 dB */ + if (chip->mgr->is_hr_stereo) { + uinfo->value.integer.min = + HR222_LINE_CAPTURE_LEVEL_MIN; /*-112 dB */ + uinfo->value.integer.max = + HR222_LINE_CAPTURE_LEVEL_MAX; /* +15.5 dB */ + } else { + uinfo->value.integer.min = + PCXHR_LINE_CAPTURE_LEVEL_MIN; /*-112 dB */ + uinfo->value.integer.max = + PCXHR_LINE_CAPTURE_LEVEL_MAX; /* +15.5 dB */ + } } return 0; } @@ -98,11 +124,11 @@ static int pcxhr_analog_vol_get(struct snd_kcontrol *kcontrol, struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); mutex_lock(&chip->mgr->mixer_mutex); if (kcontrol->private_value == 0) { /* playback */ - ucontrol->value.integer.value[0] = chip->analog_playback_volume[0]; - ucontrol->value.integer.value[1] = chip->analog_playback_volume[1]; + ucontrol->value.integer.value[0] = chip->analog_playback_volume[0]; + ucontrol->value.integer.value[1] = chip->analog_playback_volume[1]; } else { /* capture */ - ucontrol->value.integer.value[0] = chip->analog_capture_volume[0]; - ucontrol->value.integer.value[1] = chip->analog_capture_volume[1]; + ucontrol->value.integer.value[0] = chip->analog_capture_volume[0]; + ucontrol->value.integer.value[1] = chip->analog_capture_volume[1]; } mutex_unlock(&chip->mgr->mixer_mutex); return 0; @@ -123,18 +149,35 @@ static int pcxhr_analog_vol_put(struct snd_kcontrol *kcontrol, &chip->analog_capture_volume[i] : &chip->analog_playback_volume[i]; if (is_capture) { - if (new_volume < PCXHR_ANALOG_CAPTURE_LEVEL_MIN || - new_volume > PCXHR_ANALOG_CAPTURE_LEVEL_MAX) - continue; + if (chip->mgr->is_hr_stereo) { + if (new_volume < HR222_LINE_CAPTURE_LEVEL_MIN || + new_volume > HR222_LINE_CAPTURE_LEVEL_MAX) + continue; + } else { + if (new_volume < PCXHR_LINE_CAPTURE_LEVEL_MIN || + new_volume > PCXHR_LINE_CAPTURE_LEVEL_MAX) + continue; + } } else { - if (new_volume < PCXHR_ANALOG_PLAYBACK_LEVEL_MIN || - new_volume > PCXHR_ANALOG_PLAYBACK_LEVEL_MAX) - continue; + if (chip->mgr->is_hr_stereo) { + if (new_volume < HR222_LINE_PLAYBACK_LEVEL_MIN || + new_volume > HR222_LINE_PLAYBACK_LEVEL_MAX) + continue; + } else { + if (new_volume < PCXHR_LINE_PLAYBACK_LEVEL_MIN || + new_volume > PCXHR_LINE_PLAYBACK_LEVEL_MAX) + continue; + } } if (*stored_volume != new_volume) { *stored_volume = new_volume; changed = 1; - pcxhr_update_analog_audio_level(chip, is_capture, i); + if (chip->mgr->is_hr_stereo) + hr222_update_analog_audio_level(chip, + is_capture, i); + else + pcxhr_update_analog_audio_level(chip, + is_capture, i); } } mutex_unlock(&chip->mgr->mixer_mutex); @@ -153,6 +196,7 @@ static struct snd_kcontrol_new pcxhr_control_analog_level = { }; /* shared */ + #define pcxhr_sw_info snd_ctl_boolean_stereo_info static int pcxhr_audio_sw_get(struct snd_kcontrol *kcontrol, @@ -180,7 +224,10 @@ static int pcxhr_audio_sw_put(struct snd_kcontrol *kcontrol, !!ucontrol->value.integer.value[i]; changed = 1; /* update playback levels */ - pcxhr_update_analog_audio_level(chip, 0, i); + if (chip->mgr->is_hr_stereo) + hr222_update_analog_audio_level(chip, 0, i); + else + pcxhr_update_analog_audio_level(chip, 0, i); } } mutex_unlock(&chip->mgr->mixer_mutex); @@ -251,7 +298,8 @@ static int pcxhr_update_playback_stream_level(struct snd_pcxhr* chip, int idx) #define VALID_AUDIO_IO_MUTE_LEVEL 0x000004 #define VALID_AUDIO_IO_MUTE_MONITOR_1 0x000008 -static int pcxhr_update_audio_pipe_level(struct snd_pcxhr* chip, int capture, int channel) +static int pcxhr_update_audio_pipe_level(struct snd_pcxhr *chip, + int capture, int channel) { int err; struct pcxhr_rmh rmh; @@ -264,18 +312,20 @@ static int pcxhr_update_audio_pipe_level(struct snd_pcxhr* chip, int capture, in pcxhr_init_rmh(&rmh, CMD_AUDIO_LEVEL_ADJUST); /* add channel mask */ - pcxhr_set_pipe_cmd_params(&rmh, capture, 0, 0, 1 << (channel + pipe->first_audio)); - /* TODO : if mask (3 << pipe->first_audio) is used, left and right channel - * will be programmed to the same params - */ + pcxhr_set_pipe_cmd_params(&rmh, capture, 0, 0, + 1 << (channel + pipe->first_audio)); + /* TODO : if mask (3 << pipe->first_audio) is used, left and right + * channel will be programmed to the same params */ if (capture) { rmh.cmd[0] |= VALID_AUDIO_IO_DIGITAL_LEVEL; - /* VALID_AUDIO_IO_MUTE_LEVEL not yet handled (capture pipe level) */ + /* VALID_AUDIO_IO_MUTE_LEVEL not yet handled + * (capture pipe level) */ rmh.cmd[2] = chip->digital_capture_volume[channel]; } else { - rmh.cmd[0] |= VALID_AUDIO_IO_MONITOR_LEVEL | VALID_AUDIO_IO_MUTE_MONITOR_1; - /* VALID_AUDIO_IO_DIGITAL_LEVEL and VALID_AUDIO_IO_MUTE_LEVEL not yet - * handled (playback pipe level) + rmh.cmd[0] |= VALID_AUDIO_IO_MONITOR_LEVEL | + VALID_AUDIO_IO_MUTE_MONITOR_1; + /* VALID_AUDIO_IO_DIGITAL_LEVEL and VALID_AUDIO_IO_MUTE_LEVEL + * not yet handled (playback pipe level) */ rmh.cmd[2] = chip->monitoring_volume[channel] << 10; if (chip->monitoring_active[channel] == 0) @@ -284,8 +334,8 @@ static int pcxhr_update_audio_pipe_level(struct snd_pcxhr* chip, int capture, in rmh.cmd_len = 3; err = pcxhr_send_msg(chip->mgr, &rmh); - if(err<0) { - snd_printk(KERN_DEBUG "error update_audio_level card(%d) err(%x)\n", + if (err < 0) { + snd_printk(KERN_DEBUG "error update_audio_level(%d) err=%x\n", chip->chip_idx, err); return -EINVAL; } @@ -309,15 +359,15 @@ static int pcxhr_pcm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); - int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */ + int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */ int *stored_volume; int is_capture = kcontrol->private_value; mutex_lock(&chip->mgr->mixer_mutex); - if (is_capture) - stored_volume = chip->digital_capture_volume; /* digital capture */ - else - stored_volume = chip->digital_playback_volume[idx]; /* digital playback */ + if (is_capture) /* digital capture */ + stored_volume = chip->digital_capture_volume; + else /* digital playback */ + stored_volume = chip->digital_playback_volume[idx]; ucontrol->value.integer.value[0] = stored_volume[0]; ucontrol->value.integer.value[1] = stored_volume[1]; mutex_unlock(&chip->mgr->mixer_mutex); @@ -328,7 +378,7 @@ static int pcxhr_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); - int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */ + int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */ int changed = 0; int is_capture = kcontrol->private_value; int *stored_volume; @@ -384,7 +434,8 @@ static int pcxhr_pcm_sw_get(struct snd_kcontrol *kcontrol, return 0; } -static int pcxhr_pcm_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int pcxhr_pcm_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); int changed = 0; @@ -444,8 +495,8 @@ static int pcxhr_monitor_vol_put(struct snd_kcontrol *kcontrol, if (chip->monitoring_volume[i] != ucontrol->value.integer.value[i]) { chip->monitoring_volume[i] = - !!ucontrol->value.integer.value[i]; - if(chip->monitoring_active[i]) + ucontrol->value.integer.value[i]; + if (chip->monitoring_active[i]) /* update monitoring volume and mute */ /* do only when monitoring is unmuted */ pcxhr_update_audio_pipe_level(chip, 0, i); @@ -460,7 +511,7 @@ static struct snd_kcontrol_new pcxhr_control_monitor_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Monitoring Volume", + .name = "Monitoring Playback Volume", .info = pcxhr_digital_vol_info, /* shared */ .get = pcxhr_monitor_vol_get, .put = pcxhr_monitor_vol_put, @@ -511,7 +562,7 @@ static int pcxhr_monitor_sw_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new pcxhr_control_monitor_sw = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Monitoring Switch", + .name = "Monitoring Playback Switch", .info = pcxhr_sw_info, /* shared */ .get = pcxhr_monitor_sw_get, .put = pcxhr_monitor_sw_put @@ -533,7 +584,7 @@ static int pcxhr_set_audio_source(struct snd_pcxhr* chip) struct pcxhr_rmh rmh; unsigned int mask, reg; unsigned int codec; - int err, use_src, changed; + int err, changed; switch (chip->chip_idx) { case 0 : mask = PCXHR_SOURCE_AUDIO01_UER; codec = CS8420_01_CS; break; @@ -542,13 +593,10 @@ static int pcxhr_set_audio_source(struct snd_pcxhr* chip) case 3 : mask = PCXHR_SOURCE_AUDIO67_UER; codec = CS8420_67_CS; break; default: return -EINVAL; } - reg = 0; /* audio source from analog plug */ - use_src = 0; /* do not activate codec SRC */ - if (chip->audio_capture_source != 0) { reg = mask; /* audio source from digital plug */ - if (chip->audio_capture_source == 2) - use_src = 1; + } else { + reg = 0; /* audio source from analog plug */ } /* set the input source */ pcxhr_write_io_num_reg_cont(chip->mgr, mask, reg, &changed); @@ -560,29 +608,61 @@ static int pcxhr_set_audio_source(struct snd_pcxhr* chip) if (err) return err; } - pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* set codec SRC on off */ - rmh.cmd_len = 3; - rmh.cmd[0] |= IO_NUM_UER_CHIP_REG; - rmh.cmd[1] = codec; - rmh.cmd[2] = (CS8420_DATA_FLOW_CTL & CHIP_SIG_AND_MAP_SPI) | (use_src ? 0x41 : 0x54); - err = pcxhr_send_msg(chip->mgr, &rmh); - if(err) - return err; - rmh.cmd[2] = (CS8420_CLOCK_SRC_CTL & CHIP_SIG_AND_MAP_SPI) | (use_src ? 0x41 : 0x49); - err = pcxhr_send_msg(chip->mgr, &rmh); + if (chip->mgr->board_aes_in_192k) { + int i; + unsigned int src_config = 0xC0; + /* update all src configs with one call */ + for (i = 0; (i < 4) && (i < chip->mgr->capture_chips); i++) { + if (chip->mgr->chip[i]->audio_capture_source == 2) + src_config |= (1 << (3 - i)); + } + /* set codec SRC on off */ + pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); + rmh.cmd_len = 2; + rmh.cmd[0] |= IO_NUM_REG_CONFIG_SRC; + rmh.cmd[1] = src_config; + err = pcxhr_send_msg(chip->mgr, &rmh); + } else { + int use_src = 0; + if (chip->audio_capture_source == 2) + use_src = 1; + /* set codec SRC on off */ + pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); + rmh.cmd_len = 3; + rmh.cmd[0] |= IO_NUM_UER_CHIP_REG; + rmh.cmd[1] = codec; + rmh.cmd[2] = ((CS8420_DATA_FLOW_CTL & CHIP_SIG_AND_MAP_SPI) | + (use_src ? 0x41 : 0x54)); + err = pcxhr_send_msg(chip->mgr, &rmh); + if (err) + return err; + rmh.cmd[2] = ((CS8420_CLOCK_SRC_CTL & CHIP_SIG_AND_MAP_SPI) | + (use_src ? 0x41 : 0x49)); + err = pcxhr_send_msg(chip->mgr, &rmh); + } return err; } static int pcxhr_audio_src_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = {"Analog", "Digital", "Digi+SRC"}; + static const char *texts[5] = { + "Line", "Digital", "Digi+SRC", "Mic", "Line+Mic" + }; + int i; + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + i = 2; /* no SRC, no Mic available */ + if (chip->mgr->board_has_aes1) { + i = 3; /* SRC available */ + if (chip->mgr->board_has_mic) + i = 5; /* Mic and MicroMix available */ + } uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) - uinfo->value.enumerated.item = 2; + uinfo->value.enumerated.items = i; + if (uinfo->value.enumerated.item > (i-1)) + uinfo->value.enumerated.item = i-1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); return 0; @@ -601,13 +681,21 @@ static int pcxhr_audio_src_put(struct snd_kcontrol *kcontrol, { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); int ret = 0; - - if (ucontrol->value.enumerated.item[0] >= 3) + int i = 2; /* no SRC, no Mic available */ + if (chip->mgr->board_has_aes1) { + i = 3; /* SRC available */ + if (chip->mgr->board_has_mic) + i = 5; /* Mic and MicroMix available */ + } + if (ucontrol->value.enumerated.item[0] >= i) return -EINVAL; mutex_lock(&chip->mgr->mixer_mutex); if (chip->audio_capture_source != ucontrol->value.enumerated.item[0]) { chip->audio_capture_source = ucontrol->value.enumerated.item[0]; - pcxhr_set_audio_source(chip); + if (chip->mgr->is_hr_stereo) + hr222_set_audio_source(chip); + else + pcxhr_set_audio_source(chip); ret = 1; } mutex_unlock(&chip->mgr->mixer_mutex); @@ -626,25 +714,46 @@ static struct snd_kcontrol_new pcxhr_control_audio_src = { /* * clock type selection * enum pcxhr_clock_type { - * PCXHR_CLOCK_TYPE_INTERNAL = 0, - * PCXHR_CLOCK_TYPE_WORD_CLOCK, - * PCXHR_CLOCK_TYPE_AES_SYNC, - * PCXHR_CLOCK_TYPE_AES_1, - * PCXHR_CLOCK_TYPE_AES_2, - * PCXHR_CLOCK_TYPE_AES_3, - * PCXHR_CLOCK_TYPE_AES_4, - * }; + * PCXHR_CLOCK_TYPE_INTERNAL = 0, + * PCXHR_CLOCK_TYPE_WORD_CLOCK, + * PCXHR_CLOCK_TYPE_AES_SYNC, + * PCXHR_CLOCK_TYPE_AES_1, + * PCXHR_CLOCK_TYPE_AES_2, + * PCXHR_CLOCK_TYPE_AES_3, + * PCXHR_CLOCK_TYPE_AES_4, + * PCXHR_CLOCK_TYPE_MAX = PCXHR_CLOCK_TYPE_AES_4, + * HR22_CLOCK_TYPE_INTERNAL = PCXHR_CLOCK_TYPE_INTERNAL, + * HR22_CLOCK_TYPE_AES_SYNC, + * HR22_CLOCK_TYPE_AES_1, + * HR22_CLOCK_TYPE_MAX = HR22_CLOCK_TYPE_AES_1, + * }; */ static int pcxhr_clock_type_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[7] = { - "Internal", "WordClock", "AES Sync", "AES 1", "AES 2", "AES 3", "AES 4" + static const char *textsPCXHR[7] = { + "Internal", "WordClock", "AES Sync", + "AES 1", "AES 2", "AES 3", "AES 4" + }; + static const char *textsHR22[3] = { + "Internal", "AES Sync", "AES 1" }; + const char **texts; struct pcxhr_mgr *mgr = snd_kcontrol_chip(kcontrol); - int clock_items = 3 + mgr->capture_chips; - + int clock_items = 2; /* at least Internal and AES Sync clock */ + if (mgr->board_has_aes1) { + clock_items += mgr->capture_chips; /* add AES x */ + if (!mgr->is_hr_stereo) + clock_items += 1; /* add word clock */ + } + if (mgr->is_hr_stereo) { + texts = textsHR22; + snd_BUG_ON(clock_items > (HR22_CLOCK_TYPE_MAX+1)); + } else { + texts = textsPCXHR; + snd_BUG_ON(clock_items > (PCXHR_CLOCK_TYPE_MAX+1)); + } uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = clock_items; @@ -667,9 +776,13 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct pcxhr_mgr *mgr = snd_kcontrol_chip(kcontrol); - unsigned int clock_items = 3 + mgr->capture_chips; int rate, ret = 0; - + unsigned int clock_items = 2; /* at least Internal and AES Sync clock */ + if (mgr->board_has_aes1) { + clock_items += mgr->capture_chips; /* add AES x */ + if (!mgr->is_hr_stereo) + clock_items += 1; /* add word clock */ + } if (ucontrol->value.enumerated.item[0] >= clock_items) return -EINVAL; mutex_lock(&mgr->mixer_mutex); @@ -677,7 +790,8 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol, mutex_lock(&mgr->setup_mutex); mgr->use_clock_type = ucontrol->value.enumerated.item[0]; if (mgr->use_clock_type) - pcxhr_get_external_clock(mgr, mgr->use_clock_type, &rate); + pcxhr_get_external_clock(mgr, mgr->use_clock_type, + &rate); else rate = mgr->sample_rate; if (rate) { @@ -686,7 +800,7 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol, mgr->sample_rate = rate; } mutex_unlock(&mgr->setup_mutex); - ret = 1; /* return 1 even if the set was not done. ok ? */ + ret = 1; /* return 1 even if the set was not done. ok ? */ } mutex_unlock(&mgr->mixer_mutex); return ret; @@ -747,14 +861,16 @@ static struct snd_kcontrol_new pcxhr_control_clock_rate = { /* * IEC958 status bits */ -static int pcxhr_iec958_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int pcxhr_iec958_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; uinfo->count = 1; return 0; } -static int pcxhr_iec958_capture_byte(struct snd_pcxhr *chip, int aes_idx, unsigned char* aes_bits) +static int pcxhr_iec958_capture_byte(struct snd_pcxhr *chip, + int aes_idx, unsigned char *aes_bits) { int i, err; unsigned char temp; @@ -763,39 +879,61 @@ static int pcxhr_iec958_capture_byte(struct snd_pcxhr *chip, int aes_idx, unsign pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); rmh.cmd[0] |= IO_NUM_UER_CHIP_REG; switch (chip->chip_idx) { - case 0: rmh.cmd[1] = CS8420_01_CS; break; /* use CS8416_01_CS for AES SYNC plug */ + /* instead of CS8420_01_CS use CS8416_01_CS for AES SYNC plug */ + case 0: rmh.cmd[1] = CS8420_01_CS; break; case 1: rmh.cmd[1] = CS8420_23_CS; break; case 2: rmh.cmd[1] = CS8420_45_CS; break; case 3: rmh.cmd[1] = CS8420_67_CS; break; default: return -EINVAL; } - switch (aes_idx) { - case 0: rmh.cmd[2] = CS8420_CSB0; break; /* use CS8416_CSBx for AES SYNC plug */ - case 1: rmh.cmd[2] = CS8420_CSB1; break; - case 2: rmh.cmd[2] = CS8420_CSB2; break; - case 3: rmh.cmd[2] = CS8420_CSB3; break; - case 4: rmh.cmd[2] = CS8420_CSB4; break; - default: return -EINVAL; + if (chip->mgr->board_aes_in_192k) { + switch (aes_idx) { + case 0: rmh.cmd[2] = CS8416_CSB0; break; + case 1: rmh.cmd[2] = CS8416_CSB1; break; + case 2: rmh.cmd[2] = CS8416_CSB2; break; + case 3: rmh.cmd[2] = CS8416_CSB3; break; + case 4: rmh.cmd[2] = CS8416_CSB4; break; + default: return -EINVAL; + } + } else { + switch (aes_idx) { + /* instead of CS8420_CSB0 use CS8416_CSBx for AES SYNC plug */ + case 0: rmh.cmd[2] = CS8420_CSB0; break; + case 1: rmh.cmd[2] = CS8420_CSB1; break; + case 2: rmh.cmd[2] = CS8420_CSB2; break; + case 3: rmh.cmd[2] = CS8420_CSB3; break; + case 4: rmh.cmd[2] = CS8420_CSB4; break; + default: return -EINVAL; + } } - rmh.cmd[1] &= 0x0fffff; /* size and code the chip id for the fpga */ - rmh.cmd[2] &= CHIP_SIG_AND_MAP_SPI; /* chip signature + map for spi read */ + /* size and code the chip id for the fpga */ + rmh.cmd[1] &= 0x0fffff; + /* chip signature + map for spi read */ + rmh.cmd[2] &= CHIP_SIG_AND_MAP_SPI; rmh.cmd_len = 3; err = pcxhr_send_msg(chip->mgr, &rmh); if (err) return err; - temp = 0; - for (i = 0; i < 8; i++) { - /* attention : reversed bit order (not with CS8416_01_CS) */ - temp <<= 1; - if (rmh.stat[1] & (1 << i)) - temp |= 1; + + if (chip->mgr->board_aes_in_192k) { + temp = (unsigned char)rmh.stat[1]; + } else { + temp = 0; + /* reversed bit order (not with CS8416_01_CS) */ + for (i = 0; i < 8; i++) { + temp <<= 1; + if (rmh.stat[1] & (1 << i)) + temp |= 1; + } } - snd_printdd("read iec958 AES %d byte %d = 0x%x\n", chip->chip_idx, aes_idx, temp); + snd_printdd("read iec958 AES %d byte %d = 0x%x\n", + chip->chip_idx, aes_idx, temp); *aes_bits = temp; return 0; } -static int pcxhr_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int pcxhr_iec958_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); unsigned char aes_bits; @@ -806,7 +944,12 @@ static int pcxhr_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v if (kcontrol->private_value == 0) /* playback */ aes_bits = chip->aes_bits[i]; else { /* capture */ - err = pcxhr_iec958_capture_byte(chip, i, &aes_bits); + if (chip->mgr->is_hr_stereo) + err = hr222_iec958_capture_byte(chip, i, + &aes_bits); + else + err = pcxhr_iec958_capture_byte(chip, i, + &aes_bits); if (err) break; } @@ -825,7 +968,8 @@ static int pcxhr_iec958_mask_get(struct snd_kcontrol *kcontrol, return 0; } -static int pcxhr_iec958_update_byte(struct snd_pcxhr *chip, int aes_idx, unsigned char aes_bits) +static int pcxhr_iec958_update_byte(struct snd_pcxhr *chip, + int aes_idx, unsigned char aes_bits) { int i, err, cmd; unsigned char new_bits = aes_bits; @@ -834,12 +978,12 @@ static int pcxhr_iec958_update_byte(struct snd_pcxhr *chip, int aes_idx, unsigne for (i = 0; i < 8; i++) { if ((old_bits & 0x01) != (new_bits & 0x01)) { - cmd = chip->chip_idx & 0x03; /* chip index 0..3 */ - if(chip->chip_idx > 3) + cmd = chip->chip_idx & 0x03; /* chip index 0..3 */ + if (chip->chip_idx > 3) /* new bit used if chip_idx>3 (PCX1222HR) */ cmd |= 1 << 22; - cmd |= ((aes_idx << 3) + i) << 2; /* add bit offset */ - cmd |= (new_bits & 0x01) << 23; /* add bit value */ + cmd |= ((aes_idx << 3) + i) << 2; /* add bit offset */ + cmd |= (new_bits & 0x01) << 23; /* add bit value */ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); rmh.cmd[0] |= IO_NUM_REG_CUER; rmh.cmd[1] = cmd; @@ -867,7 +1011,12 @@ static int pcxhr_iec958_put(struct snd_kcontrol *kcontrol, mutex_lock(&chip->mgr->mixer_mutex); for (i = 0; i < 5; i++) { if (ucontrol->value.iec958.status[i] != chip->aes_bits[i]) { - pcxhr_iec958_update_byte(chip, i, ucontrol->value.iec958.status[i]); + if (chip->mgr->is_hr_stereo) + hr222_iec958_update_byte(chip, i, + ucontrol->value.iec958.status[i]); + else + pcxhr_iec958_update_byte(chip, i, + ucontrol->value.iec958.status[i]); changed = 1; } } @@ -917,29 +1066,53 @@ static void pcxhr_init_audio_levels(struct snd_pcxhr *chip) /* at boot time the digital volumes are unmuted 0dB */ for (j = 0; j < PCXHR_PLAYBACK_STREAMS; j++) { chip->digital_playback_active[j][i] = 1; - chip->digital_playback_volume[j][i] = PCXHR_DIGITAL_ZERO_LEVEL; + chip->digital_playback_volume[j][i] = + PCXHR_DIGITAL_ZERO_LEVEL; } - /* after boot, only two bits are set on the uer interface */ - chip->aes_bits[0] = IEC958_AES0_PROFESSIONAL | IEC958_AES0_PRO_FS_48000; -/* only for test purpose, remove later */ + /* after boot, only two bits are set on the uer + * interface + */ + chip->aes_bits[0] = (IEC958_AES0_PROFESSIONAL | + IEC958_AES0_PRO_FS_48000); #ifdef CONFIG_SND_DEBUG - /* analog volumes for playback (is LEVEL_MIN after boot) */ + /* analog volumes for playback + * (is LEVEL_MIN after boot) + */ chip->analog_playback_active[i] = 1; - chip->analog_playback_volume[i] = PCXHR_ANALOG_PLAYBACK_ZERO_LEVEL; - pcxhr_update_analog_audio_level(chip, 0, i); + if (chip->mgr->is_hr_stereo) + chip->analog_playback_volume[i] = + HR222_LINE_PLAYBACK_ZERO_LEVEL; + else { + chip->analog_playback_volume[i] = + PCXHR_LINE_PLAYBACK_ZERO_LEVEL; + pcxhr_update_analog_audio_level(chip, 0, i); + } #endif -/* test end */ + /* stereo cards need to be initialised after boot */ + if (chip->mgr->is_hr_stereo) + hr222_update_analog_audio_level(chip, 0, i); } if (chip->nb_streams_capt) { /* at boot time the digital volumes are unmuted 0dB */ - chip->digital_capture_volume[i] = PCXHR_DIGITAL_ZERO_LEVEL; -/* only for test purpose, remove later */ + chip->digital_capture_volume[i] = + PCXHR_DIGITAL_ZERO_LEVEL; + chip->analog_capture_active = 1; #ifdef CONFIG_SND_DEBUG - /* analog volumes for playback (is LEVEL_MIN after boot) */ - chip->analog_capture_volume[i] = PCXHR_ANALOG_CAPTURE_ZERO_LEVEL; - pcxhr_update_analog_audio_level(chip, 1, i); + /* analog volumes for playback + * (is LEVEL_MIN after boot) + */ + if (chip->mgr->is_hr_stereo) + chip->analog_capture_volume[i] = + HR222_LINE_CAPTURE_ZERO_LEVEL; + else { + chip->analog_capture_volume[i] = + PCXHR_LINE_CAPTURE_ZERO_LEVEL; + pcxhr_update_analog_audio_level(chip, 1, i); + } #endif -/* test end */ + /* stereo cards need to be initialised after boot */ + if (chip->mgr->is_hr_stereo) + hr222_update_analog_audio_level(chip, 1, i); } } @@ -963,90 +1136,125 @@ int pcxhr_create_mixer(struct pcxhr_mgr *mgr) temp = pcxhr_control_analog_level; temp.name = "Master Playback Volume"; temp.private_value = 0; /* playback */ - temp.tlv.p = db_scale_analog_playback; - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0) + if (mgr->is_hr_stereo) + temp.tlv.p = db_scale_a_hr222_playback; + else + temp.tlv.p = db_scale_analog_playback; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&temp, chip)); + if (err < 0) return err; + /* output mute controls */ - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_output_switch, - chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_output_switch, + chip)); + if (err < 0) return err; - + temp = snd_pcxhr_pcm_vol; temp.name = "PCM Playback Volume"; temp.count = PCXHR_PLAYBACK_STREAMS; temp.private_value = 0; /* playback */ - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&temp, chip)); + if (err < 0) return err; - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_pcm_switch, - chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_pcm_switch, chip)); + if (err < 0) return err; /* IEC958 controls */ - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_playback_iec958_mask, - chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_playback_iec958_mask, + chip)); + if (err < 0) return err; - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_playback_iec958, - chip))) < 0) + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_playback_iec958, + chip)); + if (err < 0) return err; } if (chip->nb_streams_capt) { - /* analog input level control only on first two chips !*/ + /* analog input level control */ temp = pcxhr_control_analog_level; - temp.name = "Master Capture Volume"; + temp.name = "Line Capture Volume"; temp.private_value = 1; /* capture */ - temp.tlv.p = db_scale_analog_capture; - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0) + if (mgr->is_hr_stereo) + temp.tlv.p = db_scale_a_hr222_capture; + else + temp.tlv.p = db_scale_analog_capture; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&temp, chip)); + if (err < 0) return err; temp = snd_pcxhr_pcm_vol; temp.name = "PCM Capture Volume"; temp.count = 1; temp.private_value = 1; /* capture */ - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0) + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&temp, chip)); + if (err < 0) return err; + /* Audio source */ - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_audio_src, - chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_audio_src, chip)); + if (err < 0) return err; + /* IEC958 controls */ - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_capture_iec958_mask, - chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_capture_iec958_mask, + chip)); + if (err < 0) return err; - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_capture_iec958, - chip))) < 0) + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_capture_iec958, + chip)); + if (err < 0) return err; + + if (mgr->is_hr_stereo) { + err = hr222_add_mic_controls(chip); + if (err < 0) + return err; + } } /* monitoring only if playback and capture device available */ if (chip->nb_streams_capt > 0 && chip->nb_streams_play > 0) { /* monitoring */ - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_monitor_vol, - chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_monitor_vol, chip)); + if (err < 0) return err; - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_monitor_sw, - chip))) < 0) + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_monitor_sw, chip)); + if (err < 0) return err; } if (i == 0) { /* clock mode only one control per pcxhr */ - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_clock_type, - mgr))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_clock_type, mgr)); + if (err < 0) return err; - /* non standard control used to scan the external clock presence/frequencies */ - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_clock_rate, - mgr))) < 0) + /* non standard control used to scan + * the external clock presence/frequencies + */ + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_clock_rate, mgr)); + if (err < 0) return err; } diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index e9f0706ed3e4..3caacfb9d8e0 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -172,7 +172,7 @@ MODULE_PARM_DESC(opl3_port, "OPL3 port # for Riptide driver."); #define MAX_WRITE_RETRY 10 /* cmd interface limits */ #define MAX_ERROR_COUNT 10 -#define CMDIF_TIMEOUT 500000 +#define CMDIF_TIMEOUT 50000 #define RESET_TRIES 5 #define READ_PORT_ULONG(p) inl((unsigned long)&(p)) @@ -1754,7 +1754,7 @@ snd_riptide_interrupt(int irq, void *dev_id) if (IS_EOBIRQ(cif->hwport) || IS_EOSIRQ(cif->hwport) || IS_EOCIRQ(cif->hwport)) { chip->handled_irqs++; - tasklet_hi_schedule(&chip->riptide_tq); + tasklet_schedule(&chip->riptide_tq); } if (chip->rmidi && IS_MPUIRQ(cif->hwport)) { chip->handled_irqs++; diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 736246f98acc..44d0c15e2b71 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -1452,7 +1452,7 @@ static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int i if (snd_rawmidi_new (card, buf, id, 1, 1, &hdsp->midi[id].rmidi) < 0) return -1; - sprintf (hdsp->midi[id].rmidi->name, "%s MIDI %d", card->id, id+1); + sprintf(hdsp->midi[id].rmidi->name, "HDSP MIDI %d", id+1); hdsp->midi[id].rmidi->private_data = &hdsp->midi[id]; snd_rawmidi_set_ops (hdsp->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_hdsp_midi_output); @@ -3750,7 +3750,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) } } if (hdsp->use_midi_tasklet && schedule) - tasklet_hi_schedule(&hdsp->midi_tasklet); + tasklet_schedule(&hdsp->midi_tasklet); return IRQ_HANDLED; } diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 98762f909d64..71231cf1b2b0 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1293,7 +1293,7 @@ static int __devinit snd_hdspm_create_midi (struct snd_card *card, if (err < 0) return err; - sprintf (hdspm->midi[id].rmidi->name, "%s MIDI %d", card->id, id+1); + sprintf(hdspm->midi[id].rmidi->name, "HDSPM MIDI %d", id+1); hdspm->midi[id].rmidi->private_data = &hdspm->midi[id]; snd_rawmidi_set_ops(hdspm->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, @@ -3476,7 +3476,7 @@ static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id) schedule = 1; } if (schedule) - tasklet_hi_schedule(&hdspm->midi_tasklet); + tasklet_schedule(&hdspm->midi_tasklet); return IRQ_HANDLED; } diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c index fa4b11398b1f..ea903c8e90dd 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c @@ -41,7 +41,7 @@ irqreturn_t pdacf_interrupt(int irq, void *dev) if (stat & PDAUDIOCF_IRQOVR) /* should never happen */ snd_printk(KERN_ERR "PDAUDIOCF SRAM buffer overrun detected!\n"); if (chip->pcm_substream) - tasklet_hi_schedule(&chip->tq); + tasklet_schedule(&chip->tq); if (!(stat & PDAUDIOCF_IRQAKM)) stat |= PDAUDIOCF_IRQAKM; /* check rate */ } diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index a38c0c790d2b..af76ee862d27 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -1033,7 +1033,7 @@ static int __init snd_pmac_detect(struct snd_pmac *chip) } if (of_device_is_compatible(sound, "tumbler")) { chip->model = PMAC_TUMBLER; - chip->can_capture = 0; /* no capture */ + chip->can_capture = machine_is_compatible("PowerMac4,2"); chip->can_duplex = 0; // chip->can_byte_swap = 0; /* FIXME: check this */ chip->num_freqs = ARRAY_SIZE(tumbler_freqs); diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index f746e15b8481..3eb223385416 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -875,7 +875,8 @@ static struct snd_kcontrol_new snapper_mixers[] __initdata = { .put = tumbler_put_master_switch }, DEFINE_SNAPPER_MIX("PCM Playback Volume", 0, VOL_IDX_PCM), - DEFINE_SNAPPER_MIX("PCM Playback Volume", 1, VOL_IDX_PCM2), + /* Alternative PCM is assigned to Mic analog loopback on iBook G4 */ + DEFINE_SNAPPER_MIX("Mic Playback Volume", 0, VOL_IDX_PCM2), DEFINE_SNAPPER_MIX("Monitor Mix Volume", 0, VOL_IDX_ADC), DEFINE_SNAPPER_MONO("Tone Control - Bass", bass), DEFINE_SNAPPER_MONO("Tone Control - Treble", treble), diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4dfda6674bec..ef025c66cc66 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -22,17 +22,16 @@ if SND_SOC config SND_SOC_AC97_BUS bool -# All the supported Soc's -source "sound/soc/at32/Kconfig" -source "sound/soc/at91/Kconfig" +# All the supported SoCs +source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" +source "sound/soc/blackfin/Kconfig" +source "sound/soc/davinci/Kconfig" +source "sound/soc/fsl/Kconfig" +source "sound/soc/omap/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" -source "sound/soc/fsl/Kconfig" -source "sound/soc/davinci/Kconfig" -source "sound/soc/omap/Kconfig" -source "sound/soc/blackfin/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d849349f2c66..86a9b1f5b0f3 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,13 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ -obj-$(CONFIG_SND_SOC) += omap/ au1x/ blackfin/ +obj-$(CONFIG_SND_SOC) += codecs/ +obj-$(CONFIG_SND_SOC) += atmel/ +obj-$(CONFIG_SND_SOC) += au1x/ +obj-$(CONFIG_SND_SOC) += blackfin/ +obj-$(CONFIG_SND_SOC) += davinci/ +obj-$(CONFIG_SND_SOC) += fsl/ +obj-$(CONFIG_SND_SOC) += omap/ +obj-$(CONFIG_SND_SOC) += pxa/ +obj-$(CONFIG_SND_SOC) += s3c24xx/ +obj-$(CONFIG_SND_SOC) += sh/ diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig deleted file mode 100644 index b0765e86c085..000000000000 --- a/sound/soc/at32/Kconfig +++ /dev/null @@ -1,34 +0,0 @@ -config SND_AT32_SOC - tristate "SoC Audio for the Atmel AT32 System-on-a-Chip" - depends on AVR32 && SND_SOC - help - Say Y or M if you want to add support for codecs attached to - the AT32 SSC interface. You will also need to - to select the audio interfaces to support below. - - -config SND_AT32_SOC_SSC - tristate - - - -config SND_AT32_SOC_PLAYPAQ - tristate "SoC Audio support for PlayPaq with WM8510" - depends on SND_AT32_SOC && BOARD_PLAYPAQ - select SND_AT32_SOC_SSC - select SND_SOC_WM8510 - help - Say Y or M here if you want to add support for SoC audio - on the LRS PlayPaq. - - - -config SND_AT32_SOC_PLAYPAQ_SLAVE - bool "Run CODEC on PlayPaq in slave mode" - depends on SND_AT32_SOC_PLAYPAQ - default n - help - Say Y if you want to run with the AT32 SSC generating the BCLK - and FRAME signals on the PlayPaq. Unless you want to play - with the AT32 as the SSC master, you probably want to say N here, - as this will give you better sound quality. diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile deleted file mode 100644 index c03e55ececeb..000000000000 --- a/sound/soc/at32/Makefile +++ /dev/null @@ -1,11 +0,0 @@ -# AT32 Platform Support -snd-soc-at32-objs := at32-pcm.o -snd-soc-at32-ssc-objs := at32-ssc.o - -obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o -obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o - -# AT32 Machine Support -snd-soc-playpaq-objs := playpaq_wm8510.o - -obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c deleted file mode 100644 index c83584f989a9..000000000000 --- a/sound/soc/at32/at32-pcm.c +++ /dev/null @@ -1,492 +0,0 @@ -/* sound/soc/at32/at32-pcm.c - * ASoC PCM interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum@acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * Note that this is basically a port of the sound/soc/at91-pcm.c to - * the AVR32 kernel. Thanks to Frank Mandarino for that code. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include "at32-pcm.h" - - - -/*--------------------------------------------------------------------------*\ - * Hardware definition -\*--------------------------------------------------------------------------*/ -/* TODO: These values were taken from the AT91 platform driver, check - * them against real values for AT32 - */ -static const struct snd_pcm_hardware at32_pcm_hardware = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE), - - .formats = SNDRV_PCM_FMTBIT_S16, - .period_bytes_min = 32, - .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */ - .periods_min = 2, - .periods_max = 1024, - .buffer_bytes_max = 32 * 1024, -}; - - - -/*--------------------------------------------------------------------------*\ - * Data types -\*--------------------------------------------------------------------------*/ -struct at32_runtime_data { - struct at32_pcm_dma_params *params; - dma_addr_t dma_buffer; /* physical address of DMA buffer */ - dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ - size_t period_size; - - dma_addr_t period_ptr; /* physical address of next period */ - int periods; /* period index of period_ptr */ - - /* Save PDC registers (for power management) */ - u32 pdc_xpr_save; - u32 pdc_xcr_save; - u32 pdc_xnpr_save; - u32 pdc_xncr_save; -}; - - - -/*--------------------------------------------------------------------------*\ - * Helper functions -\*--------------------------------------------------------------------------*/ -static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *dmabuf = &substream->dma_buffer; - size_t size = at32_pcm_hardware.buffer_bytes_max; - - dmabuf->dev.type = SNDRV_DMA_TYPE_DEV; - dmabuf->dev.dev = pcm->card->dev; - dmabuf->private_data = NULL; - dmabuf->area = dma_alloc_coherent(pcm->card->dev, size, - &dmabuf->addr, GFP_KERNEL); - pr_debug("at32_pcm: preallocate_dma_buffer: " - "area=%p, addr=%p, size=%ld\n", - (void *)dmabuf->area, (void *)dmabuf->addr, size); - - if (!dmabuf->area) - return -ENOMEM; - - dmabuf->bytes = size; - return 0; -} - - - -/*--------------------------------------------------------------------------*\ - * ISR -\*--------------------------------------------------------------------------*/ -static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *rtd = substream->runtime; - struct at32_runtime_data *prtd = rtd->private_data; - struct at32_pcm_dma_params *params = prtd->params; - static int count; - - count++; - if (ssc_sr & params->mask->ssc_endbuf) { - pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - "underrun" : "overrun", params->name, ssc_sr, count); - - /* re-start the PDC */ - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) - prtd->period_ptr = prtd->dma_buffer; - - - ssc_writex(params->ssc->regs, params->pdc->xpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_enable); - } - - - if (ssc_sr & params->mask->ssc_endx) { - /* Load the PDC next pointer and counter registers */ - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) - prtd->period_ptr = prtd->dma_buffer; - ssc_writex(params->ssc->regs, params->pdc->xnpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - } - - - snd_pcm_period_elapsed(substream); -} - - - -/*--------------------------------------------------------------------------*\ - * PCM operations -\*--------------------------------------------------------------------------*/ -static int at32_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - /* this may get called several times by oss emulation - * with different params - */ - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->params = rtd->dai->cpu_dai->dma_data; - prtd->params->dma_intr_handler = at32_pcm_dma_irq; - - prtd->dma_buffer = runtime->dma_addr; - prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; - prtd->period_size = params_period_bytes(params); - - pr_debug("hw_params: DMA for %s initialized " - "(dma_bytes=%ld, period_size=%ld)\n", - prtd->params->name, runtime->dma_bytes, prtd->period_size); - - return 0; -} - - - -static int at32_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - - if (params != NULL) { - ssc_writex(params->ssc->regs, SSC_PDC_PTCR, - params->mask->pdc_disable); - prtd->params->dma_intr_handler = NULL; - } - - return 0; -} - - - -static int at32_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - - ssc_writex(params->ssc->regs, SSC_IDR, - params->mask->ssc_endx | params->mask->ssc_endbuf); - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - - return 0; -} - - -static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *rtd = substream->runtime; - struct at32_runtime_data *prtd = rtd->private_data; - struct at32_pcm_dma_params *params = prtd->params; - int ret = 0; - - pr_debug("at32_pcm_trigger: buffer_size = %ld, " - "dma_area = %p, dma_bytes = %ld\n", - rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->period_ptr = prtd->dma_buffer; - - ssc_writex(params->ssc->regs, params->pdc->xpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - prtd->period_ptr += prtd->period_size; - ssc_writex(params->ssc->regs, params->pdc->xnpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - - pr_debug("trigger: period_ptr=%lx, xpr=%x, " - "xcr=%d, xnpr=%x, xncr=%d\n", - (unsigned long)prtd->period_ptr, - ssc_readx(params->ssc->regs, params->pdc->xpr), - ssc_readx(params->ssc->regs, params->pdc->xcr), - ssc_readx(params->ssc->regs, params->pdc->xnpr), - ssc_readx(params->ssc->regs, params->pdc->xncr)); - - ssc_writex(params->ssc->regs, SSC_IER, - params->mask->ssc_endx | params->mask->ssc_endbuf); - ssc_writex(params->ssc->regs, SSC_PDC_PTCR, - params->mask->pdc_enable); - - pr_debug("sr=%x, imr=%x\n", - ssc_readx(params->ssc->regs, SSC_SR), - ssc_readx(params->ssc->regs, SSC_IER)); - break; /* SNDRV_PCM_TRIGGER_START */ - - - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - break; - - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_enable); - break; - - default: - ret = -EINVAL; - } - - return ret; -} - - - -static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd = runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; - dma_addr_t ptr; - snd_pcm_uframes_t x; - - ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); - x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); - - if (x == runtime->buffer_size) - x = 0; - - return x; -} - - - -static int at32_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware); - - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); - if (prtd == NULL) { - ret = -ENOMEM; - goto out; - } - runtime->private_data = prtd; - - -out: - return ret; -} - - - -static int at32_pcm_close(struct snd_pcm_substream *substream) -{ - struct at32_runtime_data *prtd = substream->runtime->private_data; - - kfree(prtd); - return 0; -} - - -static int at32_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, - vma->vm_end - vma->vm_start, vma->vm_page_prot); -} - - - -static struct snd_pcm_ops at32_pcm_ops = { - .open = at32_pcm_open, - .close = at32_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = at32_pcm_hw_params, - .hw_free = at32_pcm_hw_free, - .prepare = at32_pcm_prepare, - .trigger = at32_pcm_trigger, - .pointer = at32_pcm_pointer, - .mmap = at32_pcm_mmap, -}; - - - -/*--------------------------------------------------------------------------*\ - * ASoC platform driver -\*--------------------------------------------------------------------------*/ -static u64 at32_pcm_dmamask = 0xffffffff; - -static int at32_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) -{ - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &at32_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = at32_pcm_preallocate_dma_buffer( - pcm, SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n"); - ret = at32_pcm_preallocate_dma_buffer( - pcm, SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - - -out: - return ret; -} - - - -static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (substream == NULL) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - - - -#ifdef CONFIG_PM -static int at32_pcm_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at32_runtime_data *prtd; - struct at32_pcm_dma_params *params; - - if (runtime == NULL) - return 0; - prtd = runtime->private_data; - params = prtd->params; - - /* Disable the PDC and save the PDC registers */ - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - - prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); - prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); - prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); - prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); - - return 0; -} - - - -static int at32_pcm_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at32_runtime_data *prtd; - struct at32_pcm_dma_params *params; - - if (runtime == NULL) - return 0; - prtd = runtime->private_data; - params = prtd->params; - - /* Restore the PDC registers and enable the PDC */ - ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); - ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); - ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); - ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); - - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, params->mask->pdc_enable); - return 0; -} -#else /* CONFIG_PM */ -# define at32_pcm_suspend NULL -# define at32_pcm_resume NULL -#endif /* CONFIG_PM */ - - - -struct snd_soc_platform at32_soc_platform = { - .name = "at32-audio", - .pcm_ops = &at32_pcm_ops, - .pcm_new = at32_pcm_new, - .pcm_free = at32_pcm_free_dma_buffers, - .suspend = at32_pcm_suspend, - .resume = at32_pcm_resume, -}; -EXPORT_SYMBOL_GPL(at32_soc_platform); - - - -MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>"); -MODULE_DESCRIPTION("Atmel AT32 PCM module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h deleted file mode 100644 index 2a52430417da..000000000000 --- a/sound/soc/at32/at32-pcm.h +++ /dev/null @@ -1,79 +0,0 @@ -/* sound/soc/at32/at32-pcm.h - * ASoC PCM interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum@acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SOC_AT32_AT32_PCM_H -#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__ - -#include <linux/atmel-ssc.h> - - -/* - * Registers and status bits that are required by the PCM driver - * TODO: Is ptcr really used? - */ -struct at32_pdc_regs { - u32 xpr; /* PDC RX/TX pointer */ - u32 xcr; /* PDC RX/TX counter */ - u32 xnpr; /* PDC next RX/TX pointer */ - u32 xncr; /* PDC next RX/TX counter */ - u32 ptcr; /* PDC transfer control */ -}; - - - -/* - * SSC mask info - */ -struct at32_ssc_mask { - u32 ssc_enable; /* SSC RX/TX enable */ - u32 ssc_disable; /* SSC RX/TX disable */ - u32 ssc_endx; /* SSC ENDTX or ENDRX */ - u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */ - u32 pdc_enable; /* PDC RX/TX enable */ - u32 pdc_disable; /* PDC RX/TX disable */ -}; - - - -/* - * This structure, shared between the PCM driver and the interface, - * contains all information required by the PCM driver to perform the - * PDC DMA operation. All fields except dma_intr_handler() are initialized - * by the interface. The dms_intr_handler() pointer is set by the PCM - * driver and called by the interface SSC interrupt handler if it is - * non-NULL. - */ -struct at32_pcm_dma_params { - char *name; /* stream identifier */ - int pdc_xfer_size; /* PDC counter increment in bytes */ - struct ssc_device *ssc; /* SSC device for stream */ - struct at32_pdc_regs *pdc; /* PDC register info */ - struct at32_ssc_mask *mask; /* SSC mask info */ - struct snd_pcm_substream *substream; - void (*dma_intr_handler) (u32, struct snd_pcm_substream *); -}; - - - -/* - * The AT32 ASoC platform driver - */ -extern struct snd_soc_platform at32_soc_platform; - - - -/* - * SSC register access (since ssc_writel() / ssc_readl() require literal name) - */ -#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) -#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) - -#endif /* __SOUND_SOC_AT32_AT32_PCM_H */ diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c deleted file mode 100644 index 4ef6492c902e..000000000000 --- a/sound/soc/at32/at32-ssc.c +++ /dev/null @@ -1,849 +0,0 @@ -/* sound/soc/at32/at32-ssc.c - * ASoC platform driver for AT32 using SSC as DAI - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum@acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * Note that this is basically a port of the sound/soc/at91-ssc.c to - * the AVR32 kernel. Thanks to Frank Mandarino for that code. - */ - -/* #define DEBUG */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/io.h> -#include <linux/atmel_pdc.h> -#include <linux/atmel-ssc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include "at32-pcm.h" -#include "at32-ssc.h" - - - -/*-------------------------------------------------------------------------*\ - * Constants -\*-------------------------------------------------------------------------*/ -#define NUM_SSC_DEVICES 3 - -/* - * SSC direction masks - */ -#define SSC_DIR_MASK_UNUSED 0 -#define SSC_DIR_MASK_PLAYBACK 1 -#define SSC_DIR_MASK_CAPTURE 2 - -/* - * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These - * are expected to be used with SSC_BF - */ -/* START bit field values */ -#define SSC_START_CONTINUOUS 0 -#define SSC_START_TX_RX 1 -#define SSC_START_LOW_RF 2 -#define SSC_START_HIGH_RF 3 -#define SSC_START_FALLING_RF 4 -#define SSC_START_RISING_RF 5 -#define SSC_START_LEVEL_RF 6 -#define SSC_START_EDGE_RF 7 -#define SSS_START_COMPARE_0 8 - -/* CKI bit field values */ -#define SSC_CKI_FALLING 0 -#define SSC_CKI_RISING 1 - -/* CKO bit field values */ -#define SSC_CKO_NONE 0 -#define SSC_CKO_CONTINUOUS 1 -#define SSC_CKO_TRANSFER 2 - -/* CKS bit field values */ -#define SSC_CKS_DIV 0 -#define SSC_CKS_CLOCK 1 -#define SSC_CKS_PIN 2 - -/* FSEDGE bit field values */ -#define SSC_FSEDGE_POSITIVE 0 -#define SSC_FSEDGE_NEGATIVE 1 - -/* FSOS bit field values */ -#define SSC_FSOS_NONE 0 -#define SSC_FSOS_NEGATIVE 1 -#define SSC_FSOS_POSITIVE 2 -#define SSC_FSOS_LOW 3 -#define SSC_FSOS_HIGH 4 -#define SSC_FSOS_TOGGLE 5 - -#define START_DELAY 1 - - - -/*-------------------------------------------------------------------------*\ - * Module data -\*-------------------------------------------------------------------------*/ -/* - * SSC PDC registered required by the PCM DMA engine - */ -static struct at32_pdc_regs pdc_tx_reg = { - .xpr = SSC_PDC_TPR, - .xcr = SSC_PDC_TCR, - .xnpr = SSC_PDC_TNPR, - .xncr = SSC_PDC_TNCR, -}; - - - -static struct at32_pdc_regs pdc_rx_reg = { - .xpr = SSC_PDC_RPR, - .xcr = SSC_PDC_RCR, - .xnpr = SSC_PDC_RNPR, - .xncr = SSC_PDC_RNCR, -}; - - - -/* - * SSC and PDC status bits for transmit and receive - */ -static struct at32_ssc_mask ssc_tx_mask = { - .ssc_enable = SSC_BIT(CR_TXEN), - .ssc_disable = SSC_BIT(CR_TXDIS), - .ssc_endx = SSC_BIT(SR_ENDTX), - .ssc_endbuf = SSC_BIT(SR_TXBUFE), - .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN), - .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS), -}; - - - -static struct at32_ssc_mask ssc_rx_mask = { - .ssc_enable = SSC_BIT(CR_RXEN), - .ssc_disable = SSC_BIT(CR_RXDIS), - .ssc_endx = SSC_BIT(SR_ENDRX), - .ssc_endbuf = SSC_BIT(SR_RXBUFF), - .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN), - .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS), -}; - - - -/* - * DMA parameters for each SSC - */ -static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { - { - { - .name = "SSC0 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC0 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, - { - { - .name = "SSC1 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC1 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, - { - { - .name = "SSC2 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC2 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, -}; - - - -static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = { - { - .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, - { - .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, - { - .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, -}; - - - - -/*-------------------------------------------------------------------------*\ - * ISR -\*-------------------------------------------------------------------------*/ -/* - * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt - * handler in the PCM driver. - */ -static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id) -{ - struct at32_ssc_info *ssc_p = dev_id; - struct at32_pcm_dma_params *dma_params; - u32 ssc_sr; - u32 ssc_substream_mask; - int i; - - ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) & - ssc_readl(ssc_p->ssc->regs, IMR)); - - /* - * Loop through substreams attached to this SSC. If a DMA-related - * interrupt occured on that substream, call the DMA interrupt - * handler function, if one has been registered in the dma_param - * structure by the PCM driver. - */ - for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { - dma_params = ssc_p->dma_params[i]; - - if ((dma_params != NULL) && - (dma_params->dma_intr_handler != NULL)) { - ssc_substream_mask = (dma_params->mask->ssc_endx | - dma_params->mask->ssc_endbuf); - if (ssc_sr & ssc_substream_mask) { - dma_params->dma_intr_handler(ssc_sr, - dma_params-> - substream); - } - } - } - - - return IRQ_HANDLED; -} - -/*-------------------------------------------------------------------------*\ - * DAI functions -\*-------------------------------------------------------------------------*/ -/* - * Startup. Only that one substream allowed in each direction. - */ -static int at32_ssc_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - int dir_mask; - - dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); - - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); - return -EBUSY; - } - ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); - - return 0; -} - - - -/* - * Shutdown. Clear DMA parameters and shutdown the SSC if there - * are no other substreams open. - */ -static void at32_ssc_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at32_pcm_dma_params *dma_params; - int dir_mask; - - dma_params = ssc_p->dma_params[substream->stream]; - - if (dma_params != NULL) { - ssc_writel(dma_params->ssc->regs, CR, - dma_params->mask->ssc_disable); - pr_debug("%s disabled SSC_SR=0x%08x\n", - (substream->stream ? "receiver" : "transmit"), - ssc_readl(ssc_p->ssc->regs, SR)); - - dma_params->ssc = NULL; - dma_params->substream = NULL; - ssc_p->dma_params[substream->stream] = NULL; - } - - - dir_mask = 1 << substream->stream; - spin_lock_irq(&ssc_p->lock); - ssc_p->dir_mask &= ~dir_mask; - if (!ssc_p->dir_mask) { - /* Shutdown the SSC clock */ - pr_debug("at32-ssc: Stopping user %d clock\n", - ssc_p->ssc->user); - clk_disable(ssc_p->ssc->clk); - - if (ssc_p->initialized) { - free_irq(ssc_p->ssc->irq, ssc_p); - ssc_p->initialized = 0; - } - - /* Reset the SSC */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); - - /* clear the SSC dividers */ - ssc_p->cmr_div = 0; - ssc_p->tcmr_period = 0; - ssc_p->rcmr_period = 0; - } - spin_unlock_irq(&ssc_p->lock); -} - - - -/* - * Set the SSC system clock rate - */ -static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - /* TODO: What the heck do I do here? */ - return 0; -} - - - -/* - * Record DAI format for use by hw_params() - */ -static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - ssc_p->daifmt = fmt; - return 0; -} - - - -/* - * Record SSC clock dividers for use in hw_params() - */ -static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - switch (div_id) { - case AT32_SSC_CMR_DIV: - /* - * The same master clock divider is used for both - * transmit and receive, so if a value has already - * been set, it must match this value - */ - if (ssc_p->cmr_div == 0) - ssc_p->cmr_div = div; - else if (div != ssc_p->cmr_div) - return -EBUSY; - break; - - case AT32_SSC_TCMR_PERIOD: - ssc_p->tcmr_period = div; - break; - - case AT32_SSC_RCMR_PERIOD: - ssc_p->rcmr_period = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - - - -/* - * Configure the SSC - */ -static int at32_ssc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int id = rtd->dai->cpu_dai->id; - struct at32_ssc_info *ssc_p = &ssc_info[id]; - struct at32_pcm_dma_params *dma_params; - int channels, bits; - u32 tfmr, rfmr, tcmr, rcmr; - int start_event; - int ret; - - - /* - * Currently, there is only one set of dma_params for each direction. - * If more are added, this code will have to be changed to select - * the proper set - */ - dma_params = &ssc_dma_params[id][substream->stream]; - dma_params->ssc = ssc_p->ssc; - dma_params->substream = substream; - - ssc_p->dma_params[substream->stream] = dma_params; - - - /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the PCM driver's hw_params() - * function. It should not be used for other purposes as it - * is common to all substreams. - */ - rtd->dai->cpu_dai->dma_data = dma_params; - - channels = params_channels(params); - - - /* - * Determine sample size in bits and the PDC increment - */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - bits = 8; - dma_params->pdc_xfer_size = 1; - break; - - case SNDRV_PCM_FORMAT_S16: - bits = 16; - dma_params->pdc_xfer_size = 2; - break; - - case SNDRV_PCM_FORMAT_S24: - bits = 24; - dma_params->pdc_xfer_size = 4; - break; - - case SNDRV_PCM_FORMAT_S32: - bits = 32; - dma_params->pdc_xfer_size = 4; - break; - - default: - pr_warning("at32-ssc: Unsupported PCM format %d", - params_format(params)); - return -EINVAL; - } - pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n", - bits, dma_params->pdc_xfer_size, channels); - - - /* - * The SSC only supports up to 16-bit samples in I2S format, due - * to the size of the Frame Mode Register FSLEN field. - */ - if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) - if (bits > 16) { - pr_warning("at32-ssc: " - "sample size %d is too large for I2S\n", - bits); - return -EINVAL; - } - - - /* - * Compute the SSC register settings - */ - switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK | - SND_SOC_DAIFMT_MASTER_MASK)) { - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: - /* - * I2S format, SSC provides BCLK and LRS clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line - */ - pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n"); - rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | - SSC_BF(RCMR_STTDLY, START_DELAY) | - SSC_BF(RCMR_START, SSC_START_FALLING_RF) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_DIV)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) | - SSC_BF(RFMR_FSLEN, bits - 1) | - SSC_BF(RFMR_DATNB, channels - 1) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | - SSC_BF(TCMR_STTDLY, START_DELAY) | - SSC_BF(TCMR_START, SSC_START_FALLING_RF) | - SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | - SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | - SSC_BF(TCMR_CKS, SSC_CKS_DIV)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) | - SSC_BF(TFMR_FSLEN, bits - 1) | - SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) | - SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* - * I2S format, CODEC supplies BCLK and LRC clock. - * - * The SSC transmit clock is obtained from the BCLK signal - * on the TK line, and the SSC receive clock is generated from - * the transmit clock. - * - * For single channel data, one sample is transferred on the - * falling edge of the LRC clock. For two channel data, one - * sample is transferred on both edges of the LRC clock. - */ - pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n"); - start_event = ((channels == 1) ? - SSC_START_FALLING_RF : SSC_START_EDGE_RF); - - rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) | - SSC_BF(RCMR_START, start_event) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_CLOCK)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) | - SSC_BF(TCMR_START, start_event) | - SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | - SSC_BF(TCMR_CKO, SSC_CKO_NONE) | - SSC_BF(TCMR_CKS, SSC_CKS_PIN)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | - SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: - /* - * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line - */ - pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n"); - rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | - SSC_BF(RCMR_STTDLY, 1) | - SSC_BF(RCMR_START, SSC_START_RISING_RF) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_DIV)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) | - SSC_BF(RFMR_DATNB, channels - 1) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | - SSC_BF(TCMR_STTDLY, 1) | - SSC_BF(TCMR_START, SSC_START_RISING_RF) | - SSC_BF(TCMR_CKI, SSC_CKI_RISING) | - SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | - SSC_BF(TCMR_CKS, SSC_CKS_DIV)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) | - SSC_BF(TFMR_DATNB, channels - 1) | - SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: - default: - pr_warning("at32-ssc: unsupported DAI format 0x%x\n", - ssc_p->daifmt); - return -EINVAL; - break; - } - pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", - rcmr, rfmr, tcmr, tfmr); - - - if (!ssc_p->initialized) { - /* enable peripheral clock */ - pr_debug("at32-ssc: Starting clock\n"); - clk_enable(ssc_p->ssc->clk); - - /* Reset the SSC and its PDC registers */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); - - ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); - - ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); - - ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0, - ssc_p->name, ssc_p); - if (ret < 0) { - pr_warning("at32-ssc: request irq failed (%d)\n", ret); - pr_debug("at32-ssc: Stopping clock\n"); - clk_disable(ssc_p->ssc->clk); - return ret; - } - - ssc_p->initialized = 1; - } - - /* Set SSC clock mode register */ - ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); - - /* set receive clock mode and format */ - ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); - ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); - - /* set transmit clock mode and format */ - ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); - ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); - - pr_debug("at32-ssc: SSC initialized\n"); - return 0; -} - - - -static int at32_ssc_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at32_pcm_dma_params *dma_params; - - dma_params = ssc_p->dma_params[substream->stream]; - - ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable); - - return 0; -} - - - -#ifdef CONFIG_PM -static int at32_ssc_suspend(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p; - - if (!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* Save the status register before disabling transmit and receive */ - ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); - - /* Save the current interrupt mask, then disable unmasked interrupts */ - ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); - ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); - - ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); - ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); - ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); - ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); - ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); - - return 0; -} - - - -static int at32_ssc_resume(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p; - u32 cr; - - if (!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* restore SSC register settings */ - ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); - ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); - ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); - ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); - ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); - - /* re-enable interrupts */ - ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); - - /* Re-enable recieve and transmit as appropriate */ - cr = 0; - cr |= - (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; - cr |= - (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; - ssc_writel(ssc_p->ssc->regs, CR, cr); - - return 0; -} -#else /* CONFIG_PM */ -# define at32_ssc_suspend NULL -# define at32_ssc_resume NULL -#endif /* CONFIG_PM */ - - -#define AT32_SSC_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - - -#define AT32_SSC_FORMATS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \ - SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) - - -struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = { - { - .name = "at32-ssc0", - .id = 0, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[0], - }, - { - .name = "at32-ssc1", - .id = 1, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[1], - }, - { - .name = "at32-ssc2", - .id = 2, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[2], - }, -}; -EXPORT_SYMBOL_GPL(at32_ssc_dai); - - -MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>"); -MODULE_DESCRIPTION("AT32 SSC ASoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h deleted file mode 100644 index 3c052dbbe460..000000000000 --- a/sound/soc/at32/at32-ssc.h +++ /dev/null @@ -1,59 +0,0 @@ -/* sound/soc/at32/at32-ssc.h - * ASoC SSC interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum@acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SOC_AT32_AT32_SSC_H -#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__ - -#include <linux/types.h> -#include <linux/atmel-ssc.h> - -#include "at32-pcm.h" - - - -struct at32_ssc_state { - u32 ssc_cmr; - u32 ssc_rcmr; - u32 ssc_rfmr; - u32 ssc_tcmr; - u32 ssc_tfmr; - u32 ssc_sr; - u32 ssc_imr; -}; - - - -struct at32_ssc_info { - char *name; - struct ssc_device *ssc; - spinlock_t lock; /* lock for dir_mask */ - unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ - unsigned short initialized; /* true if SSC has been initialized */ - unsigned short daifmt; - unsigned short cmr_div; - unsigned short tcmr_period; - unsigned short rcmr_period; - struct at32_pcm_dma_params *dma_params[2]; - struct at32_ssc_state ssc_state; -}; - - -/* SSC divider ids */ -#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */ -#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ -#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ - - -extern struct snd_soc_dai at32_ssc_dai[]; - - - -#endif /* __SOUND_SOC_AT32_AT32_SSC_H */ diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig deleted file mode 100644 index 85a883299c2e..000000000000 --- a/sound/soc/at91/Kconfig +++ /dev/null @@ -1,10 +0,0 @@ -config SND_AT91_SOC - tristate "SoC Audio for the Atmel AT91 System-on-Chip" - depends on ARCH_AT91 - help - Say Y or M if you want to add support for codecs attached to - the AT91 SSC interface. You will also need - to select the audio interfaces to support below. - -config SND_AT91_SOC_SSC - tristate diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile deleted file mode 100644 index b817f11df286..000000000000 --- a/sound/soc/at91/Makefile +++ /dev/null @@ -1,6 +0,0 @@ -# AT91 Platform Support -snd-soc-at91-objs := at91-pcm.o -snd-soc-at91-ssc-objs := at91-ssc.o - -obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o -obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c deleted file mode 100644 index 7ab48bd25e4c..000000000000 --- a/sound/soc/at91/at91-pcm.c +++ /dev/null @@ -1,434 +0,0 @@ -/* - * at91-pcm.c -- ALSA PCM interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino@endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 3, 2006 - * - * Based on pxa2xx-pcm.c by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/at91_ssc.h> - -#include "at91-pcm.h" - -#if 0 -#define DBG(x...) printk(KERN_INFO "at91-pcm: " x) -#else -#define DBG(x...) -#endif - -static const struct snd_pcm_hardware at91_pcm_hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_PAUSE, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 2, - .periods_max = 1024, - .buffer_bytes_max = 32 * 1024, -}; - -struct at91_runtime_data { - struct at91_pcm_dma_params *params; - dma_addr_t dma_buffer; /* physical address of dma buffer */ - dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ - size_t period_size; - dma_addr_t period_ptr; /* physical address of next period */ - u32 pdc_xpr_save; /* PDC register save */ - u32 pdc_xcr_save; - u32 pdc_xnpr_save; - u32 pdc_xncr_save; -}; - -static void at91_pcm_dma_irq(u32 ssc_sr, - struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - static int count = 0; - - count++; - - if (ssc_sr & params->mask->ssc_endbuf) { - - printk(KERN_WARNING - "at91-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? "underrun" : "overrun", - params->name, ssc_sr, count); - - /* re-start the PDC */ - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) { - prtd->period_ptr = prtd->dma_buffer; - } - - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - } - - if (ssc_sr & params->mask->ssc_endx) { - - /* Load the PDC next pointer and counter registers */ - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) { - prtd->period_ptr = prtd->dma_buffer; - } - at91_ssc_write(params->ssc_base + params->pdc->xnpr, - prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - } - - snd_pcm_period_elapsed(substream); -} - -static int at91_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - /* this may get called several times by oss emulation - * with different params */ - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->params = rtd->dai->cpu_dai->dma_data; - prtd->params->dma_intr_handler = at91_pcm_dma_irq; - - prtd->dma_buffer = runtime->dma_addr; - prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; - prtd->period_size = params_period_bytes(params); - - DBG("hw_params: DMA for %s initialized (dma_bytes=%d, period_size=%d)\n", - prtd->params->name, runtime->dma_bytes, prtd->period_size); - return 0; -} - -static int at91_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - - if (params != NULL) { - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - prtd->params->dma_intr_handler = NULL; - } - - return 0; -} - -static int at91_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - - at91_ssc_write(params->ssc_base + AT91_SSC_IDR, - params->mask->ssc_endx | params->mask->ssc_endbuf); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - return 0; -} - -static int at91_pcm_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->period_ptr = prtd->dma_buffer; - - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - prtd->period_ptr += prtd->period_size; - at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - - DBG("trigger: period_ptr=%lx, xpr=%lx, xcr=%ld, xnpr=%lx, xncr=%ld\n", - (unsigned long) prtd->period_ptr, - at91_ssc_read(params->ssc_base + params->pdc->xpr), - at91_ssc_read(params->ssc_base + params->pdc->xcr), - at91_ssc_read(params->ssc_base + params->pdc->xnpr), - at91_ssc_read(params->ssc_base + params->pdc->xncr)); - - at91_ssc_write(params->ssc_base + AT91_SSC_IER, - params->mask->ssc_endx | params->mask->ssc_endbuf); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, - params->mask->pdc_enable); - - DBG("sr=%lx imr=%lx\n", - at91_ssc_read(params->ssc_base + AT91_SSC_SR), - at91_ssc_read(params->ssc_base + AT91_SSC_IMR)); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - break; - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - break; - - default: - ret = -EINVAL; - } - - return ret; -} - -static snd_pcm_uframes_t at91_pcm_pointer( - struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd = runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - dma_addr_t ptr; - snd_pcm_uframes_t x; - - ptr = (dma_addr_t) at91_ssc_read(params->ssc_base + params->pdc->xpr); - x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); - - if (x == runtime->buffer_size) - x = 0; - return x; -} - -static int at91_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &at91_pcm_hardware); - - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - prtd = kzalloc(sizeof(struct at91_runtime_data), GFP_KERNEL); - if (prtd == NULL) { - ret = -ENOMEM; - goto out; - } - runtime->private_data = prtd; - - out: - return ret; -} - -static int at91_pcm_close(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - - kfree(prtd); - return 0; -} - -static int at91_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -struct snd_pcm_ops at91_pcm_ops = { - .open = at91_pcm_open, - .close = at91_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = at91_pcm_hw_params, - .hw_free = at91_pcm_hw_free, - .prepare = at91_pcm_prepare, - .trigger = at91_pcm_trigger, - .pointer = at91_pcm_pointer, - .mmap = at91_pcm_mmap, -}; - -static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, - int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = at91_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - - DBG("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", - (void *) buf->area, - (void *) buf->addr, - size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -static u64 at91_pcm_dmamask = 0xffffffff; - -static int at91_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) -{ - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &at91_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = at91_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - ret = at91_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -#ifdef CONFIG_PM -static int at91_pcm_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at91_runtime_data *prtd; - struct at91_pcm_dma_params *params; - - if (!runtime) - return 0; - - prtd = runtime->private_data; - params = prtd->params; - - /* disable the PDC and save the PDC registers */ - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - - prtd->pdc_xpr_save = at91_ssc_read(params->ssc_base + params->pdc->xpr); - prtd->pdc_xcr_save = at91_ssc_read(params->ssc_base + params->pdc->xcr); - prtd->pdc_xnpr_save = at91_ssc_read(params->ssc_base + params->pdc->xnpr); - prtd->pdc_xncr_save = at91_ssc_read(params->ssc_base + params->pdc->xncr); - - return 0; -} - -static int at91_pcm_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at91_runtime_data *prtd; - struct at91_pcm_dma_params *params; - - if (!runtime) - return 0; - - prtd = runtime->private_data; - params = prtd->params; - - /* restore the PDC registers and enable the PDC */ - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->pdc_xpr_save); - at91_ssc_write(params->ssc_base + params->pdc->xcr, prtd->pdc_xcr_save); - at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->pdc_xnpr_save); - at91_ssc_write(params->ssc_base + params->pdc->xncr, prtd->pdc_xncr_save); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - return 0; -} -#else -#define at91_pcm_suspend NULL -#define at91_pcm_resume NULL -#endif - -struct snd_soc_platform at91_soc_platform = { - .name = "at91-audio", - .pcm_ops = &at91_pcm_ops, - .pcm_new = at91_pcm_new, - .pcm_free = at91_pcm_free_dma_buffers, - .suspend = at91_pcm_suspend, - .resume = at91_pcm_resume, -}; - -EXPORT_SYMBOL_GPL(at91_soc_platform); - -MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>"); -MODULE_DESCRIPTION("Atmel AT91 PCM module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h deleted file mode 100644 index e5aada2cb102..000000000000 --- a/sound/soc/at91/at91-pcm.h +++ /dev/null @@ -1,72 +0,0 @@ -/* - * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino@endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 3, 2006 - * - * Based on pxa2xx-pcm.h by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _AT91_PCM_H -#define _AT91_PCM_H - -#include <mach/hardware.h> - -struct at91_ssc_periph { - void __iomem *base; - u32 pid; -}; - -/* - * Registers and status bits that are required by the PCM driver. - */ -struct at91_pdc_regs { - unsigned int xpr; /* PDC recv/trans pointer */ - unsigned int xcr; /* PDC recv/trans counter */ - unsigned int xnpr; /* PDC next recv/trans pointer */ - unsigned int xncr; /* PDC next recv/trans counter */ - unsigned int ptcr; /* PDC transfer control */ -}; - -struct at91_ssc_mask { - u32 ssc_enable; /* SSC recv/trans enable */ - u32 ssc_disable; /* SSC recv/trans disable */ - u32 ssc_endx; /* SSC ENDTX or ENDRX */ - u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ - u32 pdc_enable; /* PDC recv/trans enable */ - u32 pdc_disable; /* PDC recv/trans disable */ -}; - -/* - * This structure, shared between the PCM driver and the interface, - * contains all information required by the PCM driver to perform the - * PDC DMA operation. All fields except dma_intr_handler() are initialized - * by the interface. The dms_intr_handler() pointer is set by the PCM - * driver and called by the interface SSC interrupt handler if it is - * non-NULL. - */ -struct at91_pcm_dma_params { - char *name; /* stream identifier */ - int pdc_xfer_size; /* PDC counter increment in bytes */ - void __iomem *ssc_base; /* SSC base address */ - struct at91_pdc_regs *pdc; /* PDC receive or transmit registers */ - struct at91_ssc_mask *mask;/* SSC & PDC status bits */ - struct snd_pcm_substream *substream; - void (*dma_intr_handler)(u32, struct snd_pcm_substream *); -}; - -extern struct snd_soc_platform at91_soc_platform; - -#define at91_ssc_read(a) ((unsigned long) __raw_readl(a)) -#define at91_ssc_write(a,v) __raw_writel((v),(a)) - -#endif /* _AT91_PCM_H */ diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c deleted file mode 100644 index 1b61cc461261..000000000000 --- a/sound/soc/at91/at91-ssc.c +++ /dev/null @@ -1,791 +0,0 @@ -/* - * at91-ssc.c -- ALSA SoC AT91 SSC Audio Layer Platform driver - * - * Author: Frank Mandarino <fmandarino@endrelia.com> - * Endrelia Technologies Inc. - * - * Based on pxa2xx Platform drivers by - * Liam Girdwood <lrg@slimlogic.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/at91_pmc.h> -#include <mach/at91_ssc.h> - -#include "at91-pcm.h" -#include "at91-ssc.h" - -#if 0 -#define DBG(x...) printk(KERN_DEBUG "at91-ssc:" x) -#else -#define DBG(x...) -#endif - -#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) -#define NUM_SSC_DEVICES 1 -#else -#define NUM_SSC_DEVICES 3 -#endif - - -/* - * SSC PDC registers required by the PCM DMA engine. - */ -static struct at91_pdc_regs pdc_tx_reg = { - .xpr = ATMEL_PDC_TPR, - .xcr = ATMEL_PDC_TCR, - .xnpr = ATMEL_PDC_TNPR, - .xncr = ATMEL_PDC_TNCR, -}; - -static struct at91_pdc_regs pdc_rx_reg = { - .xpr = ATMEL_PDC_RPR, - .xcr = ATMEL_PDC_RCR, - .xnpr = ATMEL_PDC_RNPR, - .xncr = ATMEL_PDC_RNCR, -}; - -/* - * SSC & PDC status bits for transmit and receive. - */ -static struct at91_ssc_mask ssc_tx_mask = { - .ssc_enable = AT91_SSC_TXEN, - .ssc_disable = AT91_SSC_TXDIS, - .ssc_endx = AT91_SSC_ENDTX, - .ssc_endbuf = AT91_SSC_TXBUFE, - .pdc_enable = ATMEL_PDC_TXTEN, - .pdc_disable = ATMEL_PDC_TXTDIS, -}; - -static struct at91_ssc_mask ssc_rx_mask = { - .ssc_enable = AT91_SSC_RXEN, - .ssc_disable = AT91_SSC_RXDIS, - .ssc_endx = AT91_SSC_ENDRX, - .ssc_endbuf = AT91_SSC_RXBUFF, - .pdc_enable = ATMEL_PDC_RXTEN, - .pdc_disable = ATMEL_PDC_RXTDIS, -}; - - -/* - * DMA parameters. - */ -static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { - {{ - .name = "SSC0 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC0 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, -#if NUM_SSC_DEVICES == 3 - {{ - .name = "SSC1 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC1 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, - {{ - .name = "SSC2 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC2 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, -#endif -}; - -struct at91_ssc_state { - u32 ssc_cmr; - u32 ssc_rcmr; - u32 ssc_rfmr; - u32 ssc_tcmr; - u32 ssc_tfmr; - u32 ssc_sr; - u32 ssc_imr; -}; - -static struct at91_ssc_info { - char *name; - struct at91_ssc_periph ssc; - spinlock_t lock; /* lock for dir_mask */ - unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ - unsigned short initialized; /* 1=SSC has been initialized */ - unsigned short daifmt; - unsigned short cmr_div; - unsigned short tcmr_period; - unsigned short rcmr_period; - struct at91_pcm_dma_params *dma_params[2]; - struct at91_ssc_state ssc_state; - -} ssc_info[NUM_SSC_DEVICES] = { - { - .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), - .dir_mask = 0, - .initialized = 0, - }, -#if NUM_SSC_DEVICES == 3 - { - .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), - .dir_mask = 0, - .initialized = 0, - }, - { - .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), - .dir_mask = 0, - .initialized = 0, - }, -#endif -}; - -static unsigned int at91_ssc_sysclk; - -/* - * SSC interrupt handler. Passes PDC interrupts to the DMA - * interrupt handler in the PCM driver. - */ -static irqreturn_t at91_ssc_interrupt(int irq, void *dev_id) -{ - struct at91_ssc_info *ssc_p = dev_id; - struct at91_pcm_dma_params *dma_params; - u32 ssc_sr; - int i; - - ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR) - & at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR); - - /* - * Loop through the substreams attached to this SSC. If - * a DMA-related interrupt occurred on that substream, call - * the DMA interrupt handler function, if one has been - * registered in the dma_params structure by the PCM driver. - */ - for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { - dma_params = ssc_p->dma_params[i]; - - if (dma_params != NULL && dma_params->dma_intr_handler != NULL && - (ssc_sr & - (dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf))) - - dma_params->dma_intr_handler(ssc_sr, dma_params->substream); - } - - return IRQ_HANDLED; -} - -/* - * Startup. Only that one substream allowed in each direction. - */ -static int at91_ssc_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - int dir_mask; - - DBG("ssc_startup: SSC_SR=0x%08lx\n", - at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)); - dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2; - - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); - return -EBUSY; - } - ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); - - return 0; -} - -/* - * Shutdown. Clear DMA parameters and shutdown the SSC if there - * are no other substreams open. - */ -static void at91_ssc_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at91_pcm_dma_params *dma_params; - int dir, dir_mask; - - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - dma_params = ssc_p->dma_params[dir]; - - if (dma_params != NULL) { - at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR, - dma_params->mask->ssc_disable); - DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"), - at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)); - - dma_params->ssc_base = NULL; - dma_params->substream = NULL; - ssc_p->dma_params[dir] = NULL; - } - - dir_mask = 1 << dir; - - spin_lock_irq(&ssc_p->lock); - ssc_p->dir_mask &= ~dir_mask; - if (!ssc_p->dir_mask) { - /* Shutdown the SSC clock. */ - DBG("Stopping pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid); - - if (ssc_p->initialized) { - free_irq(ssc_p->ssc.pid, ssc_p); - ssc_p->initialized = 0; - } - - /* Reset the SSC */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST); - - /* Clear the SSC dividers */ - ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; - } - spin_unlock_irq(&ssc_p->lock); -} - -/* - * Record the SSC system clock rate. - */ -static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - /* - * The only clock supplied to the SSC is the AT91 master clock, - * which is only used if the SSC is generating BCLK and/or - * LRC clocks. - */ - switch (clk_id) { - case AT91_SYSCLK_MCK: - at91_ssc_sysclk = freq; - break; - default: - return -EINVAL; - } - - return 0; -} - -/* - * Record the DAI format for use in hw_params(). - */ -static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - ssc_p->daifmt = fmt; - return 0; -} - -/* - * Record SSC clock dividers for use in hw_params(). - */ -static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - switch (div_id) { - case AT91SSC_CMR_DIV: - /* - * The same master clock divider is used for both - * transmit and receive, so if a value has already - * been set, it must match this value. - */ - if (ssc_p->cmr_div == 0) - ssc_p->cmr_div = div; - else - if (div != ssc_p->cmr_div) - return -EBUSY; - break; - - case AT91SSC_TCMR_PERIOD: - ssc_p->tcmr_period = div; - break; - - case AT91SSC_RCMR_PERIOD: - ssc_p->rcmr_period = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - -/* - * Configure the SSC. - */ -static int at91_ssc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int id = rtd->dai->cpu_dai->id; - struct at91_ssc_info *ssc_p = &ssc_info[id]; - struct at91_pcm_dma_params *dma_params; - int dir, channels, bits; - u32 tfmr, rfmr, tcmr, rcmr; - int start_event; - int ret; - - /* - * Currently, there is only one set of dma params for - * each direction. If more are added, this code will - * have to be changed to select the proper set. - */ - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - - dma_params = &ssc_dma_params[id][dir]; - dma_params->ssc_base = ssc_p->ssc.base; - dma_params->substream = substream; - - ssc_p->dma_params[dir] = dma_params; - - /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the pcm driver hw_params() - * function. It should not be used for other purposes - * as it is common to all substreams. - */ - rtd->dai->cpu_dai->dma_data = dma_params; - - channels = params_channels(params); - - /* - * Determine sample size in bits and the PDC increment. - */ - switch(params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - bits = 8; - dma_params->pdc_xfer_size = 1; - break; - case SNDRV_PCM_FORMAT_S16_LE: - bits = 16; - dma_params->pdc_xfer_size = 2; - break; - case SNDRV_PCM_FORMAT_S24_LE: - bits = 24; - dma_params->pdc_xfer_size = 4; - break; - case SNDRV_PCM_FORMAT_S32_LE: - bits = 32; - dma_params->pdc_xfer_size = 4; - break; - default: - printk(KERN_WARNING "at91-ssc: unsupported PCM format\n"); - return -EINVAL; - } - - /* - * The SSC only supports up to 16-bit samples in I2S format, due - * to the size of the Frame Mode Register FSLEN field. - */ - if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S - && bits > 16) { - printk(KERN_WARNING - "at91-ssc: sample size %d is too large for I2S\n", bits); - return -EINVAL; - } - - /* - * Compute SSC register settings. - */ - switch (ssc_p->daifmt - & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: - /* - * I2S format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line. - */ - rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS) - | (((bits - 1) << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS) - | (((bits - 1) << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - break; - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* - * I2S format, CODEC supplies BCLK and LRC clocks. - * - * The SSC transmit clock is obtained from the BCLK signal on - * on the TK line, and the SSC receive clock is generated from the - * transmit clock. - * - * For single channel data, one sample is transferred on the falling - * edge of the LRC clock. For two channel data, one sample is - * transferred on both edges of the LRC clock. - */ - start_event = channels == 1 - ? AT91_SSC_START_FALLING_RF - : AT91_SSC_START_EDGE_RF; - - rcmr = (( 0 << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( start_event ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_CLOCK ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (( 0 << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( 0 << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( start_event ) & AT91_SSC_START) - | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_PIN ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (( 0 << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - break; - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: - /* - * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line. - */ - rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - - - break; - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: - default: - printk(KERN_WARNING "at91-ssc: unsupported DAI format 0x%x.\n", - ssc_p->daifmt); - return -EINVAL; - break; - } - DBG("RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr); - - if (!ssc_p->initialized) { - - /* Enable PMC peripheral clock for this SSC */ - DBG("Starting pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid); - - /* Reset the SSC and its PDC registers */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST); - - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNCR, 0); - - if ((ret = request_irq(ssc_p->ssc.pid, at91_ssc_interrupt, - 0, ssc_p->name, ssc_p)) < 0) { - printk(KERN_WARNING "at91-ssc: request_irq failure\n"); - - DBG("Stopping pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid); - return ret; - } - - ssc_p->initialized = 1; - } - - /* set SSC clock mode register */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->cmr_div); - - /* set receive clock mode and format */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, rcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, rfmr); - - /* set transmit clock mode and format */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, tcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, tfmr); - - DBG("hw_params: SSC initialized\n"); - return 0; -} - - -static int at91_ssc_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at91_pcm_dma_params *dma_params; - int dir; - - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - dma_params = ssc_p->dma_params[dir]; - - at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR, - dma_params->mask->ssc_enable); - - DBG("%s enabled SSC_SR=0x%08lx\n", dir ? "receive" : "transmit", - at91_ssc_read(dma_params->ssc_base + AT91_SSC_SR)); - return 0; -} - - -#ifdef CONFIG_PM -static int at91_ssc_suspend(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at91_ssc_info *ssc_p; - - if(!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* Save the status register before disabling transmit and receive. */ - ssc_p->ssc_state.ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, - AT91_SSC_TXDIS | AT91_SSC_RXDIS); - - /* Save the current interrupt mask, then disable unmasked interrupts. */ - ssc_p->ssc_state.ssc_imr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IDR, ssc_p->ssc_state.ssc_imr); - - ssc_p->ssc_state.ssc_cmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_CMR); - ssc_p->ssc_state.ssc_rcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RCMR); - ssc_p->ssc_state.ssc_rfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RFMR); - ssc_p->ssc_state.ssc_tcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TCMR); - ssc_p->ssc_state.ssc_tfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TFMR); - - return 0; -} - -static int at91_ssc_resume(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at91_ssc_info *ssc_p; - - if(!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, ssc_p->ssc_state.ssc_tfmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, ssc_p->ssc_state.ssc_tcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, ssc_p->ssc_state.ssc_rfmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, ssc_p->ssc_state.ssc_rcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->ssc_state.ssc_cmr); - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IER, ssc_p->ssc_state.ssc_imr); - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, - ((ssc_p->ssc_state.ssc_sr & AT91_SSC_RXENA) ? AT91_SSC_RXEN : 0) | - ((ssc_p->ssc_state.ssc_sr & AT91_SSC_TXENA) ? AT91_SSC_TXEN : 0)); - - return 0; -} - -#else -#define at91_ssc_suspend NULL -#define at91_ssc_resume NULL -#endif - -#define AT91_SSC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) - -#define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) - -struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = { - { .name = "at91-ssc0", - .id = 0, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[0].ssc, - }, -#if NUM_SSC_DEVICES == 3 - { .name = "at91-ssc1", - .id = 1, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[1].ssc, - }, - { .name = "at91-ssc2", - .id = 2, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[2].ssc, - }, -#endif -}; - -EXPORT_SYMBOL_GPL(at91_ssc_dai); - -/* Module information */ -MODULE_AUTHOR("Frank Mandarino, fmandarino@endrelia.com, www.endrelia.com"); -MODULE_DESCRIPTION("AT91 SSC ASoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h deleted file mode 100644 index 6b7bf382d06f..000000000000 --- a/sound/soc/at91/at91-ssc.h +++ /dev/null @@ -1,27 +0,0 @@ -/* - * at91-ssc.h - ALSA SSC interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino@endrelia.com> - * Endrelia Technologies Inc. - * Created: Jan 9, 2007 - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _AT91_SSC_H -#define _AT91_SSC_H - -/* SSC system clock ids */ -#define AT91_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ - -/* SSC divider ids */ -#define AT91SSC_CMR_DIV 0 /* MCK divider for BCLK */ -#define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ -#define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ - -extern struct snd_soc_dai at91_ssc_dai[]; - -#endif /* _AT91_SSC_H */ - diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig new file mode 100644 index 000000000000..a608d7009dbd --- /dev/null +++ b/sound/soc/atmel/Kconfig @@ -0,0 +1,43 @@ +config SND_ATMEL_SOC + tristate "SoC Audio for the Atmel System-on-Chip" + depends on ARCH_AT91 || AVR32 + help + Say Y or M if you want to add support for codecs attached to + the ATMEL SSC interface. You will also need + to select the audio interfaces to support below. + +config SND_ATMEL_SOC_SSC + tristate + depends on SND_ATMEL_SOC + help + Say Y or M if you want to add support for codecs the + ATMEL SSC interface. You will also needs to select the individual + machine drivers to support below. + +config SND_AT91_SOC_SAM9G20_WM8731 + tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" + depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8731 + help + Say Y if you want to add support for SoC audio on WM8731-based + AT91sam9g20 evaluation board. + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_ATMEL_SOC && BOARD_PLAYPAQ + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile new file mode 100644 index 000000000000..f54a7cc68e66 --- /dev/null +++ b/sound/soc/atmel/Makefile @@ -0,0 +1,15 @@ +# AT91 Platform Support +snd-soc-atmel-pcm-objs := atmel-pcm.o +snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o + +obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o + +# AT91 Machine Support +snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c new file mode 100644 index 000000000000..1fac5efd285b --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.c @@ -0,0 +1,494 @@ +/* + * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino@endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware atmel_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct atmel_runtime_data { + struct atmel_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of dma buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* PDC register save */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = atmel_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + pr_debug("atmel-pcm:" + "preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *) buf->area, + (void *) buf->addr, + size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void atmel_pcm_dma_irq(u32 ssc_sr, + struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + static int count; + + count++; + + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("atmel-pcm: buffer %s on %s" + " (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "underrun" : "overrun", + params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + snd_pcm_period_elapsed(substream); +} + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params */ + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = atmel_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("atmel-pcm: " + "hw_params: DMA for %s initialized " + "(dma_bytes=%u, period_size=%u)\n", + prtd->params->name, + runtime->dma_bytes, + prtd->period_size); + return 0; +} + +static int atmel_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + +static int atmel_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + return 0; +} + +static int atmel_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct atmel_runtime_data *prtd = rtd->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("atmel-pcm:buffer_size = %ld," + "dma_area = %p, dma_bytes = %u\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("atmel-pcm: trigger: " + "period_ptr=%lx, xpr=%u, " + "xcr=%u, xnpr=%u, xncr=%u\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%u imr=%u\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t atmel_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + +static int atmel_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &atmel_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(struct atmel_runtime_data), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + out: + return ret; +} + +static int atmel_pcm_close(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + +static int atmel_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +struct snd_pcm_ops atmel_pcm_ops = { + .open = atmel_pcm_open, + .close = atmel_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_pcm_hw_params, + .hw_free = atmel_pcm_hw_free, + .prepare = atmel_pcm_prepare, + .trigger = atmel_pcm_trigger, + .pointer = atmel_pcm_pointer, + .mmap = atmel_pcm_mmap, +}; + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 atmel_pcm_dmamask = 0xffffffff; + +static int atmel_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &atmel_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm:" + "Allocating PCM capture DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +#ifdef CONFIG_PM +static int atmel_pcm_suspend(struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* disable the PDC and save the PDC registers */ + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + +static int atmel_pcm_resume(struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else +#define atmel_pcm_suspend NULL +#define atmel_pcm_resume NULL +#endif + +struct snd_soc_platform atmel_soc_platform = { + .name = "atmel-audio", + .pcm_ops = &atmel_pcm_ops, + .pcm_new = atmel_pcm_new, + .pcm_free = atmel_pcm_free_dma_buffers, + .suspend = atmel_pcm_suspend, + .resume = atmel_pcm_resume, +}; +EXPORT_SYMBOL_GPL(atmel_soc_platform); + +static int __init atmel_pcm_modinit(void) +{ + return snd_soc_register_platform(&atmel_soc_platform); +} +module_init(atmel_pcm_modinit); + +static void __exit atmel_pcm_modexit(void) +{ + snd_soc_unregister_platform(&atmel_soc_platform); +} +module_exit(atmel_pcm_modexit); + +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou@atmel.com>"); +MODULE_DESCRIPTION("Atmel PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h new file mode 100644 index 000000000000..ec9b2824b663 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.h @@ -0,0 +1,86 @@ +/* + * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino@endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_PCM_H +#define _ATMEL_PCM_H + +#include <linux/atmel-ssc.h> + +/* + * Registers and status bits that are required by the PCM driver. + */ +struct atmel_pdc_regs { + unsigned int xpr; /* PDC recv/trans pointer */ + unsigned int xcr; /* PDC recv/trans counter */ + unsigned int xnpr; /* PDC next recv/trans pointer */ + unsigned int xncr; /* PDC next recv/trans counter */ + unsigned int ptcr; /* PDC transfer control */ +}; + +struct atmel_ssc_mask { + u32 ssc_enable; /* SSC recv/trans enable */ + u32 ssc_disable; /* SSC recv/trans disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ + u32 pdc_enable; /* PDC recv/trans enable */ + u32 pdc_disable; /* PDC recv/trans disable */ +}; + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct atmel_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct atmel_pdc_regs *pdc; /* PDC receive or transmit registers */ + struct atmel_ssc_mask *mask; /* SSC & PDC status bits */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler)(u32, struct snd_pcm_substream *); +}; + +extern struct snd_soc_platform atmel_soc_platform; + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* _ATMEL_PCM_H */ diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c new file mode 100644 index 000000000000..c5d67900d666 --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -0,0 +1,790 @@ +/* + * atmel_ssc_dai.c -- ALSA SoC ATMEL SSC Audio Layer Platform driver + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou@atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino@endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/atmel_pdc.h> + +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) +#define NUM_SSC_DEVICES 1 +#else +#define NUM_SSC_DEVICES 3 +#endif + +/* + * SSC PDC registers required by the PCM DMA engine. + */ +static struct atmel_pdc_regs pdc_tx_reg = { + .xpr = ATMEL_PDC_TPR, + .xcr = ATMEL_PDC_TCR, + .xnpr = ATMEL_PDC_TNPR, + .xncr = ATMEL_PDC_TNCR, +}; + +static struct atmel_pdc_regs pdc_rx_reg = { + .xpr = ATMEL_PDC_RPR, + .xcr = ATMEL_PDC_RCR, + .xnpr = ATMEL_PDC_RNPR, + .xncr = ATMEL_PDC_RNCR, +}; + +/* + * SSC & PDC status bits for transmit and receive. + */ +static struct atmel_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = ATMEL_PDC_TXTEN, + .pdc_disable = ATMEL_PDC_TXTDIS, +}; + +static struct atmel_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = ATMEL_PDC_RXTEN, + .pdc_disable = ATMEL_PDC_RXTDIS, +}; + + +/* + * DMA parameters. + */ +static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + {{ + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#if NUM_SSC_DEVICES == 3 + {{ + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, + {{ + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#endif +}; + + +static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#if NUM_SSC_DEVICES == 3 + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#endif +}; + + +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA + * interrupt handler in the PCM driver. + */ +static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) +{ + struct atmel_ssc_info *ssc_p = dev_id; + struct atmel_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (unsigned long)ssc_readl(ssc_p->ssc->regs, SR) + & (unsigned long)ssc_readl(ssc_p->ssc->regs, IMR); + + /* + * Loop through the substreams attached to this SSC. If + * a DMA-related interrupt occurred on that substream, call + * the DMA interrupt handler function, if one has been + * registered in the dma_params structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + return IRQ_HANDLED; +} + + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int atmel_ssc_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", + ssc_readl(ssc_p->ssc->regs, SR)); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir_mask = SSC_DIR_MASK_PLAYBACK; + else + dir_mask = SSC_DIR_MASK_CAPTURE; + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir, dir_mask; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + if (dma_params != NULL) { + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n", + (dir ? "receive" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[dir] = NULL; + } + + dir_mask = 1 << dir; + + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + if (ssc_p->initialized) { + /* Shutdown the SSC clock. */ + pr_debug("atmel_ssc_dau: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Clear the SSC dividers */ + ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} + + +/* + * Record the DAI format for use in hw_params(). + */ +static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + +/* + * Record SSC clock dividers for use in hw_params(). + */ +static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case ATMEL_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value. + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else + if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case ATMEL_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case ATMEL_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* + * Configure the SSC. + */ +static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + int id = rtd->dai->cpu_dai->id; + struct atmel_ssc_info *ssc_p = &ssc_info[id]; + struct atmel_pcm_dma_params *dma_params; + int dir, channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + /* + * Currently, there is only one set of dma params for + * each direction. If more are added, this code will + * have to be changed to select the proper set. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = &ssc_dma_params[id][dir]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[dir] = dma_params; + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the pcm driver hw_params() + * function. It should not be used for other purposes + * as it is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + /* + * Determine sample size in bits and the PDC increment. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format"); + return -EINVAL; + } + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S + && bits > 16) { + printk(KERN_WARNING + "atmel_ssc_dai: sample size %d" + "is too large for I2S\n", bits); + return -EINVAL; + } + + /* + * Compute SSC register settings. + */ + switch (ssc_p->daifmt + & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated + * from the MCK divider, and the BCLK signal + * is output on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(RFMR_FSLEN, (bits - 1)) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(TFMR_FSLEN, (bits - 1)) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clocks. + * + * The SSC transmit clock is obtained from the BCLK signal on + * on the TK line, and the SSC receive clock is + * generated from the transmit clock. + * + * For single channel data, one sample is transferred + * on the falling edge of the LRC clock. + * For two channel data, one sample is + * transferred on both edges of the LRC clock. + */ + start_event = ((channels == 1) + ? SSC_START_FALLING_RF + : SSC_START_EDGE_RF); + + rcmr = SSC_BF(RCMR_PERIOD, 0) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_CLOCK); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, 0) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, 0) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_NONE) + | SSC_BF(TCMR_CKS, SSC_CKS_PIN); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, 0) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output + * on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, SSC_START_RISING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, SSC_START_RISING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_RISING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("atmel_ssc_hw_params: " + "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + if (!ssc_p->initialized) { + + /* Enable PMC peripheral clock for this SSC */ + pr_debug("atmel_ssc_dai: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + printk(KERN_WARNING + "atmel_ssc_dai: request_irq failure\n"); + pr_debug("Atmel_ssc_dai: Stoping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("atmel_ssc_dai,hw_params: SSC initialized\n"); + return 0; +} + + +static int atmel_ssc_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + + pr_debug("%s enabled SSC_SR=0x%08x\n", + dir ? "receive" : "transmit", + ssc_readl(ssc_p->ssc->regs, SR)); + return 0; +} + + +#ifdef CONFIG_PM +static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define atmel_ssc_suspend NULL +# define atmel_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) + +#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { + { .name = "atmel-ssc0", + .id = 0, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params, + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[0], + }, +#if NUM_SSC_DEVICES == 3 + { .name = "atmel-ssc1", + .id = 1, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params, + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[1], + }, + { .name = "atmel-ssc2", + .id = 2, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params, + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[2], + }, +#endif +}; +EXPORT_SYMBOL_GPL(atmel_ssc_dai); + +static int __init atmel_ssc_modinit(void) +{ + return snd_soc_register_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai)); +} +module_init(atmel_ssc_modinit); + +static void __exit atmel_ssc_modexit(void) +{ + snd_soc_unregister_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai)); +} +module_exit(atmel_ssc_modexit); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou@atmel.com, www.atmel.com"); +MODULE_DESCRIPTION("ATMEL SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h new file mode 100644 index 000000000000..a828746e8a2f --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -0,0 +1,121 @@ +/* + * atmel_ssc_dai.h - ALSA SSC interface for the Atmel SoC + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou@atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino@endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_SSC_DAI_H +#define _ATMEL_SSC_DAI_H + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "atmel-pcm.h" + +/* SSC system clock ids */ +#define ATMEL_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ + +/* SSC divider ids */ +#define ATMEL_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define ATMEL_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define ATMEL_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + +struct atmel_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + +struct atmel_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct atmel_pcm_dma_params *dma_params[2]; + struct atmel_ssc_state ssc_state; +}; +extern struct snd_soc_dai atmel_ssc_dai[]; + +#endif /* _AT91_SSC_DAI_H */ diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index b1966e4dfcd3..43dd8cee83c6 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -22,7 +22,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/errno.h> #include <linux/clk.h> @@ -40,8 +39,8 @@ #include <mach/portmux.h> #include "../codecs/wm8510.h" -#include "at32-pcm.h" -#include "at32-ssc.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" /*-------------------------------------------------------------------------*\ @@ -362,8 +361,9 @@ static struct snd_soc_dai_link playpaq_wm8510_dai = { -static struct snd_soc_machine snd_soc_machine_playpaq = { +static struct snd_soc_card snd_soc_playpaq = { .name = "LRS_PlayPaq_WM8510", + .platform = &at32_soc_platform, .dai_link = &playpaq_wm8510_dai, .num_links = 1, }; @@ -378,8 +378,7 @@ static struct wm8510_setup_data playpaq_wm8510_setup = { static struct snd_soc_device playpaq_wm8510_snd_devdata = { - .machine = &snd_soc_machine_playpaq, - .platform = &at32_soc_platform, + .card = &snd_soc_playpaq, .codec_dev = &soc_codec_dev_wm8510, .codec_data = &playpaq_wm8510_setup, }; diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c new file mode 100644 index 000000000000..1fb59a9d3719 --- /dev/null +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -0,0 +1,328 @@ +/* + * sam9g20_wm8731 -- SoC audio for AT91SAM9G20-based + * ATMEL AT91SAM9G20ek board. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com> + * + * Based on ati_b1_wm8731.c by: + * Frank Mandarino <fmandarino@endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * Based on corgi.c by: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/hardware.h> +#include <mach/gpio.h> + +#include "../codecs/wm8731.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + + /* codec system clock is supplied by PCK0, set to 12MHz */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + + dev_dbg(rtd->socdev->dev, "shutdown"); +} + +static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct atmel_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + int ret; + + unsigned int rate; + int cmr_div, period; + + if (ssc == NULL) { + printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* + * The SSC clock dividers depend on the sample rate. The CMR.DIV + * field divides the system master clock MCK to drive the SSC TK + * signal which provides the codec BCLK. The TCMR.PERIOD and + * RCMR.PERIOD fields further divide the BCLK signal to drive + * the SSC TF and RF signals which provide the codec DACLRC and + * ADCLRC clocks. + * + * The dividers were determined through trial and error, where a + * CMR.DIV value is chosen such that the resulting BCLK value is + * divisible, or almost divisible, by (2 * sample rate), and then + * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. + */ + rate = params_rate(params); + + switch (rate) { + case 8000: + cmr_div = 55; /* BCLK = 133MHz/(2*55) = 1.209MHz */ + period = 74; /* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */ + break; + case 11025: + cmr_div = 67; /* BCLK = 133MHz/(2*60) = 1.108MHz */ + period = 45; /* LRC = BCLK/(2*(49+1)) = 11083,3Hz */ + break; + case 16000: + cmr_div = 63; /* BCLK = 133MHz/(2*63) = 1.055MHz */ + period = 32; /* LRC = BCLK/(2*(32+1)) = 15993,2Hz */ + break; + case 22050: + cmr_div = 52; /* BCLK = 133MHz/(2*52) = 1.278MHz */ + period = 28; /* LRC = BCLK/(2*(28+1)) = 22049Hz */ + break; + case 32000: + cmr_div = 66; /* BCLK = 133MHz/(2*66) = 1.007MHz */ + period = 15; /* LRC = BCLK/(2*(15+1)) = 31486,742Hz */ + break; + case 44100: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 25; /* LRC = BCLK/(2*(25+1)) = 44098Hz */ + break; + case 48000: + cmr_div = 33; /* BCLK = 133MHz/(2*33) = 2.015MHz */ + period = 20; /* LRC = BCLK/(2*(20+1)) = 47979,79Hz */ + break; + case 88200: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 12; /* LRC = BCLK/(2*(12+1)) = 88196Hz */ + break; + case 96000: + cmr_div = 23; /* BCLK = 133MHz/(2*23) = 2.891MHz */ + period = 14; /* LRC = BCLK/(2*(14+1)) = 96376Hz */ + break; + default: + printk(KERN_WARNING "unsupported rate %d" + " on at91sam9g20ek board\n", rate); + return -EINVAL; + } + + /* set the MCK divider for BCLK */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set the BCLK divider for DACLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_TCMR_PERIOD, period); + } else { + /* set the BCLK divider for ADCLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_RCMR_PERIOD, period); + } + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops at91sam9g20ek_ops = { + .startup = at91sam9g20ek_startup, + .hw_params = at91sam9g20ek_hw_params, + .shutdown = at91sam9g20ek_shutdown, +}; + + +static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + /* speaker connected to LHPOUT */ + {"Ext Spk", NULL, "LHPOUT"}, + + /* mic is connected to Mic Jack, with WM8731 Mic Bias */ + {"MICIN", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Int Mic"}, +}; + +/* + * Logic for a wm8731 as connected on a at91sam9g20ek board. + */ +static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec) +{ + printk(KERN_DEBUG + "at91sam9g20ek_wm8731 " + ": at91sam9g20ek_wm8731_init() called\n"); + + /* Add specific widgets */ + snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets, + ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); + /* Set up specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + /* not connected */ + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link at91sam9g20ek_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &wm8731_dai, + .init = at91sam9g20ek_wm8731_init, + .ops = &at91sam9g20ek_ops, +}; + +static struct snd_soc_card snd_soc_at91sam9g20ek = { + .name = "WM8731", + .platform = &atmel_soc_platform, + .dai_link = &at91sam9g20ek_dai, + .num_links = 1, +}; + +static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = { + .i2c_bus = 0, + .i2c_address = 0x1b, +}; + +static struct snd_soc_device at91sam9g20ek_snd_devdata = { + .card = &snd_soc_at91sam9g20ek, + .codec_dev = &soc_codec_dev_wm8731, + .codec_data = &at91sam9g20ek_wm8731_setup, +}; + +static struct platform_device *at91sam9g20ek_snd_device; + +static int __init at91sam9g20ek_init(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + int ret; + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); + if (!at91sam9g20ek_snd_device) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + ret = -ENOMEM; + } + + platform_set_drvdata(at91sam9g20ek_snd_device, + &at91sam9g20ek_snd_devdata); + at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev; + + ret = platform_device_add(at91sam9g20ek_snd_device); + if (ret) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + platform_device_put(at91sam9g20ek_snd_device); + } + + return ret; + +err_ssc: + return ret; +} + +static void __exit at91sam9g20ek_exit(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + platform_device_unregister(at91sam9g20ek_snd_device); + at91sam9g20ek_snd_device = NULL; +} + +module_init(at91sam9g20ek_init); +module_exit(at91sam9g20ek_exit); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou@atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 1466d9328800..74c823d60f91 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -406,11 +406,12 @@ static int __init au1xpsc_audio_dbdma_init(void) { au1xpsc_audio_pcmdma[PCM_TX] = NULL; au1xpsc_audio_pcmdma[PCM_RX] = NULL; - return 0; + return snd_soc_register_platform(&au1xpsc_soc_platform); } static void __exit au1xpsc_audio_dbdma_exit(void) { + snd_soc_unregister_platform(&au1xpsc_soc_platform); } module_init(au1xpsc_audio_dbdma_init); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 57facbad6825..f0e30aec7f23 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -160,7 +160,8 @@ struct snd_ac97_bus_ops soc_ac97_ops = { EXPORT_SYMBOL_GPL(soc_ac97_ops); static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; @@ -210,7 +211,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, } static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, - int cmd) + int cmd, struct snd_soc_dai *dai) { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; @@ -313,8 +314,7 @@ static void au1xpsc_ac97_remove(struct platform_device *pdev, au1xpsc_ac97_workdata = NULL; } -static int au1xpsc_ac97_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai) { /* save interesting registers and disable PSC */ au1xpsc_ac97_workdata->pm[0] = @@ -328,8 +328,7 @@ static int au1xpsc_ac97_suspend(struct platform_device *pdev, return 0; } -static int au1xpsc_ac97_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) { /* restore PSC clock config */ au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE, @@ -345,7 +344,7 @@ static int au1xpsc_ac97_resume(struct platform_device *pdev, struct snd_soc_dai au1xpsc_ac97_dai = { .name = "au1xpsc_ac97", - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .probe = au1xpsc_ac97_probe, .remove = au1xpsc_ac97_remove, .suspend = au1xpsc_ac97_suspend, @@ -372,11 +371,12 @@ EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); static int __init au1xpsc_ac97_init(void) { au1xpsc_ac97_workdata = NULL; - return 0; + return snd_soc_register_dai(&au1xpsc_ac97_dai); } static void __exit au1xpsc_ac97_exit(void) { + snd_soc_unregister_dai(&au1xpsc_ac97_dai); } module_init(au1xpsc_ac97_init); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 9384702c7ebd..f916de4400ed 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -116,7 +116,8 @@ out: } static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; @@ -240,7 +241,8 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) return 0; } -static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; int ret, stype = SUBSTREAM_TYPE(substream); @@ -337,8 +339,7 @@ static void au1xpsc_i2s_remove(struct platform_device *pdev, au1xpsc_i2s_workdata = NULL; } -static int au1xpsc_i2s_suspend(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) +static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai) { /* save interesting register and disable PSC */ au1xpsc_i2s_workdata->pm[0] = @@ -352,8 +353,7 @@ static int au1xpsc_i2s_suspend(struct platform_device *pdev, return 0; } -static int au1xpsc_i2s_resume(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) +static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) { /* select I2S mode and PSC clock */ au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); @@ -369,7 +369,6 @@ static int au1xpsc_i2s_resume(struct platform_device *pdev, struct snd_soc_dai au1xpsc_i2s_dai = { .name = "au1xpsc_i2s", - .type = SND_SOC_DAI_I2S, .probe = au1xpsc_i2s_probe, .remove = au1xpsc_i2s_remove, .suspend = au1xpsc_i2s_suspend, @@ -389,8 +388,6 @@ struct snd_soc_dai au1xpsc_i2s_dai = { .ops = { .trigger = au1xpsc_i2s_trigger, .hw_params = au1xpsc_i2s_hw_params, - }, - .dai_ops = { .set_fmt = au1xpsc_i2s_set_fmt, }, }; @@ -399,11 +396,12 @@ EXPORT_SYMBOL(au1xpsc_i2s_dai); static int __init au1xpsc_i2s_init(void) { au1xpsc_i2s_workdata = NULL; - return 0; + return snd_soc_register_dai(&au1xpsc_i2s_dai); } static void __exit au1xpsc_i2s_exit(void) { + snd_soc_unregister_dai(&au1xpsc_i2s_dai); } module_init(au1xpsc_i2s_init); diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c index f75ae7f62c3d..27683eb7905e 100644 --- a/sound/soc/au1x/sample-ac97.c +++ b/sound/soc/au1x/sample-ac97.c @@ -42,14 +42,14 @@ static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = { .ops = NULL, }; -static struct snd_soc_machine au1xpsc_sample_ac97_machine = { +static struct snd_soc_card au1xpsc_sample_ac97_machine = { .name = "Au1xxx PSC AC97 Audio", .dai_link = &au1xpsc_sample_ac97_dai, .num_links = 1, }; static struct snd_soc_device au1xpsc_sample_ac97_devdata = { - .machine = &au1xpsc_sample_ac97_machine, + .card = &au1xpsc_sample_ac97_machine, .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ .codec_dev = &soc_codec_dev_ac97, }; diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index dc006206f622..0a2f8f9eff53 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -1,6 +1,6 @@ config SND_BF5XX_I2S tristate "SoC I2S Audio for the ADI BF5xx chip" - depends on BLACKFIN && SND_SOC + depends on BLACKFIN help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in I2S @@ -13,7 +13,6 @@ config SND_BF5XX_SOC_SSM2602 select SND_BF5XX_SOC_I2S select SND_SOC_SSM2602 select I2C - select I2C_BLACKFIN_TWI help Say Y if you want to add support for SoC audio on BF527-EZKIT. @@ -35,7 +34,7 @@ config SND_BFIN_AD73311_SE config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" - depends on BLACKFIN && SND_SOC + depends on BLACKFIN help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in slot 16 @@ -47,7 +46,7 @@ config SND_BF5XX_AC97 properly with this driver. This driver is known to work with the Analog Devices line of AC97 codecs. -config SND_MMAP_SUPPORT +config SND_BF5XX_MMAP_SUPPORT bool "Enable MMAP Support" depends on SND_BF5XX_AC97 default y @@ -55,9 +54,17 @@ config SND_MMAP_SUPPORT Say y if you want AC97 driver to support mmap mode. We introduce an intermediate buffer to simulate mmap. +config SND_BF5XX_MULTICHAN_SUPPORT + bool "Enable Multichannel Support" + depends on SND_BF5XX_AC97 + default n + help + Say y if you want AC97 driver to support up to 5.1 channel audio. + this mode will consume much more memory for DMA. + config SND_BF5XX_SOC_SPORT tristate - + config SND_BF5XX_SOC_I2S tristate select SND_BF5XX_SOC_SPORT @@ -80,7 +87,7 @@ config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" depends on (SND_BF5XX_I2S || SND_BF5XX_AC97) range 0 3 if BF54x - range 0 1 if (BF53x || BF561) + range 0 1 if !BF54x default 0 help Set the correct SPORT for sound chip. @@ -90,12 +97,13 @@ config SND_BF5XX_HAVE_COLD_RESET depends on SND_BF5XX_AC97 default y if BFIN548_EZKIT default n if !BFIN548_EZKIT - + config SND_BF5XX_RESET_GPIO_NUM int "Set a GPIO for cold reset" depends on SND_BF5XX_HAVE_COLD_RESET range 0 159 default 19 if BFIN548_EZKIT default 5 if BFIN537_STAMP + default 0 help Set the correct GPIO for RESET the sound chip. diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 25e50d2ea1ec..8067cfafa3a7 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -43,24 +43,34 @@ #include "bf5xx-ac97.h" #include "bf5xx-sport.h" -#if defined(CONFIG_SND_MMAP_SUPPORT) +static unsigned int ac97_chan_mask[] = { + SP_FL, /* Mono */ + SP_STEREO, /* Stereo */ + SP_2DOT1, /* 2.1*/ + SP_QUAD,/*Quadraquic*/ + SP_FL | SP_FR | SP_FC | SP_SL | SP_SR,/*5 channels */ + SP_5DOT1, /* 5.1 */ +}; + +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) static void bf5xx_mmap_copy(struct snd_pcm_substream *substream, snd_pcm_uframes_t count) { struct snd_pcm_runtime *runtime = substream->runtime; struct sport_device *sport = runtime->private_data; + unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1]; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - bf5xx_pcm_to_ac97( - (struct ac97_frame *)sport->tx_dma_buf + sport->tx_pos, - (__u32 *)runtime->dma_area + sport->tx_pos, count); + bf5xx_pcm_to_ac97((struct ac97_frame *)sport->tx_dma_buf + + sport->tx_pos, (__u16 *)runtime->dma_area + sport->tx_pos * + runtime->channels, count, chan_mask); sport->tx_pos += runtime->period_size; if (sport->tx_pos >= runtime->buffer_size) sport->tx_pos %= runtime->buffer_size; sport->tx_delay_pos = sport->tx_pos; } else { - bf5xx_ac97_to_pcm( - (struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos, - (__u32 *)runtime->dma_area + sport->rx_pos, count); + bf5xx_ac97_to_pcm((struct ac97_frame *)sport->rx_dma_buf + + sport->rx_pos, (__u16 *)runtime->dma_area + sport->rx_pos * + runtime->channels, count); sport->rx_pos += runtime->period_size; if (sport->rx_pos >= runtime->buffer_size) sport->rx_pos %= runtime->buffer_size; @@ -71,7 +81,7 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream, static void bf5xx_dma_irq(void *data) { struct snd_pcm_substream *pcm = data; -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) struct snd_pcm_runtime *runtime = pcm->runtime; struct sport_device *sport = runtime->private_data; bf5xx_mmap_copy(pcm, runtime->period_size); @@ -90,17 +100,14 @@ static void bf5xx_dma_irq(void *data) * The total rx/tx buffer is for ac97 frame to hold all pcm data * is 0x20000 * sizeof(struct ac97_frame) / 4. */ -#ifdef CONFIG_SND_MMAP_SUPPORT static const struct snd_pcm_hardware bf5xx_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BLOCK_TRANSFER, -#else -static const struct snd_pcm_hardware bf5xx_pcm_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER, #endif + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE, .period_bytes_min = 32, .period_bytes_max = 0x10000, @@ -123,10 +130,20 @@ static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) { +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - memset(runtime->dma_area, 0, runtime->buffer_size); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sport->once = 0; + if (runtime->dma_area) + memset(runtime->dma_area, 0, runtime->buffer_size); + memset(sport->tx_dma_buf, 0, runtime->buffer_size * + sizeof(struct ac97_frame)); + } else + memset(sport->rx_dma_buf, 0, runtime->buffer_size * + sizeof(struct ac97_frame)); +#endif snd_pcm_lib_free_pages(substream); return 0; } @@ -139,7 +156,7 @@ static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) /* An intermediate buffer is introduced for implementing mmap for * SPORT working in TMD mode(include AC97). */ -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { sport_set_tx_callback(sport, bf5xx_dma_irq, substream); sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods, @@ -173,24 +190,24 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) bf5xx_mmap_copy(substream, runtime->period_size); - snd_pcm_period_elapsed(substream); sport->tx_delay_pos = 0; +#endif sport_tx_start(sport); - } - else + } else sport_rx_start(sport); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) sport->tx_pos = 0; #endif sport_tx_stop(sport); } else { -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) sport->rx_pos = 0; #endif sport_rx_stop(sport); @@ -208,7 +225,7 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) struct sport_device *sport = runtime->private_data; unsigned int curr; -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) curr = sport->tx_delay_pos; else @@ -249,22 +266,7 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) return ret; } -static int bf5xx_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - - pr_debug("%s enter\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - sport->once = 0; - memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame)); - } else - memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame)); - - return 0; -} - -#ifdef CONFIG_SND_MMAP_SUPPORT +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { @@ -281,32 +283,29 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, void __user *buf, snd_pcm_uframes_t count) { struct snd_pcm_runtime *runtime = substream->runtime; - + unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1]; pr_debug("%s copy pos:0x%lx count:0x%lx\n", substream->stream ? "Capture" : "Playback", pos, count); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - bf5xx_pcm_to_ac97( - (struct ac97_frame *)runtime->dma_area + pos, - buf, count); + bf5xx_pcm_to_ac97((struct ac97_frame *)runtime->dma_area + pos, + (__u16 *)buf, count, chan_mask); else - bf5xx_ac97_to_pcm( - (struct ac97_frame *)runtime->dma_area + pos, - buf, count); + bf5xx_ac97_to_pcm((struct ac97_frame *)runtime->dma_area + pos, + (__u16 *)buf, count); return 0; } #endif struct snd_pcm_ops bf5xx_pcm_ac97_ops = { .open = bf5xx_pcm_open, - .close = bf5xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = bf5xx_pcm_hw_params, .hw_free = bf5xx_pcm_hw_free, .prepare = bf5xx_pcm_prepare, .trigger = bf5xx_pcm_trigger, .pointer = bf5xx_pcm_pointer, -#ifdef CONFIG_SND_MMAP_SUPPORT +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) .mmap = bf5xx_pcm_mmap, #else .copy = bf5xx_pcm_copy, @@ -344,7 +343,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) * Need to allocate local buffer when enable * MMAP for SPORT working in TMD mode (include AC97). */ -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (!sport_handle->tx_dma_buf) { sport_handle->tx_dma_buf = dma_alloc_coherent(NULL, \ @@ -381,7 +380,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) struct snd_pcm_substream *substream; struct snd_dma_buffer *buf; int stream; -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) size_t size = bf5xx_pcm_hardware.buffer_bytes_max * sizeof(struct ac97_frame) / 4; #endif @@ -395,7 +394,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) continue; dma_free_coherent(NULL, buf->bytes, buf->area, 0); buf->area = NULL; -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (sport_handle->tx_dma_buf) dma_free_coherent(NULL, size, \ @@ -452,6 +451,18 @@ struct snd_soc_platform bf5xx_ac97_soc_platform = { }; EXPORT_SYMBOL_GPL(bf5xx_ac97_soc_platform); +static int __init bfin_ac97_init(void) +{ + return snd_soc_register_platform(&bf5xx_ac97_soc_platform); +} +module_init(bfin_ac97_init); + +static void __exit bfin_ac97_exit(void) +{ + snd_soc_unregister_platform(&bf5xx_ac97_soc_platform); +} +module_exit(bfin_ac97_exit); + MODULE_AUTHOR("Cliff Cai"); MODULE_DESCRIPTION("ADI Blackfin AC97 PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 5e5aafb6485f..3be2be60576d 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -54,71 +54,103 @@ static int *cmd_count; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; -#if defined(CONFIG_BF54x) +static u16 sport_req[][7] = { + PIN_REQ_SPORT_0, +#ifdef PIN_REQ_SPORT_1 + PIN_REQ_SPORT_1, +#endif +#ifdef PIN_REQ_SPORT_2 + PIN_REQ_SPORT_2, +#endif +#ifdef PIN_REQ_SPORT_3 + PIN_REQ_SPORT_3, +#endif + }; + static struct sport_param sport_params[4] = { { .dma_rx_chan = CH_SPORT0_RX, .dma_tx_chan = CH_SPORT0_TX, - .err_irq = IRQ_SPORT0_ERR, + .err_irq = IRQ_SPORT0_ERROR, .regs = (struct sport_register *)SPORT0_TCR1, }, +#ifdef PIN_REQ_SPORT_1 { .dma_rx_chan = CH_SPORT1_RX, .dma_tx_chan = CH_SPORT1_TX, - .err_irq = IRQ_SPORT1_ERR, + .err_irq = IRQ_SPORT1_ERROR, .regs = (struct sport_register *)SPORT1_TCR1, }, +#endif +#ifdef PIN_REQ_SPORT_2 { .dma_rx_chan = CH_SPORT2_RX, .dma_tx_chan = CH_SPORT2_TX, - .err_irq = IRQ_SPORT2_ERR, + .err_irq = IRQ_SPORT2_ERROR, .regs = (struct sport_register *)SPORT2_TCR1, }, +#endif +#ifdef PIN_REQ_SPORT_3 { .dma_rx_chan = CH_SPORT3_RX, .dma_tx_chan = CH_SPORT3_TX, - .err_irq = IRQ_SPORT3_ERR, + .err_irq = IRQ_SPORT3_ERROR, .regs = (struct sport_register *)SPORT3_TCR1, } -}; -#else -static struct sport_param sport_params[2] = { - { - .dma_rx_chan = CH_SPORT0_RX, - .dma_tx_chan = CH_SPORT0_TX, - .err_irq = IRQ_SPORT0_ERROR, - .regs = (struct sport_register *)SPORT0_TCR1, - }, - { - .dma_rx_chan = CH_SPORT1_RX, - .dma_tx_chan = CH_SPORT1_TX, - .err_irq = IRQ_SPORT1_ERROR, - .regs = (struct sport_register *)SPORT1_TCR1, - } -}; #endif +}; -void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \ - size_t count) +void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src, + size_t count, unsigned int chan_mask) { while (count--) { - dst->ac97_tag = TAG_VALID | TAG_PCM; - (dst++)->ac97_pcm = *src++; + dst->ac97_tag = TAG_VALID; + if (chan_mask & SP_FL) { + dst->ac97_pcm_r = *src++; + dst->ac97_tag |= TAG_PCM_RIGHT; + } + if (chan_mask & SP_FR) { + dst->ac97_pcm_l = *src++; + dst->ac97_tag |= TAG_PCM_LEFT; + + } +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + if (chan_mask & SP_SR) { + dst->ac97_sl = *src++; + dst->ac97_tag |= TAG_PCM_SL; + } + if (chan_mask & SP_SL) { + dst->ac97_sr = *src++; + dst->ac97_tag |= TAG_PCM_SR; + } + if (chan_mask & SP_LFE) { + dst->ac97_lfe = *src++; + dst->ac97_tag |= TAG_PCM_LFE; + } + if (chan_mask & SP_FC) { + dst->ac97_center = *src++; + dst->ac97_tag |= TAG_PCM_CENTER; + } +#endif + dst++; } } EXPORT_SYMBOL(bf5xx_pcm_to_ac97); -void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \ +void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst, size_t count) { - while (count--) - *(dst++) = (src++)->ac97_pcm; + while (count--) { + *(dst++) = src->ac97_pcm_l; + *(dst++) = src->ac97_pcm_r; + src++; + } } EXPORT_SYMBOL(bf5xx_ac97_to_pcm); static unsigned int sport_tx_curr_frag(struct sport_device *sport) { - return sport->tx_curr_frag = sport_curr_offset_tx(sport) / \ + return sport->tx_curr_frag = sport_curr_offset_tx(sport) / sport->tx_fragsize; } @@ -130,7 +162,7 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data) sport_incfrag(sport, &nextfrag, 1); - nextwrite = (struct ac97_frame *)(sport->tx_buf + \ + nextwrite = (struct ac97_frame *)(sport->tx_buf + nextfrag * sport->tx_fragsize); pr_debug("sport->tx_buf:%p, nextfrag:0x%x nextwrite:%p, cmd_count:%d\n", sport->tx_buf, nextfrag, nextwrite, cmd_count[nextfrag]); @@ -237,8 +269,7 @@ struct snd_ac97_bus_ops soc_ac97_ops = { EXPORT_SYMBOL_GPL(soc_ac97_ops); #ifdef CONFIG_PM -static int bf5xx_ac97_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) { struct sport_device *sport = (struct sport_device *)dai->private_data; @@ -253,8 +284,7 @@ static int bf5xx_ac97_suspend(struct platform_device *pdev, return 0; } -static int bf5xx_ac97_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int bf5xx_ac97_resume(struct snd_soc_dai *dai) { int ret; struct sport_device *sport = @@ -297,20 +327,15 @@ static int bf5xx_ac97_resume(struct platform_device *pdev, static int bf5xx_ac97_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - int ret; -#if defined(CONFIG_BF54x) - u16 sport_req[][7] = {PIN_REQ_SPORT_0, PIN_REQ_SPORT_1, - PIN_REQ_SPORT_2, PIN_REQ_SPORT_3}; -#else - u16 sport_req[][7] = {PIN_REQ_SPORT_0, PIN_REQ_SPORT_1}; -#endif + int ret = 0; cmd_count = (int *)get_zeroed_page(GFP_KERNEL); if (cmd_count == NULL) return -ENOMEM; if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { pr_err("Requesting Peripherals failed\n"); - return -EFAULT; + ret = -EFAULT; + goto peripheral_err; } #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET @@ -318,54 +343,54 @@ static int bf5xx_ac97_probe(struct platform_device *pdev, if (gpio_request(CONFIG_SND_BF5XX_RESET_GPIO_NUM, "SND_AD198x RESET")) { pr_err("Failed to request GPIO_%d for reset\n", CONFIG_SND_BF5XX_RESET_GPIO_NUM); - peripheral_free_list(&sport_req[sport_num][0]); - return -1; + ret = -1; + goto gpio_err; } gpio_direction_output(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1); #endif sport_handle = sport_init(&sport_params[sport_num], 2, \ sizeof(struct ac97_frame), NULL); if (!sport_handle) { - peripheral_free_list(&sport_req[sport_num][0]); -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -#endif - return -ENODEV; + ret = -ENODEV; + goto sport_err; } /*SPORT works in TDM mode to simulate AC97 transfers*/ ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1); if (ret) { pr_err("SPORT is busy!\n"); - kfree(sport_handle); - peripheral_free_list(&sport_req[sport_num][0]); -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -#endif - return -EBUSY; + ret = -EBUSY; + goto sport_config_err; } ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1)); if (ret) { pr_err("SPORT is busy!\n"); - kfree(sport_handle); - peripheral_free_list(&sport_req[sport_num][0]); -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -#endif - return -EBUSY; + ret = -EBUSY; + goto sport_config_err; } ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1)); if (ret) { pr_err("SPORT is busy!\n"); - kfree(sport_handle); - peripheral_free_list(&sport_req[sport_num][0]); -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -#endif - return -EBUSY; + ret = -EBUSY; + goto sport_config_err; } + return 0; + +sport_config_err: + kfree(sport_handle); +sport_err: +#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET + gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); +#endif +gpio_err: + peripheral_free_list(&sport_req[sport_num][0]); +peripheral_err: + free_page((unsigned long)cmd_count); + cmd_count = NULL; + + return ret; } static void bf5xx_ac97_remove(struct platform_device *pdev, @@ -373,6 +398,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev, { free_page((unsigned long)cmd_count); cmd_count = NULL; + peripheral_free_list(&sport_req[sport_num][0]); #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); #endif @@ -381,7 +407,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev, struct snd_soc_dai bfin_ac97_dai = { .name = "bf5xx-ac97", .id = 0, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .probe = bf5xx_ac97_probe, .remove = bf5xx_ac97_remove, .suspend = bf5xx_ac97_suspend, @@ -389,7 +415,11 @@ struct snd_soc_dai bfin_ac97_dai = { .playback = { .stream_name = "AC97 Playback", .channels_min = 2, +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + .channels_max = 6, +#else .channels_max = 2, +#endif .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { @@ -401,6 +431,18 @@ struct snd_soc_dai bfin_ac97_dai = { }; EXPORT_SYMBOL_GPL(bfin_ac97_dai); +static int __init bfin_ac97_init(void) +{ + return snd_soc_register_dai(&bfin_ac97_dai); +} +module_init(bfin_ac97_init); + +static void __exit bfin_ac97_exit(void) +{ + snd_soc_unregister_dai(&bfin_ac97_dai); +} +module_exit(bfin_ac97_exit); + MODULE_AUTHOR("Roy Huang"); MODULE_DESCRIPTION("AC97 driver for ADI Blackfin"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h index 3f77cc558dc0..3f2a911fe0cb 100644 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ b/sound/soc/blackfin/bf5xx-ac97.h @@ -16,21 +16,46 @@ struct ac97_frame { u16 ac97_tag; /* slot 0 */ u16 ac97_addr; /* slot 1 */ u16 ac97_data; /* slot 2 */ - u32 ac97_pcm; /* slot 3 and 4: left and right pcm data */ + u16 ac97_pcm_l; /*slot 3:front left*/ + u16 ac97_pcm_r; /*slot 4:front left*/ +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + u16 ac97_mdm_l1; + u16 ac97_center; /*slot 6:center*/ + u16 ac97_sl; /*slot 7:surround left*/ + u16 ac97_sr; /*slot 8:surround right*/ + u16 ac97_lfe; /*slot 9:lfe*/ +#endif } __attribute__ ((packed)); +/* Speaker location */ +#define SP_FL 0x0001 +#define SP_FR 0x0010 +#define SP_FC 0x0002 +#define SP_LFE 0x0020 +#define SP_SL 0x0004 +#define SP_SR 0x0040 + +#define SP_STEREO (SP_FL | SP_FR) +#define SP_2DOT1 (SP_FL | SP_FR | SP_LFE) +#define SP_QUAD (SP_FL | SP_FR | SP_SL | SP_SR) +#define SP_5DOT1 (SP_FL | SP_FR | SP_FC | SP_LFE | SP_SL | SP_SR) + #define TAG_VALID 0x8000 #define TAG_CMD 0x6000 #define TAG_PCM_LEFT 0x1000 #define TAG_PCM_RIGHT 0x0800 -#define TAG_PCM (TAG_PCM_LEFT | TAG_PCM_RIGHT) +#define TAG_PCM_MDM_L1 0x0400 +#define TAG_PCM_CENTER 0x0200 +#define TAG_PCM_SL 0x0100 +#define TAG_PCM_SR 0x0080 +#define TAG_PCM_LFE 0x0040 extern struct snd_soc_dai bfin_ac97_dai; -void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \ - size_t count); +void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src, \ + size_t count, unsigned int chan_mask); -void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \ +void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst, \ size_t count); #endif diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index 124425d22320..d8f591273778 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -43,7 +43,7 @@ #include "bf5xx-ac97-pcm.h" #include "bf5xx-ac97.h" -static struct snd_soc_machine bf5xx_board; +static struct snd_soc_card bf5xx_board; static int bf5xx_board_startup(struct snd_pcm_substream *substream) { @@ -67,15 +67,15 @@ static struct snd_soc_dai_link bf5xx_board_dai = { .ops = &bf5xx_board_ops, }; -static struct snd_soc_machine bf5xx_board = { +static struct snd_soc_card bf5xx_board = { .name = "bf5xx-board", + .platform = &bf5xx_ac97_soc_platform, .dai_link = &bf5xx_board_dai, .num_links = 1, }; static struct snd_soc_device bf5xx_board_snd_devdata = { - .machine = &bf5xx_board, - .platform = &bf5xx_ac97_soc_platform, + .card = &bf5xx_board, .codec_dev = &soc_codec_dev_ad1980, }; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 622c9b909532..7f2a5e199075 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -65,7 +65,7 @@ #define GPIO_SE CONFIG_SND_BFIN_AD73311_SE -static struct snd_soc_machine bf5xx_ad73311; +static struct snd_soc_card bf5xx_ad73311; static int snd_ad73311_startup(void) { @@ -168,7 +168,7 @@ static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream, params_format(params)); /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; @@ -190,16 +190,16 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai = { .ops = &bf5xx_ad73311_ops, }; -static struct snd_soc_machine bf5xx_ad73311 = { +static struct snd_soc_card bf5xx_ad73311 = { .name = "bf5xx_ad73311", + .platform = &bf5xx_i2s_soc_platform, .probe = bf5xx_probe, .dai_link = &bf5xx_ad73311_dai, .num_links = 1, }; static struct snd_soc_device bf5xx_ad73311_snd_devdata = { - .machine = &bf5xx_ad73311, - .platform = &bf5xx_i2s_soc_platform, + .card = &bf5xx_ad73311, .codec_dev = &soc_codec_dev_ad73311, }; diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 61fccf925192..53d290b3ea47 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -283,6 +283,18 @@ struct snd_soc_platform bf5xx_i2s_soc_platform = { }; EXPORT_SYMBOL_GPL(bf5xx_i2s_soc_platform); +static int __init bfin_i2s_init(void) +{ + return snd_soc_register_platform(&bf5xx_i2s_soc_platform); +} +module_init(bfin_i2s_init); + +static void __exit bfin_i2s_exit(void) +{ + snd_soc_unregister_platform(&bf5xx_i2s_soc_platform); +} +module_exit(bfin_i2s_exit); + MODULE_AUTHOR("Cliff Cai"); MODULE_DESCRIPTION("ADI Blackfin I2S PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index e020c160ee44..d1d95d2393fe 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -132,7 +132,8 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return ret; } -static int bf5xx_i2s_startup(struct snd_pcm_substream *substream) +static int bf5xx_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); @@ -142,7 +143,8 @@ static int bf5xx_i2s_startup(struct snd_pcm_substream *substream) } static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { int ret = 0; @@ -193,7 +195,8 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream) +static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); bf5xx_i2s.counter--; @@ -219,16 +222,14 @@ static int bf5xx_i2s_probe(struct platform_device *pdev, return 0; } -static void bf5xx_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static void bf5xx_i2s_remove(struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); peripheral_free_list(&sport_req[sport_num][0]); } #ifdef CONFIG_PM -static int bf5xx_i2s_suspend(struct platform_device *dev, - struct snd_soc_dai *dai) +static int bf5xx_i2s_suspend(struct snd_soc_dai *dai) { struct sport_device *sport = (struct sport_device *)dai->private_data; @@ -289,7 +290,6 @@ static int bf5xx_i2s_resume(struct platform_device *pdev, struct snd_soc_dai bf5xx_i2s_dai = { .name = "bf5xx-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .probe = bf5xx_i2s_probe, .remove = bf5xx_i2s_remove, .suspend = bf5xx_i2s_suspend, @@ -307,13 +307,24 @@ struct snd_soc_dai bf5xx_i2s_dai = { .ops = { .startup = bf5xx_i2s_startup, .shutdown = bf5xx_i2s_shutdown, - .hw_params = bf5xx_i2s_hw_params,}, - .dai_ops = { + .hw_params = bf5xx_i2s_hw_params, .set_fmt = bf5xx_i2s_set_dai_fmt, }, }; EXPORT_SYMBOL_GPL(bf5xx_i2s_dai); +static int __init bfin_i2s_init(void) +{ + return snd_soc_register_dai(&bf5xx_i2s_dai); +} +module_init(bfin_i2s_init); + +static void __exit bfin_i2s_exit(void) +{ + snd_soc_unregister_dai(&bf5xx_i2s_dai); +} +module_exit(bfin_i2s_exit); + /* Module information */ MODULE_AUTHOR("Cliff Cai"); MODULE_DESCRIPTION("I2S driver for ADI Blackfin"); diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h index fcadcc081f7f..2e63dea73e9c 100644 --- a/sound/soc/blackfin/bf5xx-sport.h +++ b/sound/soc/blackfin/bf5xx-sport.h @@ -116,7 +116,7 @@ struct sport_device { void *err_data; unsigned char *tx_dma_buf; unsigned char *rx_dma_buf; -#ifdef CONFIG_SND_MMAP_SUPPORT +#ifdef CONFIG_SND_BF5XX_MMAP_SUPPORT dma_addr_t tx_dma_phy; dma_addr_t rx_dma_phy; int tx_pos;/*pcm sample count*/ diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index e15f67fd7769..bc0cdded7116 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -44,7 +44,7 @@ #include "bf5xx-i2s-pcm.h" #include "bf5xx-i2s.h" -static struct snd_soc_machine bf5xx_ssm2602; +static struct snd_soc_card bf5xx_ssm2602; static int bf5xx_ssm2602_startup(struct snd_pcm_substream *substream) { @@ -92,17 +92,17 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, */ /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; - ret = codec_dai->dai_ops.set_sysclk(codec_dai, SSM2602_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -135,15 +135,15 @@ static struct ssm2602_setup_data bf5xx_ssm2602_setup = { .i2c_address = 0x1b, }; -static struct snd_soc_machine bf5xx_ssm2602 = { +static struct snd_soc_card bf5xx_ssm2602 = { .name = "bf5xx_ssm2602", + .platform = &bf5xx_i2s_soc_platform, .dai_link = &bf5xx_ssm2602_dai, .num_links = 1, }; static struct snd_soc_device bf5xx_ssm2602_snd_devdata = { - .machine = &bf5xx_ssm2602, - .platform = &bf5xx_i2s_soc_platform, + .card = &bf5xx_ssm2602, .codec_dev = &soc_codec_dev_ssm2602, .codec_data = &bf5xx_ssm2602_setup, }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 38a0e3b620a7..c41289b5f586 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1,31 +1,40 @@ config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" - depends on I2C - select SPI - select SPI_MASTER - select SND_SOC_AD73311 - select SND_SOC_AK4535 - select SND_SOC_CS4270 - select SND_SOC_SSM2602 - select SND_SOC_TLV320AIC23 - select SND_SOC_TLV320AIC26 - select SND_SOC_TLV320AIC3X - select SND_SOC_UDA1380 - select SND_SOC_WM8510 - select SND_SOC_WM8580 - select SND_SOC_WM8731 - select SND_SOC_WM8750 - select SND_SOC_WM8753 - select SND_SOC_WM8900 - select SND_SOC_WM8903 - select SND_SOC_WM8971 - select SND_SOC_WM8990 + select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS + select SND_SOC_AD1980 if SND_SOC_AC97_BUS + select SND_SOC_AD73311 if I2C + select SND_SOC_AK4535 if I2C + select SND_SOC_CS4270 if I2C + select SND_SOC_PCM3008 + select SND_SOC_SSM2602 if I2C + select SND_SOC_TLV320AIC23 if I2C + select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TLV320AIC3X if I2C + select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_UDA134X + select SND_SOC_UDA1380 if I2C + select SND_SOC_WM8350 if MFD_WM8350 + select SND_SOC_WM8510 if (I2C || SPI_MASTER) + select SND_SOC_WM8580 if I2C + select SND_SOC_WM8728 if (I2C || SPI_MASTER) + select SND_SOC_WM8731 if (I2C || SPI_MASTER) + select SND_SOC_WM8750 if (I2C || SPI_MASTER) + select SND_SOC_WM8753 if (I2C || SPI_MASTER) + select SND_SOC_WM8900 if I2C + select SND_SOC_WM8903 if I2C + select SND_SOC_WM8971 if I2C + select SND_SOC_WM8990 if I2C + select SND_SOC_WM9712 if SND_SOC_AC97_BUS + select SND_SOC_WM9713 if SND_SOC_AC97_BUS help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine driver. Selecting this option will allow these drivers to be built without an explicit machine driver for test and development purposes. + Support for the bus types used to access the codecs to be built must + be selected separately. + If unsure select "N". @@ -60,6 +69,12 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 +config SND_SOC_L3 + tristate + +config SND_SOC_PCM3008 + tristate + config SND_SOC_SSM2602 tristate @@ -75,15 +90,29 @@ config SND_SOC_TLV320AIC3X tristate depends on I2C +config SND_SOC_TWL4030 + tristate + depends on TWL4030_CORE + +config SND_SOC_UDA134X + tristate + select SND_SOC_L3 + config SND_SOC_UDA1380 tristate +config SND_SOC_WM8350 + tristate + config SND_SOC_WM8510 tristate config SND_SOC_WM8580 tristate +config SND_SOC_WM8728 + tristate + config SND_SOC_WM8731 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 90f0a585fc70..c4ddc9aa2bbd 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,13 +3,19 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o +snd-soc-l3-objs := l3.o +snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-twl4030-objs := twl4030.o +snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o +snd-soc-wm8350-objs := wm8350.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8580-objs := wm8580.o +snd-soc-wm8728-objs := wm8728.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o @@ -25,13 +31,19 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o +obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o +obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o +obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index bd1ebdc6c86c..fb53e6511af2 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -24,7 +24,8 @@ #define AC97_VERSION "0.6" -static int ac97_prepare(struct snd_pcm_substream *substream) +static int ac97_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -42,7 +43,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream) struct snd_soc_dai ac97_dai = { .name = "AC97 HiFi", - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .playback = { .stream_name = "AC97 Playback", .channels_min = 1, @@ -113,7 +114,7 @@ static int ac97_soc_probe(struct platform_device *pdev) if (ret < 0) goto bus_err; - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) goto bus_err; return 0; diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 1397b8e06c0b..73fdbb4d4a3d 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -85,6 +85,9 @@ SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0), SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1), SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1), +SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1), +SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1), + SOC_ENUM("Capture Source", ad1980_cap_src), SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), @@ -142,10 +145,11 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, struct snd_soc_dai ad1980_dai = { .name = "AC97", + .ac97_control = 1, .playback = { .stream_name = "Playback", .channels_min = 2, - .channels_max = 2, + .channels_max = 6, .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { @@ -192,6 +196,7 @@ static int ad1980_soc_probe(struct platform_device *pdev) struct snd_soc_codec *codec; int ret = 0; u16 vendor_id2; + u16 ext_status; printk(KERN_INFO "AD1980 SoC Audio Codec\n"); @@ -234,7 +239,7 @@ static int ad1980_soc_probe(struct platform_device *pdev) ret = ad1980_reset(codec, 0); if (ret < 0) { - printk(KERN_ERR "AC97 link error\n"); + printk(KERN_ERR "Failed to reset AD1980: AC97 link error\n"); goto reset_err; } @@ -253,12 +258,19 @@ static int ad1980_soc_probe(struct platform_device *pdev) "supported\n"); } - ac97_write(codec, AC97_MASTER, 0x0000); /* unmute line out volume */ - ac97_write(codec, AC97_PCM, 0x0000); /* unmute PCM out volume */ - ac97_write(codec, AC97_REC_GAIN, 0x0000);/* unmute record volume */ + /* unmute captures and playbacks volume */ + ac97_write(codec, AC97_MASTER, 0x0000); + ac97_write(codec, AC97_PCM, 0x0000); + ac97_write(codec, AC97_REC_GAIN, 0x0000); + ac97_write(codec, AC97_CENTER_LFE_MASTER, 0x0000); + ac97_write(codec, AC97_SURROUND_MASTER, 0x0000); + + /*power on LFE/CENTER/Surround DACs*/ + ext_status = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); ad1980_add_controls(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register card\n"); goto reset_err; diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index 37af8607b00a..b09289a1e55a 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -8,14 +8,10 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * Revision history - * 25th Sep 2008 Initial version. */ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> @@ -68,7 +64,7 @@ static int ad73311_soc_probe(struct platform_device *pdev) goto pcm_err; } - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "ad73311: failed to register card\n"); goto register_err; @@ -102,6 +98,18 @@ struct snd_soc_codec_device soc_codec_dev_ad73311 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311); +static int __init ad73311_init(void) +{ + return snd_soc_register_dai(&ad73311_dai); +} +module_init(ad73311_init); + +static void __exit ad73311_exit(void) +{ + snd_soc_unregister_dai(&ad73311_dai); +} +module_exit(ad73311_exit); + MODULE_DESCRIPTION("ASoC ad73311 driver"); MODULE_AUTHOR("Cliff Cai "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 2a89b5888e11..81300d8d42ca 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -339,7 +339,8 @@ static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai, } static int ak4535_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -451,8 +452,6 @@ struct snd_soc_dai ak4535_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .ops = { .hw_params = ak4535_hw_params, - }, - .dai_ops = { .set_fmt = ak4535_set_dai_fmt, .digital_mute = ak4535_mute, .set_sysclk = ak4535_set_dai_sysclk, @@ -513,7 +512,7 @@ static int ak4535_init(struct snd_soc_device *socdev) ak4535_add_controls(codec); ak4535_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "ak4535: failed to register card\n"); goto card_err; @@ -689,6 +688,18 @@ struct snd_soc_codec_device soc_codec_dev_ak4535 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535); +static int __init ak4535_modinit(void) +{ + return snd_soc_register_dai(&ak4535_dai); +} +module_init(ak4535_modinit); + +static void __exit ak4535_exit(void) +{ + snd_soc_unregister_dai(&ak4535_dai); +} +module_exit(ak4535_exit); + MODULE_DESCRIPTION("Soc AK4535 driver"); MODULE_AUTHOR("Richard Purdie"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 0bbd94501d7e..f1aa0c34421c 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -360,13 +360,14 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, /* * Program the CS4270 with the given hardware parameters. * - * The .dai_ops functions are used to provide board-specific data, like + * The .ops functions are used to provide board-specific data, like * input frequencies, to this driver. This function takes that information, * combines it with the hardware parameters provided, and programs the * hardware accordingly. */ static int cs4270_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -450,6 +451,19 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, return ret; } + /* Disable automatic volume control. It's enabled by default, and + * it causes volume change commands to be delayed, sometimes until + * after playback has started. + */ + + reg = cs4270_read_reg_cache(codec, CS4270_TRANS); + reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO); + ret = cs4270_i2c_write(codec, CS4270_TRANS, reg); + if (ret < 0) { + printk(KERN_ERR "I2C write failed\n"); + return ret; + } + /* Thaw and power-up the codec */ ret = snd_soc_write(codec, CS4270_PWRCTL, 0); @@ -697,10 +711,10 @@ static int cs4270_probe(struct platform_device *pdev) if (codec->control_data) { /* Initialize codec ops */ cs4270_dai.ops.hw_params = cs4270_hw_params; - cs4270_dai.dai_ops.set_sysclk = cs4270_set_dai_sysclk; - cs4270_dai.dai_ops.set_fmt = cs4270_set_dai_fmt; + cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk; + cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt; #ifdef CONFIG_SND_SOC_CS4270_HWMUTE - cs4270_dai.dai_ops.digital_mute = cs4270_mute; + cs4270_dai.ops.digital_mute = cs4270_mute; #endif } else printk(KERN_INFO "cs4270: no I2C device found, " @@ -709,7 +723,7 @@ static int cs4270_probe(struct platform_device *pdev) printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n"); #endif - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "cs4270: failed to register card\n"); goto error_del_driver; @@ -760,6 +774,18 @@ struct snd_soc_codec_device soc_codec_device_cs4270 = { }; EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); +static int __init cs4270_init(void) +{ + return snd_soc_register_dai(&cs4270_dai); +} +module_init(cs4270_init); + +static void __exit cs4270_exit(void) +{ + snd_soc_unregister_dai(&cs4270_dai); +} +module_exit(cs4270_exit); + MODULE_AUTHOR("Timur Tabi <timur@freescale.com>"); MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/l3.c b/sound/soc/codecs/l3.c new file mode 100644 index 000000000000..5353af58862c --- /dev/null +++ b/sound/soc/codecs/l3.c @@ -0,0 +1,91 @@ +/* + * L3 code + * + * Copyright (C) 2008, Christian Pellegrin <chripell@evolware.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * + * based on: + * + * L3 bus algorithm module. + * + * Copyright (C) 2001 Russell King, All Rights Reserved. + * + * + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/delay.h> + +#include <sound/l3.h> + +/* + * Send one byte of data to the chip. Data is latched into the chip on + * the rising edge of the clock. + */ +static void sendbyte(struct l3_pins *adap, unsigned int byte) +{ + int i; + + for (i = 0; i < 8; i++) { + adap->setclk(0); + udelay(adap->data_hold); + adap->setdat(byte & 1); + udelay(adap->data_setup); + adap->setclk(1); + udelay(adap->clock_high); + byte >>= 1; + } +} + +/* + * Send a set of bytes to the chip. We need to pulse the MODE line + * between each byte, but never at the start nor at the end of the + * transfer. + */ +static void sendbytes(struct l3_pins *adap, const u8 *buf, + int len) +{ + int i; + + for (i = 0; i < len; i++) { + if (i) { + udelay(adap->mode_hold); + adap->setmode(0); + udelay(adap->mode); + } + adap->setmode(1); + udelay(adap->mode_setup); + sendbyte(adap, buf[i]); + } +} + +int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len) +{ + adap->setclk(1); + adap->setdat(1); + adap->setmode(1); + udelay(adap->mode); + + adap->setmode(0); + udelay(adap->mode_setup); + sendbyte(adap, addr); + udelay(adap->mode_hold); + + sendbytes(adap, data, len); + + adap->setclk(1); + adap->setdat(1); + adap->setmode(0); + + return len; +} +EXPORT_SYMBOL_GPL(l3_write); + +MODULE_DESCRIPTION("L3 bit-banging driver"); +MODULE_AUTHOR("Christian Pellegrin <chripell@evolware.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c new file mode 100644 index 000000000000..9a3e67e5319c --- /dev/null +++ b/sound/soc/codecs/pcm3008.c @@ -0,0 +1,212 @@ +/* + * ALSA Soc PCM3008 codec support + * + * Author: Hugo Villeneuve + * Copyright (C) 2008 Lyrtech inc + * + * Based on AC97 Soc codec, original copyright follow: + * Copyright 2005 Wolfson Microelectronics PLC. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Generic PCM3008 support. + */ + +#include <linux/init.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "pcm3008.h" + +#define PCM3008_VERSION "0.2" + +#define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +struct snd_soc_dai pcm3008_dai = { + .name = "PCM3008 HiFi", + .playback = { + .stream_name = "PCM3008 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = PCM3008_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "PCM3008 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = PCM3008_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; +EXPORT_SYMBOL_GPL(pcm3008_dai); + +static void pcm3008_gpio_free(struct pcm3008_setup_data *setup) +{ + gpio_free(setup->dem0_pin); + gpio_free(setup->dem1_pin); + gpio_free(setup->pdad_pin); + gpio_free(setup->pdda_pin); +} + +static int pcm3008_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct pcm3008_setup_data *setup = socdev->codec_data; + int ret = 0; + + printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (!socdev->codec) + return -ENOMEM; + + codec = socdev->codec; + mutex_init(&codec->mutex); + + codec->name = "PCM3008"; + codec->owner = THIS_MODULE; + codec->dai = &pcm3008_dai; + codec->num_dai = 1; + codec->write = NULL; + codec->read = NULL; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + /* Register PCMs. */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "pcm3008: failed to create pcms\n"); + goto pcm_err; + } + + /* Register Card. */ + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "pcm3008: failed to register card\n"); + goto card_err; + } + + /* DEM1 DEM0 DE-EMPHASIS_MODE + * Low Low De-emphasis 44.1 kHz ON + * Low High De-emphasis OFF + * High Low De-emphasis 48 kHz ON + * High High De-emphasis 32 kHz ON + */ + + /* Configure DEM0 GPIO (turning OFF DAC De-emphasis). */ + ret = gpio_request(setup->dem0_pin, "codec_dem0"); + if (ret == 0) + ret = gpio_direction_output(setup->dem0_pin, 1); + if (ret != 0) + goto gpio_err; + + /* Configure DEM1 GPIO (turning OFF DAC De-emphasis). */ + ret = gpio_request(setup->dem1_pin, "codec_dem1"); + if (ret == 0) + ret = gpio_direction_output(setup->dem1_pin, 0); + if (ret != 0) + goto gpio_err; + + /* Configure PDAD GPIO. */ + ret = gpio_request(setup->pdad_pin, "codec_pdad"); + if (ret == 0) + ret = gpio_direction_output(setup->pdad_pin, 1); + if (ret != 0) + goto gpio_err; + + /* Configure PDDA GPIO. */ + ret = gpio_request(setup->pdda_pin, "codec_pdda"); + if (ret == 0) + ret = gpio_direction_output(setup->pdda_pin, 1); + if (ret != 0) + goto gpio_err; + + return ret; + +gpio_err: + pcm3008_gpio_free(setup); +card_err: + snd_soc_free_pcms(socdev); +pcm_err: + kfree(socdev->codec); + + return ret; +} + +static int pcm3008_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct pcm3008_setup_data *setup = socdev->codec_data; + + if (!codec) + return 0; + + pcm3008_gpio_free(setup); + snd_soc_free_pcms(socdev); + kfree(socdev->codec); + + return 0; +} + +#ifdef CONFIG_PM +static int pcm3008_soc_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct pcm3008_setup_data *setup = socdev->codec_data; + + gpio_set_value(setup->pdad_pin, 0); + gpio_set_value(setup->pdda_pin, 0); + + return 0; +} + +static int pcm3008_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct pcm3008_setup_data *setup = socdev->codec_data; + + gpio_set_value(setup->pdad_pin, 1); + gpio_set_value(setup->pdda_pin, 1); + + return 0; +} +#else +#define pcm3008_soc_suspend NULL +#define pcm3008_soc_resume NULL +#endif + +struct snd_soc_codec_device soc_codec_dev_pcm3008 = { + .probe = pcm3008_soc_probe, + .remove = pcm3008_soc_remove, + .suspend = pcm3008_soc_suspend, + .resume = pcm3008_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_pcm3008); + +static int __init pcm3008_init(void) +{ + return snd_soc_register_dai(&pcm3008_dai); +} +module_init(pcm3008_init); + +static void __exit pcm3008_exit(void) +{ + snd_soc_unregister_dai(&pcm3008_dai); +} +module_exit(pcm3008_exit); + +MODULE_DESCRIPTION("Soc PCM3008 driver"); +MODULE_AUTHOR("Hugo Villeneuve"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm3008.h b/sound/soc/codecs/pcm3008.h new file mode 100644 index 000000000000..d04e87d3c060 --- /dev/null +++ b/sound/soc/codecs/pcm3008.h @@ -0,0 +1,25 @@ +/* + * PCM3008 ALSA SoC Layer + * + * Author: Hugo Villeneuve + * Copyright (C) 2008 Lyrtech inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_SOC_PCM3008_H +#define __LINUX_SND_SOC_PCM3008_H + +struct pcm3008_setup_data { + unsigned dem0_pin; + unsigned dem1_pin; + unsigned pdad_pin; + unsigned pdda_pin; +}; + +extern struct snd_soc_codec_device soc_codec_dev_pcm3008; +extern struct snd_soc_dai pcm3008_dai; + +#endif diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 44ef0dacd564..cac373616768 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -285,16 +285,23 @@ static inline int get_coeff(int mclk, int rate) } static int ssm2602_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { u16 srate; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct ssm2602_priv *ssm2602 = codec->private_data; + struct i2c_client *i2c = codec->control_data; u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; int i = get_coeff(ssm2602->sysclk, params_rate(params)); + if (substream == ssm2602->slave_substream) { + dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n"); + return 0; + } + /*no match is found*/ if (i == ARRAY_SIZE(coeff_div)) return -EINVAL; @@ -324,19 +331,26 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ssm2602_startup(struct snd_pcm_substream *substream) +static int ssm2602_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct ssm2602_priv *ssm2602 = codec->private_data; + struct i2c_client *i2c = codec->control_data; struct snd_pcm_runtime *master_runtime; /* The DAI has shared clocks so if we already have a playback or * capture going then constrain this substream to match it. + * TODO: the ssm2602 allows pairs of non-matching PB/REC rates */ if (ssm2602->master_substream) { master_runtime = ssm2602->master_substream->runtime; + dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", + master_runtime->sample_bits, + master_runtime->rate); + snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, master_runtime->rate, @@ -354,7 +368,8 @@ static int ssm2602_startup(struct snd_pcm_substream *substream) return 0; } -static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream) +static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -365,14 +380,21 @@ static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream) return 0; } -static void ssm2602_shutdown(struct snd_pcm_substream *substream) +static void ssm2602_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; /* deactivate */ if (!codec->active) ssm2602_write(codec, SSM2602_ACTIVE, 0); + + if (ssm2602->master_substream == substream) + ssm2602->master_substream = ssm2602->slave_substream; + + ssm2602->slave_substream = NULL; } static int ssm2602_mute(struct snd_soc_dai *dai, int mute) @@ -432,10 +454,10 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0001; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x0013; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x0003; break; default: return -EINVAL; @@ -496,6 +518,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ SNDRV_PCM_RATE_96000) +#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + struct snd_soc_dai ssm2602_dai = { .name = "SSM2602", .playback = { @@ -503,20 +528,18 @@ struct snd_soc_dai ssm2602_dai = { .channels_min = 2, .channels_max = 2, .rates = SSM2602_RATES, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .formats = SSM2602_FORMATS,}, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SSM2602_RATES, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .formats = SSM2602_FORMATS,}, .ops = { .startup = ssm2602_startup, .prepare = ssm2602_pcm_prepare, .hw_params = ssm2602_hw_params, .shutdown = ssm2602_shutdown, - }, - .dai_ops = { .digital_mute = ssm2602_mute, .set_sysclk = ssm2602_set_dai_sysclk, .set_fmt = ssm2602_set_dai_fmt, @@ -601,7 +624,7 @@ static int ssm2602_init(struct snd_soc_device *socdev) ssm2602_add_controls(codec); ssm2602_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { pr_err("ssm2602: failed to register card\n"); goto card_err; @@ -770,6 +793,18 @@ struct snd_soc_codec_device soc_codec_dev_ssm2602 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602); +static int __init ssm2602_modinit(void) +{ + return snd_soc_register_dai(&ssm2602_dai); +} +module_init(ssm2602_modinit); + +static void __exit ssm2602_exit(void) +{ + snd_soc_unregister_dai(&ssm2602_dai); +} +module_exit(ssm2602_exit); + MODULE_DESCRIPTION("ASoC ssm2602 driver"); MODULE_AUTHOR("Cliff Cai"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 44308dac9e18..cfdea007c4cb 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -37,12 +37,6 @@ #define AIC23_VERSION "0.1" -struct tlv320aic23_srate_reg_info { - u32 sample_rate; - u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ - u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ -}; - /* * AIC23 register cache */ @@ -261,20 +255,156 @@ static const struct snd_soc_dapm_route intercon[] = { }; -/* tlv320aic23 related */ -static const struct tlv320aic23_srate_reg_info srate_reg_info[] = { - {4000, 0x06, 1}, /* 4000 */ - {8000, 0x06, 0}, /* 8000 */ - {16000, 0x0C, 1}, /* 16000 */ - {22050, 0x11, 1}, /* 22050 */ - {24000, 0x00, 1}, /* 24000 */ - {32000, 0x0C, 0}, /* 32000 */ - {44100, 0x11, 0}, /* 44100 */ - {48000, 0x00, 0}, /* 48000 */ - {88200, 0x1F, 0}, /* 88200 */ - {96000, 0x0E, 0}, /* 96000 */ +/* AIC23 driver data */ +struct aic23 { + struct snd_soc_codec codec; + int mclk; + int requested_adc; + int requested_dac; +}; + +/* + * Common Crystals used + * 11.2896 Mhz /128 = *88.2k /192 = 58.8k + * 12.0000 Mhz /125 = *96k /136 = 88.235K + * 12.2880 Mhz /128 = *96k /192 = 64k + * 16.9344 Mhz /128 = 132.3k /192 = *88.2k + * 18.4320 Mhz /128 = 144k /192 = *96k + */ + +/* + * Normal BOSR 0-256/2 = 128, 1-384/2 = 192 + * USB BOSR 0-250/2 = 125, 1-272/2 = 136 + */ +static const int bosr_usb_divisor_table[] = { + 128, 125, 192, 136 +}; +#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7)) +#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15)) +static const unsigned short sr_valid_mask[] = { + LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/ + LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/ + LOWER_GROUP, /* Usb, bosr - 0*/ + UPPER_GROUP, /* Usb, bosr - 1*/ +}; +/* + * Every divisor is a factor of 11*12 + */ +#define SR_MULT (11*12) +#define A(x) (x) ? (SR_MULT/x) : 0 +static const unsigned char sr_adc_mult_table[] = { + A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1), + A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1) +}; +static const unsigned char sr_dac_mult_table[] = { + A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1), + A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1) }; +static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc, + int dac, int dac_l, int dac_h, int need_dac) +{ + if ((adc >= adc_l) && (adc <= adc_h) && + (dac >= dac_l) && (dac <= dac_h)) { + int diff_adc = need_adc - adc; + int diff_dac = need_dac - dac; + return abs(diff_adc) + abs(diff_dac); + } + return UINT_MAX; +} + +static int find_rate(int mclk, u32 need_adc, u32 need_dac) +{ + int i, j; + int best_i = -1; + int best_j = -1; + int best_div = 0; + unsigned best_score = UINT_MAX; + int adc_l, adc_h, dac_l, dac_h; + + need_adc *= SR_MULT; + need_dac *= SR_MULT; + /* + * rates given are +/- 1/32 + */ + adc_l = need_adc - (need_adc >> 5); + adc_h = need_adc + (need_adc >> 5); + dac_l = need_dac - (need_dac >> 5); + dac_h = need_dac + (need_dac >> 5); + for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) { + int base = mclk / bosr_usb_divisor_table[i]; + int mask = sr_valid_mask[i]; + for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table); + j++, mask >>= 1) { + int adc; + int dac; + int score; + if ((mask & 1) == 0) + continue; + adc = base * sr_adc_mult_table[j]; + dac = base * sr_dac_mult_table[j]; + score = get_score(adc, adc_l, adc_h, need_adc, + dac, dac_l, dac_h, need_dac); + if (best_score > score) { + best_score = score; + best_i = i; + best_j = j; + best_div = 0; + } + score = get_score((adc >> 1), adc_l, adc_h, need_adc, + (dac >> 1), dac_l, dac_h, need_dac); + /* prefer to have a /2 */ + if ((score != UINT_MAX) && (best_score >= score)) { + best_score = score; + best_i = i; + best_j = j; + best_div = 1; + } + } + } + return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT); +} + +#ifdef DEBUG +static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, + u32 *sample_rate_adc, u32 *sample_rate_dac) +{ + int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE); + int sr = (src >> 2) & 0x0f; + int val = (mclk / bosr_usb_divisor_table[src & 3]); + int adc = (val * sr_adc_mult_table[sr]) / SR_MULT; + int dac = (val * sr_dac_mult_table[sr]) / SR_MULT; + if (src & TLV320AIC23_CLKIN_HALF) { + adc >>= 1; + dac >>= 1; + } + *sample_rate_adc = adc; + *sample_rate_dac = dac; +} +#endif + +static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, + u32 sample_rate_adc, u32 sample_rate_dac) +{ + /* Search for the right sample rate */ + int data = find_rate(mclk, sample_rate_adc, sample_rate_dac); + if (data < 0) { + printk(KERN_ERR "%s:Invalid rate %u,%u requested\n", + __func__, sample_rate_adc, sample_rate_dac); + return -EINVAL; + } + tlv320aic23_write(codec, TLV320AIC23_SRATE, data); +#ifdef DEBUG + { + u32 adc, dac; + get_current_sample_rates(codec, mclk, &adc, &dac); + printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n", + adc, dac, data); + } +#endif + return 0; +} + static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) { snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, @@ -288,32 +418,36 @@ static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) } static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; - u16 iface_reg, data; - u8 count = 0; + u16 iface_reg; + int ret; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); + u32 sample_rate_adc = aic23->requested_adc; + u32 sample_rate_dac = aic23->requested_dac; + u32 sample_rate = params_rate(params); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + aic23->requested_dac = sample_rate_dac = sample_rate; + if (!sample_rate_adc) + sample_rate_adc = sample_rate; + } else { + aic23->requested_adc = sample_rate_adc = sample_rate; + if (!sample_rate_dac) + sample_rate_dac = sample_rate; + } + ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc, + sample_rate_dac); + if (ret < 0) + return ret; iface_reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); - - /* Search for the right sample rate */ - /* Verify what happens if the rate is not supported - * now it goes to 96Khz */ - while ((srate_reg_info[count].sample_rate != params_rate(params)) && - (count < ARRAY_SIZE(srate_reg_info))) { - count++; - } - - data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) | - (srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) | - TLV320AIC23_USB_CLK_ON; - - tlv320aic23_write(codec, TLV320AIC23_SRATE, data); - switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; @@ -332,7 +466,8 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, return 0; } -static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream) +static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -344,17 +479,23 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream) return 0; } -static void tlv320aic23_shutdown(struct snd_pcm_substream *substream) +static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); /* deactivate */ if (!codec->active) { udelay(50); tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + aic23->requested_dac = 0; + else + aic23->requested_adc = 0; } static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) @@ -400,7 +541,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_I2S: iface_reg |= TLV320AIC23_FOR_I2S; break; - case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: iface_reg |= TLV320AIC23_FOR_DSP; break; case SND_SOC_DAIFMT_RIGHT_J: @@ -422,12 +563,9 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; - - switch (freq) { - case 12000000: - return 0; - } - return -EINVAL; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); + aic23->mclk = freq; + return 0; } static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, @@ -478,12 +616,10 @@ struct snd_soc_dai tlv320aic23_dai = { .prepare = tlv320aic23_pcm_prepare, .hw_params = tlv320aic23_hw_params, .shutdown = tlv320aic23_shutdown, - }, - .dai_ops = { - .digital_mute = tlv320aic23_mute, - .set_fmt = tlv320aic23_set_dai_fmt, - .set_sysclk = tlv320aic23_set_dai_sysclk, - } + .digital_mute = tlv320aic23_mute, + .set_fmt = tlv320aic23_set_dai_fmt, + .set_sysclk = tlv320aic23_set_dai_sysclk, + } }; EXPORT_SYMBOL_GPL(tlv320aic23_dai); @@ -584,7 +720,7 @@ static int tlv320aic23_init(struct snd_soc_device *socdev) tlv320aic23_add_controls(codec); tlv320aic23_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "tlv320aic23: failed to register card\n"); goto card_err; @@ -659,14 +795,15 @@ static int tlv320aic23_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; + struct aic23 *aic23; int ret = 0; printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) + aic23 = kzalloc(sizeof(struct aic23), GFP_KERNEL); + if (aic23 == NULL) return -ENOMEM; - + codec = &aic23->codec; socdev->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); @@ -687,6 +824,7 @@ static int tlv320aic23_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); if (codec->control_data) tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -697,7 +835,7 @@ static int tlv320aic23_remove(struct platform_device *pdev) i2c_del_driver(&tlv320aic23_i2c_driver); #endif kfree(codec->reg_cache); - kfree(codec); + kfree(aic23); return 0; } @@ -709,6 +847,18 @@ struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23); +static int __init tlv320aic23_modinit(void) +{ + return snd_soc_register_dai(&tlv320aic23_dai); +} +module_init(tlv320aic23_modinit); + +static void __exit tlv320aic23_exit(void) +{ + snd_soc_unregister_dai(&tlv320aic23_dai); +} +module_exit(tlv320aic23_exit); + MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index bed8a9e63ddc..29f2f1a017fd 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -125,7 +125,8 @@ static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg, * Digital Audio Interface Operations */ static int aic26_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -287,8 +288,6 @@ struct snd_soc_dai aic26_dai = { }, .ops = { .hw_params = aic26_hw_params, - }, - .dai_ops = { .digital_mute = aic26_mute, .set_sysclk = aic26_set_sysclk, .set_fmt = aic26_set_fmt, @@ -360,7 +359,7 @@ static int aic26_probe(struct platform_device *pdev) /* CODEC is setup, we can register the card now */ dev_dbg(&pdev->dev, "Registering card\n"); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { dev_err(&pdev->dev, "aic26: failed to register card\n"); goto card_err; @@ -427,7 +426,7 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); static int aic26_spi_probe(struct spi_device *spi) { struct aic26 *aic26; - int rc, i, reg; + int ret, i, reg; dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n"); @@ -457,6 +456,14 @@ static int aic26_spi_probe(struct spi_device *spi) aic26->codec.reg_cache_size = AIC26_NUM_REGS; aic26->codec.reg_cache = aic26->reg_cache; + aic26_dai.dev = &spi->dev; + ret = snd_soc_register_dai(&aic26_dai); + if (ret != 0) { + dev_err(&spi->dev, "Failed to register DAI: %d\n", ret); + kfree(aic26); + return ret; + } + /* Reset the codec to power on defaults */ aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00); @@ -475,8 +482,8 @@ static int aic26_spi_probe(struct spi_device *spi) /* Register the sysfs files for debugging */ /* Create SysFS files */ - rc = device_create_file(&spi->dev, &dev_attr_keyclick); - if (rc) + ret = device_create_file(&spi->dev, &dev_attr_keyclick); + if (ret) dev_info(&spi->dev, "error creating sysfs files\n"); #if defined(CONFIG_SND_SOC_OF_SIMPLE) @@ -493,6 +500,7 @@ static int aic26_spi_remove(struct spi_device *spi) { struct aic26 *aic26 = dev_get_drvdata(&spi->dev); + snd_soc_unregister_dai(&aic26_dai); kfree(aic26); return 0; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index cff276ee261e..b47a749c5ea2 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -253,11 +253,17 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("Line DAC Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3, - 0x01, 0), - SOC_DOUBLE_R("Line PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, - PGAR_2_RLOPM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("Line Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, + SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0), + SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL, + DACR1_2_LLOPM_VOL, 0, 0x7f, 1), + SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, + 0, 0x7f, 1), + SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL, + 0, 0x7f, 1), + SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, + LINE2R_2_LLOPM_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL, LINE2R_2_RLOPM_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL, @@ -272,8 +278,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACR1_2_HPROUT_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), - SOC_DOUBLE_R("HP PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, + SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL, 0, 0x7f, 1), + SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, + 0, 0x7f, 1), + SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, + 0, 0x7f, 1), SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, 0, 0x7f, 1), @@ -281,8 +291,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACR1_2_HPRCOM_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), - SOC_DOUBLE_R("HPCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, - PGAR_2_HPRCOM_VOL, 0, 0x7f, 1), + SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, + 0, 0x7f, 1), + SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL, + 0, 0x7f, 1), SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1), @@ -333,7 +345,8 @@ SOC_DAPM_ENUM("Route", aic3x_enum[RHPCOM_ENUM]); /* Left DAC_L1 Mixer */ static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", DACL1_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", DACL1_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", DACL1_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", DACL1_2_MONOLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HP Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HPCOM Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0), @@ -341,7 +354,8 @@ static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = { /* Right DAC_R1 Mixer */ static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", DACR1_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", DACR1_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", DACR1_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", DACR1_2_MONOLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HP Switch", DACR1_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HPCOM Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0), @@ -350,14 +364,18 @@ static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = { /* Left PGA Mixer */ static const struct snd_kcontrol_new aic3x_left_pga_mixer_controls[] = { SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_LADC_CTRL, 3, 1, 1), SOC_DAPM_SINGLE_AIC3X("Line2L Switch", LINE2L_2_LADC_CTRL, 3, 1, 1), SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_LADC_CTRL, 0, 1, 1), }; /* Right PGA Mixer */ static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = { SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_RADC_CTRL, 3, 1, 1), SOC_DAPM_SINGLE_AIC3X("Line2R Switch", LINE2R_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_RADC_CTRL, 4, 1, 1), SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1), }; @@ -379,34 +397,42 @@ SOC_DAPM_ENUM("Route", aic3x_enum[LINE2R_ENUM]); /* Left PGA Bypass Mixer */ static const struct snd_kcontrol_new aic3x_left_pga_bp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", PGAL_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", PGAL_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", PGAL_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", PGAL_2_MONOLOPM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HP Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HPCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPL Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPR Switch", PGAL_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPLCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPRCOM Switch", PGAL_2_HPRCOM_VOL, 7, 1, 0), }; /* Right PGA Bypass Mixer */ static const struct snd_kcontrol_new aic3x_right_pga_bp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", PGAR_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", PGAR_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", PGAR_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", PGAR_2_MONOLOPM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HP Switch", PGAR_2_HPROUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HPCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPL Switch", PGAR_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPR Switch", PGAR_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPLCOM Switch", PGAR_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPRCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0), }; /* Left Line2 Bypass Mixer */ static const struct snd_kcontrol_new aic3x_left_line2_bp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", LINE2L_2_MONOLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HP Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HPCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPLCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), }; /* Right Line2 Bypass Mixer */ static const struct snd_kcontrol_new aic3x_right_line2_bp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", LINE2R_2_MONOLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HP Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HPCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPRCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), }; static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { @@ -439,22 +465,26 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { /* Mono Output */ SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), - /* Left Inputs to Left ADC */ + /* Inputs to Left ADC */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0), SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, &aic3x_left_pga_mixer_controls[0], ARRAY_SIZE(aic3x_left_pga_mixer_controls)), SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line1_mux_controls), + SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_line1_mux_controls), SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line2_mux_controls), - /* Right Inputs to Right ADC */ + /* Inputs to Right ADC */ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", LINE1R_2_RADC_CTRL, 2, 0), SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, &aic3x_right_pga_mixer_controls[0], ARRAY_SIZE(aic3x_right_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0, + &aic3x_right_line1_mux_controls), SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line1_mux_controls), SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, @@ -531,7 +561,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left DAC Mux", "DAC_L2", "Left DAC"}, {"Left DAC Mux", "DAC_L3", "Left DAC"}, - {"Left DAC_L1 Mixer", "Line Switch", "Left DAC Mux"}, + {"Left DAC_L1 Mixer", "LineL Switch", "Left DAC Mux"}, + {"Left DAC_L1 Mixer", "LineR Switch", "Left DAC Mux"}, {"Left DAC_L1 Mixer", "Mono Switch", "Left DAC Mux"}, {"Left DAC_L1 Mixer", "HP Switch", "Left DAC Mux"}, {"Left DAC_L1 Mixer", "HPCOM Switch", "Left DAC Mux"}, @@ -557,7 +588,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right DAC Mux", "DAC_R2", "Right DAC"}, {"Right DAC Mux", "DAC_R3", "Right DAC"}, - {"Right DAC_R1 Mixer", "Line Switch", "Right DAC Mux"}, + {"Right DAC_R1 Mixer", "LineL Switch", "Right DAC Mux"}, + {"Right DAC_R1 Mixer", "LineR Switch", "Right DAC Mux"}, {"Right DAC_R1 Mixer", "Mono Switch", "Right DAC Mux"}, {"Right DAC_R1 Mixer", "HP Switch", "Right DAC Mux"}, {"Right DAC_R1 Mixer", "HPCOM Switch", "Right DAC Mux"}, @@ -592,8 +624,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left Line2L Mux", "differential", "LINE2L"}, {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, + {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"}, {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, + {"Left PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Left ADC", NULL, "Left PGA Mixer"}, {"Left ADC", NULL, "GPIO1 dmic modclk"}, @@ -605,18 +639,23 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line2R Mux", "single-ended", "LINE2R"}, {"Right Line2R Mux", "differential", "LINE2R"}, + {"Right PGA Mixer", "Line1L Switch", "Right Line1L Mux"}, {"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"}, {"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"}, + {"Right PGA Mixer", "Mic3L Switch", "MIC3L"}, {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Right ADC", NULL, "Right PGA Mixer"}, {"Right ADC", NULL, "GPIO1 dmic modclk"}, /* Left PGA Bypass */ - {"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "LineL Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "LineR Switch", "Left PGA Mixer"}, {"Left PGA Bypass Mixer", "Mono Switch", "Left PGA Mixer"}, - {"Left PGA Bypass Mixer", "HP Switch", "Left PGA Mixer"}, - {"Left PGA Bypass Mixer", "HPCOM Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "HPL Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "HPR Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "HPLCOM Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "HPRCOM Switch", "Left PGA Mixer"}, {"Left HPCOM Mux", "differential of HPLOUT", "Left PGA Bypass Mixer"}, {"Left HPCOM Mux", "constant VCM", "Left PGA Bypass Mixer"}, @@ -627,10 +666,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left HP Out", NULL, "Left PGA Bypass Mixer"}, /* Right PGA Bypass */ - {"Right PGA Bypass Mixer", "Line Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "LineL Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "LineR Switch", "Right PGA Mixer"}, {"Right PGA Bypass Mixer", "Mono Switch", "Right PGA Mixer"}, - {"Right PGA Bypass Mixer", "HP Switch", "Right PGA Mixer"}, - {"Right PGA Bypass Mixer", "HPCOM Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "HPL Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "HPR Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "HPLCOM Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "HPRCOM Switch", "Right PGA Mixer"}, {"Right HPCOM Mux", "differential of HPROUT", "Right PGA Bypass Mixer"}, {"Right HPCOM Mux", "constant VCM", "Right PGA Bypass Mixer"}, @@ -643,10 +685,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right HP Out", NULL, "Right PGA Bypass Mixer"}, /* Left Line2 Bypass */ - {"Left Line2 Bypass Mixer", "Line Switch", "Left Line2L Mux"}, + {"Left Line2 Bypass Mixer", "LineL Switch", "Left Line2L Mux"}, + {"Left Line2 Bypass Mixer", "LineR Switch", "Left Line2L Mux"}, {"Left Line2 Bypass Mixer", "Mono Switch", "Left Line2L Mux"}, {"Left Line2 Bypass Mixer", "HP Switch", "Left Line2L Mux"}, - {"Left Line2 Bypass Mixer", "HPCOM Switch", "Left Line2L Mux"}, + {"Left Line2 Bypass Mixer", "HPLCOM Switch", "Left Line2L Mux"}, {"Left HPCOM Mux", "differential of HPLOUT", "Left Line2 Bypass Mixer"}, {"Left HPCOM Mux", "constant VCM", "Left Line2 Bypass Mixer"}, @@ -657,10 +700,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left HP Out", NULL, "Left Line2 Bypass Mixer"}, /* Right Line2 Bypass */ - {"Right Line2 Bypass Mixer", "Line Switch", "Right Line2R Mux"}, + {"Right Line2 Bypass Mixer", "LineL Switch", "Right Line2R Mux"}, + {"Right Line2 Bypass Mixer", "LineR Switch", "Right Line2R Mux"}, {"Right Line2 Bypass Mixer", "Mono Switch", "Right Line2R Mux"}, {"Right Line2 Bypass Mixer", "HP Switch", "Right Line2R Mux"}, - {"Right Line2 Bypass Mixer", "HPCOM Switch", "Right Line2R Mux"}, + {"Right Line2 Bypass Mixer", "HPRCOM Switch", "Right Line2R Mux"}, {"Right HPCOM Mux", "differential of HPROUT", "Right Line2 Bypass Mixer"}, {"Right HPCOM Mux", "constant VCM", "Right Line2 Bypass Mixer"}, @@ -694,7 +738,8 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) } static int aic3x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -846,6 +891,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = codec->private_data; u8 iface_areg, iface_breg; + int delay = 0; iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; @@ -871,6 +917,8 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, SND_SOC_DAIFMT_INV_MASK)) { case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; + case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF): + delay = 1; case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): iface_breg |= (0x01 << 6); break; @@ -887,6 +935,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, /* set iface */ aic3x_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg); aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg); + aic3x_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } @@ -981,14 +1030,41 @@ int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio) } EXPORT_SYMBOL_GPL(aic3x_get_gpio); +void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect, + int headset_debounce, int button_debounce) +{ + u8 val; + + val = ((detect & AIC3X_HEADSET_DETECT_MASK) + << AIC3X_HEADSET_DETECT_SHIFT) | + ((headset_debounce & AIC3X_HEADSET_DEBOUNCE_MASK) + << AIC3X_HEADSET_DEBOUNCE_SHIFT) | + ((button_debounce & AIC3X_BUTTON_DEBOUNCE_MASK) + << AIC3X_BUTTON_DEBOUNCE_SHIFT); + + if (detect & AIC3X_HEADSET_DETECT_MASK) + val |= AIC3X_HEADSET_DETECT_ENABLED; + + aic3x_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val); +} +EXPORT_SYMBOL_GPL(aic3x_set_headset_detection); + int aic3x_headset_detected(struct snd_soc_codec *codec) { u8 val; - aic3x_read(codec, AIC3X_RT_IRQ_FLAGS_REG, &val); - return (val >> 2) & 1; + aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val); + return (val >> 4) & 1; } EXPORT_SYMBOL_GPL(aic3x_headset_detected); +int aic3x_button_pressed(struct snd_soc_codec *codec) +{ + u8 val; + aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val); + return (val >> 5) & 1; +} +EXPORT_SYMBOL_GPL(aic3x_button_pressed); + #define AIC3X_RATES SNDRV_PCM_RATE_8000_96000 #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -1009,8 +1085,6 @@ struct snd_soc_dai aic3x_dai = { .formats = AIC3X_FORMATS,}, .ops = { .hw_params = aic3x_hw_params, - }, - .dai_ops = { .digital_mute = aic3x_mute, .set_sysclk = aic3x_set_dai_sysclk, .set_fmt = aic3x_set_dai_fmt, @@ -1152,7 +1226,7 @@ static int aic3x_init(struct snd_soc_device *socdev) aic3x_add_controls(codec); aic3x_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "aic3x: failed to register card\n"); goto card_err; @@ -1341,6 +1415,18 @@ struct snd_soc_codec_device soc_codec_dev_aic3x = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_aic3x); +static int __init aic3x_modinit(void) +{ + return snd_soc_register_dai(&aic3x_dai); +} +module_init(aic3x_modinit); + +static void __exit aic3x_exit(void) +{ + snd_soc_unregister_dai(&aic3x_dai); +} +module_exit(aic3x_exit); + MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver"); MODULE_AUTHOR("Vladimir Barinov"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 00a195aa02e4..ac827e578c4d 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -35,11 +35,15 @@ #define AIC3X_ASD_INTF_CTRLA 8 /* Audio serial data interface control register B */ #define AIC3X_ASD_INTF_CTRLB 9 +/* Audio serial data interface control register C */ +#define AIC3X_ASD_INTF_CTRLC 10 /* Audio overflow status and PLL R value programming register */ #define AIC3X_OVRF_STATUS_AND_PLLR_REG 11 /* Audio codec digital filter control register */ #define AIC3X_CODEC_DFILT_CTRL 12 - +/* Headset/button press detection register */ +#define AIC3X_HEADSET_DETECT_CTRL_A 13 +#define AIC3X_HEADSET_DETECT_CTRL_B 14 /* ADC PGA Gain control registers */ #define LADC_VOL 15 #define RADC_VOL 16 @@ -48,7 +52,9 @@ #define MIC3LR_2_RADC_CTRL 18 /* Line1 Input control registers */ #define LINE1L_2_LADC_CTRL 19 +#define LINE1R_2_LADC_CTRL 21 #define LINE1R_2_RADC_CTRL 22 +#define LINE1L_2_RADC_CTRL 24 /* Line2 Input control registers */ #define LINE2L_2_LADC_CTRL 20 #define LINE2R_2_RADC_CTRL 23 @@ -79,6 +85,8 @@ #define LINE2L_2_HPLOUT_VOL 45 #define LINE2R_2_HPROUT_VOL 62 #define PGAL_2_HPLOUT_VOL 46 +#define PGAL_2_HPROUT_VOL 60 +#define PGAR_2_HPLOUT_VOL 49 #define PGAR_2_HPROUT_VOL 63 #define DACL1_2_HPLOUT_VOL 47 #define DACR1_2_HPROUT_VOL 64 @@ -88,6 +96,8 @@ #define LINE2L_2_HPLCOM_VOL 52 #define LINE2R_2_HPRCOM_VOL 69 #define PGAL_2_HPLCOM_VOL 53 +#define PGAR_2_HPLCOM_VOL 56 +#define PGAL_2_HPRCOM_VOL 67 #define PGAR_2_HPRCOM_VOL 70 #define DACL1_2_HPLCOM_VOL 54 #define DACR1_2_HPRCOM_VOL 71 @@ -103,11 +113,17 @@ #define MONOLOPM_CTRL 79 /* Line Output Plus/Minus control registers */ #define LINE2L_2_LLOPM_VOL 80 +#define LINE2L_2_RLOPM_VOL 87 +#define LINE2R_2_LLOPM_VOL 83 #define LINE2R_2_RLOPM_VOL 90 #define PGAL_2_LLOPM_VOL 81 +#define PGAL_2_RLOPM_VOL 88 +#define PGAR_2_LLOPM_VOL 84 #define PGAR_2_RLOPM_VOL 91 #define DACL1_2_LLOPM_VOL 82 +#define DACL1_2_RLOPM_VOL 89 #define DACR1_2_RLOPM_VOL 92 +#define DACR1_2_LLOPM_VOL 85 #define LLOPM_CTRL 86 #define RLOPM_CTRL 93 /* GPIO/IRQ registers */ @@ -221,7 +237,49 @@ enum { void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state); int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio); + +/* headset detection / button API */ + +/* The AIC3x supports detection of stereo headsets (GND + left + right signal) + * and cellular headsets (GND + speaker output + microphone input). + * It is recommended to enable MIC bias for this function to work properly. + * For more information, please refer to the datasheet. */ +enum { + AIC3X_HEADSET_DETECT_OFF = 0, + AIC3X_HEADSET_DETECT_STEREO = 1, + AIC3X_HEADSET_DETECT_CELLULAR = 2, + AIC3X_HEADSET_DETECT_BOTH = 3 +}; + +enum { + AIC3X_HEADSET_DEBOUNCE_16MS = 0, + AIC3X_HEADSET_DEBOUNCE_32MS = 1, + AIC3X_HEADSET_DEBOUNCE_64MS = 2, + AIC3X_HEADSET_DEBOUNCE_128MS = 3, + AIC3X_HEADSET_DEBOUNCE_256MS = 4, + AIC3X_HEADSET_DEBOUNCE_512MS = 5 +}; + +enum { + AIC3X_BUTTON_DEBOUNCE_0MS = 0, + AIC3X_BUTTON_DEBOUNCE_8MS = 1, + AIC3X_BUTTON_DEBOUNCE_16MS = 2, + AIC3X_BUTTON_DEBOUNCE_32MS = 3 +}; + +#define AIC3X_HEADSET_DETECT_ENABLED 0x80 +#define AIC3X_HEADSET_DETECT_SHIFT 5 +#define AIC3X_HEADSET_DETECT_MASK 3 +#define AIC3X_HEADSET_DEBOUNCE_SHIFT 2 +#define AIC3X_HEADSET_DEBOUNCE_MASK 7 +#define AIC3X_BUTTON_DEBOUNCE_SHIFT 0 +#define AIC3X_BUTTON_DEBOUNCE_MASK 3 + +/* see the enums above for valid parameters to this function */ +void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect, + int headset_debounce, int button_debounce); int aic3x_headset_detected(struct snd_soc_codec *codec); +int aic3x_button_pressed(struct snd_soc_codec *codec); struct aic3x_setup_data { int i2c_bus; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c new file mode 100644 index 000000000000..51848880504a --- /dev/null +++ b/sound/soc/codecs/twl4030.c @@ -0,0 +1,1317 @@ +/* + * ALSA SoC TWL4030 codec driver + * + * Author: Steve Sakoman, <steve@sakoman.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/i2c/twl4030.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "twl4030.h" + +/* + * twl4030 register cache & default register settings + */ +static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { + 0x00, /* this register not used */ + 0x93, /* REG_CODEC_MODE (0x1) */ + 0xc3, /* REG_OPTION (0x2) */ + 0x00, /* REG_UNKNOWN (0x3) */ + 0x00, /* REG_MICBIAS_CTL (0x4) */ + 0x20, /* REG_ANAMICL (0x5) */ + 0x00, /* REG_ANAMICR (0x6) */ + 0x00, /* REG_AVADC_CTL (0x7) */ + 0x00, /* REG_ADCMICSEL (0x8) */ + 0x00, /* REG_DIGMIXING (0x9) */ + 0x0c, /* REG_ATXL1PGA (0xA) */ + 0x0c, /* REG_ATXR1PGA (0xB) */ + 0x00, /* REG_AVTXL2PGA (0xC) */ + 0x00, /* REG_AVTXR2PGA (0xD) */ + 0x01, /* REG_AUDIO_IF (0xE) */ + 0x00, /* REG_VOICE_IF (0xF) */ + 0x00, /* REG_ARXR1PGA (0x10) */ + 0x00, /* REG_ARXL1PGA (0x11) */ + 0x6c, /* REG_ARXR2PGA (0x12) */ + 0x6c, /* REG_ARXL2PGA (0x13) */ + 0x00, /* REG_VRXPGA (0x14) */ + 0x00, /* REG_VSTPGA (0x15) */ + 0x00, /* REG_VRX2ARXPGA (0x16) */ + 0x0c, /* REG_AVDAC_CTL (0x17) */ + 0x00, /* REG_ARX2VTXPGA (0x18) */ + 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ + 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ + 0x4b, /* REG_ARXL2_APGA_CTL (0x1B) */ + 0x4b, /* REG_ARXR2_APGA_CTL (0x1C) */ + 0x00, /* REG_ATX2ARXPGA (0x1D) */ + 0x00, /* REG_BT_IF (0x1E) */ + 0x00, /* REG_BTPGA (0x1F) */ + 0x00, /* REG_BTSTPGA (0x20) */ + 0x00, /* REG_EAR_CTL (0x21) */ + 0x24, /* REG_HS_SEL (0x22) */ + 0x0a, /* REG_HS_GAIN_SET (0x23) */ + 0x00, /* REG_HS_POPN_SET (0x24) */ + 0x00, /* REG_PREDL_CTL (0x25) */ + 0x00, /* REG_PREDR_CTL (0x26) */ + 0x00, /* REG_PRECKL_CTL (0x27) */ + 0x00, /* REG_PRECKR_CTL (0x28) */ + 0x00, /* REG_HFL_CTL (0x29) */ + 0x00, /* REG_HFR_CTL (0x2A) */ + 0x00, /* REG_ALC_CTL (0x2B) */ + 0x00, /* REG_ALC_SET1 (0x2C) */ + 0x00, /* REG_ALC_SET2 (0x2D) */ + 0x00, /* REG_BOOST_CTL (0x2E) */ + 0x00, /* REG_SOFTVOL_CTL (0x2F) */ + 0x00, /* REG_DTMF_FREQSEL (0x30) */ + 0x00, /* REG_DTMF_TONEXT1H (0x31) */ + 0x00, /* REG_DTMF_TONEXT1L (0x32) */ + 0x00, /* REG_DTMF_TONEXT2H (0x33) */ + 0x00, /* REG_DTMF_TONEXT2L (0x34) */ + 0x00, /* REG_DTMF_TONOFF (0x35) */ + 0x00, /* REG_DTMF_WANONOFF (0x36) */ + 0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */ + 0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */ + 0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */ + 0x16, /* REG_APLL_CTL (0x3A) */ + 0x00, /* REG_DTMF_CTL (0x3B) */ + 0x00, /* REG_DTMF_PGA_CTL2 (0x3C) */ + 0x00, /* REG_DTMF_PGA_CTL1 (0x3D) */ + 0x00, /* REG_MISC_SET_1 (0x3E) */ + 0x00, /* REG_PCMBTMUX (0x3F) */ + 0x00, /* not used (0x40) */ + 0x00, /* not used (0x41) */ + 0x00, /* not used (0x42) */ + 0x00, /* REG_RX_PATH_SEL (0x43) */ + 0x00, /* REG_VDL_APGA_CTL (0x44) */ + 0x00, /* REG_VIBRA_CTL (0x45) */ + 0x00, /* REG_VIBRA_SET (0x46) */ + 0x00, /* REG_VIBRA_PWM_SET (0x47) */ + 0x00, /* REG_ANAMIC_GAIN (0x48) */ + 0x00, /* REG_MISC_SET_2 (0x49) */ +}; + +/* + * read twl4030 register cache + */ +static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + return cache[reg]; +} + +/* + * write twl4030 register cache + */ +static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL4030_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the twl4030 register space + */ +static int twl4030_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + twl4030_write_reg_cache(codec, reg, value); + return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); +} + +static void twl4030_clear_codecpdz(struct snd_soc_codec *codec) +{ + u8 mode; + + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, + mode & ~TWL4030_CODECPDZ); + + /* REVISIT: this delay is present in TI sample drivers */ + /* but there seems to be no TRM requirement for it */ + udelay(10); +} + +static void twl4030_set_codecpdz(struct snd_soc_codec *codec) +{ + u8 mode; + + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, + mode | TWL4030_CODECPDZ); + + /* REVISIT: this delay is present in TI sample drivers */ + /* but there seems to be no TRM requirement for it */ + udelay(10); +} + +static void twl4030_init_chip(struct snd_soc_codec *codec) +{ + int i; + + /* clear CODECPDZ prior to setting register defaults */ + twl4030_clear_codecpdz(codec); + + /* set all audio section registers to reasonable defaults */ + for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++) + twl4030_write(codec, i, twl4030_reg[i]); + +} + +/* Earpiece */ +static const char *twl4030_earpiece_texts[] = + {"Off", "DACL1", "DACL2", "Invalid", "DACR1"}; + +static const struct soc_enum twl4030_earpiece_enum = + SOC_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, + ARRAY_SIZE(twl4030_earpiece_texts), + twl4030_earpiece_texts); + +static const struct snd_kcontrol_new twl4030_dapm_earpiece_control = +SOC_DAPM_ENUM("Route", twl4030_earpiece_enum); + +/* PreDrive Left */ +static const char *twl4030_predrivel_texts[] = + {"Off", "DACL1", "DACL2", "Invalid", "DACR2"}; + +static const struct soc_enum twl4030_predrivel_enum = + SOC_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, + ARRAY_SIZE(twl4030_predrivel_texts), + twl4030_predrivel_texts); + +static const struct snd_kcontrol_new twl4030_dapm_predrivel_control = +SOC_DAPM_ENUM("Route", twl4030_predrivel_enum); + +/* PreDrive Right */ +static const char *twl4030_predriver_texts[] = + {"Off", "DACR1", "DACR2", "Invalid", "DACL2"}; + +static const struct soc_enum twl4030_predriver_enum = + SOC_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, + ARRAY_SIZE(twl4030_predriver_texts), + twl4030_predriver_texts); + +static const struct snd_kcontrol_new twl4030_dapm_predriver_control = +SOC_DAPM_ENUM("Route", twl4030_predriver_enum); + +/* Headset Left */ +static const char *twl4030_hsol_texts[] = + {"Off", "DACL1", "DACL2"}; + +static const struct soc_enum twl4030_hsol_enum = + SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1, + ARRAY_SIZE(twl4030_hsol_texts), + twl4030_hsol_texts); + +static const struct snd_kcontrol_new twl4030_dapm_hsol_control = +SOC_DAPM_ENUM("Route", twl4030_hsol_enum); + +/* Headset Right */ +static const char *twl4030_hsor_texts[] = + {"Off", "DACR1", "DACR2"}; + +static const struct soc_enum twl4030_hsor_enum = + SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4, + ARRAY_SIZE(twl4030_hsor_texts), + twl4030_hsor_texts); + +static const struct snd_kcontrol_new twl4030_dapm_hsor_control = +SOC_DAPM_ENUM("Route", twl4030_hsor_enum); + +/* Carkit Left */ +static const char *twl4030_carkitl_texts[] = + {"Off", "DACL1", "DACL2"}; + +static const struct soc_enum twl4030_carkitl_enum = + SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1, + ARRAY_SIZE(twl4030_carkitl_texts), + twl4030_carkitl_texts); + +static const struct snd_kcontrol_new twl4030_dapm_carkitl_control = +SOC_DAPM_ENUM("Route", twl4030_carkitl_enum); + +/* Carkit Right */ +static const char *twl4030_carkitr_texts[] = + {"Off", "DACR1", "DACR2"}; + +static const struct soc_enum twl4030_carkitr_enum = + SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1, + ARRAY_SIZE(twl4030_carkitr_texts), + twl4030_carkitr_texts); + +static const struct snd_kcontrol_new twl4030_dapm_carkitr_control = +SOC_DAPM_ENUM("Route", twl4030_carkitr_enum); + +/* Handsfree Left */ +static const char *twl4030_handsfreel_texts[] = + {"Voice", "DACL1", "DACL2", "DACR2"}; + +static const struct soc_enum twl4030_handsfreel_enum = + SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0, + ARRAY_SIZE(twl4030_handsfreel_texts), + twl4030_handsfreel_texts); + +static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control = +SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); + +/* Handsfree Right */ +static const char *twl4030_handsfreer_texts[] = + {"Voice", "DACR1", "DACR2", "DACL2"}; + +static const struct soc_enum twl4030_handsfreer_enum = + SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0, + ARRAY_SIZE(twl4030_handsfreer_texts), + twl4030_handsfreer_texts); + +static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control = +SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum); + +static int outmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + int ret = 0; + int val; + + switch (e->reg) { + case TWL4030_REG_PREDL_CTL: + case TWL4030_REG_PREDR_CTL: + case TWL4030_REG_EAR_CTL: + val = w->value >> e->shift_l; + if (val == 3) { + printk(KERN_WARNING + "Invalid MUX setting for register 0x%02x (%d)\n", + e->reg, val); + ret = -1; + } + break; + } + + return ret; +} + +static int handsfree_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value; + unsigned char hs_ctl; + + hs_ctl = twl4030_read_reg_cache(w->codec, e->reg); + + if (hs_ctl & TWL4030_HF_CTL_REF_EN) { + hs_ctl |= TWL4030_HF_CTL_RAMP_EN; + twl4030_write(w->codec, e->reg, hs_ctl); + hs_ctl |= TWL4030_HF_CTL_LOOP_EN; + twl4030_write(w->codec, e->reg, hs_ctl); + hs_ctl |= TWL4030_HF_CTL_HB_EN; + twl4030_write(w->codec, e->reg, hs_ctl); + } else { + hs_ctl &= ~(TWL4030_HF_CTL_RAMP_EN | TWL4030_HF_CTL_LOOP_EN + | TWL4030_HF_CTL_HB_EN); + twl4030_write(w->codec, e->reg, hs_ctl); + } + + return 0; +} + +/* + * Some of the gain controls in TWL (mostly those which are associated with + * the outputs) are implemented in an interesting way: + * 0x0 : Power down (mute) + * 0x1 : 6dB + * 0x2 : 0 dB + * 0x3 : -6 dB + * Inverting not going to help with these. + * Custom volsw and volsw_2r get/put functions to handle these gain bits. + */ +#define SOC_DOUBLE_TLV_TWL4030(xname, xreg, shift_left, shift_right, xmax,\ + xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw_twl4030, \ + .put = snd_soc_put_volsw_twl4030, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right,\ + .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_R_TLV_TWL4030(xname, reg_left, reg_right, xshift, xmax,\ + xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r, \ + .get = snd_soc_get_volsw_r2_twl4030,\ + .put = snd_soc_put_volsw_r2_twl4030, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .rshift = xshift, .max = xmax, .invert = xinvert} } +#define SOC_SINGLE_TLV_TWL4030(xname, xreg, xshift, xmax, xinvert, tlv_array) \ + SOC_DOUBLE_TLV_TWL4030(xname, xreg, xshift, xshift, xmax, \ + xinvert, tlv_array) + +static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int mask = (1 << fls(max)) - 1; + + ucontrol->value.integer.value[0] = + (snd_soc_read(codec, reg) >> shift) & mask; + if (ucontrol->value.integer.value[0]) + ucontrol->value.integer.value[0] = + max + 1 - ucontrol->value.integer.value[0]; + + if (shift != rshift) { + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, reg) >> rshift) & mask; + if (ucontrol->value.integer.value[1]) + ucontrol->value.integer.value[1] = + max + 1 - ucontrol->value.integer.value[1]; + } + + return 0; +} + +static int snd_soc_put_volsw_twl4030(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int mask = (1 << fls(max)) - 1; + unsigned short val, val2, val_mask; + + val = (ucontrol->value.integer.value[0] & mask); + + val_mask = mask << shift; + if (val) + val = max + 1 - val; + val = val << shift; + if (shift != rshift) { + val2 = (ucontrol->value.integer.value[1] & mask); + val_mask |= mask << rshift; + if (val2) + val2 = max + 1 - val2; + val |= val2 << rshift; + } + return snd_soc_update_bits(codec, reg, val_mask, val); +} + +static int snd_soc_get_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + int max = mc->max; + int mask = (1<<fls(max))-1; + + ucontrol->value.integer.value[0] = + (snd_soc_read(codec, reg) >> shift) & mask; + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, reg2) >> shift) & mask; + + if (ucontrol->value.integer.value[0]) + ucontrol->value.integer.value[0] = + max + 1 - ucontrol->value.integer.value[0]; + if (ucontrol->value.integer.value[1]) + ucontrol->value.integer.value[1] = + max + 1 - ucontrol->value.integer.value[1]; + + return 0; +} + +static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + int max = mc->max; + int mask = (1 << fls(max)) - 1; + int err; + unsigned short val, val2, val_mask; + + val_mask = mask << shift; + val = (ucontrol->value.integer.value[0] & mask); + val2 = (ucontrol->value.integer.value[1] & mask); + + if (val) + val = max + 1 - val; + if (val2) + val2 = max + 1 - val2; + + val = val << shift; + val2 = val2 << shift; + + err = snd_soc_update_bits(codec, reg, val_mask, val); + if (err < 0) + return err; + + err = snd_soc_update_bits(codec, reg2, val_mask, val2); + return err; +} + +static int twl4030_get_left_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + int result = 0; + + /* one bit must be set a time */ + reg &= TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN + | TWL4030_MAINMIC_EN; + if (reg != 0) { + result++; + while ((reg & 1) == 0) { + result++; + reg >>= 1; + } + } + + ucontrol->value.integer.value[0] = result; + return 0; +} + +static int twl4030_put_left_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + int value = ucontrol->value.integer.value[0]; + u8 anamicl, micbias, avadc_ctl; + + anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + anamicl &= ~(TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN + | TWL4030_MAINMIC_EN); + micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL); + micbias &= ~(TWL4030_HSMICBIAS_EN | TWL4030_MICBIAS1_EN); + avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL); + + switch (value) { + case 1: + anamicl |= TWL4030_MAINMIC_EN; + micbias |= TWL4030_MICBIAS1_EN; + break; + case 2: + anamicl |= TWL4030_HSMIC_EN; + micbias |= TWL4030_HSMICBIAS_EN; + break; + case 3: + anamicl |= TWL4030_AUXL_EN; + break; + case 4: + anamicl |= TWL4030_CKMIC_EN; + break; + default: + break; + } + + /* If some input is selected, enable amp and ADC */ + if (value != 0) { + anamicl |= TWL4030_MICAMPL_EN; + avadc_ctl |= TWL4030_ADCL_EN; + } else { + anamicl &= ~TWL4030_MICAMPL_EN; + avadc_ctl &= ~TWL4030_ADCL_EN; + } + + twl4030_write(codec, TWL4030_REG_ANAMICL, anamicl); + twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); + twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl); + + return 1; +} + +static int twl4030_get_right_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR); + int value = 0; + + reg &= TWL4030_SUBMIC_EN|TWL4030_AUXR_EN; + switch (reg) { + case TWL4030_SUBMIC_EN: + value = 1; + break; + case TWL4030_AUXR_EN: + value = 2; + break; + default: + break; + } + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int twl4030_put_right_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + int value = ucontrol->value.integer.value[0]; + u8 anamicr, micbias, avadc_ctl; + + anamicr = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR); + anamicr &= ~(TWL4030_SUBMIC_EN|TWL4030_AUXR_EN); + micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL); + micbias &= ~TWL4030_MICBIAS2_EN; + avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL); + + switch (value) { + case 1: + anamicr |= TWL4030_SUBMIC_EN; + micbias |= TWL4030_MICBIAS2_EN; + break; + case 2: + anamicr |= TWL4030_AUXR_EN; + break; + default: + break; + } + + if (value != 0) { + anamicr |= TWL4030_MICAMPR_EN; + avadc_ctl |= TWL4030_ADCR_EN; + } else { + anamicr &= ~TWL4030_MICAMPR_EN; + avadc_ctl &= ~TWL4030_ADCR_EN; + } + + twl4030_write(codec, TWL4030_REG_ANAMICR, anamicr); + twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); + twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl); + + return 1; +} + +static const char *twl4030_left_in_sel[] = { + "None", + "Main Mic", + "Headset Mic", + "Line In", + "Carkit Mic", +}; + +static const char *twl4030_right_in_sel[] = { + "None", + "Sub Mic", + "Line In", +}; + +static const struct soc_enum twl4030_left_input_mux = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_left_in_sel), + twl4030_left_in_sel); + +static const struct soc_enum twl4030_right_input_mux = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_right_in_sel), + twl4030_right_in_sel); + +/* + * FGAIN volume control: + * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) + */ +static DECLARE_TLV_DB_SCALE(digital_fine_tlv, -6300, 100, 1); + +/* + * CGAIN volume control: + * 0 dB to 12 dB in 6 dB steps + * value 2 and 3 means 12 dB + */ +static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0); + +/* + * Analog playback gain + * -24 dB to 12 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0); + +/* + * Gain controls tied to outputs + * -6 dB to 6 dB in 6 dB steps (mute instead of -12) + */ +static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1); + +/* + * Capture gain after the ADCs + * from 0 dB to 31 dB in 1 dB steps + */ +static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); + +/* + * Gain control for input amplifiers + * 0 dB to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new twl4030_snd_controls[] = { + /* Common playback gain controls */ + SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", + TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA, + 0, 0x3f, 0, digital_fine_tlv), + SOC_DOUBLE_R_TLV("DAC2 Digital Fine Playback Volume", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 0, 0x3f, 0, digital_fine_tlv), + + SOC_DOUBLE_R_TLV("DAC1 Digital Coarse Playback Volume", + TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA, + 6, 0x2, 0, digital_coarse_tlv), + SOC_DOUBLE_R_TLV("DAC2 Digital Coarse Playback Volume", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 6, 0x2, 0, digital_coarse_tlv), + + SOC_DOUBLE_R_TLV("DAC1 Analog Playback Volume", + TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL, + 3, 0x12, 1, analog_tlv), + SOC_DOUBLE_R_TLV("DAC2 Analog Playback Volume", + TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL, + 3, 0x12, 1, analog_tlv), + SOC_DOUBLE_R("DAC1 Analog Playback Switch", + TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL, + 1, 1, 0), + SOC_DOUBLE_R("DAC2 Analog Playback Switch", + TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL, + 1, 1, 0), + + /* Separate output gain controls */ + SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume", + TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL, + 4, 3, 0, output_tvl), + + SOC_DOUBLE_TLV_TWL4030("Headset Playback Volume", + TWL4030_REG_HS_GAIN_SET, 0, 2, 3, 0, output_tvl), + + SOC_DOUBLE_R_TLV_TWL4030("Carkit Playback Volume", + TWL4030_REG_PRECKL_CTL, TWL4030_REG_PRECKR_CTL, + 4, 3, 0, output_tvl), + + SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume", + TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl), + + /* Common capture gain controls */ + SOC_DOUBLE_R_TLV("Capture Volume", + TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, + 0, 0x1f, 0, digital_capture_tlv), + + SOC_DOUBLE_TLV("Input Boost Volume", TWL4030_REG_ANAMIC_GAIN, + 0, 3, 5, 0, input_gain_tlv), + + /* Input source controls */ + SOC_ENUM_EXT("Left Input Source", twl4030_left_input_mux, + twl4030_get_left_input, twl4030_put_left_input), + SOC_ENUM_EXT("Right Input Source", twl4030_right_input_mux, + twl4030_get_right_input, twl4030_put_right_input), +}; + +/* add non dapm controls */ +static int twl4030_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&twl4030_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("INL"), + SND_SOC_DAPM_INPUT("INR"), + + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), + SND_SOC_DAPM_OUTPUT("EARPIECE"), + SND_SOC_DAPM_OUTPUT("PREDRIVEL"), + SND_SOC_DAPM_OUTPUT("PREDRIVER"), + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("CARKITL"), + SND_SOC_DAPM_OUTPUT("CARKITR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback", + TWL4030_REG_AVDAC_CTL, 0, 0), + SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback", + TWL4030_REG_AVDAC_CTL, 1, 0), + SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback", + TWL4030_REG_AVDAC_CTL, 2, 0), + SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", + TWL4030_REG_AVDAC_CTL, 3, 0), + + /* Analog PGAs */ + SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ARXR2_APGA", TWL4030_REG_ARXR2_APGA_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL, + 0, 0, NULL, 0), + + /* Output MUX controls */ + /* Earpiece */ + SND_SOC_DAPM_MUX_E("Earpiece Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_earpiece_control, outmixer_event, + SND_SOC_DAPM_PRE_REG), + /* PreDrivL/R */ + SND_SOC_DAPM_MUX_E("PredriveL Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predrivel_control, outmixer_event, + SND_SOC_DAPM_PRE_REG), + SND_SOC_DAPM_MUX_E("PredriveR Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predriver_control, outmixer_event, + SND_SOC_DAPM_PRE_REG), + /* HeadsetL/R */ + SND_SOC_DAPM_MUX("HeadsetL Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsol_control), + SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsor_control), + /* CarkitL/R */ + SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitl_control), + SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitr_control), + /* HandsfreeL/R */ + SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, + &twl4030_dapm_handsfreel_control, handsfree_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0, + &twl4030_dapm_handsfreer_control, handsfree_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"ARXL1_APGA", NULL, "DAC Left1"}, + {"ARXR1_APGA", NULL, "DAC Right1"}, + {"ARXL2_APGA", NULL, "DAC Left2"}, + {"ARXR2_APGA", NULL, "DAC Right2"}, + + /* Internal playback routings */ + /* Earpiece */ + {"Earpiece Mux", "DACL1", "ARXL1_APGA"}, + {"Earpiece Mux", "DACL2", "ARXL2_APGA"}, + {"Earpiece Mux", "DACR1", "ARXR1_APGA"}, + /* PreDrivL */ + {"PredriveL Mux", "DACL1", "ARXL1_APGA"}, + {"PredriveL Mux", "DACL2", "ARXL2_APGA"}, + {"PredriveL Mux", "DACR2", "ARXR2_APGA"}, + /* PreDrivR */ + {"PredriveR Mux", "DACR1", "ARXR1_APGA"}, + {"PredriveR Mux", "DACR2", "ARXR2_APGA"}, + {"PredriveR Mux", "DACL2", "ARXL2_APGA"}, + /* HeadsetL */ + {"HeadsetL Mux", "DACL1", "ARXL1_APGA"}, + {"HeadsetL Mux", "DACL2", "ARXL2_APGA"}, + /* HeadsetR */ + {"HeadsetR Mux", "DACR1", "ARXR1_APGA"}, + {"HeadsetR Mux", "DACR2", "ARXR2_APGA"}, + /* CarkitL */ + {"CarkitL Mux", "DACL1", "ARXL1_APGA"}, + {"CarkitL Mux", "DACL2", "ARXL2_APGA"}, + /* CarkitR */ + {"CarkitR Mux", "DACR1", "ARXR1_APGA"}, + {"CarkitR Mux", "DACR2", "ARXR2_APGA"}, + /* HandsfreeL */ + {"HandsfreeL Mux", "DACL1", "ARXL1_APGA"}, + {"HandsfreeL Mux", "DACL2", "ARXL2_APGA"}, + {"HandsfreeL Mux", "DACR2", "ARXR2_APGA"}, + /* HandsfreeR */ + {"HandsfreeR Mux", "DACR1", "ARXR1_APGA"}, + {"HandsfreeR Mux", "DACR2", "ARXR2_APGA"}, + {"HandsfreeR Mux", "DACL2", "ARXL2_APGA"}, + + /* outputs */ + {"OUTL", NULL, "ARXL2_APGA"}, + {"OUTR", NULL, "ARXR2_APGA"}, + {"EARPIECE", NULL, "Earpiece Mux"}, + {"PREDRIVEL", NULL, "PredriveL Mux"}, + {"PREDRIVER", NULL, "PredriveR Mux"}, + {"HSOL", NULL, "HeadsetL Mux"}, + {"HSOR", NULL, "HeadsetR Mux"}, + {"CARKITL", NULL, "CarkitL Mux"}, + {"CARKITR", NULL, "CarkitR Mux"}, + {"HFL", NULL, "HandsfreeL Mux"}, + {"HFR", NULL, "HandsfreeR Mux"}, + + /* inputs */ + {"ADCL", NULL, "INL"}, + {"ADCR", NULL, "INR"}, +}; + +static int twl4030_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl4030_dapm_widgets, + ARRAY_SIZE(twl4030_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static void twl4030_power_up(struct snd_soc_codec *codec) +{ + u8 anamicl, regmisc1, byte, popn; + int i = 0; + + /* set CODECPDZ to turn on codec */ + twl4030_set_codecpdz(codec); + + /* initiate offset cancellation */ + anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + twl4030_write(codec, TWL4030_REG_ANAMICL, + anamicl | TWL4030_CNCL_OFFSET_START); + + /* wait for offset cancellation to complete */ + do { + /* this takes a little while, so don't slam i2c */ + udelay(2000); + twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, + TWL4030_REG_ANAMICL); + } while ((i++ < 100) && + ((byte & TWL4030_CNCL_OFFSET_START) == + TWL4030_CNCL_OFFSET_START)); + + /* anti-pop when changing analog gain */ + regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1); + twl4030_write(codec, TWL4030_REG_MISC_SET_1, + regmisc1 | TWL4030_SMOOTH_ANAVOL_EN); + + /* toggle CODECPDZ as per TRM */ + twl4030_clear_codecpdz(codec); + twl4030_set_codecpdz(codec); + + /* program anti-pop with bias ramp delay */ + popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + popn &= TWL4030_RAMP_DELAY; + popn |= TWL4030_RAMP_DELAY_645MS; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + popn |= TWL4030_VMID_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + + /* enable anti-pop ramp */ + popn |= TWL4030_RAMP_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); +} + +static void twl4030_power_down(struct snd_soc_codec *codec) +{ + u8 popn; + + /* disable anti-pop ramp */ + popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + popn &= ~TWL4030_RAMP_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + + /* disable bias out */ + popn &= ~TWL4030_VMID_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + + /* power down */ + twl4030_clear_codecpdz(codec); +} + +static int twl4030_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + twl4030_power_up(codec); + break; + case SND_SOC_BIAS_PREPARE: + /* TODO: develop a twl4030_prepare function */ + break; + case SND_SOC_BIAS_STANDBY: + /* TODO: develop a twl4030_standby function */ + twl4030_power_down(codec); + break; + case SND_SOC_BIAS_OFF: + twl4030_power_down(codec); + break; + } + codec->bias_level = level; + + return 0; +} + +static int twl4030_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u8 mode, old_mode, format, old_format; + + + /* bit rate */ + old_mode = twl4030_read_reg_cache(codec, + TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; + mode = old_mode & ~TWL4030_APLL_RATE; + + switch (params_rate(params)) { + case 8000: + mode |= TWL4030_APLL_RATE_8000; + break; + case 11025: + mode |= TWL4030_APLL_RATE_11025; + break; + case 12000: + mode |= TWL4030_APLL_RATE_12000; + break; + case 16000: + mode |= TWL4030_APLL_RATE_16000; + break; + case 22050: + mode |= TWL4030_APLL_RATE_22050; + break; + case 24000: + mode |= TWL4030_APLL_RATE_24000; + break; + case 32000: + mode |= TWL4030_APLL_RATE_32000; + break; + case 44100: + mode |= TWL4030_APLL_RATE_44100; + break; + case 48000: + mode |= TWL4030_APLL_RATE_48000; + break; + default: + printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n", + params_rate(params)); + return -EINVAL; + } + + if (mode != old_mode) { + /* change rate and set CODECPDZ */ + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_set_codecpdz(codec); + } + + /* sample size */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + format = old_format; + format &= ~TWL4030_DATA_WIDTH; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + format |= TWL4030_DATA_WIDTH_16S_16W; + break; + case SNDRV_PCM_FORMAT_S24_LE: + format |= TWL4030_DATA_WIDTH_32S_24W; + break; + default: + printk(KERN_ERR "TWL4030 hw params: unknown format %d\n", + params_format(params)); + return -EINVAL; + } + + if (format != old_format) { + + /* clear CODECPDZ before changing format (codec requirement) */ + twl4030_clear_codecpdz(codec); + + /* change format */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + + /* set CODECPDZ afterwards */ + twl4030_set_codecpdz(codec); + } + return 0; +} + +static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 infreq; + + switch (freq) { + case 19200000: + infreq = TWL4030_APLL_INFREQ_19200KHZ; + break; + case 26000000: + infreq = TWL4030_APLL_INFREQ_26000KHZ; + break; + case 38400000: + infreq = TWL4030_APLL_INFREQ_38400KHZ; + break; + default: + printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n", + freq); + return -EINVAL; + } + + infreq |= TWL4030_APLL_EN; + twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq); + + return 0; +} + +static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 old_format, format; + + /* get format */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + format = old_format; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + format &= ~(TWL4030_AIF_SLAVE_EN); + format &= ~(TWL4030_CLK256FS_EN); + break; + case SND_SOC_DAIFMT_CBS_CFS: + format |= TWL4030_AIF_SLAVE_EN; + format |= TWL4030_CLK256FS_EN; + break; + default: + return -EINVAL; + } + + /* interface format */ + format &= ~TWL4030_AIF_FORMAT; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format |= TWL4030_AIF_FORMAT_CODEC; + break; + default: + return -EINVAL; + } + + if (format != old_format) { + + /* clear CODECPDZ before changing format (codec requirement) */ + twl4030_clear_codecpdz(codec); + + /* change format */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + + /* set CODECPDZ afterwards */ + twl4030_set_codecpdz(codec); + } + + return 0; +} + +#define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) +#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) + +struct snd_soc_dai twl4030_dai = { + .name = "twl4030", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TWL4030_RATES, + .formats = TWL4030_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = TWL4030_RATES, + .formats = TWL4030_FORMATS,}, + .ops = { + .hw_params = twl4030_hw_params, + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(twl4030_dai); + +static int twl4030_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl4030_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl4030_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialize the driver + * register the mixer and dsp interfaces with the kernel + */ + +static int twl4030_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + printk(KERN_INFO "TWL4030 Audio Codec init \n"); + + codec->name = "twl4030"; + codec->owner = THIS_MODULE; + codec->read = twl4030_read_reg_cache; + codec->write = twl4030_write; + codec->set_bias_level = twl4030_set_bias_level; + codec->dai = &twl4030_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(twl4030_reg); + codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "twl4030: failed to create pcms\n"); + goto pcm_err; + } + + twl4030_init_chip(codec); + + /* power on device */ + twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + twl4030_add_controls(codec); + twl4030_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "twl4030: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *twl4030_socdev; + +static int twl4030_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + twl4030_socdev = socdev; + twl4030_init(socdev); + + return 0; +} + +static int twl4030_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + printk(KERN_INFO "TWL4030 Audio Codec remove\n"); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl4030 = { + .probe = twl4030_probe, + .remove = twl4030_remove, + .suspend = twl4030_suspend, + .resume = twl4030_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); + +static int __init twl4030_modinit(void) +{ + return snd_soc_register_dai(&twl4030_dai); +} +module_init(twl4030_modinit); + +static void __exit twl4030_exit(void) +{ + snd_soc_unregister_dai(&twl4030_dai); +} +module_exit(twl4030_exit); + +MODULE_DESCRIPTION("ASoC TWL4030 codec driver"); +MODULE_AUTHOR("Steve Sakoman"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h new file mode 100644 index 000000000000..54615c76802b --- /dev/null +++ b/sound/soc/codecs/twl4030.h @@ -0,0 +1,219 @@ +/* + * ALSA SoC TWL4030 codec driver + * + * Author: Steve Sakoman <steve@sakoman.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TWL4030_AUDIO_H__ +#define __TWL4030_AUDIO_H__ + +#define TWL4030_REG_CODEC_MODE 0x1 +#define TWL4030_REG_OPTION 0x2 +#define TWL4030_REG_UNKNOWN 0x3 +#define TWL4030_REG_MICBIAS_CTL 0x4 +#define TWL4030_REG_ANAMICL 0x5 +#define TWL4030_REG_ANAMICR 0x6 +#define TWL4030_REG_AVADC_CTL 0x7 +#define TWL4030_REG_ADCMICSEL 0x8 +#define TWL4030_REG_DIGMIXING 0x9 +#define TWL4030_REG_ATXL1PGA 0xA +#define TWL4030_REG_ATXR1PGA 0xB +#define TWL4030_REG_AVTXL2PGA 0xC +#define TWL4030_REG_AVTXR2PGA 0xD +#define TWL4030_REG_AUDIO_IF 0xE +#define TWL4030_REG_VOICE_IF 0xF +#define TWL4030_REG_ARXR1PGA 0x10 +#define TWL4030_REG_ARXL1PGA 0x11 +#define TWL4030_REG_ARXR2PGA 0x12 +#define TWL4030_REG_ARXL2PGA 0x13 +#define TWL4030_REG_VRXPGA 0x14 +#define TWL4030_REG_VSTPGA 0x15 +#define TWL4030_REG_VRX2ARXPGA 0x16 +#define TWL4030_REG_AVDAC_CTL 0x17 +#define TWL4030_REG_ARX2VTXPGA 0x18 +#define TWL4030_REG_ARXL1_APGA_CTL 0x19 +#define TWL4030_REG_ARXR1_APGA_CTL 0x1A +#define TWL4030_REG_ARXL2_APGA_CTL 0x1B +#define TWL4030_REG_ARXR2_APGA_CTL 0x1C +#define TWL4030_REG_ATX2ARXPGA 0x1D +#define TWL4030_REG_BT_IF 0x1E +#define TWL4030_REG_BTPGA 0x1F +#define TWL4030_REG_BTSTPGA 0x20 +#define TWL4030_REG_EAR_CTL 0x21 +#define TWL4030_REG_HS_SEL 0x22 +#define TWL4030_REG_HS_GAIN_SET 0x23 +#define TWL4030_REG_HS_POPN_SET 0x24 +#define TWL4030_REG_PREDL_CTL 0x25 +#define TWL4030_REG_PREDR_CTL 0x26 +#define TWL4030_REG_PRECKL_CTL 0x27 +#define TWL4030_REG_PRECKR_CTL 0x28 +#define TWL4030_REG_HFL_CTL 0x29 +#define TWL4030_REG_HFR_CTL 0x2A +#define TWL4030_REG_ALC_CTL 0x2B +#define TWL4030_REG_ALC_SET1 0x2C +#define TWL4030_REG_ALC_SET2 0x2D +#define TWL4030_REG_BOOST_CTL 0x2E +#define TWL4030_REG_SOFTVOL_CTL 0x2F +#define TWL4030_REG_DTMF_FREQSEL 0x30 +#define TWL4030_REG_DTMF_TONEXT1H 0x31 +#define TWL4030_REG_DTMF_TONEXT1L 0x32 +#define TWL4030_REG_DTMF_TONEXT2H 0x33 +#define TWL4030_REG_DTMF_TONEXT2L 0x34 +#define TWL4030_REG_DTMF_TONOFF 0x35 +#define TWL4030_REG_DTMF_WANONOFF 0x36 +#define TWL4030_REG_I2S_RX_SCRAMBLE_H 0x37 +#define TWL4030_REG_I2S_RX_SCRAMBLE_M 0x38 +#define TWL4030_REG_I2S_RX_SCRAMBLE_L 0x39 +#define TWL4030_REG_APLL_CTL 0x3A +#define TWL4030_REG_DTMF_CTL 0x3B +#define TWL4030_REG_DTMF_PGA_CTL2 0x3C +#define TWL4030_REG_DTMF_PGA_CTL1 0x3D +#define TWL4030_REG_MISC_SET_1 0x3E +#define TWL4030_REG_PCMBTMUX 0x3F +#define TWL4030_REG_RX_PATH_SEL 0x43 +#define TWL4030_REG_VDL_APGA_CTL 0x44 +#define TWL4030_REG_VIBRA_CTL 0x45 +#define TWL4030_REG_VIBRA_SET 0x46 +#define TWL4030_REG_VIBRA_PWM_SET 0x47 +#define TWL4030_REG_ANAMIC_GAIN 0x48 +#define TWL4030_REG_MISC_SET_2 0x49 + +#define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1) + +/* Bitfield Definitions */ + +/* TWL4030_CODEC_MODE (0x01) Fields */ + +#define TWL4030_APLL_RATE 0xF0 +#define TWL4030_APLL_RATE_8000 0x00 +#define TWL4030_APLL_RATE_11025 0x10 +#define TWL4030_APLL_RATE_12000 0x20 +#define TWL4030_APLL_RATE_16000 0x40 +#define TWL4030_APLL_RATE_22050 0x50 +#define TWL4030_APLL_RATE_24000 0x60 +#define TWL4030_APLL_RATE_32000 0x80 +#define TWL4030_APLL_RATE_44100 0x90 +#define TWL4030_APLL_RATE_48000 0xA0 +#define TWL4030_SEL_16K 0x04 +#define TWL4030_CODECPDZ 0x02 +#define TWL4030_OPT_MODE 0x01 + +/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ + +#define TWL4030_MICBIAS2_CTL 0x40 +#define TWL4030_MICBIAS1_CTL 0x20 +#define TWL4030_HSMICBIAS_EN 0x04 +#define TWL4030_MICBIAS2_EN 0x02 +#define TWL4030_MICBIAS1_EN 0x01 + +/* ANAMICL (0x05) Fields */ + +#define TWL4030_CNCL_OFFSET_START 0x80 +#define TWL4030_OFFSET_CNCL_SEL 0x60 +#define TWL4030_OFFSET_CNCL_SEL_ARX1 0x00 +#define TWL4030_OFFSET_CNCL_SEL_ARX2 0x20 +#define TWL4030_OFFSET_CNCL_SEL_VRX 0x40 +#define TWL4030_OFFSET_CNCL_SEL_ALL 0x60 +#define TWL4030_MICAMPL_EN 0x10 +#define TWL4030_CKMIC_EN 0x08 +#define TWL4030_AUXL_EN 0x04 +#define TWL4030_HSMIC_EN 0x02 +#define TWL4030_MAINMIC_EN 0x01 + +/* ANAMICR (0x06) Fields */ + +#define TWL4030_MICAMPR_EN 0x10 +#define TWL4030_AUXR_EN 0x04 +#define TWL4030_SUBMIC_EN 0x01 + +/* AVADC_CTL (0x07) Fields */ + +#define TWL4030_ADCL_EN 0x08 +#define TWL4030_AVADC_CLK_PRIORITY 0x04 +#define TWL4030_ADCR_EN 0x02 + +/* AUDIO_IF (0x0E) Fields */ + +#define TWL4030_AIF_SLAVE_EN 0x80 +#define TWL4030_DATA_WIDTH 0x60 +#define TWL4030_DATA_WIDTH_16S_16W 0x00 +#define TWL4030_DATA_WIDTH_32S_16W 0x40 +#define TWL4030_DATA_WIDTH_32S_24W 0x60 +#define TWL4030_AIF_FORMAT 0x18 +#define TWL4030_AIF_FORMAT_CODEC 0x00 +#define TWL4030_AIF_FORMAT_LEFT 0x08 +#define TWL4030_AIF_FORMAT_RIGHT 0x10 +#define TWL4030_AIF_FORMAT_TDM 0x18 +#define TWL4030_AIF_TRI_EN 0x04 +#define TWL4030_CLK256FS_EN 0x02 +#define TWL4030_AIF_EN 0x01 + +/* HS_GAIN_SET (0x23) Fields */ + +#define TWL4030_HSR_GAIN 0x0C +#define TWL4030_HSR_GAIN_PWR_DOWN 0x00 +#define TWL4030_HSR_GAIN_PLUS_6DB 0x04 +#define TWL4030_HSR_GAIN_0DB 0x08 +#define TWL4030_HSR_GAIN_MINUS_6DB 0x0C +#define TWL4030_HSL_GAIN 0x03 +#define TWL4030_HSL_GAIN_PWR_DOWN 0x00 +#define TWL4030_HSL_GAIN_PLUS_6DB 0x01 +#define TWL4030_HSL_GAIN_0DB 0x02 +#define TWL4030_HSL_GAIN_MINUS_6DB 0x03 + +/* HS_POPN_SET (0x24) Fields */ + +#define TWL4030_VMID_EN 0x40 +#define TWL4030_EXTMUTE 0x20 +#define TWL4030_RAMP_DELAY 0x1C +#define TWL4030_RAMP_DELAY_20MS 0x00 +#define TWL4030_RAMP_DELAY_40MS 0x04 +#define TWL4030_RAMP_DELAY_81MS 0x08 +#define TWL4030_RAMP_DELAY_161MS 0x0C +#define TWL4030_RAMP_DELAY_323MS 0x10 +#define TWL4030_RAMP_DELAY_645MS 0x14 +#define TWL4030_RAMP_DELAY_1291MS 0x18 +#define TWL4030_RAMP_DELAY_2581MS 0x1C +#define TWL4030_RAMP_EN 0x02 + +/* HFL_CTL (0x29, 0x2A) Fields */ +#define TWL4030_HF_CTL_HB_EN 0x04 +#define TWL4030_HF_CTL_LOOP_EN 0x08 +#define TWL4030_HF_CTL_RAMP_EN 0x10 +#define TWL4030_HF_CTL_REF_EN 0x20 + +/* APLL_CTL (0x3A) Fields */ + +#define TWL4030_APLL_EN 0x10 +#define TWL4030_APLL_INFREQ 0x0F +#define TWL4030_APLL_INFREQ_19200KHZ 0x05 +#define TWL4030_APLL_INFREQ_26000KHZ 0x06 +#define TWL4030_APLL_INFREQ_38400KHZ 0x0F + +/* REG_MISC_SET_1 (0x3E) Fields */ + +#define TWL4030_CLK64_EN 0x80 +#define TWL4030_SCRAMBLE_EN 0x40 +#define TWL4030_FMLOOP_EN 0x20 +#define TWL4030_SMOOTH_ANAVOL_EN 0x02 +#define TWL4030_DIGMIC_LR_SWAP_EN 0x01 + +extern struct snd_soc_dai twl4030_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl4030; + +#endif /* End of __TWL4030_AUDIO_H__ */ diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c new file mode 100644 index 000000000000..a2c5064a774b --- /dev/null +++ b/sound/soc/codecs/uda134x.c @@ -0,0 +1,668 @@ +/* + * uda134x.c -- UDA134X ALSA SoC Codec driver + * + * Modifications by Christian Pellegrin <chripell@evolware.org> + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <sound/uda134x.h> +#include <sound/l3.h> + +#include "uda134x.h" + + +#define POWER_OFF_ON_STANDBY 1 +/* + ALSA SOC usually puts the device in standby mode when it's not used + for sometime. If you define POWER_OFF_ON_STANDBY the driver will + turn off the ADC/DAC when this callback is invoked and turn it back + on when needed. Unfortunately this will result in a very light bump + (it can be audible only with good earphones). If this bothers you + just comment this line, you will have slightly higher power + consumption . Please note that sending the L3 command for ADC is + enough to make the bump, so it doesn't make difference if you + completely take off power from the codec. + */ + +#define UDA134X_RATES SNDRV_PCM_RATE_8000_48000 +#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) + +struct uda134x_priv { + int sysclk; + int dai_fmt; + + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; +}; + +/* In-data addresses are hard-coded into the reg-cache values */ +static const char uda134x_reg[UDA134X_REGS_NUM] = { + /* Extended address registers */ + 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, + /* Status, data regs */ + 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, +}; + +/* + * The codec has no support for reading its registers except for peak level... + */ +static inline unsigned int uda134x_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= UDA134X_REGS_NUM) + return -1; + return cache[reg]; +} + +/* + * Write the register cache + */ +static inline void uda134x_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, unsigned int value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= UDA134X_REGS_NUM) + return; + cache[reg] = value; +} + +/* + * Write to the uda134x registers + * + */ +static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + int ret; + u8 addr; + u8 data = value; + struct uda134x_platform_data *pd = codec->control_data; + + pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value); + + if (reg >= UDA134X_REGS_NUM) { + printk(KERN_ERR "%s unkown register: reg: %d", + __func__, reg); + return -EINVAL; + } + + uda134x_write_reg_cache(codec, reg, value); + + switch (reg) { + case UDA134X_STATUS0: + case UDA134X_STATUS1: + addr = UDA134X_STATUS_ADDR; + break; + case UDA134X_DATA000: + case UDA134X_DATA001: + case UDA134X_DATA010: + addr = UDA134X_DATA0_ADDR; + break; + case UDA134X_DATA1: + addr = UDA134X_DATA1_ADDR; + break; + default: + /* It's an extended address register */ + addr = (reg | UDA134X_EXTADDR_PREFIX); + + ret = l3_write(&pd->l3, + UDA134X_DATA0_ADDR, &addr, 1); + if (ret != 1) + return -EIO; + + addr = UDA134X_DATA0_ADDR; + data = (value | UDA134X_EXTDATA_PREFIX); + break; + } + + ret = l3_write(&pd->l3, + addr, &data, 1); + if (ret != 1) + return -EIO; + + return 0; +} + +static inline void uda134x_reset(struct snd_soc_codec *codec) +{ + u8 reset_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS0); + uda134x_write(codec, UDA134X_STATUS0, reset_reg | (1<<6)); + msleep(1); + uda134x_write(codec, UDA134X_STATUS0, reset_reg & ~(1<<6)); +} + +static int uda134x_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = uda134x_read_reg_cache(codec, UDA134X_DATA010); + + pr_debug("%s mute: %d\n", __func__, mute); + + if (mute) + mute_reg |= (1<<2); + else + mute_reg &= ~(1<<2); + + uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2)); + + return 0; +} + +static int uda134x_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + struct snd_pcm_runtime *master_runtime; + + if (uda134x->master_substream) { + master_runtime = uda134x->master_substream->runtime; + + pr_debug("%s constraining to %d bits at %d\n", __func__, + master_runtime->sample_bits, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + uda134x->slave_substream = substream; + } else + uda134x->master_substream = substream; + + return 0; +} + +static void uda134x_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + + if (uda134x->master_substream == substream) + uda134x->master_substream = uda134x->slave_substream; + + uda134x->slave_substream = NULL; +} + +static int uda134x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + u8 hw_params; + + if (substream == uda134x->slave_substream) { + pr_debug("%s ignoring hw_params for slave substream\n", + __func__); + return 0; + } + + hw_params = uda134x_read_reg_cache(codec, UDA134X_STATUS0); + hw_params &= STATUS0_SYSCLK_MASK; + hw_params &= STATUS0_DAIFMT_MASK; + + pr_debug("%s sysclk: %d, rate:%d\n", __func__, + uda134x->sysclk, params_rate(params)); + + /* set SYSCLK / fs ratio */ + switch (uda134x->sysclk / params_rate(params)) { + case 512: + break; + case 384: + hw_params |= (1<<4); + break; + case 256: + hw_params |= (1<<5); + break; + default: + printk(KERN_ERR "%s unsupported fs\n", __func__); + return -EINVAL; + } + + pr_debug("%s dai_fmt: %d, params_format:%d\n", __func__, + uda134x->dai_fmt, params_format(params)); + + /* set DAI format and word length */ + switch (uda134x->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + hw_params |= (1<<1); + break; + case SNDRV_PCM_FORMAT_S18_3LE: + hw_params |= (1<<2); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + hw_params |= ((1<<2) | (1<<1)); + break; + default: + printk(KERN_ERR "%s unsupported format (right)\n", + __func__); + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_LEFT_J: + hw_params |= (1<<3); + break; + default: + printk(KERN_ERR "%s unsupported format\n", __func__); + return -EINVAL; + } + + uda134x_write(codec, UDA134X_STATUS0, hw_params); + + return 0; +} + +static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda134x_priv *uda134x = codec->private_data; + + pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__, + clk_id, freq, dir); + + /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable + because the codec is slave. Of course limitations of the clock + master (the IIS controller) apply. + We'll error out on set_hw_params if it's not OK */ + if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) { + uda134x->sysclk = freq; + return 0; + } + + printk(KERN_ERR "%s unsupported sysclk\n", __func__); + return -EINVAL; +} + +static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda134x_priv *uda134x = codec->private_data; + + pr_debug("%s fmt: %08X\n", __func__, fmt); + + /* codec supports only full slave mode */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + printk(KERN_ERR "%s unsupported slave mode\n", __func__); + return -EINVAL; + } + + /* no support for clock inversion */ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) { + printk(KERN_ERR "%s unsupported clock inversion\n", __func__); + return -EINVAL; + } + + /* We can't setup DAI format here as it depends on the word bit num */ + /* so let's just store the value for later */ + uda134x->dai_fmt = fmt; + + return 0; +} + +static int uda134x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 reg; + struct uda134x_platform_data *pd = codec->control_data; + int i; + u8 *cache = codec->reg_cache; + + pr_debug("%s bias level %d\n", __func__, level); + + switch (level) { + case SND_SOC_BIAS_ON: + /* ADC, DAC on */ + reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); + break; + case SND_SOC_BIAS_PREPARE: + /* power on */ + if (pd->power) { + pd->power(1); + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(uda134x_reg); i++) + codec->write(codec, i, *cache++); + } + break; + case SND_SOC_BIAS_STANDBY: + /* ADC, DAC power off */ + reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); + break; + case SND_SOC_BIAS_OFF: + /* power off */ + if (pd->power) + pd->power(0); + break; + } + codec->bias_level = level; + return 0; +} + +static const char *uda134x_dsp_setting[] = {"Flat", "Minimum1", + "Minimum2", "Maximum"}; +static const char *uda134x_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *uda134x_mixmode[] = {"Differential", "Analog1", + "Analog2", "Both"}; + +static const struct soc_enum uda134x_mixer_enum[] = { +SOC_ENUM_SINGLE(UDA134X_DATA010, 0, 0x04, uda134x_dsp_setting), +SOC_ENUM_SINGLE(UDA134X_DATA010, 3, 0x04, uda134x_deemph), +SOC_ENUM_SINGLE(UDA134X_EA010, 0, 0x04, uda134x_mixmode), +}; + +static const struct snd_kcontrol_new uda1341_snd_controls[] = { +SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), +SOC_SINGLE("Capture Volume", UDA134X_EA010, 2, 0x07, 0), +SOC_SINGLE("Analog1 Volume", UDA134X_EA000, 0, 0x1F, 1), +SOC_SINGLE("Analog2 Volume", UDA134X_EA001, 0, 0x1F, 1), + +SOC_SINGLE("Mic Sensitivity", UDA134X_EA010, 2, 7, 0), +SOC_SINGLE("Mic Volume", UDA134X_EA101, 0, 0x1F, 0), + +SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0), +SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0), + +SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]), +SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), +SOC_ENUM("Input Mux", uda134x_mixer_enum[2]), + +SOC_SINGLE("AGC Switch", UDA134X_EA100, 4, 1, 0), +SOC_SINGLE("AGC Target Volume", UDA134X_EA110, 0, 0x03, 1), +SOC_SINGLE("AGC Timing", UDA134X_EA110, 2, 0x07, 0), + +SOC_SINGLE("DAC +6dB Switch", UDA134X_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC +6dB Switch", UDA134X_STATUS1, 5, 1, 0), +SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0), +SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0), +SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0), +SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), +}; + +static const struct snd_kcontrol_new uda1340_snd_controls[] = { +SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), + +SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0), +SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0), + +SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]), +SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), + +SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), +}; + +static int uda134x_add_controls(struct snd_soc_codec *codec) +{ + int err, i, n; + const struct snd_kcontrol_new *ctrls; + struct uda134x_platform_data *pd = codec->control_data; + + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1344: + n = ARRAY_SIZE(uda1340_snd_controls); + ctrls = uda1340_snd_controls; + break; + case UDA134X_UDA1341: + n = ARRAY_SIZE(uda1341_snd_controls); + ctrls = uda1341_snd_controls; + break; + default: + printk(KERN_ERR "%s unkown codec type: %d", + __func__, pd->model); + return -EINVAL; + } + + for (i = 0; i < n; i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&ctrls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +struct snd_soc_dai uda134x_dai = { + .name = "UDA134X", + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA134X_RATES, + .formats = UDA134X_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA134X_RATES, + .formats = UDA134X_FORMATS, + }, + /* pcm operations */ + .ops = { + .startup = uda134x_startup, + .shutdown = uda134x_shutdown, + .hw_params = uda134x_hw_params, + .digital_mute = uda134x_mute, + .set_sysclk = uda134x_set_dai_sysclk, + .set_fmt = uda134x_set_dai_fmt, + } +}; +EXPORT_SYMBOL(uda134x_dai); + + +static int uda134x_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct uda134x_priv *uda134x; + void *codec_setup_data = socdev->codec_data; + int ret = -ENOMEM; + struct uda134x_platform_data *pd; + + printk(KERN_INFO "UDA134X SoC Audio Codec\n"); + + if (!codec_setup_data) { + printk(KERN_ERR "UDA134X SoC codec: " + "missing L3 bitbang function\n"); + return -ENODEV; + } + + pd = codec_setup_data; + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1341: + case UDA134X_UDA1344: + break; + default: + printk(KERN_ERR "UDA134X SoC codec: " + "unsupported model %d\n", + pd->model); + return -EINVAL; + } + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return ret; + + codec = socdev->codec; + + uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL); + if (uda134x == NULL) + goto priv_err; + codec->private_data = uda134x; + + codec->reg_cache = kmemdup(uda134x_reg, sizeof(uda134x_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + goto reg_err; + + mutex_init(&codec->mutex); + + codec->reg_cache_size = sizeof(uda134x_reg); + codec->reg_cache_step = 1; + + codec->name = "UDA134X"; + codec->owner = THIS_MODULE; + codec->dai = &uda134x_dai; + codec->num_dai = 1; + codec->read = uda134x_read_reg_cache; + codec->write = uda134x_write; +#ifdef POWER_OFF_ON_STANDBY + codec->set_bias_level = uda134x_set_bias_level; +#endif + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->control_data = codec_setup_data; + + if (pd->power) + pd->power(1); + + uda134x_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register pcms\n"); + goto pcm_err; + } + + ret = uda134x_add_controls(codec); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register controls\n"); + goto pcm_err; + } + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register card\n"); + goto card_err; + } + + return 0; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); +reg_err: + kfree(codec->private_data); +priv_err: + kfree(codec); + return ret; +} + +/* power down chip */ +static int uda134x_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} + +#if defined(CONFIG_PM) +static int uda134x_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int uda134x_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + uda134x_set_bias_level(codec, SND_SOC_BIAS_ON); + return 0; +} +#else +#define uda134x_soc_suspend NULL +#define uda134x_soc_resume NULL +#endif /* CONFIG_PM */ + +struct snd_soc_codec_device soc_codec_dev_uda134x = { + .probe = uda134x_soc_probe, + .remove = uda134x_soc_remove, + .suspend = uda134x_soc_suspend, + .resume = uda134x_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_uda134x); + +static int __init uda134x_init(void) +{ + return snd_soc_register_dai(&uda134x_dai); +} +module_init(uda134x_init); + +static void __exit uda134x_exit(void) +{ + snd_soc_unregister_dai(&uda134x_dai); +} +module_exit(uda134x_exit); + +MODULE_DESCRIPTION("UDA134X ALSA soc codec driver"); +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h new file mode 100644 index 000000000000..94f440490b31 --- /dev/null +++ b/sound/soc/codecs/uda134x.h @@ -0,0 +1,36 @@ +#ifndef _UDA134X_CODEC_H +#define _UDA134X_CODEC_H + +#define UDA134X_L3ADDR 5 +#define UDA134X_DATA0_ADDR ((UDA134X_L3ADDR << 2) | 0) +#define UDA134X_DATA1_ADDR ((UDA134X_L3ADDR << 2) | 1) +#define UDA134X_STATUS_ADDR ((UDA134X_L3ADDR << 2) | 2) + +#define UDA134X_EXTADDR_PREFIX 0xC0 +#define UDA134X_EXTDATA_PREFIX 0xE0 + +/* UDA134X registers */ +#define UDA134X_EA000 0 +#define UDA134X_EA001 1 +#define UDA134X_EA010 2 +#define UDA134X_EA011 3 +#define UDA134X_EA100 4 +#define UDA134X_EA101 5 +#define UDA134X_EA110 6 +#define UDA134X_EA111 7 +#define UDA134X_STATUS0 8 +#define UDA134X_STATUS1 9 +#define UDA134X_DATA000 10 +#define UDA134X_DATA001 11 +#define UDA134X_DATA010 12 +#define UDA134X_DATA1 13 + +#define UDA134X_REGS_NUM 14 + +#define STATUS0_DAIFMT_MASK (~(7<<1)) +#define STATUS0_SYSCLK_MASK (~(3<<4)) + +extern struct snd_soc_dai uda134x_dai; +extern struct snd_soc_codec_device soc_codec_dev_uda134x; + +#endif diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index a69ee72a7af5..e6bf0844fbf3 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -407,7 +407,8 @@ static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai, * when the DAI is being clocked by the CPU DAI. It's up to the * machine and cpu DAI driver to do this before we are called. */ -static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) +static int uda1380_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -439,7 +440,8 @@ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) } static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -477,7 +479,8 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream) +static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -560,8 +563,6 @@ struct snd_soc_dai uda1380_dai[] = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, - }, - .dai_ops = { .digital_mute = uda1380_mute, .set_fmt = uda1380_set_dai_fmt, }, @@ -579,8 +580,6 @@ struct snd_soc_dai uda1380_dai[] = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, - }, - .dai_ops = { .digital_mute = uda1380_mute, .set_fmt = uda1380_set_dai_fmt, }, @@ -598,8 +597,6 @@ struct snd_soc_dai uda1380_dai[] = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, - }, - .dai_ops = { .set_fmt = uda1380_set_dai_fmt, }, }, @@ -680,7 +677,7 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) /* uda1380 init */ uda1380_add_controls(codec); uda1380_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { pr_err("uda1380: failed to register card\n"); goto card_err; @@ -844,6 +841,18 @@ struct snd_soc_codec_device soc_codec_dev_uda1380 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); +static int __init uda1380_modinit(void) +{ + return snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); +} +module_init(uda1380_modinit); + +static void __exit uda1380_exit(void) +{ + snd_soc_unregister_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); +} +module_exit(uda1380_exit); + MODULE_AUTHOR("Giorgio Padrin"); MODULE_DESCRIPTION("Audio support for codec Philips UDA1380"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c new file mode 100644 index 000000000000..e3989d406f54 --- /dev/null +++ b/sound/soc/codecs/wm8350.c @@ -0,0 +1,1583 @@ +/* + * wm8350.c -- WM8350 ALSA SoC audio driver + * + * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC. + * + * Author: Liam Girdwood <lg@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/platform_device.h> +#include <linux/mfd/wm8350/audio.h> +#include <linux/mfd/wm8350/core.h> +#include <linux/regulator/consumer.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8350.h" + +#define WM8350_OUTn_0dB 0x39 + +#define WM8350_RAMP_NONE 0 +#define WM8350_RAMP_UP 1 +#define WM8350_RAMP_DOWN 2 + +/* We only include the analogue supplies here; the digital supplies + * need to be available well before this driver can be probed. + */ +static const char *supply_names[] = { + "AVDD", + "HPVDD", +}; + +struct wm8350_output { + u16 active; + u16 left_vol; + u16 right_vol; + u16 ramp; + u16 mute; +}; + +struct wm8350_data { + struct snd_soc_codec codec; + struct wm8350_output out1; + struct wm8350_output out2; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; +}; + +static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct wm8350 *wm8350 = codec->control_data; + return wm8350->reg_cache[reg]; +} + +static unsigned int wm8350_codec_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct wm8350 *wm8350 = codec->control_data; + return wm8350_reg_read(wm8350, reg); +} + +static int wm8350_codec_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct wm8350 *wm8350 = codec->control_data; + return wm8350_reg_write(wm8350, reg, value); +} + +/* + * Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown. + */ +static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec) +{ + struct wm8350_data *wm8350_data = codec->private_data; + struct wm8350_output *out1 = &wm8350_data->out1; + struct wm8350 *wm8350 = codec->control_data; + int left_complete = 0, right_complete = 0; + u16 reg, val; + + /* left channel */ + reg = wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME); + val = (reg & WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT; + + if (out1->ramp == WM8350_RAMP_UP) { + /* ramp step up */ + if (val < out1->left_vol) { + val++; + reg &= ~WM8350_OUT1L_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME, + reg | (val << WM8350_OUT1L_VOL_SHIFT)); + } else + left_complete = 1; + } else if (out1->ramp == WM8350_RAMP_DOWN) { + /* ramp step down */ + if (val > 0) { + val--; + reg &= ~WM8350_OUT1L_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME, + reg | (val << WM8350_OUT1L_VOL_SHIFT)); + } else + left_complete = 1; + } else + return 1; + + /* right channel */ + reg = wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME); + val = (reg & WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT; + if (out1->ramp == WM8350_RAMP_UP) { + /* ramp step up */ + if (val < out1->right_vol) { + val++; + reg &= ~WM8350_OUT1R_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME, + reg | (val << WM8350_OUT1R_VOL_SHIFT)); + } else + right_complete = 1; + } else if (out1->ramp == WM8350_RAMP_DOWN) { + /* ramp step down */ + if (val > 0) { + val--; + reg &= ~WM8350_OUT1R_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME, + reg | (val << WM8350_OUT1R_VOL_SHIFT)); + } else + right_complete = 1; + } + + /* only hit the update bit if either volume has changed this step */ + if (!left_complete || !right_complete) + wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME, WM8350_OUT1_VU); + + return left_complete & right_complete; +} + +/* + * Ramp OUT2 PGA volume to minimise pops at stream startup and shutdown. + */ +static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec) +{ + struct wm8350_data *wm8350_data = codec->private_data; + struct wm8350_output *out2 = &wm8350_data->out2; + struct wm8350 *wm8350 = codec->control_data; + int left_complete = 0, right_complete = 0; + u16 reg, val; + + /* left channel */ + reg = wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME); + val = (reg & WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT; + if (out2->ramp == WM8350_RAMP_UP) { + /* ramp step up */ + if (val < out2->left_vol) { + val++; + reg &= ~WM8350_OUT2L_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME, + reg | (val << WM8350_OUT1L_VOL_SHIFT)); + } else + left_complete = 1; + } else if (out2->ramp == WM8350_RAMP_DOWN) { + /* ramp step down */ + if (val > 0) { + val--; + reg &= ~WM8350_OUT2L_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME, + reg | (val << WM8350_OUT1L_VOL_SHIFT)); + } else + left_complete = 1; + } else + return 1; + + /* right channel */ + reg = wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME); + val = (reg & WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT; + if (out2->ramp == WM8350_RAMP_UP) { + /* ramp step up */ + if (val < out2->right_vol) { + val++; + reg &= ~WM8350_OUT2R_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME, + reg | (val << WM8350_OUT1R_VOL_SHIFT)); + } else + right_complete = 1; + } else if (out2->ramp == WM8350_RAMP_DOWN) { + /* ramp step down */ + if (val > 0) { + val--; + reg &= ~WM8350_OUT2R_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME, + reg | (val << WM8350_OUT1R_VOL_SHIFT)); + } else + right_complete = 1; + } + + /* only hit the update bit if either volume has changed this step */ + if (!left_complete || !right_complete) + wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME, WM8350_OUT2_VU); + + return left_complete & right_complete; +} + +/* + * This work ramps both output PGAs at stream start/stop time to + * minimise pop associated with DAPM power switching. + * It's best to enable Zero Cross when ramping occurs to minimise any + * zipper noises. + */ +static void wm8350_pga_work(struct work_struct *work) +{ + struct snd_soc_codec *codec = + container_of(work, struct snd_soc_codec, delayed_work.work); + struct wm8350_data *wm8350_data = codec->private_data; + struct wm8350_output *out1 = &wm8350_data->out1, + *out2 = &wm8350_data->out2; + int i, out1_complete, out2_complete; + + /* do we need to ramp at all ? */ + if (out1->ramp == WM8350_RAMP_NONE && out2->ramp == WM8350_RAMP_NONE) + return; + + /* PGA volumes have 6 bits of resolution to ramp */ + for (i = 0; i <= 63; i++) { + out1_complete = 1, out2_complete = 1; + if (out1->ramp != WM8350_RAMP_NONE) + out1_complete = wm8350_out1_ramp_step(codec); + if (out2->ramp != WM8350_RAMP_NONE) + out2_complete = wm8350_out2_ramp_step(codec); + + /* ramp finished ? */ + if (out1_complete && out2_complete) + break; + + /* we need to delay longer on the up ramp */ + if (out1->ramp == WM8350_RAMP_UP || + out2->ramp == WM8350_RAMP_UP) { + /* delay is longer over 0dB as increases are larger */ + if (i >= WM8350_OUTn_0dB) + schedule_timeout_interruptible(msecs_to_jiffies + (2)); + else + schedule_timeout_interruptible(msecs_to_jiffies + (1)); + } else + udelay(50); /* doesn't matter if we delay longer */ + } + + out1->ramp = WM8350_RAMP_NONE; + out2->ramp = WM8350_RAMP_NONE; +} + +/* + * WM8350 Controls + */ + +static int pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8350_data *wm8350_data = codec->private_data; + struct wm8350_output *out; + + switch (w->shift) { + case 0: + case 1: + out = &wm8350_data->out1; + break; + case 2: + case 3: + out = &wm8350_data->out2; + break; + + default: + BUG(); + return -1; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + out->ramp = WM8350_RAMP_UP; + out->active = 1; + + if (!delayed_work_pending(&codec->delayed_work)) + schedule_delayed_work(&codec->delayed_work, + msecs_to_jiffies(1)); + break; + + case SND_SOC_DAPM_PRE_PMD: + out->ramp = WM8350_RAMP_DOWN; + out->active = 0; + + if (!delayed_work_pending(&codec->delayed_work)) + schedule_delayed_work(&codec->delayed_work, + msecs_to_jiffies(1)); + break; + } + + return 0; +} + +static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8350_data *wm8350_priv = codec->private_data; + struct wm8350_output *out = NULL; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int ret; + unsigned int reg = mc->reg; + u16 val; + + /* For OUT1 and OUT2 we shadow the values and only actually write + * them out when active in order to ensure the amplifier comes on + * as quietly as possible. */ + switch (reg) { + case WM8350_LOUT1_VOLUME: + out = &wm8350_priv->out1; + break; + case WM8350_LOUT2_VOLUME: + out = &wm8350_priv->out2; + break; + default: + break; + } + + if (out) { + out->left_vol = ucontrol->value.integer.value[0]; + out->right_vol = ucontrol->value.integer.value[1]; + if (!out->active) + return 1; + } + + ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + if (ret < 0) + return ret; + + /* now hit the volume update bits (always bit 8) */ + val = wm8350_codec_read(codec, reg); + wm8350_codec_write(codec, reg, val | WM8350_OUT1_VU); + return 1; +} + +static int wm8350_get_volsw_2r(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8350_data *wm8350_priv = codec->private_data; + struct wm8350_output *out1 = &wm8350_priv->out1; + struct wm8350_output *out2 = &wm8350_priv->out2; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + + /* If these are cached registers use the cache */ + switch (reg) { + case WM8350_LOUT1_VOLUME: + ucontrol->value.integer.value[0] = out1->left_vol; + ucontrol->value.integer.value[1] = out1->right_vol; + return 0; + + case WM8350_LOUT2_VOLUME: + ucontrol->value.integer.value[0] = out2->left_vol; + ucontrol->value.integer.value[1] = out2->right_vol; + return 0; + + default: + break; + } + + return snd_soc_get_volsw_2r(kcontrol, ucontrol); +} + +/* double control with volume update */ +#define SOC_WM8350_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \ + xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r, \ + .get = wm8350_get_volsw_2r, .put = wm8350_put_volsw_2r_vu, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .rshift = xshift, .max = xmax, .invert = xinvert}, } + +static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; +static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" }; +static const char *wm8350_dacmutem[] = { "Normal", "Soft" }; +static const char *wm8350_dacmutes[] = { "Fast", "Slow" }; +static const char *wm8350_dacfilter[] = { "Normal", "Sloping" }; +static const char *wm8350_adcfilter[] = { "None", "High Pass" }; +static const char *wm8350_adchp[] = { "44.1kHz", "8kHz", "16kHz", "32kHz" }; +static const char *wm8350_lr[] = { "Left", "Right" }; + +static const struct soc_enum wm8350_enum[] = { + SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 4, 4, wm8350_deemp), + SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 0, 4, wm8350_pol), + SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 14, 2, wm8350_dacmutem), + SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 13, 2, wm8350_dacmutes), + SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 12, 2, wm8350_dacfilter), + SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 15, 2, wm8350_adcfilter), + SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 8, 4, wm8350_adchp), + SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 0, 4, wm8350_pol), + SOC_ENUM_SINGLE(WM8350_INPUT_MIXER_VOLUME, 15, 2, wm8350_lr), +}; + +static DECLARE_TLV_DB_LINEAR(pre_amp_tlv, -1200, 3525); +static DECLARE_TLV_DB_LINEAR(out_pga_tlv, -5700, 600); +static DECLARE_TLV_DB_SCALE(dac_pcm_tlv, -7163, 36, 1); +static DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -12700, 50, 1); +static DECLARE_TLV_DB_SCALE(out_mix_tlv, -1500, 300, 1); + +static const unsigned int capture_sd_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 12, TLV_DB_SCALE_ITEM(-3600, 300, 1), + 13, 15, TLV_DB_SCALE_ITEM(0, 0, 0), +}; + +static const struct snd_kcontrol_new wm8350_snd_controls[] = { + SOC_ENUM("Playback Deemphasis", wm8350_enum[0]), + SOC_ENUM("Playback DAC Inversion", wm8350_enum[1]), + SOC_WM8350_DOUBLE_R_TLV("Playback PCM Volume", + WM8350_DAC_DIGITAL_VOLUME_L, + WM8350_DAC_DIGITAL_VOLUME_R, + 0, 255, 0, dac_pcm_tlv), + SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]), + SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]), + SOC_ENUM("Playback PCM Filter", wm8350_enum[4]), + SOC_ENUM("Capture PCM Filter", wm8350_enum[5]), + SOC_ENUM("Capture PCM HP Filter", wm8350_enum[6]), + SOC_ENUM("Capture ADC Inversion", wm8350_enum[7]), + SOC_WM8350_DOUBLE_R_TLV("Capture PCM Volume", + WM8350_ADC_DIGITAL_VOLUME_L, + WM8350_ADC_DIGITAL_VOLUME_R, + 0, 255, 0, adc_pcm_tlv), + SOC_DOUBLE_TLV("Capture Sidetone Volume", + WM8350_ADC_DIVIDER, + 8, 4, 15, 1, capture_sd_tlv), + SOC_WM8350_DOUBLE_R_TLV("Capture Volume", + WM8350_LEFT_INPUT_VOLUME, + WM8350_RIGHT_INPUT_VOLUME, + 2, 63, 0, pre_amp_tlv), + SOC_DOUBLE_R("Capture ZC Switch", + WM8350_LEFT_INPUT_VOLUME, + WM8350_RIGHT_INPUT_VOLUME, 13, 1, 0), + SOC_SINGLE_TLV("Left Input Left Sidetone Volume", + WM8350_OUTPUT_LEFT_MIXER_VOLUME, 1, 7, 0, out_mix_tlv), + SOC_SINGLE_TLV("Left Input Right Sidetone Volume", + WM8350_OUTPUT_LEFT_MIXER_VOLUME, + 5, 7, 0, out_mix_tlv), + SOC_SINGLE_TLV("Left Input Bypass Volume", + WM8350_OUTPUT_LEFT_MIXER_VOLUME, + 9, 7, 0, out_mix_tlv), + SOC_SINGLE_TLV("Right Input Left Sidetone Volume", + WM8350_OUTPUT_RIGHT_MIXER_VOLUME, + 1, 7, 0, out_mix_tlv), + SOC_SINGLE_TLV("Right Input Right Sidetone Volume", + WM8350_OUTPUT_RIGHT_MIXER_VOLUME, + 5, 7, 0, out_mix_tlv), + SOC_SINGLE_TLV("Right Input Bypass Volume", + WM8350_OUTPUT_RIGHT_MIXER_VOLUME, + 13, 7, 0, out_mix_tlv), + SOC_SINGLE("Left Input Mixer +20dB Switch", + WM8350_INPUT_MIXER_VOLUME_L, 0, 1, 0), + SOC_SINGLE("Right Input Mixer +20dB Switch", + WM8350_INPUT_MIXER_VOLUME_R, 0, 1, 0), + SOC_SINGLE_TLV("Out4 Capture Volume", + WM8350_INPUT_MIXER_VOLUME, + 1, 7, 0, out_mix_tlv), + SOC_WM8350_DOUBLE_R_TLV("Out1 Playback Volume", + WM8350_LOUT1_VOLUME, + WM8350_ROUT1_VOLUME, + 2, 63, 0, out_pga_tlv), + SOC_DOUBLE_R("Out1 Playback ZC Switch", + WM8350_LOUT1_VOLUME, + WM8350_ROUT1_VOLUME, 13, 1, 0), + SOC_WM8350_DOUBLE_R_TLV("Out2 Playback Volume", + WM8350_LOUT2_VOLUME, + WM8350_ROUT2_VOLUME, + 2, 63, 0, out_pga_tlv), + SOC_DOUBLE_R("Out2 Playback ZC Switch", WM8350_LOUT2_VOLUME, + WM8350_ROUT2_VOLUME, 13, 1, 0), + SOC_SINGLE("Out2 Right Invert Switch", WM8350_ROUT2_VOLUME, 10, 1, 0), + SOC_SINGLE_TLV("Out2 Beep Volume", WM8350_BEEP_VOLUME, + 5, 7, 0, out_mix_tlv), + + SOC_DOUBLE_R("Out1 Playback Switch", + WM8350_LOUT1_VOLUME, + WM8350_ROUT1_VOLUME, + 14, 1, 1), + SOC_DOUBLE_R("Out2 Playback Switch", + WM8350_LOUT2_VOLUME, + WM8350_ROUT2_VOLUME, + 14, 1, 1), +}; + +/* + * DAPM Controls + */ + +/* Left Playback Mixer */ +static const struct snd_kcontrol_new wm8350_left_play_mixer_controls[] = { + SOC_DAPM_SINGLE("Playback Switch", + WM8350_LEFT_MIXER_CONTROL, 11, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", + WM8350_LEFT_MIXER_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("Right Playback Switch", + WM8350_LEFT_MIXER_CONTROL, 12, 1, 0), + SOC_DAPM_SINGLE("Left Sidetone Switch", + WM8350_LEFT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("Right Sidetone Switch", + WM8350_LEFT_MIXER_CONTROL, 1, 1, 0), +}; + +/* Right Playback Mixer */ +static const struct snd_kcontrol_new wm8350_right_play_mixer_controls[] = { + SOC_DAPM_SINGLE("Playback Switch", + WM8350_RIGHT_MIXER_CONTROL, 12, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", + WM8350_RIGHT_MIXER_CONTROL, 3, 1, 0), + SOC_DAPM_SINGLE("Left Playback Switch", + WM8350_RIGHT_MIXER_CONTROL, 11, 1, 0), + SOC_DAPM_SINGLE("Left Sidetone Switch", + WM8350_RIGHT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("Right Sidetone Switch", + WM8350_RIGHT_MIXER_CONTROL, 1, 1, 0), +}; + +/* Out4 Mixer */ +static const struct snd_kcontrol_new wm8350_out4_mixer_controls[] = { + SOC_DAPM_SINGLE("Right Playback Switch", + WM8350_OUT4_MIXER_CONTROL, 12, 1, 0), + SOC_DAPM_SINGLE("Left Playback Switch", + WM8350_OUT4_MIXER_CONTROL, 11, 1, 0), + SOC_DAPM_SINGLE("Right Capture Switch", + WM8350_OUT4_MIXER_CONTROL, 9, 1, 0), + SOC_DAPM_SINGLE("Out3 Playback Switch", + WM8350_OUT4_MIXER_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("Right Mixer Switch", + WM8350_OUT4_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("Left Mixer Switch", + WM8350_OUT4_MIXER_CONTROL, 0, 1, 0), +}; + +/* Out3 Mixer */ +static const struct snd_kcontrol_new wm8350_out3_mixer_controls[] = { + SOC_DAPM_SINGLE("Left Playback Switch", + WM8350_OUT3_MIXER_CONTROL, 11, 1, 0), + SOC_DAPM_SINGLE("Left Capture Switch", + WM8350_OUT3_MIXER_CONTROL, 8, 1, 0), + SOC_DAPM_SINGLE("Out4 Playback Switch", + WM8350_OUT3_MIXER_CONTROL, 3, 1, 0), + SOC_DAPM_SINGLE("Left Mixer Switch", + WM8350_OUT3_MIXER_CONTROL, 0, 1, 0), +}; + +/* Left Input Mixer */ +static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = { + SOC_DAPM_SINGLE_TLV("L2 Capture Volume", + WM8350_INPUT_MIXER_VOLUME_L, 1, 7, 0, out_mix_tlv), + SOC_DAPM_SINGLE_TLV("L3 Capture Volume", + WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv), + SOC_DAPM_SINGLE("PGA Capture Switch", + WM8350_LEFT_INPUT_VOLUME, 14, 1, 0), +}; + +/* Right Input Mixer */ +static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = { + SOC_DAPM_SINGLE_TLV("L2 Capture Volume", + WM8350_INPUT_MIXER_VOLUME_R, 5, 7, 0, out_mix_tlv), + SOC_DAPM_SINGLE_TLV("L3 Capture Volume", + WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv), + SOC_DAPM_SINGLE("PGA Capture Switch", + WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0), +}; + +/* Left Mic Mixer */ +static const struct snd_kcontrol_new wm8350_left_mic_mixer_controls[] = { + SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 2, 1, 0), +}; + +/* Right Mic Mixer */ +static const struct snd_kcontrol_new wm8350_right_mic_mixer_controls[] = { + SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 9, 1, 0), + SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 8, 1, 0), + SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 10, 1, 0), +}; + +/* Beep Switch */ +static const struct snd_kcontrol_new wm8350_beep_switch_controls = +SOC_DAPM_SINGLE("Switch", WM8350_BEEP_VOLUME, 15, 1, 1); + +/* Out4 Capture Mux */ +static const struct snd_kcontrol_new wm8350_out4_capture_controls = +SOC_DAPM_ENUM("Route", wm8350_enum[8]); + +static const struct snd_soc_dapm_widget wm8350_dapm_widgets[] = { + + SND_SOC_DAPM_PGA("IN3R PGA", WM8350_POWER_MGMT_2, 11, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN3L PGA", WM8350_POWER_MGMT_2, 10, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("Right Out2 PGA", WM8350_POWER_MGMT_3, 3, 0, NULL, + 0, pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("Left Out2 PGA", WM8350_POWER_MGMT_3, 2, 0, NULL, 0, + pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("Right Out1 PGA", WM8350_POWER_MGMT_3, 1, 0, NULL, + 0, pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("Left Out1 PGA", WM8350_POWER_MGMT_3, 0, 0, NULL, 0, + pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_MIXER("Right Capture Mixer", WM8350_POWER_MGMT_2, + 7, 0, &wm8350_right_capt_mixer_controls[0], + ARRAY_SIZE(wm8350_right_capt_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Capture Mixer", WM8350_POWER_MGMT_2, + 6, 0, &wm8350_left_capt_mixer_controls[0], + ARRAY_SIZE(wm8350_left_capt_mixer_controls)), + + SND_SOC_DAPM_MIXER("Out4 Mixer", WM8350_POWER_MGMT_2, 5, 0, + &wm8350_out4_mixer_controls[0], + ARRAY_SIZE(wm8350_out4_mixer_controls)), + + SND_SOC_DAPM_MIXER("Out3 Mixer", WM8350_POWER_MGMT_2, 4, 0, + &wm8350_out3_mixer_controls[0], + ARRAY_SIZE(wm8350_out3_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Playback Mixer", WM8350_POWER_MGMT_2, 1, 0, + &wm8350_right_play_mixer_controls[0], + ARRAY_SIZE(wm8350_right_play_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Playback Mixer", WM8350_POWER_MGMT_2, 0, 0, + &wm8350_left_play_mixer_controls[0], + ARRAY_SIZE(wm8350_left_play_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Mic Mixer", WM8350_POWER_MGMT_2, 8, 0, + &wm8350_left_mic_mixer_controls[0], + ARRAY_SIZE(wm8350_left_mic_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Mic Mixer", WM8350_POWER_MGMT_2, 9, 0, + &wm8350_right_mic_mixer_controls[0], + ARRAY_SIZE(wm8350_right_mic_mixer_controls)), + + /* virtual mixer for Beep and Out2R */ + SND_SOC_DAPM_MIXER("Out2 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Beep", WM8350_POWER_MGMT_3, 7, 0, + &wm8350_beep_switch_controls), + + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", + WM8350_POWER_MGMT_4, 3, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", + WM8350_POWER_MGMT_4, 2, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", + WM8350_POWER_MGMT_4, 5, 0), + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", + WM8350_POWER_MGMT_4, 4, 0), + + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8350_POWER_MGMT_1, 4, 0), + + SND_SOC_DAPM_MUX("Out4 Capture Channel", SND_SOC_NOPM, 0, 0, + &wm8350_out4_capture_controls), + + SND_SOC_DAPM_OUTPUT("OUT1R"), + SND_SOC_DAPM_OUTPUT("OUT1L"), + SND_SOC_DAPM_OUTPUT("OUT2R"), + SND_SOC_DAPM_OUTPUT("OUT2L"), + SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_OUTPUT("OUT4"), + + SND_SOC_DAPM_INPUT("IN1RN"), + SND_SOC_DAPM_INPUT("IN1RP"), + SND_SOC_DAPM_INPUT("IN2R"), + SND_SOC_DAPM_INPUT("IN1LP"), + SND_SOC_DAPM_INPUT("IN1LN"), + SND_SOC_DAPM_INPUT("IN2L"), + SND_SOC_DAPM_INPUT("IN3R"), + SND_SOC_DAPM_INPUT("IN3L"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + + /* left playback mixer */ + {"Left Playback Mixer", "Playback Switch", "Left DAC"}, + {"Left Playback Mixer", "Left Bypass Switch", "IN3L PGA"}, + {"Left Playback Mixer", "Right Playback Switch", "Right DAC"}, + {"Left Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"}, + {"Left Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"}, + + /* right playback mixer */ + {"Right Playback Mixer", "Playback Switch", "Right DAC"}, + {"Right Playback Mixer", "Right Bypass Switch", "IN3R PGA"}, + {"Right Playback Mixer", "Left Playback Switch", "Left DAC"}, + {"Right Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"}, + {"Right Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"}, + + /* out4 playback mixer */ + {"Out4 Mixer", "Right Playback Switch", "Right DAC"}, + {"Out4 Mixer", "Left Playback Switch", "Left DAC"}, + {"Out4 Mixer", "Right Capture Switch", "Right Capture Mixer"}, + {"Out4 Mixer", "Out3 Playback Switch", "Out3 Mixer"}, + {"Out4 Mixer", "Right Mixer Switch", "Right Playback Mixer"}, + {"Out4 Mixer", "Left Mixer Switch", "Left Playback Mixer"}, + {"OUT4", NULL, "Out4 Mixer"}, + + /* out3 playback mixer */ + {"Out3 Mixer", "Left Playback Switch", "Left DAC"}, + {"Out3 Mixer", "Left Capture Switch", "Left Capture Mixer"}, + {"Out3 Mixer", "Left Mixer Switch", "Left Playback Mixer"}, + {"Out3 Mixer", "Out4 Playback Switch", "Out4 Mixer"}, + {"OUT3", NULL, "Out3 Mixer"}, + + /* out2 */ + {"Right Out2 PGA", NULL, "Right Playback Mixer"}, + {"Left Out2 PGA", NULL, "Left Playback Mixer"}, + {"OUT2L", NULL, "Left Out2 PGA"}, + {"OUT2R", NULL, "Right Out2 PGA"}, + + /* out1 */ + {"Right Out1 PGA", NULL, "Right Playback Mixer"}, + {"Left Out1 PGA", NULL, "Left Playback Mixer"}, + {"OUT1L", NULL, "Left Out1 PGA"}, + {"OUT1R", NULL, "Right Out1 PGA"}, + + /* ADCs */ + {"Left ADC", NULL, "Left Capture Mixer"}, + {"Right ADC", NULL, "Right Capture Mixer"}, + + /* Left capture mixer */ + {"Left Capture Mixer", "L2 Capture Volume", "IN2L"}, + {"Left Capture Mixer", "L3 Capture Volume", "IN3L PGA"}, + {"Left Capture Mixer", "PGA Capture Switch", "Left Mic Mixer"}, + {"Left Capture Mixer", NULL, "Out4 Capture Channel"}, + + /* Right capture mixer */ + {"Right Capture Mixer", "L2 Capture Volume", "IN2R"}, + {"Right Capture Mixer", "L3 Capture Volume", "IN3R PGA"}, + {"Right Capture Mixer", "PGA Capture Switch", "Right Mic Mixer"}, + {"Right Capture Mixer", NULL, "Out4 Capture Channel"}, + + /* L3 Inputs */ + {"IN3L PGA", NULL, "IN3L"}, + {"IN3R PGA", NULL, "IN3R"}, + + /* Left Mic mixer */ + {"Left Mic Mixer", "INN Capture Switch", "IN1LN"}, + {"Left Mic Mixer", "INP Capture Switch", "IN1LP"}, + {"Left Mic Mixer", "IN2 Capture Switch", "IN2L"}, + + /* Right Mic mixer */ + {"Right Mic Mixer", "INN Capture Switch", "IN1RN"}, + {"Right Mic Mixer", "INP Capture Switch", "IN1RP"}, + {"Right Mic Mixer", "IN2 Capture Switch", "IN2R"}, + + /* out 4 capture */ + {"Out4 Capture Channel", NULL, "Out4 Mixer"}, + + /* Beep */ + {"Beep", NULL, "IN3R PGA"}, +}; + +static int wm8350_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8350_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +static int wm8350_add_widgets(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, + wm8350_dapm_widgets, + ARRAY_SIZE(wm8350_dapm_widgets)); + if (ret != 0) { + dev_err(codec->dev, "dapm control register failed\n"); + return ret; + } + + /* set up audio paths */ + ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + if (ret != 0) { + dev_err(codec->dev, "DAPM route register failed\n"); + return ret; + } + + return snd_soc_dapm_new_widgets(codec); +} + +static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8350 *wm8350 = codec->control_data; + u16 fll_4; + + switch (clk_id) { + case WM8350_MCLK_SEL_MCLK: + wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_1, + WM8350_MCLK_SEL); + break; + case WM8350_MCLK_SEL_PLL_MCLK: + case WM8350_MCLK_SEL_PLL_DAC: + case WM8350_MCLK_SEL_PLL_ADC: + case WM8350_MCLK_SEL_PLL_32K: + wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_1, + WM8350_MCLK_SEL); + fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) & + ~WM8350_FLL_CLK_SRC_MASK; + wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id); + break; + } + + /* MCLK direction */ + if (dir == WM8350_MCLK_DIR_OUT) + wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_2, + WM8350_MCLK_DIR); + else + wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_2, + WM8350_MCLK_DIR); + + return 0; +} + +static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 val; + + switch (div_id) { + case WM8350_ADC_CLKDIV: + val = wm8350_codec_read(codec, WM8350_ADC_DIVIDER) & + ~WM8350_ADC_CLKDIV_MASK; + wm8350_codec_write(codec, WM8350_ADC_DIVIDER, val | div); + break; + case WM8350_DAC_CLKDIV: + val = wm8350_codec_read(codec, WM8350_DAC_CLOCK_CONTROL) & + ~WM8350_DAC_CLKDIV_MASK; + wm8350_codec_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div); + break; + case WM8350_BCLK_CLKDIV: + val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + ~WM8350_BCLK_DIV_MASK; + wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + break; + case WM8350_OPCLK_CLKDIV: + val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + ~WM8350_OPCLK_DIV_MASK; + wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + break; + case WM8350_SYS_CLKDIV: + val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + ~WM8350_MCLK_DIV_MASK; + wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + break; + case WM8350_DACLR_CLKDIV: + val = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) & + ~WM8350_DACLRC_RATE_MASK; + wm8350_codec_write(codec, WM8350_DAC_LR_RATE, val | div); + break; + case WM8350_ADCLR_CLKDIV: + val = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) & + ~WM8350_ADCLRC_RATE_MASK; + wm8350_codec_write(codec, WM8350_ADC_LR_RATE, val | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) & + ~(WM8350_AIF_BCLK_INV | WM8350_AIF_LRCLK_INV | WM8350_AIF_FMT_MASK); + u16 master = wm8350_codec_read(codec, WM8350_AI_DAC_CONTROL) & + ~WM8350_BCLK_MSTR; + u16 dac_lrc = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) & + ~WM8350_DACLRC_ENA; + u16 adc_lrc = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) & + ~WM8350_ADCLRC_ENA; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + master |= WM8350_BCLK_MSTR; + dac_lrc |= WM8350_DACLRC_ENA; + adc_lrc |= WM8350_ADCLRC_ENA; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x2 << 8; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x1 << 8; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x3 << 8; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x3 << 8; /* lg not sure which mode */ + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= WM8350_AIF_LRCLK_INV | WM8350_AIF_BCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= WM8350_AIF_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= WM8350_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + + wm8350_codec_write(codec, WM8350_AI_FORMATING, iface); + wm8350_codec_write(codec, WM8350_AI_DAC_CONTROL, master); + wm8350_codec_write(codec, WM8350_DAC_LR_RATE, dac_lrc); + wm8350_codec_write(codec, WM8350_ADC_LR_RATE, adc_lrc); + return 0; +} + +static int wm8350_pcm_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int master = wm8350_codec_cache_read(codec, WM8350_AI_DAC_CONTROL) & + WM8350_BCLK_MSTR; + int enabled = 0; + + /* Check that the DACs or ADCs are enabled since they are + * required for LRC in master mode. The DACs or ADCs need a + * valid audio path i.e. pin -> ADC or DAC -> pin before + * the LRC will be enabled in master mode. */ + if (!master && cmd != SNDRV_PCM_TRIGGER_START) + return 0; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) & + (WM8350_ADCR_ENA | WM8350_ADCL_ENA); + } else { + enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) & + (WM8350_DACR_ENA | WM8350_DACL_ENA); + } + + if (!enabled) { + dev_err(codec->dev, + "%s: invalid audio path - no clocks available\n", + __func__); + return -EINVAL; + } + return 0; +} + +static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) & + ~WM8350_AIF_WL_MASK; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x1 << 10; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x2 << 10; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x3 << 10; + break; + } + + wm8350_codec_write(codec, WM8350_AI_FORMATING, iface); + return 0; +} + +static int wm8350_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8350 *wm8350 = codec->control_data; + + if (mute) + wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA); + else + wm8350_clear_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA); + return 0; +} + +/* FLL divisors */ +struct _fll_div { + int div; /* FLL_OUTDIV */ + int n; + int k; + int ratio; /* FLL_FRATIO */ +}; + +/* The size in bits of the fll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_FLL_SIZE ((1 << 16) * 10) + +static inline int fll_factors(struct _fll_div *fll_div, unsigned int input, + unsigned int output) +{ + u64 Kpart; + unsigned int t1, t2, K, Nmod; + + if (output >= 2815250 && output <= 3125000) + fll_div->div = 0x4; + else if (output >= 5625000 && output <= 6250000) + fll_div->div = 0x3; + else if (output >= 11250000 && output <= 12500000) + fll_div->div = 0x2; + else if (output >= 22500000 && output <= 25000000) + fll_div->div = 0x1; + else { + printk(KERN_ERR "wm8350: fll freq %d out of range\n", output); + return -EINVAL; + } + + if (input > 48000) + fll_div->ratio = 1; + else + fll_div->ratio = 8; + + t1 = output * (1 << (fll_div->div + 1)); + t2 = input * fll_div->ratio; + + fll_div->n = t1 / t2; + Nmod = t1 % t2; + + if (Nmod) { + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + do_div(Kpart, t2); + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + fll_div->k = K; + } else + fll_div->k = 0; + + return 0; +} + +static int wm8350_set_fll(struct snd_soc_dai *codec_dai, + int pll_id, unsigned int freq_in, + unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8350 *wm8350 = codec->control_data; + struct _fll_div fll_div; + int ret = 0; + u16 fll_1, fll_4; + + /* power down FLL - we need to do this for reconfiguration */ + wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, + WM8350_FLL_ENA | WM8350_FLL_OSC_ENA); + + if (freq_out == 0 || freq_in == 0) + return ret; + + ret = fll_factors(&fll_div, freq_in, freq_out); + if (ret < 0) + return ret; + dev_dbg(wm8350->dev, + "FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d", + freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div, + fll_div.ratio); + + /* set up N.K & dividers */ + fll_1 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_1) & + ~(WM8350_FLL_OUTDIV_MASK | WM8350_FLL_RSP_RATE_MASK | 0xc000); + wm8350_codec_write(codec, WM8350_FLL_CONTROL_1, + fll_1 | (fll_div.div << 8) | 0x50); + wm8350_codec_write(codec, WM8350_FLL_CONTROL_2, + (fll_div.ratio << 11) | (fll_div. + n & WM8350_FLL_N_MASK)); + wm8350_codec_write(codec, WM8350_FLL_CONTROL_3, fll_div.k); + fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) & + ~(WM8350_FLL_FRAC | WM8350_FLL_SLOW_LOCK_REF); + wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, + fll_4 | (fll_div.k ? WM8350_FLL_FRAC : 0) | + (fll_div.ratio == 8 ? WM8350_FLL_SLOW_LOCK_REF : 0)); + + /* power FLL on */ + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_OSC_ENA); + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_ENA); + + return 0; +} + +static int wm8350_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8350 *wm8350 = codec->control_data; + struct wm8350_data *priv = codec->private_data; + struct wm8350_audio_platform_data *platform = + wm8350->codec.platform_data; + u16 pm1; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) & + ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK); + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, + pm1 | WM8350_VMID_50K | + platform->codec_current_on << 14); + break; + + case SND_SOC_BIAS_PREPARE: + pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1); + pm1 &= ~WM8350_VMID_MASK; + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, + pm1 | WM8350_VMID_50K); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret != 0) + return ret; + + /* Enable the system clock */ + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, + WM8350_SYSCLK_ENA); + + /* mute DAC & outputs */ + wm8350_set_bits(wm8350, WM8350_DAC_MUTE, + WM8350_DAC_MUTE_ENA); + + /* discharge cap memory */ + wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, + platform->dis_out1 | + (platform->dis_out2 << 2) | + (platform->dis_out3 << 4) | + (platform->dis_out4 << 6)); + + /* wait for discharge */ + schedule_timeout_interruptible(msecs_to_jiffies + (platform-> + cap_discharge_msecs)); + + /* enable antipop */ + wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, + (platform->vmid_s_curve << 8)); + + /* ramp up vmid */ + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, + (platform-> + codec_current_charge << 14) | + WM8350_VMID_5K | WM8350_VMIDEN | + WM8350_VBUFEN); + + /* wait for vmid */ + schedule_timeout_interruptible(msecs_to_jiffies + (platform-> + vmid_charge_msecs)); + + /* turn on vmid 300k */ + pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) & + ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK); + pm1 |= WM8350_VMID_300K | + (platform->codec_current_standby << 14); + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, + pm1); + + + /* enable analogue bias */ + pm1 |= WM8350_BIASEN; + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1); + + /* disable antipop */ + wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0); + + } else { + /* turn on vmid 300k and reduce current */ + pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) & + ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK); + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, + pm1 | WM8350_VMID_300K | + (platform-> + codec_current_standby << 14)); + + } + break; + + case SND_SOC_BIAS_OFF: + + /* mute DAC & enable outputs */ + wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA); + + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_3, + WM8350_OUT1L_ENA | WM8350_OUT1R_ENA | + WM8350_OUT2L_ENA | WM8350_OUT2R_ENA); + + /* enable anti pop S curve */ + wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, + (platform->vmid_s_curve << 8)); + + /* turn off vmid */ + pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) & + ~WM8350_VMIDEN; + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1); + + /* wait */ + schedule_timeout_interruptible(msecs_to_jiffies + (platform-> + vmid_discharge_msecs)); + + wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, + (platform->vmid_s_curve << 8) | + platform->dis_out1 | + (platform->dis_out2 << 2) | + (platform->dis_out3 << 4) | + (platform->dis_out4 << 6)); + + /* turn off VBuf and drain */ + pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) & + ~(WM8350_VBUFEN | WM8350_VMID_MASK); + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, + pm1 | WM8350_OUTPUT_DRAIN_EN); + + /* wait */ + schedule_timeout_interruptible(msecs_to_jiffies + (platform->drain_msecs)); + + pm1 &= ~WM8350_BIASEN; + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1); + + /* disable anti-pop */ + wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0); + + wm8350_clear_bits(wm8350, WM8350_LOUT1_VOLUME, + WM8350_OUT1L_ENA); + wm8350_clear_bits(wm8350, WM8350_ROUT1_VOLUME, + WM8350_OUT1R_ENA); + wm8350_clear_bits(wm8350, WM8350_LOUT2_VOLUME, + WM8350_OUT2L_ENA); + wm8350_clear_bits(wm8350, WM8350_ROUT2_VOLUME, + WM8350_OUT2R_ENA); + + /* disable clock gen */ + wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, + WM8350_SYSCLK_ENA); + + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), + priv->supplies); + break; + } + codec->bias_level = level; + return 0; +} + +static int wm8350_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8350_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + wm8350_set_bias_level(codec, SND_SOC_BIAS_ON); + + return 0; +} + +static struct snd_soc_codec *wm8350_codec; + +static int wm8350_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct wm8350 *wm8350; + struct wm8350_data *priv; + int ret; + struct wm8350_output *out1; + struct wm8350_output *out2; + + BUG_ON(!wm8350_codec); + + socdev->codec = wm8350_codec; + codec = socdev->codec; + wm8350 = codec->control_data; + priv = codec->private_data; + + /* Enable the codec */ + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); + + /* Enable robust clocking mode in ADC */ + wm8350_codec_write(codec, WM8350_SECURITY, 0xa7); + wm8350_codec_write(codec, 0xde, 0x13); + wm8350_codec_write(codec, WM8350_SECURITY, 0); + + /* read OUT1 & OUT2 volumes */ + out1 = &priv->out1; + out2 = &priv->out2; + out1->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME) & + WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT; + out1->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME) & + WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT; + out2->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME) & + WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT; + out2->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME) & + WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT; + wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME, 0); + wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME, 0); + wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME, 0); + wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME, 0); + + /* Latch VU bits & mute */ + wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME, + WM8350_OUT1_VU | WM8350_OUT1L_MUTE); + wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME, + WM8350_OUT2_VU | WM8350_OUT2L_MUTE); + wm8350_set_bits(wm8350, WM8350_ROUT1_VOLUME, + WM8350_OUT1_VU | WM8350_OUT1R_MUTE); + wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME, + WM8350_OUT2_VU | WM8350_OUT2R_MUTE); + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; + } + + wm8350_add_controls(codec); + wm8350_add_widgets(codec); + + wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return 0; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + return ret; +} + +static int wm8350_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct wm8350 *wm8350 = codec->control_data; + int ret; + + /* cancel any work waiting to be queued. */ + ret = cancel_delayed_work(&codec->delayed_work); + + /* if there was any work waiting then we run it now and + * wait for its completion */ + if (ret) { + schedule_delayed_work(&codec->delayed_work, 0); + flush_scheduled_work(); + } + + wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); + + wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); + + return 0; +} + +#define WM8350_RATES (SNDRV_PCM_RATE_8000_96000) + +#define WM8350_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +struct snd_soc_dai wm8350_dai = { + .name = "WM8350", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8350_RATES, + .formats = WM8350_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8350_RATES, + .formats = WM8350_FORMATS, + }, + .ops = { + .hw_params = wm8350_pcm_hw_params, + .digital_mute = wm8350_mute, + .trigger = wm8350_pcm_trigger, + .set_fmt = wm8350_set_dai_fmt, + .set_sysclk = wm8350_set_dai_sysclk, + .set_pll = wm8350_set_fll, + .set_clkdiv = wm8350_set_clkdiv, + }, +}; +EXPORT_SYMBOL_GPL(wm8350_dai); + +struct snd_soc_codec_device soc_codec_dev_wm8350 = { + .probe = wm8350_probe, + .remove = wm8350_remove, + .suspend = wm8350_suspend, + .resume = wm8350_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8350); + +static int wm8350_codec_probe(struct platform_device *pdev) +{ + struct wm8350 *wm8350 = platform_get_drvdata(pdev); + struct wm8350_data *priv; + struct snd_soc_codec *codec; + int ret, i; + + if (wm8350->codec.platform_data == NULL) { + dev_err(&pdev->dev, "No audio platform data supplied\n"); + return -EINVAL; + } + + priv = kzalloc(sizeof(struct wm8350_data), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = regulator_bulk_get(wm8350->dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret != 0) + goto err_priv; + + codec = &priv->codec; + wm8350->codec.codec = codec; + + wm8350_dai.dev = &pdev->dev; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &pdev->dev; + codec->name = "WM8350"; + codec->owner = THIS_MODULE; + codec->read = wm8350_codec_read; + codec->write = wm8350_codec_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8350_set_bias_level; + codec->dai = &wm8350_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8350_MAX_REGISTER; + codec->private_data = priv; + codec->control_data = wm8350; + + /* Put the codec into reset if it wasn't already */ + wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); + + INIT_DELAYED_WORK(&codec->delayed_work, wm8350_pga_work); + ret = snd_soc_register_codec(codec); + if (ret != 0) + goto err_supply; + + wm8350_codec = codec; + + ret = snd_soc_register_dai(&wm8350_dai); + if (ret != 0) + goto err_codec; + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err_supply: + regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies); +err_priv: + kfree(priv); + wm8350_codec = NULL; + return ret; +} + +static int __devexit wm8350_codec_remove(struct platform_device *pdev) +{ + struct wm8350 *wm8350 = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = wm8350->codec.codec; + struct wm8350_data *priv = codec->private_data; + + snd_soc_unregister_dai(&wm8350_dai); + snd_soc_unregister_codec(codec); + regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies); + kfree(priv); + wm8350_codec = NULL; + return 0; +} + +static struct platform_driver wm8350_codec_driver = { + .driver = { + .name = "wm8350-codec", + .owner = THIS_MODULE, + }, + .probe = wm8350_codec_probe, + .remove = __devexit_p(wm8350_codec_remove), +}; + +static __init int wm8350_init(void) +{ + return platform_driver_register(&wm8350_codec_driver); +} +module_init(wm8350_init); + +static __exit void wm8350_exit(void) +{ + platform_driver_unregister(&wm8350_codec_driver); +} +module_exit(wm8350_exit); + +MODULE_DESCRIPTION("ASoC WM8350 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm8350-codec"); diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h new file mode 100644 index 000000000000..cc2887aa6c38 --- /dev/null +++ b/sound/soc/codecs/wm8350.h @@ -0,0 +1,20 @@ +/* + * wm8350.h - WM8903 audio codec interface + * + * Copyright 2008 Wolfson Microelectronics PLC. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _WM8350_H +#define _WM8350_H + +#include <sound/soc.h> + +extern struct snd_soc_dai wm8350_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8350; + +#endif diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index d8ca2da8d634..40f8238df717 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -463,7 +463,8 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -585,8 +586,6 @@ struct snd_soc_dai wm8510_dai = { .formats = WM8510_FORMATS,}, .ops = { .hw_params = wm8510_pcm_hw_params, - }, - .dai_ops = { .digital_mute = wm8510_mute, .set_fmt = wm8510_set_dai_fmt, .set_clkdiv = wm8510_set_dai_clkdiv, @@ -659,7 +658,7 @@ static int wm8510_init(struct snd_soc_device *socdev) wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm8510_add_controls(codec); wm8510_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8510: failed to register card\n"); goto card_err; @@ -890,6 +889,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8510 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510); +static int __init wm8510_modinit(void) +{ + return snd_soc_register_dai(&wm8510_dai); +} +module_init(wm8510_modinit); + +static void __exit wm8510_exit(void) +{ + snd_soc_unregister_dai(&wm8510_dai); +} +module_exit(wm8510_exit); + MODULE_DESCRIPTION("ASoC WM8510 driver"); MODULE_AUTHOR("Liam Girdwood"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 627ebfb4209b..d004e5845298 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -548,13 +548,13 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, * Set PCM DAI bit size and sample rate. */ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai_link *dai = rtd->dai; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; - u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->codec_dai->id); + u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id); paifb &= ~WM8580_AIF_LENGTH_MASK; /* bit size */ @@ -574,7 +574,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - wm8580_write(codec, WM8580_PAIF3 + dai->codec_dai->id, paifb); + wm8580_write(codec, WM8580_PAIF3 + dai->id, paifb); return 0; } @@ -798,8 +798,6 @@ struct snd_soc_dai wm8580_dai[] = { }, .ops = { .hw_params = wm8580_paif_hw_params, - }, - .dai_ops = { .set_fmt = wm8580_set_paif_dai_fmt, .set_clkdiv = wm8580_set_dai_clkdiv, .set_pll = wm8580_set_dai_pll, @@ -818,8 +816,6 @@ struct snd_soc_dai wm8580_dai[] = { }, .ops = { .hw_params = wm8580_paif_hw_params, - }, - .dai_ops = { .set_fmt = wm8580_set_paif_dai_fmt, .set_clkdiv = wm8580_set_dai_clkdiv, .set_pll = wm8580_set_dai_pll, @@ -873,7 +869,7 @@ static int wm8580_init(struct snd_soc_device *socdev) wm8580_add_controls(codec); wm8580_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8580: failed to register card\n"); goto card_err; @@ -900,85 +896,85 @@ static struct snd_soc_device *wm8580_socdev; * low = 0x1a * high = 0x1b */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; - -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; -static struct i2c_driver wm8580_i2c_driver; -static struct i2c_client client_template; - -static int wm8580_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8580_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8580_socdev; - struct wm8580_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); - return -ENOMEM; - } i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - dev_err(&i2c->dev, "failed to attach codec at addr %x\n", addr); - goto err; - } - ret = wm8580_init(socdev); - if (ret < 0) { + if (ret < 0) dev_err(&i2c->dev, "failed to initialise WM8580\n"); - goto err; - } - - return ret; - -err: - kfree(codec); - kfree(i2c); return ret; } -static int wm8580_i2c_detach(struct i2c_client *client) +static int wm8580_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8580_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8580_codec_probe); -} +static const struct i2c_device_id wm8580_i2c_id[] = { + { "wm8580", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver wm8580_i2c_driver = { .driver = { .name = "WM8580 I2C Codec", .owner = THIS_MODULE, }, - .attach_adapter = wm8580_i2c_attach, - .detach_client = wm8580_i2c_detach, - .command = NULL, + .probe = wm8580_i2c_probe, + .remove = wm8580_i2c_remove, + .id_table = wm8580_i2c_id, }; -static struct i2c_client client_template = { - .name = "WM8580", - .driver = &wm8580_i2c_driver, -}; +static int wm8580_add_i2c_device(struct platform_device *pdev, + const struct wm8580_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8580_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8580", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8580_i2c_driver); + return -ENODEV; +} #endif static int wm8580_probe(struct platform_device *pdev) @@ -1011,11 +1007,8 @@ static int wm8580_probe(struct platform_device *pdev) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8580_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = wm8580_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -1034,6 +1027,7 @@ static int wm8580_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8580_i2c_driver); #endif kfree(codec->private_data); @@ -1048,6 +1042,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8580 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580); +static int __init wm8580_modinit(void) +{ + return snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); +} +module_init(wm8580_modinit); + +static void __exit wm8580_exit(void) +{ + snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); +} +module_exit(wm8580_exit); + MODULE_DESCRIPTION("ASoC WM8580 driver"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h index 589ddaba21d7..09e4422f6f2f 100644 --- a/sound/soc/codecs/wm8580.h +++ b/sound/soc/codecs/wm8580.h @@ -29,6 +29,7 @@ #define WM8580_CLKSRC_NONE 5 struct wm8580_setup_data { + int i2c_bus; unsigned short i2c_address; }; diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c new file mode 100644 index 000000000000..80b11983e137 --- /dev/null +++ b/sound/soc/codecs/wm8728.c @@ -0,0 +1,585 @@ +/* + * wm8728.c -- WM8728 ALSA SoC Audio driver + * + * Copyright 2008 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8728.h" + +struct snd_soc_codec_device soc_codec_dev_wm8728; + +/* + * We can't read the WM8728 register space so we cache them instead. + * Note that the defaults here aren't the physical defaults, we latch + * the volume update bits, mute the output and enable infinite zero + * detect. + */ +static const u16 wm8728_reg_defaults[] = { + 0x1ff, + 0x1ff, + 0x001, + 0x100, +}; + +static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults)); + return cache[reg]; +} + +static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults)); + cache[reg] = value; +} + +/* + * write to the WM8728 register space + */ +static int wm8728_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8728 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8728_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1); + +static const struct snd_kcontrol_new wm8728_snd_controls[] = { + +SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL, + 0, 255, 0, wm8728_tlv), + +SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0), +}; + +static int wm8728_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8728_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* + * DAPM controls. + */ +static const struct snd_soc_dapm_widget wm8728_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"VOUTL", NULL, "DAC"}, + {"VOUTR", NULL, "DAC"}, +}; + +static int wm8728_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8728_dapm_widgets, + ARRAY_SIZE(wm8728_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int wm8728_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); + + if (mute) + wm8728_write(codec, WM8728_DACCTL, mute_reg | 1); + else + wm8728_write(codec, WM8728_DACCTL, mute_reg & ~1); + + return 0; +} + +static int wm8728_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL); + + dac &= ~0x18; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + dac |= 0x10; + break; + case SNDRV_PCM_FORMAT_S24_LE: + dac |= 0x08; + break; + default: + return -EINVAL; + } + + wm8728_write(codec, WM8728_DACCTL, dac); + + return 0; +} + +static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = wm8728_read_reg_cache(codec, WM8728_IFCTL); + + /* Currently only I2S is supported by the driver, though the + * hardware is more flexible. + */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 1; + break; + default: + return -EINVAL; + } + + /* The hardware only support full slave mode */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + iface &= ~0x22; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x20; + iface &= ~0x02; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x02; + iface &= ~0x20; + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x22; + break; + default: + return -EINVAL; + } + + wm8728_write(codec, WM8728_IFCTL, iface); + return 0; +} + +static int wm8728_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg; + int i; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Power everything up... */ + reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); + wm8728_write(codec, WM8728_DACCTL, reg & ~0x4); + + /* ..then sync in the register cache. */ + for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++) + wm8728_write(codec, i, + wm8728_read_reg_cache(codec, i)); + } + break; + + case SND_SOC_BIAS_OFF: + reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); + wm8728_write(codec, WM8728_DACCTL, reg | 0x4); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8728_RATES (SNDRV_PCM_RATE_8000_192000) + +#define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +struct snd_soc_dai wm8728_dai = { + .name = "WM8728", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8728_RATES, + .formats = WM8728_FORMATS, + }, + .ops = { + .hw_params = wm8728_hw_params, + .digital_mute = wm8728_mute, + .set_fmt = wm8728_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(wm8728_dai); + +static int wm8728_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8728_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8728_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +/* + * initialise the WM8728 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8728_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "WM8728"; + codec->owner = THIS_MODULE; + codec->read = wm8728_read_reg_cache; + codec->write = wm8728_write; + codec->set_bias_level = wm8728_set_bias_level; + codec->dai = &wm8728_dai; + codec->num_dai = 1; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->reg_cache_size = ARRAY_SIZE(wm8728_reg_defaults); + codec->reg_cache = kmemdup(wm8728_reg_defaults, + sizeof(wm8728_reg_defaults), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8728: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8728_add_controls(codec); + wm8728_add_widgets(codec); + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8728: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *wm8728_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM8728 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ + +static int wm8728_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct snd_soc_device *socdev = wm8728_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = wm8728_init(socdev); + if (ret < 0) + pr_err("failed to initialise WM8728\n"); + + return ret; +} + +static int wm8728_i2c_remove(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + kfree(codec->reg_cache); + return 0; +} + +static const struct i2c_device_id wm8728_i2c_id[] = { + { "wm8728", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id); + +static struct i2c_driver wm8728_i2c_driver = { + .driver = { + .name = "WM8728 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8728_i2c_probe, + .remove = wm8728_i2c_remove, + .id_table = wm8728_i2c_id, +}; + +static int wm8728_add_i2c_device(struct platform_device *pdev, + const struct wm8728_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8728_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8728", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8728_i2c_driver); + return -ENODEV; +} +#endif + +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8728_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8728_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = spi; + + ret = wm8728_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8728\n"); + + return ret; +} + +static int __devexit wm8728_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8728_spi_driver = { + .driver = { + .name = "wm8728", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8728_spi_probe, + .remove = __devexit_p(wm8728_spi_remove), +}; + +static int wm8728_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif /* CONFIG_SPI_MASTER */ + +static int wm8728_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8728_setup_data *setup; + struct snd_soc_codec *codec; + int ret = 0; + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + wm8728_socdev = socdev; + ret = -ENODEV; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + codec->hw_write = (hw_write_t)i2c_master_send; + ret = wm8728_add_i2c_device(pdev, setup); + } +#endif +#if defined(CONFIG_SPI_MASTER) + if (setup->spi) { + codec->hw_write = (hw_write_t)wm8728_spi_write; + ret = spi_register_driver(&wm8728_spi_driver); + if (ret != 0) + printk(KERN_ERR "can't add spi driver"); + } +#endif + + if (ret != 0) + kfree(codec); + + return ret; +} + +/* power down chip */ +static int wm8728_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); + i2c_del_driver(&wm8728_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8728_spi_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8728 = { + .probe = wm8728_probe, + .remove = wm8728_remove, + .suspend = wm8728_suspend, + .resume = wm8728_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8728); + +static int __init wm8728_modinit(void) +{ + return snd_soc_register_dai(&wm8728_dai); +} +module_init(wm8728_modinit); + +static void __exit wm8728_exit(void) +{ + snd_soc_unregister_dai(&wm8728_dai); +} +module_exit(wm8728_exit); + +MODULE_DESCRIPTION("ASoC WM8728 driver"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8728.h b/sound/soc/codecs/wm8728.h new file mode 100644 index 000000000000..d269c132474b --- /dev/null +++ b/sound/soc/codecs/wm8728.h @@ -0,0 +1,30 @@ +/* + * wm8728.h -- WM8728 ASoC codec driver + * + * Copyright 2008 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8728_H +#define _WM8728_H + +#define WM8728_DACLVOL 0x00 +#define WM8728_DACRVOL 0x01 +#define WM8728_DACCTL 0x02 +#define WM8728_IFCTL 0x03 + +struct wm8728_setup_data { + int spi; + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai wm8728_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8728; + +#endif diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7f8a7e36b33e..c444b9f2701e 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -264,7 +264,8 @@ static inline int get_coeff(int mclk, int rate) } static int wm8731_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -293,7 +294,8 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8731_pcm_prepare(struct snd_pcm_substream *substream) +static int wm8731_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -305,7 +307,8 @@ static int wm8731_pcm_prepare(struct snd_pcm_substream *substream) return 0; } -static void wm8731_shutdown(struct snd_pcm_substream *substream) +static void wm8731_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -461,8 +464,6 @@ struct snd_soc_dai wm8731_dai = { .prepare = wm8731_pcm_prepare, .hw_params = wm8731_hw_params, .shutdown = wm8731_shutdown, - }, - .dai_ops = { .digital_mute = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, .set_fmt = wm8731_set_dai_fmt, @@ -544,7 +545,7 @@ static int wm8731_init(struct snd_soc_device *socdev) wm8731_add_controls(codec); wm8731_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8731: failed to register card\n"); goto card_err; @@ -792,6 +793,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); +static int __init wm8731_modinit(void) +{ + return snd_soc_register_dai(&wm8731_dai); +} +module_init(wm8731_modinit); + +static void __exit wm8731_exit(void) +{ + snd_soc_unregister_dai(&wm8731_dai); +} +module_exit(wm8731_exit); + MODULE_DESCRIPTION("ASoC WM8731 driver"); MODULE_AUTHOR("Richard Purdie"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 9b7296ee5b08..5997fa68e0d5 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -614,7 +614,8 @@ static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -709,8 +710,6 @@ struct snd_soc_dai wm8750_dai = { .formats = WM8750_FORMATS,}, .ops = { .hw_params = wm8750_pcm_hw_params, - }, - .dai_ops = { .digital_mute = wm8750_mute, .set_fmt = wm8750_set_dai_fmt, .set_sysclk = wm8750_set_dai_sysclk, @@ -819,7 +818,7 @@ static int wm8750_init(struct snd_soc_device *socdev) wm8750_add_controls(codec); wm8750_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8750: failed to register card\n"); goto card_err; @@ -1086,6 +1085,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8750 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); +static int __init wm8750_modinit(void) +{ + return snd_soc_register_dai(&wm8750_dai); +} +module_init(wm8750_modinit); + +static void __exit wm8750_exit(void) +{ + snd_soc_unregister_dai(&wm8750_dai); +} +module_exit(wm8750_exit); + MODULE_DESCRIPTION("ASoC WM8750 driver"); MODULE_AUTHOR("Liam Girdwood"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d426eaa22185..6c21b50c9375 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -922,7 +922,8 @@ static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai, * Set PCM DAI bit size and sample rate. */ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1155,7 +1156,8 @@ static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, * Set PCM DAI bit size and sample rate. */ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1323,16 +1325,15 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, + .formats = WM8753_FORMATS}, .capture = { /* dummy for fast DAI switching */ .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, + .formats = WM8753_FORMATS}, .ops = { - .hw_params = wm8753_i2s_hw_params,}, - .dai_ops = { + .hw_params = wm8753_i2s_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode1h_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, @@ -1356,8 +1357,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, .ops = { - .hw_params = wm8753_pcm_hw_params,}, - .dai_ops = { + .hw_params = wm8753_pcm_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode1v_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, @@ -1385,8 +1385,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, .ops = { - .hw_params = wm8753_pcm_hw_params,}, - .dai_ops = { + .hw_params = wm8753_pcm_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode2_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, @@ -1410,8 +1409,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, .ops = { - .hw_params = wm8753_i2s_hw_params,}, - .dai_ops = { + .hw_params = wm8753_i2s_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode3_4_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, @@ -1439,8 +1437,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, .ops = { - .hw_params = wm8753_i2s_hw_params,}, - .dai_ops = { + .hw_params = wm8753_i2s_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode3_4_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, @@ -1608,7 +1605,7 @@ static int wm8753_init(struct snd_soc_device *socdev) wm8753_add_controls(codec); wm8753_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8753: failed to register card\n"); goto card_err; @@ -1877,6 +1874,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8753 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753); +static int __init wm8753_modinit(void) +{ + return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai)); +} +module_init(wm8753_modinit); + +static void __exit wm8753_exit(void) +{ + snd_soc_unregister_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai)); +} +module_exit(wm8753_exit); + MODULE_DESCRIPTION("ASoC WM8753 driver"); MODULE_AUTHOR("Liam Girdwood"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 3b326c9b5586..6767de10ded0 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -138,6 +138,10 @@ struct snd_soc_codec_device soc_codec_dev_wm8900; struct wm8900_priv { + struct snd_soc_codec codec; + + u16 reg_cache[WM8900_MAXREG]; + u32 fll_in; /* FLL input frequency */ u32 fll_out; /* FLL output frequency */ }; @@ -727,7 +731,8 @@ static int wm8900_add_widgets(struct snd_soc_codec *codec) } static int wm8900_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1117,8 +1122,6 @@ struct snd_soc_dai wm8900_dai = { }, .ops = { .hw_params = wm8900_hw_params, - }, - .dai_ops = { .set_clkdiv = wm8900_set_dai_clkdiv, .set_pll = wm8900_set_dai_pll, .set_fmt = wm8900_set_dai_fmt, @@ -1283,16 +1286,28 @@ static int wm8900_resume(struct platform_device *pdev) return 0; } -/* - * initialise the WM8900 driver - * register the mixer and dsp interfaces with the kernel - */ -static int wm8900_init(struct snd_soc_device *socdev) +static struct snd_soc_codec *wm8900_codec; + +static int wm8900_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { - struct snd_soc_codec *codec = socdev->codec; - int ret = 0; + struct wm8900_priv *wm8900; + struct snd_soc_codec *codec; unsigned int reg; - struct i2c_client *i2c_client = socdev->codec->control_data; + int ret; + + wm8900 = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL); + if (wm8900 == NULL) + return -ENOMEM; + + codec = &wm8900->codec; + codec->private_data = wm8900; + codec->reg_cache = &wm8900->reg_cache[0]; + codec->reg_cache_size = WM8900_MAXREG; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); codec->name = "WM8900"; codec->owner = THIS_MODULE; @@ -1300,33 +1315,28 @@ static int wm8900_init(struct snd_soc_device *socdev) codec->write = wm8900_write; codec->dai = &wm8900_dai; codec->num_dai = 1; - codec->reg_cache_size = WM8900_MAXREG; - codec->reg_cache = kmemdup(wm8900_reg_defaults, - sizeof(wm8900_reg_defaults), GFP_KERNEL); - - if (codec->reg_cache == NULL) - return -ENOMEM; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->control_data = i2c; + codec->set_bias_level = wm8900_set_bias_level; + codec->dev = &i2c->dev; reg = wm8900_read(codec, WM8900_REG_ID); if (reg != 0x8900) { - dev_err(&i2c_client->dev, "Device is not a WM8900 - ID %x\n", - reg); - return -ENODEV; - } - - codec->private_data = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL); - if (codec->private_data == NULL) { - ret = -ENOMEM; - goto priv_err; + dev_err(&i2c->dev, "Device is not a WM8900 - ID %x\n", reg); + ret = -ENODEV; + goto err; } /* Read back from the chip */ reg = wm8900_chip_read(codec, WM8900_REG_POWER1); reg = (reg >> 12) & 0xf; - dev_info(&i2c_client->dev, "WM8900 revision %d\n", reg); + dev_info(&i2c->dev, "WM8900 revision %d\n", reg); wm8900_reset(codec); + /* Turn the chip on */ + wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Latch the volume update bits */ wm8900_write(codec, WM8900_REG_LINVOL, wm8900_read(codec, WM8900_REG_LINVOL) | 0x100); @@ -1352,160 +1362,98 @@ static int wm8900_init(struct snd_soc_device *socdev) /* Set the DAC and mixer output bias */ wm8900_write(codec, WM8900_REG_OUTBIASCTL, 0x81); - /* Register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - dev_err(&i2c_client->dev, "Failed to register new PCMs\n"); - goto pcm_err; - } - - /* Turn the chip on */ - codec->bias_level = SND_SOC_BIAS_OFF; - wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - wm8900_add_controls(codec); - wm8900_add_widgets(codec); - - ret = snd_soc_register_card(socdev); - if (ret < 0) { - dev_err(&i2c_client->dev, "Failed to register card\n"); - goto card_err; - } - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); -pcm_err: - kfree(codec->reg_cache); -priv_err: - kfree(codec->private_data); - return ret; -} + wm8900_dai.dev = &i2c->dev; -static struct snd_soc_device *wm8900_socdev; + wm8900_codec = codec; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; - -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver wm8900_i2c_driver; -static struct i2c_client client_template; - -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ -static int wm8900_codec_probe(struct i2c_adapter *adap, int addr, int kind) -{ - struct snd_soc_device *socdev = wm8900_socdev; - struct wm8900_setup_data *setup = socdev->codec_data; - struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; - int ret; - - if (addr != setup->i2c_address) - return -ENODEV; - - dev_err(&adap->dev, "Probe on %x\n", addr); - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); - return -ENOMEM; - } - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; - - ret = i2c_attach_client(i2c); - if (ret < 0) { - dev_err(&adap->dev, - "failed to attach codec at addr %x\n", addr); + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); goto err; } - ret = wm8900_init(socdev); - if (ret < 0) { - dev_err(&adap->dev, "failed to initialise WM8900\n"); - goto err; + ret = snd_soc_register_dai(&wm8900_dai); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; } + return ret; +err_codec: + snd_soc_unregister_codec(codec); err: - kfree(codec); - kfree(i2c); + kfree(wm8900); + wm8900_codec = NULL; return ret; } -static int wm8900_i2c_detach(struct i2c_client *client) +static int wm8900_i2c_remove(struct i2c_client *client) { - struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); - kfree(codec->reg_cache); - kfree(client); + snd_soc_unregister_dai(&wm8900_dai); + snd_soc_unregister_codec(wm8900_codec); + + wm8900_set_bias_level(wm8900_codec, SND_SOC_BIAS_OFF); + + wm8900_dai.dev = NULL; + kfree(wm8900_codec->private_data); + wm8900_codec = NULL; + return 0; } -static int wm8900_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8900_codec_probe); -} +static const struct i2c_device_id wm8900_i2c_id[] = { + { "wm8900", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver wm8900_i2c_driver = { .driver = { - .name = "WM8900 I2C codec", + .name = "WM8900", .owner = THIS_MODULE, }, - .attach_adapter = wm8900_i2c_attach, - .detach_client = wm8900_i2c_detach, - .command = NULL, -}; - -static struct i2c_client client_template = { - .name = "WM8900", - .driver = &wm8900_i2c_driver, + .probe = wm8900_i2c_probe, + .remove = wm8900_i2c_remove, + .id_table = wm8900_i2c_id, }; -#endif static int wm8900_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8900_setup_data *setup; struct snd_soc_codec *codec; int ret = 0; - dev_info(&pdev->dev, "WM8900 Audio Codec\n"); - - setup = socdev->codec_data; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); + if (!wm8900_codec) { + dev_err(&pdev->dev, "I2C client not yet instantiated\n"); + return -ENODEV; + } + codec = wm8900_codec; socdev->codec = codec; - codec->set_bias_level = wm8900_set_bias_level; + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register new PCMs\n"); + goto pcm_err; + } - wm8900_socdev = socdev; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; - codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8900_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + wm8900_add_controls(codec); + wm8900_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register card\n"); + goto card_err; } -#else -#error Non-I2C interfaces not yet supported -#endif + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: return ret; } @@ -1513,17 +1461,9 @@ static int wm8900_probe(struct platform_device *pdev) static int wm8900_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; - - if (codec->control_data) - wm8900_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&wm8900_i2c_driver); -#endif - kfree(codec); return 0; } @@ -1536,6 +1476,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8900 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8900); +static int __init wm8900_modinit(void) +{ + return i2c_add_driver(&wm8900_i2c_driver); +} +module_init(wm8900_modinit); + +static void __exit wm8900_exit(void) +{ + i2c_del_driver(&wm8900_i2c_driver); +} +module_exit(wm8900_exit); + MODULE_DESCRIPTION("ASoC WM8900 driver"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8900.h b/sound/soc/codecs/wm8900.h index ba450d99e902..fd15007d10c7 100644 --- a/sound/soc/codecs/wm8900.h +++ b/sound/soc/codecs/wm8900.h @@ -52,12 +52,6 @@ #define WM8900_DAC_CLKDIV_5_5 0x14 #define WM8900_DAC_CLKDIV_6 0x18 -#define WM8900_ - -struct wm8900_setup_data { - unsigned short i2c_address; -}; - extern struct snd_soc_dai wm8900_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8900; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index ce40d7877605..bde74546db4a 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -33,19 +33,6 @@ #include "wm8903.h" -struct wm8903_priv { - int sysclk; - - /* Reference counts */ - int charge_pump_users; - int class_w_users; - int playback_active; - int capture_active; - - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; -}; - /* Register defaults at reset */ static u16 wm8903_reg_defaults[] = { 0x8903, /* R0 - SW Reset and ID */ @@ -223,6 +210,23 @@ static u16 wm8903_reg_defaults[] = { 0x0000, /* R172 - Analogue Output Bias 0 */ }; +struct wm8903_priv { + struct snd_soc_codec codec; + u16 reg_cache[ARRAY_SIZE(wm8903_reg_defaults)]; + + int sysclk; + + /* Reference counts */ + int charge_pump_users; + int class_w_users; + int playback_active; + int capture_active; + + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; +}; + + static unsigned int wm8903_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { @@ -360,6 +364,8 @@ static void wm8903_sync_reg_cache(struct snd_soc_codec *codec, u16 *cache) static void wm8903_reset(struct snd_soc_codec *codec) { wm8903_write(codec, WM8903_SW_RESET_AND_ID, 0); + memcpy(codec->reg_cache, wm8903_reg_defaults, + sizeof(wm8903_reg_defaults)); } #define WM8903_OUTPUT_SHORT 0x8 @@ -392,6 +398,7 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, break; default: BUG(); + return -EINVAL; /* Spurious warning from some compilers */ } switch (w->shift) { @@ -403,6 +410,7 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, break; default: BUG(); + return -EINVAL; /* Spurious warning from some compilers */ } if (event & SND_SOC_DAPM_PRE_PMU) { @@ -773,14 +781,14 @@ static const struct snd_kcontrol_new left_output_mixer[] = { SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0), SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0), SOC_DAPM_SINGLE_W("Left Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 1, 1, 0), -SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 1, 1, 0), +SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 0, 1, 0), }; static const struct snd_kcontrol_new right_output_mixer[] = { SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 3, 1, 0), SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 2, 1, 0), SOC_DAPM_SINGLE_W("Left Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 1, 1, 0), -SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 1, 1, 0), +SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 0, 1, 0), }; static const struct snd_kcontrol_new left_speaker_mixer[] = { @@ -788,7 +796,7 @@ SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 3, 1, 0), SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 2, 1, 0), SOC_DAPM_SINGLE("Left Bypass Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 1, 1, 0), SOC_DAPM_SINGLE("Right Bypass Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, - 1, 1, 0), + 0, 1, 0), }; static const struct snd_kcontrol_new right_speaker_mixer[] = { @@ -797,7 +805,7 @@ SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0, 2, 1, 0), SOC_DAPM_SINGLE("Left Bypass Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0, 1, 1, 0), SOC_DAPM_SINGLE("Right Bypass Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0, - 1, 1, 0), + 0, 1, 0), }; static const struct snd_soc_dapm_widget wm8903_dapm_widgets[] = { @@ -989,6 +997,9 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { + wm8903_write(codec, WM8903_CLOCK_RATES_2, + WM8903_CLK_SYS_ENA); + wm8903_run_sequence(codec, 0); wm8903_sync_reg_cache(codec, codec->reg_cache); @@ -1019,6 +1030,9 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: wm8903_run_sequence(codec, 32); + reg = wm8903_read(codec, WM8903_CLOCK_RATES_2); + reg &= ~WM8903_CLK_SYS_ENA; + wm8903_write(codec, WM8903_CLOCK_RATES_2, reg); break; } @@ -1257,7 +1271,8 @@ static struct { { 0, 0 }, }; -static int wm8903_startup(struct snd_pcm_substream *substream) +static int wm8903_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1298,7 +1313,8 @@ static int wm8903_startup(struct snd_pcm_substream *substream) return 0; } -static void wm8903_shutdown(struct snd_pcm_substream *substream) +static void wm8903_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1317,7 +1333,8 @@ static void wm8903_shutdown(struct snd_pcm_substream *substream) } static int wm8903_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1515,8 +1532,6 @@ struct snd_soc_dai wm8903_dai = { .startup = wm8903_startup, .shutdown = wm8903_shutdown, .hw_params = wm8903_hw_params, - }, - .dai_ops = { .digital_mute = wm8903_digital_mute, .set_fmt = wm8903_set_dai_fmt, .set_sysclk = wm8903_set_dai_sysclk @@ -1560,39 +1575,48 @@ static int wm8903_resume(struct platform_device *pdev) return 0; } -/* - * initialise the WM8903 driver - * register the mixer and dsp interfaces with the kernel - */ -static int wm8903_init(struct snd_soc_device *socdev) +static struct snd_soc_codec *wm8903_codec; + +static int wm8903_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { - struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c = codec->control_data; - int ret = 0; + struct wm8903_priv *wm8903; + struct snd_soc_codec *codec; + int ret; u16 val; - val = wm8903_hw_read(codec, WM8903_SW_RESET_AND_ID); - if (val != wm8903_reg_defaults[WM8903_SW_RESET_AND_ID]) { - dev_err(&i2c->dev, - "Device with ID register %x is not a WM8903\n", val); - return -ENODEV; - } + wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL); + if (wm8903 == NULL) + return -ENOMEM; + + codec = &wm8903->codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->dev = &i2c->dev; codec->name = "WM8903"; codec->owner = THIS_MODULE; codec->read = wm8903_read; codec->write = wm8903_write; + codec->hw_write = (hw_write_t)i2c_master_send; codec->bias_level = SND_SOC_BIAS_OFF; codec->set_bias_level = wm8903_set_bias_level; codec->dai = &wm8903_dai; codec->num_dai = 1; - codec->reg_cache_size = ARRAY_SIZE(wm8903_reg_defaults); - codec->reg_cache = kmemdup(wm8903_reg_defaults, - sizeof(wm8903_reg_defaults), - GFP_KERNEL); - if (codec->reg_cache == NULL) { - dev_err(&i2c->dev, "Failed to allocate register cache\n"); - return -ENOMEM; + codec->reg_cache_size = ARRAY_SIZE(wm8903->reg_cache); + codec->reg_cache = &wm8903->reg_cache[0]; + codec->private_data = wm8903; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + val = wm8903_hw_read(codec, WM8903_SW_RESET_AND_ID); + if (val != wm8903_reg_defaults[WM8903_SW_RESET_AND_ID]) { + dev_err(&i2c->dev, + "Device with ID register %x is not a WM8903\n", val); + return -ENODEV; } val = wm8903_read(codec, WM8903_REVISION_NUMBER); @@ -1601,16 +1625,6 @@ static int wm8903_init(struct snd_soc_device *socdev) wm8903_reset(codec); - /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - dev_err(&i2c->dev, "failed to create pcms\n"); - goto pcm_err; - } - - /* SYSCLK is required for pretty much anything */ - wm8903_write(codec, WM8903_CLOCK_RATES_2, WM8903_CLK_SYS_ENA); - /* power on device */ wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1645,47 +1659,45 @@ static int wm8903_init(struct snd_soc_device *socdev) val |= WM8903_DAC_MUTEMODE; wm8903_write(codec, WM8903_DAC_DIGITAL_1, val); - wm8903_add_controls(codec); - wm8903_add_widgets(codec); - ret = snd_soc_register_card(socdev); - if (ret < 0) { - dev_err(&i2c->dev, "wm8903: failed to register card\n"); - goto card_err; + wm8903_dai.dev = &i2c->dev; + wm8903_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&wm8903_dai); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); -pcm_err: - kfree(codec->reg_cache); +err_codec: + snd_soc_unregister_codec(codec); +err: + wm8903_codec = NULL; + kfree(wm8903); return ret; } -static struct snd_soc_device *wm8903_socdev; - -static int wm8903_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static int wm8903_i2c_remove(struct i2c_client *client) { - struct snd_soc_device *socdev = wm8903_socdev; - struct snd_soc_codec *codec = socdev->codec; - int ret; + struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; + snd_soc_unregister_dai(&wm8903_dai); + snd_soc_unregister_codec(codec); - ret = wm8903_init(socdev); - if (ret < 0) - dev_err(&i2c->dev, "Device initialisation failed\n"); + wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); - return ret; -} + kfree(codec->private_data); + + wm8903_codec = NULL; + wm8903_dai.dev = NULL; -static int wm8903_i2c_remove(struct i2c_client *client) -{ - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); return 0; } @@ -1709,75 +1721,37 @@ static struct i2c_driver wm8903_i2c_driver = { static int wm8903_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8903_setup_data *setup; - struct snd_soc_codec *codec; - struct wm8903_priv *wm8903; - struct i2c_board_info board_info; - struct i2c_adapter *adapter; - struct i2c_client *i2c_client; int ret = 0; - setup = socdev->codec_data; - - if (!setup->i2c_address) { - dev_err(&pdev->dev, "No codec address provided\n"); - return -ENODEV; + if (!wm8903_codec) { + dev_err(&pdev->dev, "I2C device not yet probed\n"); + goto err; } - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; + socdev->codec = wm8903_codec; - wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL); - if (wm8903 == NULL) { - ret = -ENOMEM; - goto err_codec; + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + goto err; } - codec->private_data = wm8903; - socdev->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - - wm8903_socdev = socdev; + wm8903_add_controls(socdev->codec); + wm8903_add_widgets(socdev->codec); - codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8903_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - goto err_priv; - } else { - memset(&board_info, 0, sizeof(board_info)); - strlcpy(board_info.type, "wm8903", I2C_NAME_SIZE); - board_info.addr = setup->i2c_address; - - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "Can't get I2C bus %d\n", - setup->i2c_bus); - ret = -ENODEV; - goto err_adapter; - } - - i2c_client = i2c_new_device(adapter, &board_info); - i2c_put_adapter(adapter); - if (i2c_client == NULL) { - dev_err(&pdev->dev, - "I2C driver registration failed\n"); - ret = -ENODEV; - goto err_adapter; - } + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "wm8903: failed to register card\n"); + goto card_err; } return ret; -err_adapter: - i2c_del_driver(&wm8903_i2c_driver); -err_priv: - kfree(codec->private_data); -err_codec: - kfree(codec); +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +err: return ret; } @@ -1792,10 +1766,6 @@ static int wm8903_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); - i2c_unregister_device(socdev->codec->control_data); - i2c_del_driver(&wm8903_i2c_driver); - kfree(codec->private_data); - kfree(codec); return 0; } @@ -1808,6 +1778,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8903 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8903); +static int __init wm8903_modinit(void) +{ + return i2c_add_driver(&wm8903_i2c_driver); +} +module_init(wm8903_modinit); + +static void __exit wm8903_exit(void) +{ + i2c_del_driver(&wm8903_i2c_driver); +} +module_exit(wm8903_exit); + MODULE_DESCRIPTION("ASoC WM8903 driver"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.cm>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h index cec622f2f660..0ea27e2b9963 100644 --- a/sound/soc/codecs/wm8903.h +++ b/sound/soc/codecs/wm8903.h @@ -18,11 +18,6 @@ extern struct snd_soc_dai wm8903_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8903; -struct wm8903_setup_data { - int i2c_bus; - int i2c_address; -}; - #define WM8903_MCLK_DIV_2 1 #define WM8903_CLK_SYS 2 #define WM8903_BCLK 3 diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index f41a578ddd4f..88ead7f8dd98 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -541,7 +541,8 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -634,8 +635,6 @@ struct snd_soc_dai wm8971_dai = { .formats = WM8971_FORMATS,}, .ops = { .hw_params = wm8971_pcm_hw_params, - }, - .dai_ops = { .digital_mute = wm8971_mute, .set_fmt = wm8971_set_dai_fmt, .set_sysclk = wm8971_set_dai_sysclk, @@ -748,7 +747,7 @@ static int wm8971_init(struct snd_soc_device *socdev) wm8971_add_controls(codec); wm8971_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8971: failed to register card\n"); goto card_err; @@ -936,6 +935,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8971 = { EXPORT_SYMBOL_GPL(soc_codec_dev_wm8971); +static int __init wm8971_modinit(void) +{ + return snd_soc_register_dai(&wm8971_dai); +} +module_init(wm8971_modinit); + +static void __exit wm8971_exit(void) +{ + snd_soc_unregister_dai(&wm8971_dai); +} +module_exit(wm8971_exit); + MODULE_DESCRIPTION("ASoC WM8971 driver"); MODULE_AUTHOR("Lab126"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 572d22b0880b..5b5afc144478 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -106,6 +106,7 @@ static const u16 wm8990_reg[] = { 0x0008, /* R60 - PLL1 */ 0x0031, /* R61 - PLL2 */ 0x0026, /* R62 - PLL3 */ + 0x0000, /* R63 - Driver internal */ }; /* @@ -126,10 +127,9 @@ static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u16 *cache = codec->reg_cache; - BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1); - /* Reset register is uncached */ - if (reg == 0) + /* Reset register and reserved registers are uncached */ + if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1) return; cache[reg] = value; @@ -1172,7 +1172,8 @@ static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai, * Set PCM DAI bit size and sample rate. */ static int wm8990_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1222,8 +1223,14 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: break; + case SND_SOC_BIAS_PREPARE: + /* VMID=2*50k */ + val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) & + ~WM8990_VMID_MODE_MASK; + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2); break; + case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Enable all output discharge bits */ @@ -1272,10 +1279,17 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN); - } else { - /* ON -> standby */ + /* Enable workaround for ADC clocking issue. */ + wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0x2); + wm8990_write(codec, WM8990_EXT_CTL1, 0xa003); + wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0); } + + /* VMID=2*250k */ + val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) & + ~WM8990_VMID_MODE_MASK; + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4); break; case SND_SOC_BIAS_OFF: @@ -1349,8 +1363,7 @@ struct snd_soc_dai wm8990_dai = { .rates = WM8990_RATES, .formats = WM8990_FORMATS,}, .ops = { - .hw_params = wm8990_hw_params,}, - .dai_ops = { + .hw_params = wm8990_hw_params, .digital_mute = wm8990_mute, .set_fmt = wm8990_set_dai_fmt, .set_clkdiv = wm8990_set_dai_clkdiv, @@ -1449,7 +1462,7 @@ static int wm8990_init(struct snd_soc_device *socdev) wm8990_add_controls(codec); wm8990_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8990: failed to register card\n"); goto card_err; @@ -1630,6 +1643,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8990 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990); +static int __init wm8990_modinit(void) +{ + return snd_soc_register_dai(&wm8990_dai); +} +module_init(wm8990_modinit); + +static void __exit wm8990_exit(void) +{ + snd_soc_unregister_dai(&wm8990_dai); +} +module_exit(wm8990_exit); + MODULE_DESCRIPTION("ASoC WM8990 driver"); MODULE_AUTHOR("Liam Girdwood"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h index 0e192f3b0788..7114ddc88b4b 100644 --- a/sound/soc/codecs/wm8990.h +++ b/sound/soc/codecs/wm8990.h @@ -80,8 +80,8 @@ #define WM8990_PLL3 0x3E #define WM8990_INTDRIVBITS 0x3F -#define WM8990_REGISTER_COUNT 60 -#define WM8990_MAX_REGISTER 0x3F +#define WM8990_EXT_ACCESS_ENA 0x75 +#define WM8990_EXT_CTL1 0x7a /* * Field Definitions. diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index ffb471e420e2..af83d629078a 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -487,7 +487,8 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } -static int ac97_prepare(struct snd_pcm_substream *substream) +static int ac97_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -507,7 +508,8 @@ static int ac97_prepare(struct snd_pcm_substream *substream) return ac97_write(codec, reg, runtime->rate); } -static int ac97_aux_prepare(struct snd_pcm_substream *substream) +static int ac97_aux_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -533,7 +535,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) struct snd_soc_dai wm9712_dai[] = { { .name = "AC97 HiFi", - .type = SND_SOC_DAI_AC97_BUS, + .ac97_control = 1, .playback = { .stream_name = "HiFi Playback", .channels_min = 1, @@ -688,7 +690,7 @@ static int wm9712_soc_probe(struct platform_device *pdev) ret = wm9712_reset(codec, 0); if (ret < 0) { - printk(KERN_ERR "AC97 link error\n"); + printk(KERN_ERR "Failed to reset WM9712: AC97 link error\n"); goto reset_err; } @@ -698,7 +700,7 @@ static int wm9712_soc_probe(struct platform_device *pdev) wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm9712_add_controls(codec); wm9712_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register card\n"); goto reset_err; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 945b32ed9884..f3ca8aaf0139 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -928,11 +928,10 @@ static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = dai->codec; u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3; switch (params_format(params)) { @@ -954,11 +953,10 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static void wm9713_voiceshutdown(struct snd_pcm_substream *substream) +static void wm9713_voiceshutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = dai->codec; u16 status; /* Gracefully shut down the voice interface. */ @@ -969,12 +967,11 @@ static void wm9713_voiceshutdown(struct snd_pcm_substream *substream) ac97_write(codec, AC97_EXTENDED_MID, status); } -static int ac97_hifi_prepare(struct snd_pcm_substream *substream) +static int ac97_hifi_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { + struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; int reg; u16 vra; @@ -989,12 +986,11 @@ static int ac97_hifi_prepare(struct snd_pcm_substream *substream) return ac97_write(codec, reg, runtime->rate); } -static int ac97_aux_prepare(struct snd_pcm_substream *substream) +static int ac97_aux_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { + struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; u16 vra, xsle; vra = ac97_read(codec, AC97_EXTENDED_STATUS); @@ -1028,7 +1024,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) struct snd_soc_dai wm9713_dai[] = { { .name = "AC97 HiFi", - .type = SND_SOC_DAI_AC97_BUS, + .ac97_control = 1, .playback = { .stream_name = "HiFi Playback", .channels_min = 1, @@ -1042,8 +1038,7 @@ struct snd_soc_dai wm9713_dai[] = { .rates = WM9713_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .ops = { - .prepare = ac97_hifi_prepare,}, - .dai_ops = { + .prepare = ac97_hifi_prepare, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll,}, }, @@ -1056,8 +1051,7 @@ struct snd_soc_dai wm9713_dai[] = { .rates = WM9713_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .ops = { - .prepare = ac97_aux_prepare,}, - .dai_ops = { + .prepare = ac97_aux_prepare, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll,}, }, @@ -1077,8 +1071,7 @@ struct snd_soc_dai wm9713_dai[] = { .formats = WM9713_PCM_FORMATS,}, .ops = { .hw_params = wm9713_pcm_hw_params, - .shutdown = wm9713_voiceshutdown,}, - .dai_ops = { + .shutdown = wm9713_voiceshutdown, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll, .set_fmt = wm9713_set_dai_fmt, @@ -1097,6 +1090,8 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) } soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); if (ac97_read(codec, 0) != wm9713_reg[0]) return -EIO; return 0; @@ -1240,7 +1235,7 @@ static int wm9713_soc_probe(struct platform_device *pdev) wm9713_reset(codec, 0); ret = wm9713_reset(codec, 1); if (ret < 0) { - printk(KERN_ERR "AC97 link error\n"); + printk(KERN_ERR "Failed to reset WM9713: AC97 link error\n"); goto reset_err; } @@ -1252,7 +1247,7 @@ static int wm9713_soc_probe(struct platform_device *pdev) wm9713_add_controls(codec); wm9713_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) goto reset_err; return 0; @@ -1288,7 +1283,6 @@ static int wm9713_soc_remove(struct platform_device *pdev) snd_soc_free_ac97_codec(codec); kfree(codec->private_data); kfree(codec->reg_cache); - kfree(codec->dai); kfree(codec); return 0; } diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 8f7e33834902..b502741692d6 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -17,3 +17,13 @@ config SND_DAVINCI_SOC_EVM help Say Y if you want to add support for SoC audio on TI DaVinci EVM platform. + +config SND_DAVINCI_SOC_SFFSDR + tristate "SoC Audio support for SFFSDR" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR + select SND_DAVINCI_SOC_I2S + select SND_SOC_PCM3008 + select SFFSDR_FPGA + help + Say Y if you want to add support for SoC audio on + Lyrtech SFFSDR board. diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index ca772e5b4637..ca8bae1fc3f6 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -7,5 +7,7 @@ obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o +snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 9e6062cd6b59..01b948bb55a1 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -28,6 +28,8 @@ #define EVM_CODEC_CLOCK 22579200 +#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ + SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -37,14 +39,12 @@ static int evm_hw_params(struct snd_pcm_substream *substream, int ret = 0; /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_CBM_CFM); + ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF); + ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); if (ret < 0) return ret; @@ -128,8 +128,9 @@ static struct snd_soc_dai_link evm_dai = { }; /* davinci-evm audio machine driver */ -static struct snd_soc_machine snd_soc_machine_evm = { +static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", + .platform = &davinci_soc_platform, .dai_link = &evm_dai, .num_links = 1, }; @@ -142,8 +143,7 @@ static struct aic3x_setup_data evm_aic3x_setup = { /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { - .machine = &snd_soc_machine_evm, - .platform = &davinci_soc_platform, + .card = &snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, .codec_data = &evm_aic3x_setup, }; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index abb5fedb0b1e..0fee779e3c76 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -59,6 +59,7 @@ #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) #define DAVINCI_MCBSP_PCR_FSRP (1 << 2) #define DAVINCI_MCBSP_PCR_FSXP (1 << 3) +#define DAVINCI_MCBSP_PCR_SCLKME (1 << 7) #define DAVINCI_MCBSP_PCR_CLKRM (1 << 8) #define DAVINCI_MCBSP_PCR_CLKXM (1 << 9) #define DAVINCI_MCBSP_PCR_FSRM (1 << 10) @@ -110,16 +111,59 @@ static void davinci_mcbsp_start(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_platform *platform = socdev->card->platform; u32 w; + int ret; /* Start the sample generator and enable transmitter/receiver */ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* Stop the DMA to avoid data loss */ + /* while the transmitter is out of reset to handle XSYNCERR */ + if (platform->pcm_ops->trigger) { + ret = platform->pcm_ops->trigger(substream, + SNDRV_PCM_TRIGGER_STOP); + if (ret < 0) + printk(KERN_DEBUG "Playback DMA stop failed\n"); + } + + /* Enable the transmitter */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1); - else + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + /* wait for any unexpected frame sync error to occur */ + udelay(100); + + /* Disable the transmitter to clear any outstanding XSYNCERR */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + /* Restart the DMA */ + if (platform->pcm_ops->trigger) { + ret = platform->pcm_ops->trigger(substream, + SNDRV_PCM_TRIGGER_START); + if (ret < 0) + printk(KERN_DEBUG "Playback DMA start failed\n"); + } + /* Enable the transmitter */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + } else { + + /* Enable the reciever */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + } + /* Start frame sync */ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -144,7 +188,8 @@ static void davinci_mcbsp_stop(struct snd_pcm_substream *substream) davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); } -static int davinci_i2s_startup(struct snd_pcm_substream *substream) +static int davinci_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -155,61 +200,138 @@ static int davinci_i2s_startup(struct snd_pcm_substream *substream) return 0; } +#define DEFAULT_BITPERSAMPLE 16 + static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct davinci_mcbsp_dev *dev = cpu_dai->private_data; - u32 w; + unsigned int pcr; + unsigned int srgr; + unsigned int rcr; + unsigned int xcr; + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | + DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, - DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM | - DAVINCI_MCBSP_PCR_CLKXM | - DAVINCI_MCBSP_PCR_CLKRM); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, - DAVINCI_MCBSP_SRGR_FSGM); + /* cpu is master */ + pcr = DAVINCI_MCBSP_PCR_FSXM | + DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; + break; + case SND_SOC_DAIFMT_CBM_CFS: + /* McBSP CLKR pin is the input for the Sample Rate Generator. + * McBSP FSR and FSX are driven by the Sample Rate Generator. */ + pcr = DAVINCI_MCBSP_PCR_SCLKME | + DAVINCI_MCBSP_PCR_FSXM | + DAVINCI_MCBSP_PCR_FSRM; break; case SND_SOC_DAIFMT_CBM_CFM: - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, 0); + /* codec is master */ + pcr = 0; break; default: + printk(KERN_ERR "%s:bad master\n", __func__); return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_IB_NF: - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP | - DAVINCI_MCBSP_PCR_CLKRP, 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + rcr = DAVINCI_MCBSP_RCR_RFRLEN1(1); + xcr = DAVINCI_MCBSP_XCR_XFIG | DAVINCI_MCBSP_XCR_XFRLEN1(1); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_B: break; - case SND_SOC_DAIFMT_NB_IF: - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_FSXP | - DAVINCI_MCBSP_PCR_FSRP, 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + case SND_SOC_DAIFMT_I2S: + /* Davinci doesn't support TRUE I2S, but some codecs will have + * the left and right channels contiguous. This allows + * dsp_a mode to be used with an inverted normal frame clk. + * If your codec is master and does not have contiguous + * channels, then you will have sound on only one channel. + * Try using a different mode, or codec as slave. + * + * The TLV320AIC33 is an example of a codec where this works. + * It has a variable bit clock frequency allowing it to have + * valid data on every bit clock. + * + * The TLV320AIC23 is an example of a codec where this does not + * work. It has a fixed bit clock frequency with progressively + * more empty bit clock slots between channels as the sample + * rate is lowered. + */ + fmt ^= SND_SOC_DAIFMT_NB_IF; + case SND_SOC_DAIFMT_DSP_A: + rcr |= DAVINCI_MCBSP_RCR_RDATDLY(1); + xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); + break; + default: + printk(KERN_ERR "%s:bad format\n", __func__); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* CLKRP Receive clock polarity, + * 1 - sampled on rising edge of CLKR + * valid on rising edge + * CLKXP Transmit clock polarity, + * 1 - clocked on falling edge of CLKX + * valid on rising edge + * FSRP Receive frame sync pol, 0 - active high + * FSXP Transmit frame sync pol, 0 - active high + */ + pcr |= (DAVINCI_MCBSP_PCR_CLKXP | DAVINCI_MCBSP_PCR_CLKRP); break; case SND_SOC_DAIFMT_IB_IF: - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP | - DAVINCI_MCBSP_PCR_CLKRP | - DAVINCI_MCBSP_PCR_FSXP | - DAVINCI_MCBSP_PCR_FSRP, 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + /* CLKRP Receive clock polarity, + * 0 - sampled on falling edge of CLKR + * valid on falling edge + * CLKXP Transmit clock polarity, + * 0 - clocked on rising edge of CLKX + * valid on falling edge + * FSRP Receive frame sync pol, 1 - active low + * FSXP Transmit frame sync pol, 1 - active low + */ + pcr |= (DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP); break; - case SND_SOC_DAIFMT_NB_NF: + case SND_SOC_DAIFMT_NB_IF: + /* CLKRP Receive clock polarity, + * 1 - sampled on rising edge of CLKR + * valid on rising edge + * CLKXP Transmit clock polarity, + * 1 - clocked on falling edge of CLKX + * valid on rising edge + * FSRP Receive frame sync pol, 1 - active low + * FSXP Transmit frame sync pol, 1 - active low + */ + pcr |= (DAVINCI_MCBSP_PCR_CLKXP | DAVINCI_MCBSP_PCR_CLKRP | + DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP); + break; + case SND_SOC_DAIFMT_IB_NF: + /* CLKRP Receive clock polarity, + * 0 - sampled on falling edge of CLKR + * valid on falling edge + * CLKXP Transmit clock polarity, + * 0 - clocked on rising edge of CLKX + * valid on falling edge + * FSRP Receive frame sync pol, 0 - active high + * FSXP Transmit frame sync pol, 0 - active high + */ break; default: return -EINVAL; } - + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, pcr); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); return 0; } static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; @@ -219,25 +341,20 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, u32 w; /* general line settings */ - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, - DAVINCI_MCBSP_SPCR_RINTM(3) | - DAVINCI_MCBSP_SPCR_XINTM(3) | - DAVINCI_MCBSP_SPCR_FREE); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, - DAVINCI_MCBSP_RCR_RFRLEN1(1) | - DAVINCI_MCBSP_RCR_RDATDLY(1)); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, - DAVINCI_MCBSP_XCR_XFRLEN1(1) | - DAVINCI_MCBSP_XCR_XDATDLY(1) | - DAVINCI_MCBSP_XCR_XFIG); + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + w |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + } else { + w |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + } i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG); + w = DAVINCI_MCBSP_SRGR_FSGM; MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1), 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w); i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG); MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1), 1); davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w); @@ -260,20 +377,24 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w); - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w); + } else { + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w); + } return 0; } -static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { int ret = 0; @@ -299,8 +420,8 @@ static int davinci_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea; struct evm_snd_platform_data *pdata; @@ -361,8 +482,8 @@ static void davinci_i2s_remove(struct platform_device *pdev, struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev = cpu_dai->private_data; struct resource *mem; @@ -381,7 +502,6 @@ static void davinci_i2s_remove(struct platform_device *pdev, struct snd_soc_dai davinci_i2s_dai = { .name = "davinci-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .probe = davinci_i2s_probe, .remove = davinci_i2s_remove, .playback = { @@ -397,13 +517,24 @@ struct snd_soc_dai davinci_i2s_dai = { .ops = { .startup = davinci_i2s_startup, .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params,}, - .dai_ops = { + .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, }, }; EXPORT_SYMBOL_GPL(davinci_i2s_dai); +static int __init davinci_i2s_init(void) +{ + return snd_soc_register_dai(&davinci_i2s_dai); +} +module_init(davinci_i2s_init); + +static void __exit davinci_i2s_exit(void) +{ + snd_soc_unregister_dai(&davinci_i2s_dai); +} +module_exit(davinci_i2s_exit); + MODULE_AUTHOR("Vladimir Barinov"); MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 76feaa657375..74abc9b4f1cc 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -14,6 +14,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/dma-mapping.h> +#include <linux/kernel.h> #include <sound/core.h> #include <sound/pcm.h> @@ -24,13 +25,6 @@ #include "davinci-pcm.h" -#define DAVINCI_PCM_DEBUG 0 -#if DAVINCI_PCM_DEBUG -#define DPRINTK(x...) printk(KERN_DEBUG x) -#else -#define DPRINTK(x...) -#endif - static struct snd_pcm_hardware davinci_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -78,8 +72,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dma_offset = prtd->period * period_size; dma_pos = runtime->dma_addr + dma_offset; - DPRINTK("audio_set_dma_params_play channel = %d dma_ptr = %x " - "period_size=%x\n", lch, dma_pos, period_size); + pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " + "dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size); data_type = prtd->params->data_type; count = period_size / data_type; @@ -112,7 +106,7 @@ static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data) struct snd_pcm_substream *substream = data; struct davinci_runtime_data *prtd = substream->runtime->private_data; - DPRINTK("lch=%d, status=0x%x\n", lch, ch_status); + pr_debug("davinci_pcm: lch=%d, status=0x%x\n", lch, ch_status); if (unlikely(ch_status != DMA_COMPLETE)) return; @@ -316,8 +310,8 @@ static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) buf->area = dma_alloc_writecombine(pcm->card->dev, size, &buf->addr, GFP_KERNEL); - DPRINTK("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", - (void *) buf->area, (void *) buf->addr, size); + pr_debug("davinci_pcm: preallocate_dma_buffer: area=%p, addr=%p, " + "size=%d\n", (void *) buf->area, (void *) buf->addr, size); if (!buf->area) return -ENOMEM; @@ -384,6 +378,18 @@ struct snd_soc_platform davinci_soc_platform = { }; EXPORT_SYMBOL_GPL(davinci_soc_platform); +static int __init davinci_soc_platform_init(void) +{ + return snd_soc_register_platform(&davinci_soc_platform); +} +module_init(davinci_soc_platform_init); + +static void __exit davinci_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&davinci_soc_platform); +} +module_exit(davinci_soc_platform_exit); + MODULE_AUTHOR("Vladimir Barinov"); MODULE_DESCRIPTION("TI DAVINCI PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c new file mode 100644 index 000000000000..f67579d52765 --- /dev/null +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -0,0 +1,157 @@ +/* + * ASoC driver for Lyrtech SFFSDR board. + * + * Author: Hugo Villeneuve + * Copyright (C) 2008 Lyrtech inc + * + * Based on ASoC driver for TI DAVINCI EVM platform, original copyright follow: + * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/dma.h> +#include <asm/plat-sffsdr/sffsdr-fpga.h> + +#include <mach/mcbsp.h> +#include <mach/edma.h> + +#include "../codecs/pcm3008.h" +#include "davinci-pcm.h" +#include "davinci-i2s.h" + +static int sffsdr_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int fs; + int ret = 0; + + /* Set cpu DAI configuration: + * CLKX and CLKR are the inputs for the Sample Rate Generator. + * FSX and FSR are outputs, driven by the sample Rate Generator. */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_RIGHT_J | + SND_SOC_DAIFMT_CBM_CFS | + SND_SOC_DAIFMT_IB_NF); + if (ret < 0) + return ret; + + /* Fsref can be 32000, 44100 or 48000. */ + fs = params_rate(params); + + pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs); + + return sffsdr_fpga_set_codec_fs(fs); +} + +static struct snd_soc_ops sffsdr_ops = { + .hw_params = sffsdr_hw_params, +}; + +/* davinci-sffsdr digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sffsdr_dai = { + .name = "PCM3008", /* Codec name */ + .stream_name = "PCM3008 HiFi", + .cpu_dai = &davinci_i2s_dai, + .codec_dai = &pcm3008_dai, + .ops = &sffsdr_ops, +}; + +/* davinci-sffsdr audio machine driver */ +static struct snd_soc_card snd_soc_sffsdr = { + .name = "DaVinci SFFSDR", + .platform = &davinci_soc_platform, + .dai_link = &sffsdr_dai, + .num_links = 1, +}; + +/* sffsdr audio private data */ +static struct pcm3008_setup_data sffsdr_pcm3008_setup = { + .dem0_pin = GPIO(45), + .dem1_pin = GPIO(46), + .pdad_pin = GPIO(47), + .pdda_pin = GPIO(38), +}; + +/* sffsdr audio subsystem */ +static struct snd_soc_device sffsdr_snd_devdata = { + .card = &snd_soc_sffsdr, + .codec_dev = &soc_codec_dev_pcm3008, + .codec_data = &sffsdr_pcm3008_setup, +}; + +static struct resource sffsdr_snd_resources[] = { + { + .start = DAVINCI_MCBSP_BASE, + .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data sffsdr_snd_data = { + .tx_dma_ch = DAVINCI_DMA_MCBSP_TX, + .rx_dma_ch = DAVINCI_DMA_MCBSP_RX, +}; + +static struct platform_device *sffsdr_snd_device; + +static int __init sffsdr_init(void) +{ + int ret; + + sffsdr_snd_device = platform_device_alloc("soc-audio", 0); + if (!sffsdr_snd_device) { + printk(KERN_ERR "platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sffsdr_snd_device, &sffsdr_snd_devdata); + sffsdr_snd_devdata.dev = &sffsdr_snd_device->dev; + sffsdr_snd_device->dev.platform_data = &sffsdr_snd_data; + + ret = platform_device_add_resources(sffsdr_snd_device, + sffsdr_snd_resources, + ARRAY_SIZE(sffsdr_snd_resources)); + if (ret) { + printk(KERN_ERR "platform device add ressources failed\n"); + goto error; + } + + ret = platform_device_add(sffsdr_snd_device); + if (ret) + goto error; + + return ret; + +error: + platform_device_put(sffsdr_snd_device); + return ret; +} + +static void __exit sffsdr_exit(void) +{ + platform_device_unregister(sffsdr_snd_device); +} + +module_init(sffsdr_init); +module_exit(sffsdr_exit); + +MODULE_AUTHOR("Hugo Villeneuve"); +MODULE_DESCRIPTION("Lyrtech SFFSDR ASoC driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 8d73edc56102..95c12b26fe37 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -20,7 +20,7 @@ config SND_SOC_MPC8610_HPCD config SND_SOC_MPC5200_I2S tristate "Freescale MPC5200 PSC in I2S mode driver" - depends on SND_SOC && PPC_MPC52xx && PPC_BESTCOMM + depends on PPC_MPC52xx && PPC_BESTCOMM select SND_SOC_OF_SIMPLE select PPC_BESTCOMM_GEN_BD help diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index d2d3da9729f2..64993eda5679 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -284,7 +284,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) * fsl_dma_new: initialize this PCM driver. * * This function is called when the codec driver calls snd_soc_new_pcms(), - * once for each .dai_link in the machine driver's snd_soc_machine + * once for each .dai_link in the machine driver's snd_soc_card * structure. */ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, @@ -853,6 +853,18 @@ int fsl_dma_configure(struct fsl_dma_info *dma_info) } EXPORT_SYMBOL_GPL(fsl_dma_configure); +static int __init fsl_soc_platform_init(void) +{ + return snd_soc_register_platform(&fsl_soc_platform); +} +module_init(fsl_soc_platform_init); + +static void __exit fsl_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&fsl_soc_platform); +} +module_exit(fsl_soc_platform_exit); + MODULE_AUTHOR("Timur Tabi <timur@freescale.com>"); MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 157a7895ffa1..c6d6eb71dc1d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -266,7 +266,8 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) * If this is the first stream open, then grab the IRQ and program most of * the SSI registers. */ -static int fsl_ssi_startup(struct snd_pcm_substream *substream) +static int fsl_ssi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; @@ -411,7 +412,8 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream) * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the * clock master. */ -static int fsl_ssi_prepare(struct snd_pcm_substream *substream) +static int fsl_ssi_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -441,7 +443,8 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream) * The DMA channel is in external master start and pause mode, which * means the SSI completely controls the flow of data. */ -static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) +static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; @@ -490,7 +493,8 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) * * Shutdown the SSI if there are no other substreams open. */ -static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) +static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; @@ -578,8 +582,6 @@ static struct snd_soc_dai fsl_ssi_dai_template = { .prepare = fsl_ssi_prepare, .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, - }, - .dai_ops = { .set_sysclk = fsl_ssi_set_sysclk, .set_fmt = fsl_ssi_set_fmt, }, @@ -671,6 +673,14 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) fsl_ssi_dai->private_data = ssi_private; fsl_ssi_dai->name = ssi_private->name; fsl_ssi_dai->id = ssi_info->id; + fsl_ssi_dai->dev = ssi_info->dev; + + ret = snd_soc_register_dai(fsl_ssi_dai); + if (ret != 0) { + dev_err(ssi_info->dev, "failed to register DAI: %d\n", ret); + kfree(fsl_ssi_dai); + return NULL; + } return fsl_ssi_dai; } @@ -688,6 +698,8 @@ void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai) device_remove_file(ssi_private->dev, &ssi_private->dev_attr); + snd_soc_unregister_dai(&ssi_private->cpu_dai); + kfree(ssi_private); } EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai); diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 94a02eaa4825..9eb1ce185bd0 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -187,7 +187,8 @@ static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream) * If this is the first stream open, then grab the IRQ and program most of * the PSC registers. */ -static int psc_i2s_startup(struct snd_pcm_substream *substream) +static int psc_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; @@ -220,7 +221,8 @@ static int psc_i2s_startup(struct snd_pcm_substream *substream) } static int psc_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; @@ -256,7 +258,8 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int psc_i2s_hw_free(struct snd_pcm_substream *substream) +static int psc_i2s_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { snd_pcm_set_runtime_buffer(substream, NULL); return 0; @@ -268,7 +271,8 @@ static int psc_i2s_hw_free(struct snd_pcm_substream *substream) * This function is called by ALSA to start, stop, pause, and resume the DMA * transfer of data. */ -static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; @@ -383,7 +387,8 @@ static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) * * Shutdown the PSC if there are no other substreams open. */ -static void psc_i2s_shutdown(struct snd_pcm_substream *substream) +static void psc_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; @@ -464,7 +469,6 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) * psc_i2s_dai_template: template CPU Digital Audio Interface */ static struct snd_soc_dai psc_i2s_dai_template = { - .type = SND_SOC_DAI_I2S, .playback = { .channels_min = 2, .channels_max = 2, @@ -483,8 +487,6 @@ static struct snd_soc_dai psc_i2s_dai_template = { .hw_free = psc_i2s_hw_free, .shutdown = psc_i2s_shutdown, .trigger = psc_i2s_trigger, - }, - .dai_ops = { .set_sysclk = psc_i2s_set_sysclk, .set_fmt = psc_i2s_set_fmt, }, @@ -826,6 +828,8 @@ static int __devinit psc_i2s_of_probe(struct of_device *op, if (rc) dev_info(psc_i2s->dev, "error creating sysfs files\n"); + snd_soc_register_platform(&psc_i2s_pcm_soc_platform); + /* Tell the ASoC OF helpers about it */ of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node, &psc_i2s->dai); @@ -839,6 +843,8 @@ static int __devexit psc_i2s_of_remove(struct of_device *op) dev_dbg(&op->dev, "psc_i2s_remove()\n"); + snd_soc_unregister_platform(&psc_i2s_pcm_soc_platform); + bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task); bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task); diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 94f89debde1f..bcec3f60bad9 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -29,7 +29,7 @@ struct mpc8610_hpcd_data { struct snd_soc_device sound_devdata; struct snd_soc_dai_link dai; - struct snd_soc_machine machine; + struct snd_soc_card machine; unsigned int dai_format; unsigned int codec_clk_direction; unsigned int cpu_clk_direction; @@ -185,7 +185,7 @@ static struct snd_soc_ops mpc8610_hpcd_ops = { /** * mpc8610_hpcd_machine: ASoC machine data */ -static struct snd_soc_machine mpc8610_hpcd_machine = { +static struct snd_soc_card mpc8610_hpcd_machine = { .probe = mpc8610_hpcd_machine_probe, .remove = mpc8610_hpcd_machine_remove, .name = "MPC8610 HPCD", @@ -465,9 +465,9 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, goto error; } - machine_data->sound_devdata.machine = &mpc8610_hpcd_machine; + machine_data->sound_devdata.card = &mpc8610_hpcd_machine; machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270; - machine_data->sound_devdata.platform = &fsl_soc_platform; + machine_data->machine.platform = &fsl_soc_platform; sound_device->dev.platform_data = machine_data; diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c index 0382fdac51cd..8bc5cd9e972f 100644 --- a/sound/soc/fsl/soc-of-simple.c +++ b/sound/soc/fsl/soc-of-simple.c @@ -31,7 +31,7 @@ struct of_snd_soc_device { int id; struct list_head list; struct snd_soc_device device; - struct snd_soc_machine machine; + struct snd_soc_card card; struct snd_soc_dai_link dai_link; struct platform_device *pdev; struct device_node *platform_node; @@ -58,9 +58,9 @@ of_snd_soc_get_device(struct device_node *codec_node) /* Initialize the structure and add it to the global list */ of_soc->codec_node = codec_node; of_soc->id = of_snd_soc_next_index++; - of_soc->machine.dai_link = &of_soc->dai_link; - of_soc->machine.num_links = 1; - of_soc->device.machine = &of_soc->machine; + of_soc->card.dai_link = &of_soc->dai_link; + of_soc->card.num_links = 1; + of_soc->device.card = &of_soc->card; of_soc->dai_link.ops = &of_snd_soc_ops; list_add(&of_soc->list, &of_snd_soc_device_list); @@ -158,8 +158,8 @@ int of_snd_soc_register_platform(struct snd_soc_platform *platform, of_soc->platform_node = node; of_soc->dai_link.cpu_dai = cpu_dai; - of_soc->device.platform = platform; - of_soc->machine.name = of_soc->dai_link.cpu_dai->name; + of_soc->card.platform = platform; + of_soc->card.name = of_soc->dai_link.cpu_dai->name; /* Now try to register the SoC device */ of_snd_soc_register_device(of_soc); diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 8b7766b998d7..a7b1d77b2105 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,6 +1,6 @@ config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" - depends on ARCH_OMAP && SND_SOC + depends on ARCH_OMAP config SND_OMAP_SOC_MCBSP tristate @@ -21,3 +21,36 @@ config SND_OMAP_SOC_OSK5912 select SND_SOC_TLV320AIC23 help Say Y if you want to add support for SoC audio on osk5912. + +config SND_OMAP_SOC_OVERO + tristate "SoC Audio support for Gumstix Overo" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OVERO + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the Gumstix Overo. + +config SND_OMAP_SOC_OMAP2EVM + tristate "SoC Audio support for OMAP2EVM board" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP2EVM + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the omap2evm board. + +config SND_OMAP_SOC_SDP3430 + tristate "SoC Audio support for Texas Instruments SDP3430" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on Texas Instruments + SDP3430. + +config SND_OMAP_SOC_OMAP3_PANDORA + tristate "SoC Audio support for OMAP3 Pandora" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP3 Pandora. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index e09d1f297f64..76fedd96e365 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -8,6 +8,14 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o # OMAP Machine Support snd-soc-n810-objs := n810.o snd-soc-osk5912-objs := osk5912.o +snd-soc-overo-objs := overo.o +snd-soc-omap2evm-objs := omap2evm.o +snd-soc-sdp3430-objs := sdp3430.o +snd-soc-omap3pandora-objs := omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o +obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o +obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index fae3ad36e0bf..25593fee9121 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -70,9 +70,13 @@ static void n810_ext_control(struct snd_soc_codec *codec) static int n810_startup(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->socdev->codec; + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); + n810_ext_control(codec); return clk_enable(sys_clkout2); } @@ -282,8 +286,9 @@ static struct snd_soc_dai_link n810_dai = { }; /* Audio machine driver */ -static struct snd_soc_machine snd_soc_machine_n810 = { +static struct snd_soc_card snd_soc_n810 = { .name = "N810", + .platform = &omap_soc_platform, .dai_link = &n810_dai, .num_links = 1, }; @@ -298,8 +303,7 @@ static struct aic3x_setup_data n810_aic33_setup = { /* Audio subsystem */ static struct snd_soc_device n810_snd_devdata = { - .machine = &snd_soc_machine_n810, - .platform = &omap_soc_platform, + .card = &snd_soc_n810, .codec_dev = &soc_codec_dev_aic3x, .codec_data = &n810_aic33_setup, }; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 8485a8a9d0ff..ec5e18a78758 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -36,9 +36,7 @@ #include "omap-mcbsp.h" #include "omap-pcm.h" -#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | \ - SNDRV_PCM_RATE_KNOT) +#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) struct omap_mcbsp_data { unsigned int bus_id; @@ -140,7 +138,8 @@ static const unsigned long omap34xx_mcbsp_port[][2] = { static const unsigned long omap34xx_mcbsp_port[][2] = {}; #endif -static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) +static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -153,7 +152,8 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) return err; } -static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) +static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -165,7 +165,8 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) } } -static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd) +static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -194,14 +195,15 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd) } static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; - int wlen; + int wlen, channels; unsigned long port; if (cpu_class_is_omap1()) { @@ -230,12 +232,17 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return 0; } - switch (params_channels(params)) { + channels = params_channels(params); + switch (channels) { case 2: - /* Set 1 word per (McBPSP) frame and use dual-phase frames */ - regs->rcr2 |= RFRLEN2(1 - 1) | RPHASE; + /* Use dual-phase frames */ + regs->rcr2 |= RPHASE; + regs->xcr2 |= XPHASE; + case 1: + /* Set 1 word per (McBSP) frame */ + regs->rcr2 |= RFRLEN2(1 - 1); regs->rcr1 |= RFRLEN1(1 - 1); - regs->xcr2 |= XFRLEN2(1 - 1) | XPHASE; + regs->xcr2 |= XFRLEN2(1 - 1); regs->xcr1 |= XFRLEN1(1 - 1); break; default: @@ -263,9 +270,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, regs->srgr2 |= FPER(wlen * 2 - 1); regs->srgr1 |= FWID(wlen - 1); break; - case SND_SOC_DAIFMT_DSP_A: - regs->srgr2 |= FPER(wlen * 2 - 1); - regs->srgr1 |= FWID(wlen * 2 - 2); + case SND_SOC_DAIFMT_DSP_B: + regs->srgr2 |= FPER(wlen * channels - 1); + regs->srgr1 |= FWID(wlen * channels - 2); break; } @@ -302,7 +309,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; - case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); @@ -452,17 +459,16 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, #define OMAP_MCBSP_DAI_BUILDER(link_id) \ { \ - .name = "omap-mcbsp-dai-(link_id)", \ + .name = "omap-mcbsp-dai-"#link_id, \ .id = (link_id), \ - .type = SND_SOC_DAI_I2S, \ .playback = { \ - .channels_min = 2, \ + .channels_min = 1, \ .channels_max = 2, \ .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ .capture = { \ - .channels_min = 2, \ + .channels_min = 1, \ .channels_max = 2, \ .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ @@ -472,8 +478,6 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, .shutdown = omap_mcbsp_dai_shutdown, \ .trigger = omap_mcbsp_dai_trigger, \ .hw_params = omap_mcbsp_dai_hw_params, \ - }, \ - .dai_ops = { \ .set_fmt = omap_mcbsp_dai_set_dai_fmt, \ .set_clkdiv = omap_mcbsp_dai_set_clkdiv, \ .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, \ @@ -495,6 +499,19 @@ struct snd_soc_dai omap_mcbsp_dai[] = { EXPORT_SYMBOL_GPL(omap_mcbsp_dai); +static int __init snd_omap_mcbsp_init(void) +{ + return snd_soc_register_dais(omap_mcbsp_dai, + ARRAY_SIZE(omap_mcbsp_dai)); +} +module_init(snd_omap_mcbsp_init); + +static void __exit snd_omap_mcbsp_exit(void) +{ + snd_soc_unregister_dais(omap_mcbsp_dai, ARRAY_SIZE(omap_mcbsp_dai)); +} +module_exit(snd_omap_mcbsp_exit); + MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>"); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index acd68efb2b75..b0362dfd5b71 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, prtd->dma_data = dma_data; err = omap_request_dma(dma_data->dma_req, dma_data->name, omap_pcm_dma_irq, substream, &prtd->dma_ch); - if (!err & !cpu_is_omap1510()) { + if (!err && !cpu_is_omap1510()) { /* * Link channel with itself so DMA doesn't need any * reprogramming while looping the buffer @@ -354,6 +354,18 @@ struct snd_soc_platform omap_soc_platform = { }; EXPORT_SYMBOL_GPL(omap_soc_platform); +static int __init omap_soc_platform_init(void) +{ + return snd_soc_register_platform(&omap_soc_platform); +} +module_init(omap_soc_platform_init); + +static void __exit omap_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&omap_soc_platform); +} +module_exit(omap_soc_platform_exit); + MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>"); MODULE_DESCRIPTION("OMAP PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c new file mode 100644 index 000000000000..0c2322dcf02a --- /dev/null +++ b/sound/soc/omap/omap2evm.c @@ -0,0 +1,151 @@ +/* + * omap2evm.c -- SoC audio machine driver for omap2evm board + * + * Author: Arun KS <arunks@mistralsolutions.com> + * + * Based on sound/soc/omap/overo.c by Steve Sakoman + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int omap2evm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap2evm_ops = { + .hw_params = omap2evm_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap2evm_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap2evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_omap2evm = { + .name = "omap2evm", + .platform = &omap_soc_platform, + .dai_link = &omap2evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap2evm_snd_devdata = { + .card = &snd_soc_omap2evm, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap2evm_snd_device; + +static int __init omap2evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap2evm()) { + pr_debug("Not omap2evm!\n"); + return -ENODEV; + } + printk(KERN_INFO "omap2evm SoC init\n"); + + omap2evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap2evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap2evm_snd_device, &omap2evm_snd_devdata); + omap2evm_snd_devdata.dev = &omap2evm_snd_device->dev; + *(unsigned int *)omap2evm_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(omap2evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap2evm_snd_device); + + return ret; +} +module_init(omap2evm_soc_init); + +static void __exit omap2evm_soc_exit(void) +{ + platform_device_unregister(omap2evm_snd_device); +} +module_exit(omap2evm_soc_exit); + +MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); +MODULE_DESCRIPTION("ALSA SoC omap2evm"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c new file mode 100644 index 000000000000..fd24a4acd2f5 --- /dev/null +++ b/sound/soc/omap/omap3beagle.c @@ -0,0 +1,149 @@ +/* + * omap3beagle.c -- SoC audio for OMAP3 Beagle + * + * Author: Steve Sakoman <steve@sakoman.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int omap3beagle_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3beagle_ops = { + .hw_params = omap3beagle_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3beagle_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3beagle_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_omap3beagle = { + .name = "omap3beagle", + .platform = &omap_soc_platform, + .dai_link = &omap3beagle_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3beagle_snd_devdata = { + .card = &snd_soc_omap3beagle, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3beagle_snd_device; + +static int __init omap3beagle_soc_init(void) +{ + int ret; + + if (!machine_is_omap3_beagle()) { + pr_debug("Not OMAP3 Beagle!\n"); + return -ENODEV; + } + pr_info("OMAP3 Beagle SoC init\n"); + + omap3beagle_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap3beagle_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap3beagle_snd_device, &omap3beagle_snd_devdata); + omap3beagle_snd_devdata.dev = &omap3beagle_snd_device->dev; + *(unsigned int *)omap3beagle_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(omap3beagle_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap3beagle_snd_device); + + return ret; +} + +static void __exit omap3beagle_soc_exit(void) +{ + platform_device_unregister(omap3beagle_snd_device); +} + +module_init(omap3beagle_soc_init); +module_exit(omap3beagle_soc_exit); + +MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 Beagle"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c new file mode 100644 index 000000000000..bd91594496b1 --- /dev/null +++ b/sound/soc/omap/omap3pandora.c @@ -0,0 +1,311 @@ +/* + * omap3pandora.c -- SoC audio for Pandora Handheld Console + * + * Author: Gražvydas Ignotas <notasas@gmail.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <linux/delay.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define OMAP3_PANDORA_DAC_POWER_GPIO 118 +#define OMAP3_PANDORA_AMP_POWER_GPIO 14 + +#define PREFIX "ASoC omap3pandora: " + +static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_err(PREFIX "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_err(PREFIX "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set codec system clock\n"); + return ret; + } + + /* Set McBSP clock to external */ + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set cpu system clock\n"); + return ret; + } + + ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8); + if (ret < 0) { + pr_err(PREFIX "can't set SRG clock divider\n"); + return ret; + } + + return 0; +} + +static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFS); +} + +static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +} + +static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) { + gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1); + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1); + } else { + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + mdelay(1); + gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + } + + return 0; +} + +/* + * Audio paths on Pandora board: + * + * |O| ---> PCM DAC +-> AMP -> Headphone Jack + * |M| A +--------> Line Out + * |A| <~~clk~~+ + * |P| <--- TWL4030 <--------- Line In and MICs + */ +static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { + SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM, + 0, 0, NULL, 0, omap3pandora_hp_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), +}; + +static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_MIC("Mic (external)", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static const struct snd_soc_dapm_route omap3pandora_out_map[] = { + {"Headphone Amplifier", NULL, "PCM DAC"}, + {"Line Out", NULL, "PCM DAC"}, + {"Headphone Jack", NULL, "Headphone Amplifier"}, +}; + +static const struct snd_soc_dapm_route omap3pandora_in_map[] = { + {"INL", NULL, "Line In"}, + {"INR", NULL, "Line In"}, + {"INL", NULL, "Mic (Internal)"}, + {"INR", NULL, "Mic (external)"}, +}; + +static int omap3pandora_out_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, + ARRAY_SIZE(omap3pandora_out_dapm_widgets)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, omap3pandora_out_map, + ARRAY_SIZE(omap3pandora_out_map)); + + return snd_soc_dapm_sync(codec); +} + +static int omap3pandora_in_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets, + ARRAY_SIZE(omap3pandora_in_dapm_widgets)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, omap3pandora_in_map, + ARRAY_SIZE(omap3pandora_in_map)); + + return snd_soc_dapm_sync(codec); +} + +static struct snd_soc_ops omap3pandora_out_ops = { + .hw_params = omap3pandora_out_hw_params, +}; + +static struct snd_soc_ops omap3pandora_in_ops = { + .hw_params = omap3pandora_in_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3pandora_dai[] = { + { + .name = "PCM1773", + .stream_name = "HiFi Out", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3pandora_out_ops, + .init = omap3pandora_out_init, + }, { + .name = "TWL4030", + .stream_name = "Line/Mic In", + .cpu_dai = &omap_mcbsp_dai[1], + .codec_dai = &twl4030_dai, + .ops = &omap3pandora_in_ops, + .init = omap3pandora_in_init, + } +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_omap3pandora = { + .name = "omap3pandora", + .platform = &omap_soc_platform, + .dai_link = omap3pandora_dai, + .num_links = ARRAY_SIZE(omap3pandora_dai), +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3pandora_snd_data = { + .card = &snd_soc_card_omap3pandora, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3pandora_snd_device; + +static int __init omap3pandora_soc_init(void) +{ + int ret; + + if (!machine_is_omap3_pandora()) { + pr_debug(PREFIX "Not OMAP3 Pandora\n"); + return -ENODEV; + } + pr_info("OMAP3 Pandora SoC init\n"); + + ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power"); + if (ret) { + pr_err(PREFIX "Failed to get DAC power GPIO\n"); + return ret; + } + + ret = gpio_direction_output(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + if (ret) { + pr_err(PREFIX "Failed to set DAC power GPIO direction\n"); + goto fail0; + } + + ret = gpio_request(OMAP3_PANDORA_AMP_POWER_GPIO, "amp_power"); + if (ret) { + pr_err(PREFIX "Failed to get amp power GPIO\n"); + goto fail0; + } + + ret = gpio_direction_output(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + if (ret) { + pr_err(PREFIX "Failed to set amp power GPIO direction\n"); + goto fail1; + } + + omap3pandora_snd_device = platform_device_alloc("soc-audio", -1); + if (omap3pandora_snd_device == NULL) { + pr_err(PREFIX "Platform device allocation failed\n"); + ret = -ENOMEM; + goto fail1; + } + + platform_set_drvdata(omap3pandora_snd_device, &omap3pandora_snd_data); + omap3pandora_snd_data.dev = &omap3pandora_snd_device->dev; + *(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */ + *(unsigned int *)omap_mcbsp_dai[1].private_data = 3; /* McBSP4 */ + + ret = platform_device_add(omap3pandora_snd_device); + if (ret) { + pr_err(PREFIX "Unable to add platform device\n"); + goto fail2; + } + + return 0; + +fail2: + platform_device_put(omap3pandora_snd_device); +fail1: + gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); +fail0: + gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); + return ret; +} +module_init(omap3pandora_soc_init); + +static void __exit omap3pandora_soc_exit(void) +{ + platform_device_unregister(omap3pandora_snd_device); + gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); + gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); +} +module_exit(omap3pandora_soc_exit); + +MODULE_AUTHOR("Grazvydas Ignotas <notasas@gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 0fe733796898..cd41a948df7b 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -61,7 +61,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream, /* Set codec DAI configuration */ err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBM_CFM); if (err < 0) { @@ -71,7 +71,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream, /* Set cpu DAI configuration */ err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBM_CFM); if (err < 0) { @@ -143,16 +143,16 @@ static struct snd_soc_dai_link osk_dai = { }; /* Audio machine driver */ -static struct snd_soc_machine snd_soc_machine_osk = { +static struct snd_soc_card snd_soc_card_osk = { .name = "OSK5912", + .platform = &omap_soc_platform, .dai_link = &osk_dai, .num_links = 1, }; /* Audio subsystem */ static struct snd_soc_device osk_snd_devdata = { - .machine = &snd_soc_machine_osk, - .platform = &omap_soc_platform, + .card = &snd_soc_card_osk, .codec_dev = &soc_codec_dev_tlv320aic23, }; diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c new file mode 100644 index 000000000000..a72dc4e159e5 --- /dev/null +++ b/sound/soc/omap/overo.c @@ -0,0 +1,148 @@ +/* + * overo.c -- SoC audio for Gumstix Overo + * + * Author: Steve Sakoman <steve@sakoman.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int overo_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops overo_ops = { + .hw_params = overo_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link overo_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &overo_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_card_overo = { + .name = "overo", + .platform = &omap_soc_platform, + .dai_link = &overo_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device overo_snd_devdata = { + .card = &snd_soc_card_overo, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *overo_snd_device; + +static int __init overo_soc_init(void) +{ + int ret; + + if (!machine_is_overo()) { + pr_debug("Not Overo!\n"); + return -ENODEV; + } + printk(KERN_INFO "overo SoC init\n"); + + overo_snd_device = platform_device_alloc("soc-audio", -1); + if (!overo_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(overo_snd_device, &overo_snd_devdata); + overo_snd_devdata.dev = &overo_snd_device->dev; + *(unsigned int *)overo_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(overo_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(overo_snd_device); + + return ret; +} +module_init(overo_soc_init); + +static void __exit overo_soc_exit(void) +{ + platform_device_unregister(overo_snd_device); +} +module_exit(overo_soc_exit); + +MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>"); +MODULE_DESCRIPTION("ALSA SoC overo"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c new file mode 100644 index 000000000000..ad97836818b1 --- /dev/null +++ b/sound/soc/omap/sdp3430.c @@ -0,0 +1,152 @@ +/* + * sdp3430.c -- SoC audio for TI OMAP3430 SDP + * + * Author: Misael Lopez Cruz <x0052729@ti.com> + * + * Based on: + * Author: Steve Sakoman <steve@sakoman.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int sdp3430_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops sdp3430_ops = { + .hw_params = sdp3430_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sdp3430_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &sdp3430_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_sdp3430 = { + .name = "SDP3430", + .platform = &omap_soc_platform, + .dai_link = &sdp3430_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device sdp3430_snd_devdata = { + .machine = &snd_soc_machine_sdp3430, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *sdp3430_snd_device; + +static int __init sdp3430_soc_init(void) +{ + int ret; + + if (!machine_is_omap_3430sdp()) { + pr_debug("Not SDP3430!\n"); + return -ENODEV; + } + printk(KERN_INFO "SDP3430 SoC init\n"); + + sdp3430_snd_device = platform_device_alloc("soc-audio", -1); + if (!sdp3430_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata); + sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev; + *(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(sdp3430_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(sdp3430_snd_device); + + return ret; +} +module_init(sdp3430_soc_init); + +static void __exit sdp3430_soc_exit(void) +{ + platform_device_unregister(sdp3430_snd_device); +} +module_exit(sdp3430_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>"); +MODULE_DESCRIPTION("ALSA SoC SDP3430"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index f8c1cdd940ac..f82e10699471 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -21,6 +21,9 @@ config SND_PXA2XX_SOC_AC97 config SND_PXA2XX_SOC_I2S tristate +config SND_PXA_SOC_SSP + tristate + config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx @@ -75,3 +78,22 @@ config SND_PXA2XX_SOC_EM_X270 help Say Y if you want to add support for SoC audio on CompuLab EM-x270. + +config SND_PXA2XX_SOC_PALM27X + bool "SoC Audio support for Palm T|X, T5 and LifeDrive" + depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || MACH_PALMT5) + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9712 + help + Say Y if you want to add support for SoC audio on + Palm T|X, T5 or LifeDrive handheld computer. + +config SND_SOC_ZYLONITE + tristate "SoC Audio support for Marvell Zylonite" + depends on SND_PXA2XX_SOC && MACH_ZYLONITE + select SND_PXA2XX_SOC_AC97 + select SND_PXA_SOC_SSP + select SND_SOC_WM9713 + help + Say Y if you want to add support for SoC audio on the + Marvell Zylonite reference platform. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 5bc8edf9dca9..08a9f2797729 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -2,10 +2,12 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o +snd-soc-pxa-ssp-objs := pxa-ssp.o obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o +obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o # PXA Machine Support snd-soc-corgi-objs := corgi.o @@ -14,6 +16,8 @@ snd-soc-tosa-objs := tosa.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o +snd-soc-palm27x-objs := palm27x.o +snd-soc-zylonite-objs := zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -21,3 +25,5 @@ obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o +obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o +obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 2718eaf7895f..1ba25a559524 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -108,15 +108,11 @@ static int corgi_startup(struct snd_pcm_substream *substream) } /* we need to unmute the HP at shutdown as the mute burns power on corgi */ -static int corgi_shutdown(struct snd_pcm_substream *substream) +static void corgi_shutdown(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; - /* set = unmute headphone */ gpio_set_value(CORGI_GPIO_MUTE_L, 1); gpio_set_value(CORGI_GPIO_MUTE_R, 1); - return 0; } static int corgi_hw_params(struct snd_pcm_substream *substream, @@ -314,8 +310,9 @@ static struct snd_soc_dai_link corgi_dai = { }; /* corgi audio machine driver */ -static struct snd_soc_machine snd_soc_machine_corgi = { +static struct snd_soc_card snd_soc_corgi = { .name = "Corgi", + .platform = &pxa2xx_soc_platform, .dai_link = &corgi_dai, .num_links = 1, }; @@ -328,8 +325,7 @@ static struct wm8731_setup_data corgi_wm8731_setup = { /* corgi audio subsystem */ static struct snd_soc_device corgi_snd_devdata = { - .machine = &snd_soc_machine_corgi, - .platform = &pxa2xx_soc_platform, + .card = &snd_soc_corgi, .codec_dev = &soc_codec_dev_wm8731, .codec_data = &corgi_wm8731_setup, }; diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 6781c5be242f..2e3386dfa0f0 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -29,7 +29,7 @@ #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -static struct snd_soc_machine e800; +static struct snd_soc_card e800; static struct snd_soc_dai_link e800_dai[] = { { @@ -40,15 +40,15 @@ static struct snd_soc_dai_link e800_dai[] = { }, }; -static struct snd_soc_machine e800 = { +static struct snd_soc_card e800 = { .name = "Toshiba e800", + .platform = &pxa2xx_soc_platform, .dai_link = e800_dai, .num_links = ARRAY_SIZE(e800_dai), }; static struct snd_soc_device e800_snd_devdata = { - .machine = &e800, - .platform = &pxa2xx_soc_platform, + .card = &e800, .codec_dev = &soc_codec_dev_wm9712, }; diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index e6ff6929ab4b..fe4a729ea648 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -23,7 +23,6 @@ #include <linux/moduleparam.h> #include <linux/device.h> -#include <sound/driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -53,15 +52,15 @@ static struct snd_soc_dai_link em_x270_dai[] = { }, }; -static struct snd_soc_machine em_x270 = { +static struct snd_soc_card em_x270 = { .name = "EM-X270", + .platform = &pxa2xx_soc_platform, .dai_link = em_x270_dai, .num_links = ARRAY_SIZE(em_x270_dai), }; static struct snd_soc_device em_x270_snd_devdata = { - .machine = &em_x270, - .platform = &pxa2xx_soc_platform, + .card = &em_x270, .codec_dev = &soc_codec_dev_wm9712, }; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c new file mode 100644 index 000000000000..4a9cf3083af0 --- /dev/null +++ b/sound/soc/pxa/palm27x.c @@ -0,0 +1,269 @@ +/* + * linux/sound/soc/pxa/palm27x.c + * + * SoC Audio driver for Palm T|X, T5 and LifeDrive + * + * based on tosa.c + * + * Copyright (C) 2008 Marek Vasut <marek.vasut@gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/gpio.h> +#include <linux/interrupt.h> +#include <linux/irq.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/audio.h> +#include <mach/palmasoc.h> + +#include "../codecs/wm9712.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + +static int palm27x_jack_func = 1; +static int palm27x_spk_func = 1; +static int palm27x_ep_gpio = -1; + +static void palm27x_ext_control(struct snd_soc_codec *codec) +{ + if (!palm27x_spk_func) + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); + + if (!palm27x_jack_func) + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + snd_soc_dapm_sync(codec); +} + +static int palm27x_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->socdev->codec; + + /* check the jack status at stream startup */ + palm27x_ext_control(codec); + return 0; +} + +static struct snd_soc_ops palm27x_ops = { + .startup = palm27x_startup, +}; + +static irqreturn_t palm27x_interrupt(int irq, void *v) +{ + palm27x_spk_func = gpio_get_value(palm27x_ep_gpio); + palm27x_jack_func = !palm27x_spk_func; + return IRQ_HANDLED; +} + +static int palm27x_get_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = palm27x_jack_func; + return 0; +} + +static int palm27x_set_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (palm27x_jack_func == ucontrol->value.integer.value[0]) + return 0; + + palm27x_jack_func = ucontrol->value.integer.value[0]; + palm27x_ext_control(codec); + return 1; +} + +static int palm27x_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = palm27x_spk_func; + return 0; +} + +static int palm27x_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (palm27x_spk_func == ucontrol->value.integer.value[0]) + return 0; + + palm27x_spk_func = ucontrol->value.integer.value[0]; + palm27x_ext_control(codec); + return 1; +} + +/* PalmTX machine dapm widgets */ +static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +/* PalmTX audio map */ +static const struct snd_soc_dapm_route audio_map[] = { + /* headphone connected to HPOUTL, HPOUTR */ + {"Headphone Jack", NULL, "HPOUTL"}, + {"Headphone Jack", NULL, "HPOUTR"}, + + /* ext speaker connected to ROUT2, LOUT2 */ + {"Speaker", NULL, "LOUT2"}, + {"Speaker", NULL, "ROUT2"}, +}; + +static const char *jack_function[] = {"Headphone", "Off"}; +static const char *spk_function[] = {"On", "Off"}; +static const struct soc_enum palm27x_enum[] = { + SOC_ENUM_SINGLE_EXT(2, jack_function), + SOC_ENUM_SINGLE_EXT(2, spk_function), +}; + +static const struct snd_kcontrol_new palm27x_controls[] = { + SOC_ENUM_EXT("Jack Function", palm27x_enum[0], palm27x_get_jack, + palm27x_set_jack), + SOC_ENUM_EXT("Speaker Function", palm27x_enum[1], palm27x_get_spk, + palm27x_set_spk), +}; + +static int palm27x_ac97_init(struct snd_soc_codec *codec) +{ + int i, err; + + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "MONOOUT"); + + /* add palm27x specific controls */ + for (i = 0; i < ARRAY_SIZE(palm27x_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&palm27x_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + /* add palm27x specific widgets */ + snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets, + ARRAY_SIZE(palm27x_dapm_widgets)); + + /* set up palm27x specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + return 0; +} + +static struct snd_soc_dai_link palm27x_dai[] = { +{ + .name = "AC97 HiFi", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + .init = palm27x_ac97_init, + .ops = &palm27x_ops, +}, +{ + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], + .ops = &palm27x_ops, +}, +}; + +static struct snd_soc_card palm27x_asoc = { + .name = "Palm/PXA27x", + .platform = &pxa2xx_soc_platform, + .dai_link = palm27x_dai, + .num_links = ARRAY_SIZE(palm27x_dai), +}; + +static struct snd_soc_device palm27x_snd_devdata = { + .card = &palm27x_asoc, + .codec_dev = &soc_codec_dev_wm9712, +}; + +static struct platform_device *palm27x_snd_device; + +static int __init palm27x_asoc_init(void) +{ + int ret; + + if (!(machine_is_palmtx() || machine_is_palmt5() || + machine_is_palmld())) + return -ENODEV; + + ret = gpio_request(palm27x_ep_gpio, "Headphone Jack"); + if (ret) + return ret; + ret = gpio_direction_input(palm27x_ep_gpio); + if (ret) + goto err_alloc; + + if (request_irq(gpio_to_irq(palm27x_ep_gpio), palm27x_interrupt, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, + "Headphone jack", NULL)) + goto err_alloc; + + palm27x_snd_device = platform_device_alloc("soc-audio", -1); + if (!palm27x_snd_device) { + ret = -ENOMEM; + goto err_dev; + } + + platform_set_drvdata(palm27x_snd_device, &palm27x_snd_devdata); + palm27x_snd_devdata.dev = &palm27x_snd_device->dev; + ret = platform_device_add(palm27x_snd_device); + + if (ret != 0) + goto put_device; + + return 0; + +put_device: + platform_device_put(palm27x_snd_device); +err_dev: + free_irq(gpio_to_irq(palm27x_ep_gpio), NULL); +err_alloc: + gpio_free(palm27x_ep_gpio); + + return ret; +} + +static void __exit palm27x_asoc_exit(void) +{ + free_irq(gpio_to_irq(palm27x_ep_gpio), NULL); + gpio_free(palm27x_ep_gpio); + platform_device_unregister(palm27x_snd_device); +} + +void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data) +{ + palm27x_ep_gpio = data->jack_gpio; +} + +module_init(palm27x_asoc_init); +module_exit(palm27x_asoc_exit); + +/* Module information */ +MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 4d9930c52789..6e9827189fff 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -276,8 +276,9 @@ static struct snd_soc_dai_link poodle_dai = { }; /* poodle audio machine driver */ -static struct snd_soc_machine snd_soc_machine_poodle = { +static struct snd_soc_card snd_soc_poodle = { .name = "Poodle", + .platform = &pxa2xx_soc_platform, .dai_link = &poodle_dai, .num_links = 1, }; @@ -290,8 +291,7 @@ static struct wm8731_setup_data poodle_wm8731_setup = { /* poodle audio subsystem */ static struct snd_soc_device poodle_snd_devdata = { - .machine = &snd_soc_machine_poodle, - .platform = &pxa2xx_soc_platform, + .card = &snd_soc_poodle, .codec_dev = &soc_codec_dev_wm8731, .codec_data = &poodle_wm8731_setup, }; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c new file mode 100644 index 000000000000..73cb6b4c2f2d --- /dev/null +++ b/sound/soc/pxa/pxa-ssp.c @@ -0,0 +1,931 @@ +#define DEBUG +/* + * pxa-ssp.c -- ALSA Soc Audio Layer + * + * Copyright 2005,2008 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * TODO: + * o Test network mode for > 16bit sample size + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/pxa2xx-lib.h> + +#include <mach/hardware.h> +#include <mach/pxa-regs.h> +#include <mach/regs-ssp.h> +#include <mach/audio.h> +#include <mach/ssp.h> + +#include "pxa2xx-pcm.h" +#include "pxa-ssp.h" + +/* + * SSP audio private data + */ +struct ssp_priv { + struct ssp_dev dev; + unsigned int sysclk; + int dai_fmt; +#ifdef CONFIG_PM + struct ssp_state state; +#endif +}; + +#define PXA2xx_SSP1_BASE 0x41000000 +#define PXA27x_SSP2_BASE 0x41700000 +#define PXA27x_SSP3_BASE 0x41900000 +#define PXA3xx_SSP4_BASE 0x41a00000 + +static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_out = { + .name = "SSP1 PCM Mono out", + .dev_addr = PXA2xx_SSP1_BASE + SSDR, + .drcmr = &DRCMR(14), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_in = { + .name = "SSP1 PCM Mono in", + .dev_addr = PXA2xx_SSP1_BASE + SSDR, + .drcmr = &DRCMR(13), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_out = { + .name = "SSP1 PCM Stereo out", + .dev_addr = PXA2xx_SSP1_BASE + SSDR, + .drcmr = &DRCMR(14), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_in = { + .name = "SSP1 PCM Stereo in", + .dev_addr = PXA2xx_SSP1_BASE + SSDR, + .drcmr = &DRCMR(13), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_out = { + .name = "SSP2 PCM Mono out", + .dev_addr = PXA27x_SSP2_BASE + SSDR, + .drcmr = &DRCMR(16), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_in = { + .name = "SSP2 PCM Mono in", + .dev_addr = PXA27x_SSP2_BASE + SSDR, + .drcmr = &DRCMR(15), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_out = { + .name = "SSP2 PCM Stereo out", + .dev_addr = PXA27x_SSP2_BASE + SSDR, + .drcmr = &DRCMR(16), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_in = { + .name = "SSP2 PCM Stereo in", + .dev_addr = PXA27x_SSP2_BASE + SSDR, + .drcmr = &DRCMR(15), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_out = { + .name = "SSP3 PCM Mono out", + .dev_addr = PXA27x_SSP3_BASE + SSDR, + .drcmr = &DRCMR(67), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_in = { + .name = "SSP3 PCM Mono in", + .dev_addr = PXA27x_SSP3_BASE + SSDR, + .drcmr = &DRCMR(66), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_out = { + .name = "SSP3 PCM Stereo out", + .dev_addr = PXA27x_SSP3_BASE + SSDR, + .drcmr = &DRCMR(67), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_in = { + .name = "SSP3 PCM Stereo in", + .dev_addr = PXA27x_SSP3_BASE + SSDR, + .drcmr = &DRCMR(66), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_out = { + .name = "SSP4 PCM Mono out", + .dev_addr = PXA3xx_SSP4_BASE + SSDR, + .drcmr = &DRCMR(67), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_in = { + .name = "SSP4 PCM Mono in", + .dev_addr = PXA3xx_SSP4_BASE + SSDR, + .drcmr = &DRCMR(66), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_out = { + .name = "SSP4 PCM Stereo out", + .dev_addr = PXA3xx_SSP4_BASE + SSDR, + .drcmr = &DRCMR(67), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_in = { + .name = "SSP4 PCM Stereo in", + .dev_addr = PXA3xx_SSP4_BASE + SSDR, + .drcmr = &DRCMR(66), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static void dump_registers(struct ssp_device *ssp) +{ + dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", + ssp_read_reg(ssp, SSCR0), ssp_read_reg(ssp, SSCR1), + ssp_read_reg(ssp, SSTO)); + + dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n", + ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR), + ssp_read_reg(ssp, SSACD)); +} + +static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = { + { + &pxa_ssp1_pcm_mono_out, &pxa_ssp1_pcm_mono_in, + &pxa_ssp1_pcm_stereo_out, &pxa_ssp1_pcm_stereo_in, + }, + { + &pxa_ssp2_pcm_mono_out, &pxa_ssp2_pcm_mono_in, + &pxa_ssp2_pcm_stereo_out, &pxa_ssp2_pcm_stereo_in, + }, + { + &pxa_ssp3_pcm_mono_out, &pxa_ssp3_pcm_mono_in, + &pxa_ssp3_pcm_stereo_out, &pxa_ssp3_pcm_stereo_in, + }, + { + &pxa_ssp4_pcm_mono_out, &pxa_ssp4_pcm_mono_in, + &pxa_ssp4_pcm_stereo_out, &pxa_ssp4_pcm_stereo_in, + }, +}; + +static int pxa_ssp_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_priv *priv = cpu_dai->private_data; + int ret = 0; + + if (!cpu_dai->active) { + ret = ssp_init(&priv->dev, cpu_dai->id + 1, SSP_NO_IRQ); + if (ret < 0) + return ret; + ssp_disable(&priv->dev); + } + return ret; +} + +static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_priv *priv = cpu_dai->private_data; + + if (!cpu_dai->active) { + ssp_disable(&priv->dev); + ssp_exit(&priv->dev); + } +} + +#ifdef CONFIG_PM + +static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) +{ + struct ssp_priv *priv = cpu_dai->private_data; + + if (!cpu_dai->active) + return 0; + + ssp_save_state(&priv->dev, &priv->state); + clk_disable(priv->dev.ssp->clk); + return 0; +} + +static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) +{ + struct ssp_priv *priv = cpu_dai->private_data; + + if (!cpu_dai->active) + return 0; + + clk_enable(priv->dev.ssp->clk); + ssp_restore_state(&priv->dev, &priv->state); + ssp_enable(&priv->dev); + + return 0; +} + +#else +#define pxa_ssp_suspend NULL +#define pxa_ssp_resume NULL +#endif + +/** + * ssp_set_clkdiv - set SSP clock divider + * @div: serial clock rate divider + */ +static void ssp_set_scr(struct ssp_dev *dev, u32 div) +{ + struct ssp_device *ssp = dev->ssp; + u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR; + + ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div))); +} + +/* + * Set the SSP ports SYSCLK. + */ +static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + int val; + + u32 sscr0 = ssp_read_reg(ssp, SSCR0) & + ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC); + + dev_dbg(&ssp->pdev->dev, + "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n", + cpu_dai->id, clk_id, freq); + + switch (clk_id) { + case PXA_SSP_CLK_NET_PLL: + sscr0 |= SSCR0_MOD; + break; + case PXA_SSP_CLK_PLL: + /* Internal PLL is fixed */ + if (cpu_is_pxa25x()) + priv->sysclk = 1843200; + else + priv->sysclk = 13000000; + break; + case PXA_SSP_CLK_EXT: + priv->sysclk = freq; + sscr0 |= SSCR0_ECS; + break; + case PXA_SSP_CLK_NET: + priv->sysclk = freq; + sscr0 |= SSCR0_NCS | SSCR0_MOD; + break; + case PXA_SSP_CLK_AUDIO: + priv->sysclk = 0; + ssp_set_scr(&priv->dev, 1); + sscr0 |= SSCR0_ADC; + break; + default: + return -ENODEV; + } + + /* The SSP clock must be disabled when changing SSP clock mode + * on PXA2xx. On PXA3xx it must be enabled when doing so. */ + if (!cpu_is_pxa3xx()) + clk_disable(priv->dev.ssp->clk); + val = ssp_read_reg(ssp, SSCR0) | sscr0; + ssp_write_reg(ssp, SSCR0, val); + if (!cpu_is_pxa3xx()) + clk_enable(priv->dev.ssp->clk); + + return 0; +} + +/* + * Set the SSP clock dividers. + */ +static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + int val; + + switch (div_id) { + case PXA_SSP_AUDIO_DIV_ACDS: + val = (ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); + ssp_write_reg(ssp, SSACD, val); + break; + case PXA_SSP_AUDIO_DIV_SCDB: + val = ssp_read_reg(ssp, SSACD); + val &= ~SSACD_SCDB; +#if defined(CONFIG_PXA3xx) + if (cpu_is_pxa3xx()) + val &= ~SSACD_SCDX8; +#endif + switch (div) { + case PXA_SSP_CLK_SCDB_1: + val |= SSACD_SCDB; + break; + case PXA_SSP_CLK_SCDB_4: + break; +#if defined(CONFIG_PXA3xx) + case PXA_SSP_CLK_SCDB_8: + if (cpu_is_pxa3xx()) + val |= SSACD_SCDX8; + else + return -EINVAL; + break; +#endif + default: + return -EINVAL; + } + ssp_write_reg(ssp, SSACD, val); + break; + case PXA_SSP_DIV_SCR: + ssp_set_scr(&priv->dev, div); + break; + default: + return -ENODEV; + } + + return 0; +} + +/* + * Configure the PLL frequency pxa27x and (afaik - pxa320 only) + */ +static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70; + +#if defined(CONFIG_PXA3xx) + if (cpu_is_pxa3xx()) + ssp_write_reg(ssp, SSACDD, 0); +#endif + + switch (freq_out) { + case 5622000: + break; + case 11345000: + ssacd |= (0x1 << 4); + break; + case 12235000: + ssacd |= (0x2 << 4); + break; + case 14857000: + ssacd |= (0x3 << 4); + break; + case 32842000: + ssacd |= (0x4 << 4); + break; + case 48000000: + ssacd |= (0x5 << 4); + break; + case 0: + /* Disable */ + break; + + default: +#ifdef CONFIG_PXA3xx + /* PXA3xx has a clock ditherer which can be used to generate + * a wider range of frequencies - calculate a value for it. + */ + if (cpu_is_pxa3xx()) { + u32 val; + u64 tmp = 19968; + tmp *= 1000000; + do_div(tmp, freq_out); + val = tmp; + + val = (val << 16) | 64;; + ssp_write_reg(ssp, SSACDD, val); + + ssacd |= (0x6 << 4); + + dev_dbg(&ssp->pdev->dev, + "Using SSACDD %x to supply %dHz\n", + val, freq_out); + break; + } +#endif + + return -EINVAL; + } + + ssp_write_reg(ssp, SSACD, ssacd); + + return 0; +} + +/* + * Set the active slots in TDM/Network mode + */ +static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, + unsigned int mask, int slots) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + u32 sscr0; + + sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); + + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + ssp_write_reg(ssp, SSCR0, sscr0); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, mask); + ssp_write_reg(ssp, SSRSA, mask); + return 0; +} + +/* + * Tristate the SSP DAI lines + */ +static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, + int tristate) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + u32 sscr1; + + sscr1 = ssp_read_reg(ssp, SSCR1); + if (tristate) + sscr1 &= ~SSCR1_TTE; + else + sscr1 |= SSCR1_TTE; + ssp_write_reg(ssp, SSCR1, sscr1); + + return 0; +} + +/* + * Set up the SSP DAI format. + * The SSP Port must be inactive before calling this function as the + * physical interface format is changed. + */ +static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + u32 sscr0; + u32 sscr1; + u32 sspsp; + + /* reset port settings */ + sscr0 = ssp_read_reg(ssp, SSCR0) & + (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC); + sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); + sspsp = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + sscr1 |= SSCR1_SCLKDIR; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + ssp_write_reg(ssp, SSCR0, sscr0); + ssp_write_reg(ssp, SSCR1, sscr1); + ssp_write_reg(ssp, SSPSP, sspsp); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspsp |= SSPSP_FSRT; + break; + case SND_SOC_DAIFMT_NB_IF: + sspsp |= SSPSP_SFRMP | SSPSP_FSRT; + break; + case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SFRMP; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_DSP_A: + sspsp |= SSPSP_FSRT; + case SND_SOC_DAIFMT_DSP_B: + sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspsp |= SSPSP_SFRMP; + break; + case SND_SOC_DAIFMT_IB_IF: + break; + default: + return -EINVAL; + } + break; + + default: + return -EINVAL; + } + + ssp_write_reg(ssp, SSCR0, sscr0); + ssp_write_reg(ssp, SSCR1, sscr1); + ssp_write_reg(ssp, SSPSP, sspsp); + + dump_registers(ssp); + + /* Since we are configuring the timings for the format by hand + * we have to defer some things until hw_params() where we + * know parameters like the sample size. + */ + priv->dai_fmt = fmt; + + return 0; +} + +/* + * Set the SSP audio DMA parameters and sample size. + * Can be called multiple times by oss emulation. + */ +static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + int dma = 0, chn = params_channels(params); + u32 sscr0; + u32 sspsp; + int width = snd_pcm_format_physical_width(params_format(params)); + + /* select correct DMA params */ + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + dma = 1; /* capture DMA offset is 1,3 */ + if (chn == 2) + dma += 2; /* stereo DMA offset is 2, mono is 0 */ + cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma]; + + dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma); + + /* we can only change the settings if the port is not in use */ + if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + return 0; + + /* clear selected SSP bits */ + sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); + ssp_write_reg(ssp, SSCR0, sscr0); + + /* bit size */ + sscr0 = ssp_read_reg(ssp, SSCR0); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: +#ifdef CONFIG_PXA3xx + if (cpu_is_pxa3xx()) + sscr0 |= SSCR0_FPCKE; +#endif + sscr0 |= SSCR0_DataSize(16); + if (params_channels(params) > 1) + sscr0 |= SSCR0_EDSS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); + /* we must be in network mode (2 slots) for 24 bit stereo */ + break; + case SNDRV_PCM_FORMAT_S32_LE: + sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); + /* we must be in network mode (2 slots) for 32 bit stereo */ + break; + } + ssp_write_reg(ssp, SSCR0, sscr0); + + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* Cleared when the DAI format is set */ + sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width); + ssp_write_reg(ssp, SSPSP, sspsp); + break; + default: + break; + } + + /* We always use a network mode so we always require TDM slots + * - complain loudly and fail if they've not been set up yet. + */ + if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) { + dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); + return -EINVAL; + } + + dump_registers(ssp); + + return 0; +} + +static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + int val; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + ssp_enable(&priv->dev); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= SSCR1_TSRE; + else + val |= SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + val = ssp_read_reg(ssp, SSSR); + ssp_write_reg(ssp, SSSR, val); + break; + case SNDRV_PCM_TRIGGER_START: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= SSCR1_TSRE; + else + val |= SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + ssp_enable(&priv->dev); + break; + case SNDRV_PCM_TRIGGER_STOP: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val &= ~SSCR1_TSRE; + else + val &= ~SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + ssp_disable(&priv->dev); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val &= ~SSCR1_TSRE; + else + val &= ~SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + break; + + default: + ret = -EINVAL; + } + + dump_registers(ssp); + + return ret; +} + +static int pxa_ssp_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct ssp_priv *priv; + int ret; + + priv = kzalloc(sizeof(struct ssp_priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->dev.ssp = ssp_request(dai->id, "SoC audio"); + if (priv->dev.ssp == NULL) { + ret = -ENODEV; + goto err_priv; + } + + dai->private_data = priv; + + return 0; + +err_priv: + kfree(priv); + return ret; +} + +static void pxa_ssp_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct ssp_priv *priv = dai->private_data; + ssp_free(priv->dev.ssp); +} + +#define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai pxa_ssp_dai[] = { + { + .name = "pxa2xx-ssp1", + .id = 0, + .probe = pxa_ssp_probe, + .remove = pxa_ssp_remove, + .suspend = pxa_ssp_suspend, + .resume = pxa_ssp_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .ops = { + .startup = pxa_ssp_startup, + .shutdown = pxa_ssp_shutdown, + .trigger = pxa_ssp_trigger, + .hw_params = pxa_ssp_hw_params, + .set_sysclk = pxa_ssp_set_dai_sysclk, + .set_clkdiv = pxa_ssp_set_dai_clkdiv, + .set_pll = pxa_ssp_set_dai_pll, + .set_fmt = pxa_ssp_set_dai_fmt, + .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, + .set_tristate = pxa_ssp_set_dai_tristate, + }, + }, + { .name = "pxa2xx-ssp2", + .id = 1, + .probe = pxa_ssp_probe, + .remove = pxa_ssp_remove, + .suspend = pxa_ssp_suspend, + .resume = pxa_ssp_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .ops = { + .startup = pxa_ssp_startup, + .shutdown = pxa_ssp_shutdown, + .trigger = pxa_ssp_trigger, + .hw_params = pxa_ssp_hw_params, + .set_sysclk = pxa_ssp_set_dai_sysclk, + .set_clkdiv = pxa_ssp_set_dai_clkdiv, + .set_pll = pxa_ssp_set_dai_pll, + .set_fmt = pxa_ssp_set_dai_fmt, + .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, + .set_tristate = pxa_ssp_set_dai_tristate, + }, + }, + { + .name = "pxa2xx-ssp3", + .id = 2, + .probe = pxa_ssp_probe, + .remove = pxa_ssp_remove, + .suspend = pxa_ssp_suspend, + .resume = pxa_ssp_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .ops = { + .startup = pxa_ssp_startup, + .shutdown = pxa_ssp_shutdown, + .trigger = pxa_ssp_trigger, + .hw_params = pxa_ssp_hw_params, + .set_sysclk = pxa_ssp_set_dai_sysclk, + .set_clkdiv = pxa_ssp_set_dai_clkdiv, + .set_pll = pxa_ssp_set_dai_pll, + .set_fmt = pxa_ssp_set_dai_fmt, + .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, + .set_tristate = pxa_ssp_set_dai_tristate, + }, + }, + { + .name = "pxa2xx-ssp4", + .id = 3, + .probe = pxa_ssp_probe, + .remove = pxa_ssp_remove, + .suspend = pxa_ssp_suspend, + .resume = pxa_ssp_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .ops = { + .startup = pxa_ssp_startup, + .shutdown = pxa_ssp_shutdown, + .trigger = pxa_ssp_trigger, + .hw_params = pxa_ssp_hw_params, + .set_sysclk = pxa_ssp_set_dai_sysclk, + .set_clkdiv = pxa_ssp_set_dai_clkdiv, + .set_pll = pxa_ssp_set_dai_pll, + .set_fmt = pxa_ssp_set_dai_fmt, + .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, + .set_tristate = pxa_ssp_set_dai_tristate, + }, + }, +}; +EXPORT_SYMBOL_GPL(pxa_ssp_dai); + +static int __init pxa_ssp_init(void) +{ + return snd_soc_register_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai)); +} +module_init(pxa_ssp_init); + +static void __exit pxa_ssp_exit(void) +{ + snd_soc_unregister_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai)); +} +module_exit(pxa_ssp_exit); + +/* Module information */ +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h new file mode 100644 index 000000000000..91deadd55675 --- /dev/null +++ b/sound/soc/pxa/pxa-ssp.h @@ -0,0 +1,47 @@ +/* + * ASoC PXA SSP port support + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _PXA_SSP_H +#define _PXA_SSP_H + +/* pxa DAI SSP IDs */ +#define PXA_DAI_SSP1 0 +#define PXA_DAI_SSP2 1 +#define PXA_DAI_SSP3 2 +#define PXA_DAI_SSP4 3 + +/* SSP clock sources */ +#define PXA_SSP_CLK_PLL 0 +#define PXA_SSP_CLK_EXT 1 +#define PXA_SSP_CLK_NET 2 +#define PXA_SSP_CLK_AUDIO 3 +#define PXA_SSP_CLK_NET_PLL 4 + +/* SSP audio dividers */ +#define PXA_SSP_AUDIO_DIV_ACDS 0 +#define PXA_SSP_AUDIO_DIV_SCDB 1 +#define PXA_SSP_DIV_SCR 2 + +/* SSP ACDS audio dividers values */ +#define PXA_SSP_CLK_AUDIO_DIV_1 0 +#define PXA_SSP_CLK_AUDIO_DIV_2 1 +#define PXA_SSP_CLK_AUDIO_DIV_4 2 +#define PXA_SSP_CLK_AUDIO_DIV_8 3 +#define PXA_SSP_CLK_AUDIO_DIV_16 4 +#define PXA_SSP_CLK_AUDIO_DIV_32 5 + +/* SSP divider bypass */ +#define PXA_SSP_CLK_SCDB_4 0 +#define PXA_SSP_CLK_SCDB_1 1 +#define PXA_SSP_CLK_SCDB_8 2 + +#define PXA_SSP_PLL_OUT 0 + +extern struct snd_soc_dai pxa_ssp_dai[4]; + +#endif diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index a7a3a9c5c6ff..780db6757ad2 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -87,14 +87,12 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = { }; #ifdef CONFIG_PM -static int pxa2xx_ac97_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int pxa2xx_ac97_suspend(struct snd_soc_dai *dai) { return pxa2xx_ac97_hw_suspend(); } -static int pxa2xx_ac97_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int pxa2xx_ac97_resume(struct snd_soc_dai *dai) { return pxa2xx_ac97_hw_resume(); } @@ -117,7 +115,8 @@ static void pxa2xx_ac97_remove(struct platform_device *pdev, } static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -131,7 +130,8 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, } static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -145,7 +145,8 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, } static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -170,7 +171,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { { .name = "pxa2xx-ac97", .id = 0, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .probe = pxa2xx_ac97_probe, .remove = pxa2xx_ac97_remove, .suspend = pxa2xx_ac97_suspend, @@ -193,7 +194,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { { .name = "pxa2xx-ac97-aux", .id = 1, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .playback = { .stream_name = "AC97 Aux Playback", .channels_min = 1, @@ -212,7 +213,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { { .name = "pxa2xx-ac97-mic", .id = 2, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .capture = { .stream_name = "AC97 Mic Capture", .channels_min = 1, @@ -227,6 +228,18 @@ struct snd_soc_dai pxa_ac97_dai[] = { EXPORT_SYMBOL_GPL(pxa_ac97_dai); EXPORT_SYMBOL_GPL(soc_ac97_ops); +static int __init pxa_ac97_init(void) +{ + return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai)); +} +module_init(pxa_ac97_init); + +static void __exit pxa_ac97_exit(void) +{ + snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai)); +} +module_exit(pxa_ac97_exit); + MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index e758034db5c3..517991fb1099 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -121,7 +121,8 @@ static struct pxa2xx_gpio gpio_bus[] = { }, }; -static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream) +static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -187,7 +188,8 @@ static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai, } static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -248,7 +250,8 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { int ret = 0; @@ -269,7 +272,8 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } -static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream) +static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { SACR1 |= SACR1_DRPL; @@ -289,8 +293,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream) } #ifdef CONFIG_PM -static int pxa2xx_i2s_suspend(struct platform_device *dev, - struct snd_soc_dai *dai) +static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) { if (!dai->active) return 0; @@ -307,8 +310,7 @@ static int pxa2xx_i2s_suspend(struct platform_device *dev, return 0; } -static int pxa2xx_i2s_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int pxa2xx_i2s_resume(struct snd_soc_dai *dai) { if (!dai->active) return 0; @@ -336,7 +338,6 @@ static int pxa2xx_i2s_resume(struct platform_device *pdev, struct snd_soc_dai pxa_i2s_dai = { .name = "pxa2xx-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .suspend = pxa2xx_i2s_suspend, .resume = pxa2xx_i2s_resume, .playback = { @@ -353,8 +354,7 @@ struct snd_soc_dai pxa_i2s_dai = { .startup = pxa2xx_i2s_startup, .shutdown = pxa2xx_i2s_shutdown, .trigger = pxa2xx_i2s_trigger, - .hw_params = pxa2xx_i2s_hw_params,}, - .dai_ops = { + .hw_params = pxa2xx_i2s_hw_params, .set_fmt = pxa2xx_i2s_set_dai_fmt, .set_sysclk = pxa2xx_i2s_set_dai_sysclk, }, @@ -364,12 +364,23 @@ EXPORT_SYMBOL_GPL(pxa_i2s_dai); static int pxa2xx_i2s_probe(struct platform_device *dev) { + int ret; + clk_i2s = clk_get(&dev->dev, "I2SCLK"); - return IS_ERR(clk_i2s) ? PTR_ERR(clk_i2s) : 0; + if (IS_ERR(clk_i2s)) + return PTR_ERR(clk_i2s); + + pxa_i2s_dai.dev = &dev->dev; + ret = snd_soc_register_dai(&pxa_i2s_dai); + if (ret != 0) + clk_put(clk_i2s); + + return ret; } static int __devexit pxa2xx_i2s_remove(struct platform_device *dev) { + snd_soc_unregister_dai(&pxa_i2s_dai); clk_put(clk_i2s); clk_i2s = ERR_PTR(-ENOENT); return 0; diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index afcd892cd2fa..c670d08e7c9e 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -69,7 +69,7 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -struct snd_pcm_ops pxa2xx_pcm_ops = { +static struct snd_pcm_ops pxa2xx_pcm_ops = { .open = __pxa2xx_pcm_open, .close = __pxa2xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, @@ -118,6 +118,18 @@ struct snd_soc_platform pxa2xx_soc_platform = { }; EXPORT_SYMBOL_GPL(pxa2xx_soc_platform); +static int __init pxa2xx_soc_platform_init(void) +{ + return snd_soc_register_platform(&pxa2xx_soc_platform); +} +module_init(pxa2xx_soc_platform_init); + +static void __exit pxa2xx_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&pxa2xx_soc_platform); +} +module_exit(pxa2xx_soc_platform_exit); + MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index d307b6757e95..a3b9e6bdf979 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -319,8 +319,9 @@ static struct snd_soc_dai_link spitz_dai = { }; /* spitz audio machine driver */ -static struct snd_soc_machine snd_soc_machine_spitz = { +static struct snd_soc_card snd_soc_spitz = { .name = "Spitz", + .platform = &pxa2xx_soc_platform, .dai_link = &spitz_dai, .num_links = 1, }; @@ -333,8 +334,7 @@ static struct wm8750_setup_data spitz_wm8750_setup = { /* spitz audio subsystem */ static struct snd_soc_device spitz_snd_devdata = { - .machine = &snd_soc_machine_spitz, - .platform = &pxa2xx_soc_platform, + .card = &snd_soc_spitz, .codec_dev = &soc_codec_dev_wm8750, .codec_data = &spitz_wm8750_setup, }; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index afefe41b8c46..c77194f74c9b 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -38,7 +38,7 @@ #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -static struct snd_soc_machine tosa; +static struct snd_soc_card tosa; #define TOSA_HP 0 #define TOSA_MIC_INT 1 @@ -230,15 +230,37 @@ static struct snd_soc_dai_link tosa_dai[] = { }, }; -static struct snd_soc_machine tosa = { +static int tosa_probe(struct platform_device *dev) +{ + int ret; + + ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack"); + if (ret) + return ret; + ret = gpio_direction_output(TOSA_GPIO_L_MUTE, 0); + if (ret) + gpio_free(TOSA_GPIO_L_MUTE); + + return ret; +} + +static int tosa_remove(struct platform_device *dev) +{ + gpio_free(TOSA_GPIO_L_MUTE); + return 0; +} + +static struct snd_soc_card tosa = { .name = "Tosa", + .platform = &pxa2xx_soc_platform, .dai_link = tosa_dai, .num_links = ARRAY_SIZE(tosa_dai), + .probe = tosa_probe, + .remove = tosa_remove, }; static struct snd_soc_device tosa_snd_devdata = { - .machine = &tosa, - .platform = &pxa2xx_soc_platform, + .card = &tosa, .codec_dev = &soc_codec_dev_wm9712, }; @@ -251,11 +273,6 @@ static int __init tosa_init(void) if (!machine_is_tosa()) return -ENODEV; - ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack"); - if (ret) - return ret; - gpio_direction_output(TOSA_GPIO_L_MUTE, 0); - tosa_snd_device = platform_device_alloc("soc-audio", -1); if (!tosa_snd_device) { ret = -ENOMEM; @@ -272,15 +289,12 @@ static int __init tosa_init(void) platform_device_put(tosa_snd_device); err_alloc: - gpio_free(TOSA_GPIO_L_MUTE); - return ret; } static void __exit tosa_exit(void) { platform_device_unregister(tosa_snd_device); - gpio_free(TOSA_GPIO_L_MUTE); } module_init(tosa_init); diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c new file mode 100644 index 000000000000..f8e9ecd589d3 --- /dev/null +++ b/sound/soc/pxa/zylonite.c @@ -0,0 +1,219 @@ +/* + * zylonite.c -- SoC audio for Zylonite + * + * Copyright 2008 Wolfson Microelectronics PLC. + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation; either version 2 of the + * License, or (at your option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm9713.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" +#include "pxa-ssp.h" + +static struct snd_soc_card zylonite; + +static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Microphone", NULL), + SND_SOC_DAPM_MIC("Handset Microphone", NULL), + SND_SOC_DAPM_SPK("Multiactor", NULL), + SND_SOC_DAPM_SPK("Headset Earpiece", NULL), +}; + +/* Currently supported audio map */ +static const struct snd_soc_dapm_route audio_map[] = { + + /* Headphone output connected to HPL/HPR */ + { "Headphone", NULL, "HPL" }, + { "Headphone", NULL, "HPR" }, + + /* On-board earpiece */ + { "Headset Earpiece", NULL, "OUT3" }, + + /* Headphone mic */ + { "MIC2A", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "Headset Microphone" }, + + /* On-board mic */ + { "MIC1", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "Handset Microphone" }, + + /* Multiactor differentially connected over SPKL/SPKR */ + { "Multiactor", NULL, "SPKL" }, + { "Multiactor", NULL, "SPKR" }, +}; + +static int zylonite_wm9713_init(struct snd_soc_codec *codec) +{ + /* Currently we only support use of the AC97 clock here. If + * CLK_POUT is selected by SW15 then the clock API will need + * to be used to request and enable it here. + */ + + snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets, + ARRAY_SIZE(zylonite_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* Static setup for now */ + snd_soc_dapm_enable_pin(codec, "Headphone"); + snd_soc_dapm_enable_pin(codec, "Headset Earpiece"); + + snd_soc_dapm_sync(codec); + return 0; +} + +static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 0; + unsigned int acds = 0; + unsigned int wm9713_div = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + wm9713_div = 12; + pll_out = 2048000; + break; + case 16000: + wm9713_div = 6; + pll_out = 4096000; + break; + case 48000: + default: + wm9713_div = 2; + pll_out = 12288000; + acds = 1; + break; + } + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, + params_channels(params), + params_channels(params)); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); + if (ret < 0) + return ret; + + /* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs + * to be set instead. + */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV, + WM9713_PCMDIV(wm9713_div)); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops zylonite_voice_ops = { + .hw_params = zylonite_voice_hw_params, +}; + +static struct snd_soc_dai_link zylonite_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], + .init = zylonite_wm9713_init, +}, +{ + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX], +}, +{ + .name = "WM9713 Voice", + .stream_name = "WM9713 Voice", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP3], + .codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE], + .ops = &zylonite_voice_ops, +}, +}; + +static struct snd_soc_card zylonite = { + .name = "Zylonite", + .platform = &pxa2xx_soc_platform, + .dai_link = zylonite_dai, + .num_links = ARRAY_SIZE(zylonite_dai), +}; + +static struct snd_soc_device zylonite_snd_ac97_devdata = { + .card = &zylonite, + .codec_dev = &soc_codec_dev_wm9713, +}; + +static struct platform_device *zylonite_snd_ac97_device; + +static int __init zylonite_init(void) +{ + int ret; + + zylonite_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!zylonite_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(zylonite_snd_ac97_device, + &zylonite_snd_ac97_devdata); + zylonite_snd_ac97_devdata.dev = &zylonite_snd_ac97_device->dev; + + ret = platform_device_add(zylonite_snd_ac97_device); + if (ret != 0) + platform_device_put(zylonite_snd_ac97_device); + + return ret; +} + +static void __exit zylonite_exit(void) +{ + platform_device_unregister(zylonite_snd_ac97_device); +} + +module_init(zylonite_init); +module_exit(zylonite_exit); + +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("ALSA SoC WM9713 Zylonite"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index b9f2353effeb..fcd03acf10f6 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -44,3 +44,8 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650 Say Y if you want to add support for SoC audio on ln2440sbc with the ALC650. +config SND_S3C24XX_SOC_S3C24XX_UDA134X + tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" + depends on SND_S3C24XX_SOC + select SND_S3C24XX_SOC_I2S + select SND_SOC_UDA134X diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 0aa5fb0b9700..96b3f3f617d4 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -13,7 +13,9 @@ obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o +snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o +obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c index 4eab2c19c454..12c71482d258 100644 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -27,7 +27,7 @@ #include "s3c24xx-pcm.h" #include "s3c24xx-ac97.h" -static struct snd_soc_machine ln2440sbc; +static struct snd_soc_card ln2440sbc; static struct snd_soc_dai_link ln2440sbc_dai[] = { { @@ -38,15 +38,15 @@ static struct snd_soc_dai_link ln2440sbc_dai[] = { }, }; -static struct snd_soc_machine ln2440sbc = { +static struct snd_soc_card ln2440sbc = { .name = "LN2440SBC", + .platform = &s3c24xx_soc_platform, .dai_link = ln2440sbc_dai, .num_links = ARRAY_SIZE(ln2440sbc_dai), }; static struct snd_soc_device ln2440sbc_snd_ac97_devdata = { - .machine = &ln2440sbc, - .platform = &s3c24xx_soc_platform, + .card = &ln2440sbc, .codec_dev = &soc_codec_dev_ac97, }; diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 87ddfefcc2fb..45bb12e8ea44 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -59,7 +59,7 @@ #define NEO_CAPTURE_HEADSET 7 #define NEO_CAPTURE_BLUETOOTH 8 -static struct snd_soc_machine neo1973; +static struct snd_soc_card neo1973; static struct i2c_client *i2c; static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, @@ -548,7 +548,6 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) static struct snd_soc_dai bt_dai = { .name = "Bluetooth", .id = 0, - .type = SND_SOC_DAI_PCM, .playback = { .channels_min = 1, .channels_max = 1, @@ -579,8 +578,9 @@ static struct snd_soc_dai_link neo1973_dai[] = { }, }; -static struct snd_soc_machine neo1973 = { +static struct snd_soc_card neo1973 = { .name = "neo1973", + .platform = &s3c24xx_soc_platform, .dai_link = neo1973_dai, .num_links = ARRAY_SIZE(neo1973_dai), }; @@ -591,8 +591,7 @@ static struct wm8753_setup_data neo1973_wm8753_setup = { }; static struct snd_soc_device neo1973_snd_devdata = { - .machine = &neo1973, - .platform = &s3c24xx_soc_platform, + .card = &neo1973, .codec_dev = &soc_codec_dev_wm8753, .codec_data = &neo1973_wm8753_setup, }; diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index ded7d995a922..f3fc0aba0aaf 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -343,7 +343,8 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, } static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; u32 iismod; @@ -373,7 +374,8 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; @@ -647,8 +649,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev, } #ifdef CONFIG_PM -static int s3c2412_i2s_suspend(struct platform_device *dev, - struct snd_soc_dai *dai) +static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; u32 iismod; @@ -663,25 +664,24 @@ static int s3c2412_i2s_suspend(struct platform_device *dev, iismod = readl(i2s->regs + S3C2412_IISMOD); if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) - dev_warn(&dev->dev, "%s: RXDMA active?\n", __func__); + pr_warning("%s: RXDMA active?\n", __func__); if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) - dev_warn(&dev->dev, "%s: TXDMA active?\n", __func__); + pr_warning("%s: TXDMA active?\n", __func__); if (iismod & S3C2412_IISCON_IIS_ACTIVE) - dev_warn(&dev->dev, "%s: IIS active\n", __func__); + pr_warning("%s: IIS active\n", __func__); } return 0; } -static int s3c2412_i2s_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int s3c2412_i2s_resume(struct snd_soc_dai *dai) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; - dev_info(&pdev->dev, "dai_active %d, IISMOD %08x, IISCON %08x\n", - dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); + pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", + dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); if (dai->active) { writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); @@ -711,7 +711,6 @@ static int s3c2412_i2s_resume(struct platform_device *pdev, struct snd_soc_dai s3c2412_i2s_dai = { .name = "s3c2412-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .probe = s3c2412_i2s_probe, .suspend = s3c2412_i2s_suspend, .resume = s3c2412_i2s_resume, @@ -730,8 +729,6 @@ struct snd_soc_dai s3c2412_i2s_dai = { .ops = { .trigger = s3c2412_i2s_trigger, .hw_params = s3c2412_i2s_hw_params, - }, - .dai_ops = { .set_fmt = s3c2412_i2s_set_fmt, .set_clkdiv = s3c2412_i2s_set_clkdiv, .set_sysclk = s3c2412_i2s_set_sysclk, @@ -739,6 +736,19 @@ struct snd_soc_dai s3c2412_i2s_dai = { }; EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); +static int __init s3c2412_i2s_init(void) +{ + return snd_soc_register_dai(&s3c2412_i2s_dai); +} +module_init(s3c2412_i2s_init); + +static void __exit s3c2412_i2s_exit(void) +{ + snd_soc_unregister_dai(&s3c2412_i2s_dai); +} +module_exit(s3c2412_i2s_exit); + + /* Module information */ MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>"); MODULE_DESCRIPTION("S3C2412 I2S SoC Interface"); diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 19c5c3cf5d8c..1bfce40bb2e4 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -271,7 +271,8 @@ static void s3c2443_ac97_remove(struct platform_device *pdev, } static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -284,7 +285,8 @@ static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream, return 0; } -static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd) +static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { u32 ac_glbctrl; @@ -313,7 +315,8 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd) } static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -327,7 +330,7 @@ static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, } static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, - int cmd) + int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; @@ -356,7 +359,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "s3c2443-ac97", .id = 0, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .probe = s3c2443_ac97_probe, .remove = s3c2443_ac97_remove, .playback = { @@ -378,7 +381,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "pxa2xx-ac97-mic", .id = 1, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .capture = { .stream_name = "AC97 Mic Capture", .channels_min = 1, @@ -393,6 +396,21 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { EXPORT_SYMBOL_GPL(s3c2443_ac97_dai); EXPORT_SYMBOL_GPL(soc_ac97_ops); +static int __init s3c2443_ac97_init(void) +{ + return snd_soc_register_dais(s3c2443_ac97_dai, + ARRAY_SIZE(s3c2443_ac97_dai)); +} +module_init(s3c2443_ac97_init); + +static void __exit s3c2443_ac97_exit(void) +{ + snd_soc_unregister_dais(s3c2443_ac97_dai, + ARRAY_SIZE(s3c2443_ac97_dai)); +} +module_exit(s3c2443_ac97_exit); + + MODULE_AUTHOR("Graeme Gregory"); MODULE_DESCRIPTION("AC97 driver for the Samsung s3c2443 chip"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index ba4476b55fbc..6f4d439b57aa 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -243,7 +243,8 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, } static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; u32 iismod; @@ -261,10 +262,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: + iismod &= ~S3C2410_IISMOD_16BIT; + ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->dma_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; + ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->dma_size = 2; break; + default: + return -EINVAL; } writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -272,7 +280,8 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { int ret = 0; @@ -410,8 +419,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev, } #ifdef CONFIG_PM -static int s3c24xx_i2s_suspend(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) +static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) { DBG("Entered %s\n", __func__); @@ -425,8 +433,7 @@ static int s3c24xx_i2s_suspend(struct platform_device *pdev, return 0; } -static int s3c24xx_i2s_resume(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) +static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) { DBG("Entered %s\n", __func__); clk_enable(s3c24xx_i2s.iis_clk); @@ -452,7 +459,6 @@ static int s3c24xx_i2s_resume(struct platform_device *pdev, struct snd_soc_dai s3c24xx_i2s_dai = { .name = "s3c24xx-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .probe = s3c24xx_i2s_probe, .suspend = s3c24xx_i2s_suspend, .resume = s3c24xx_i2s_resume, @@ -468,8 +474,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = { .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, .ops = { .trigger = s3c24xx_i2s_trigger, - .hw_params = s3c24xx_i2s_hw_params,}, - .dai_ops = { + .hw_params = s3c24xx_i2s_hw_params, .set_fmt = s3c24xx_i2s_set_fmt, .set_clkdiv = s3c24xx_i2s_set_clkdiv, .set_sysclk = s3c24xx_i2s_set_sysclk, @@ -477,6 +482,18 @@ struct snd_soc_dai s3c24xx_i2s_dai = { }; EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai); +static int __init s3c24xx_i2s_init(void) +{ + return snd_soc_register_dai(&s3c24xx_i2s_dai); +} +module_init(s3c24xx_i2s_init); + +static void __exit s3c24xx_i2s_exit(void) +{ + snd_soc_unregister_dai(&s3c24xx_i2s_dai); +} +module_exit(s3c24xx_i2s_exit); + /* Module information */ MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>"); MODULE_DESCRIPTION("s3c24xx I2S SoC Interface"); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index e13e614bada9..7c64d31d067e 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -465,6 +465,18 @@ struct snd_soc_platform s3c24xx_soc_platform = { }; EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); +static int __init s3c24xx_soc_platform_init(void) +{ + return snd_soc_register_platform(&s3c24xx_soc_platform); +} +module_init(s3c24xx_soc_platform_init); + +static void __exit s3c24xx_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&s3c24xx_soc_platform); +} +module_exit(s3c24xx_soc_platform_exit); + MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>"); MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c new file mode 100644 index 000000000000..a0a4d1832a14 --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -0,0 +1,373 @@ +/* + * Modifications by Christian Pellegrin <chripell@evolware.org> + * + * s3c24xx_uda134x.c -- S3C24XX_UDA134X ALSA SoC Audio board driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/mutex.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/s3c24xx_uda134x.h> +#include <sound/uda134x.h> + +#include <asm/plat-s3c24xx/regs-iis.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda134x.h" + + +/* #define ENFORCE_RATES 1 */ +/* + Unfortunately the S3C24XX in master mode has a limited capacity of + generating the clock for the codec. If you define this only rates + that are really available will be enforced. But be careful, most + user level application just want the usual sampling frequencies (8, + 11.025, 22.050, 44.1 kHz) and anyway resampling is a costly + operation for embedded systems. So if you aren't very lucky or your + hardware engineer wasn't very forward-looking it's better to leave + this undefined. If you do so an approximate value for the requested + sampling rate in the range -/+ 5% will be chosen. If this in not + possible an error will be returned. +*/ + +static struct clk *xtal; +static struct clk *pclk; +/* this is need because we don't have a place where to keep the + * pointers to the clocks in each substream. We get the clocks only + * when we are actually using them so we don't block stuff like + * frequency change or oscillator power-off */ +static int clk_users; +static DEFINE_MUTEX(clk_lock); + +static unsigned int rates[33 * 2]; +#ifdef ENFORCE_RATES +static struct snd_pcm_hw_constraint_list hw_constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; +#endif + +static struct platform_device *s3c24xx_uda134x_snd_device; + +static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) +{ + int ret = 0; +#ifdef ENFORCE_RATES + struct snd_pcm_runtime *runtime = substream->runtime;; +#endif + + mutex_lock(&clk_lock); + pr_debug("%s %d\n", __func__, clk_users); + if (clk_users == 0) { + xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); + if (!xtal) { + printk(KERN_ERR "%s cannot get xtal\n", __func__); + ret = -EBUSY; + } else { + pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, + "pclk"); + if (!pclk) { + printk(KERN_ERR "%s cannot get pclk\n", + __func__); + clk_put(xtal); + ret = -EBUSY; + } + } + if (!ret) { + int i, j; + + for (i = 0; i < 2; i++) { + int fs = i ? 256 : 384; + + rates[i*33] = clk_get_rate(xtal) / fs; + for (j = 1; j < 33; j++) + rates[i*33 + j] = clk_get_rate(pclk) / + (j * fs); + } + } + } + clk_users += 1; + mutex_unlock(&clk_lock); + if (!ret) { +#ifdef ENFORCE_RATES + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &hw_constraints_rates); + if (ret < 0) + printk(KERN_ERR "%s cannot set constraints\n", + __func__); +#endif + } + return ret; +} + +static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream) +{ + mutex_lock(&clk_lock); + pr_debug("%s %d\n", __func__, clk_users); + clk_users -= 1; + if (clk_users == 0) { + clk_put(xtal); + xtal = NULL; + clk_put(pclk); + pclk = NULL; + } + mutex_unlock(&clk_lock); +} + +static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int clk = 0; + int ret = 0; + int clk_source, fs_mode; + unsigned long rate = params_rate(params); + long err, cerr; + unsigned int div; + int i, bi; + + err = 999999; + bi = 0; + for (i = 0; i < 2*33; i++) { + cerr = rates[i] - rate; + if (cerr < 0) + cerr = -cerr; + if (cerr < err) { + err = cerr; + bi = i; + } + } + if (bi / 33 == 1) + fs_mode = S3C2410_IISMOD_256FS; + else + fs_mode = S3C2410_IISMOD_384FS; + if (bi % 33 == 0) { + clk_source = S3C24XX_CLKSRC_MPLL; + div = 1; + } else { + clk_source = S3C24XX_CLKSRC_PCLK; + div = bi % 33; + } + pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi); + + clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; + pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__, + fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", + clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", + div, clk, err); + + if ((err * 100 / rate) > 5) { + printk(KERN_ERR "S3C24XX_UDA134X: effective frequency " + "too different from desired (%ld%%)\n", + err * 100 / rate); + return -EINVAL; + } + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, fs_mode); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops s3c24xx_uda134x_ops = { + .startup = s3c24xx_uda134x_startup, + .shutdown = s3c24xx_uda134x_shutdown, + .hw_params = s3c24xx_uda134x_hw_params, +}; + +static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { + .name = "UDA134X", + .stream_name = "UDA134X", + .codec_dai = &uda134x_dai, + .cpu_dai = &s3c24xx_i2s_dai, + .ops = &s3c24xx_uda134x_ops, +}; + +static struct snd_soc_card snd_soc_s3c24xx_uda134x = { + .name = "S3C24XX_UDA134X", + .platform = &s3c24xx_soc_platform, + .dai_link = &s3c24xx_uda134x_dai_link, + .num_links = 1, +}; + +static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins; + +static void setdat(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0); +} + +static void setclk(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0); +} + +static void setmode(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0); +} + +static struct uda134x_platform_data s3c24xx_uda134x = { + .l3 = { + .setdat = setdat, + .setclk = setclk, + .setmode = setmode, + .data_hold = 1, + .data_setup = 1, + .clock_high = 1, + .mode_hold = 1, + .mode = 1, + .mode_setup = 1, + }, +}; + +static struct snd_soc_device s3c24xx_uda134x_snd_devdata = { + .card = &snd_soc_s3c24xx_uda134x, + .codec_dev = &soc_codec_dev_uda134x, + .codec_data = &s3c24xx_uda134x, +}; + +static int s3c24xx_uda134x_setup_pin(int pin, char *fun) +{ + if (gpio_request(pin, "s3c24xx_uda134x") < 0) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "l3 %s pin already in use", fun); + return -EBUSY; + } + gpio_direction_output(pin, 0); + return 0; +} + +static int s3c24xx_uda134x_probe(struct platform_device *pdev) +{ + int ret; + + printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n"); + + s3c24xx_uda134x_l3_pins = pdev->dev.platform_data; + if (s3c24xx_uda134x_l3_pins == NULL) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "unable to find platform data\n"); + return -ENODEV; + } + s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power; + s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model; + + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data, + "data") < 0) + return -EBUSY; + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk, + "clk") < 0) { + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + return -EBUSY; + } + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode, + "mode") < 0) { + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); + return -EBUSY; + } + + s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_uda134x_snd_device) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "Unable to register\n"); + return -ENOMEM; + } + + platform_set_drvdata(s3c24xx_uda134x_snd_device, + &s3c24xx_uda134x_snd_devdata); + s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev; + ret = platform_device_add(s3c24xx_uda134x_snd_device); + if (ret) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n"); + platform_device_put(s3c24xx_uda134x_snd_device); + } + + return ret; +} + +static int s3c24xx_uda134x_remove(struct platform_device *pdev) +{ + platform_device_unregister(s3c24xx_uda134x_snd_device); + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); + gpio_free(s3c24xx_uda134x_l3_pins->l3_mode); + return 0; +} + +static struct platform_driver s3c24xx_uda134x_driver = { + .probe = s3c24xx_uda134x_probe, + .remove = s3c24xx_uda134x_remove, + .driver = { + .name = "s3c24xx_uda134x", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c24xx_uda134x_init(void) +{ + return platform_driver_register(&s3c24xx_uda134x_driver); +} + +static void __exit s3c24xx_uda134x_exit(void) +{ + platform_driver_unregister(&s3c24xx_uda134x_driver); +} + + +module_init(s3c24xx_uda134x_init); +module_exit(s3c24xx_uda134x_exit); + +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>"); +MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index 8515d6ff03f2..a2a4f5323c17 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -23,7 +23,7 @@ #include "s3c24xx-pcm.h" #include "s3c24xx-ac97.h" -static struct snd_soc_machine smdk2443; +static struct snd_soc_card smdk2443; static struct snd_soc_dai_link smdk2443_dai[] = { { @@ -34,15 +34,15 @@ static struct snd_soc_dai_link smdk2443_dai[] = { }, }; -static struct snd_soc_machine smdk2443 = { +static struct snd_soc_card smdk2443 = { .name = "SMDK2443", + .platform = &s3c24xx_soc_platform, .dai_link = smdk2443_dai, .num_links = ARRAY_SIZE(smdk2443_dai), }; static struct snd_soc_device smdk2443_snd_ac97_devdata = { - .machine = &smdk2443, - .platform = &s3c24xx_soc_platform, + .card = &smdk2443, .codec_dev = &soc_codec_dev_ac97, }; diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 9faa12622d09..0dad3a0bb920 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -348,6 +348,18 @@ struct snd_soc_platform sh7760_soc_platform = { }; EXPORT_SYMBOL_GPL(sh7760_soc_platform); +static int __init sh7760_soc_platform_init(void) +{ + return snd_soc_register_platform(&sh7760_soc_platform); +} +module_init(sh7760_soc_platform_init); + +static void __exit sh7760_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&sh7760_soc_platform); +} +module_exit(sh7760_soc_platform_exit); + MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver"); MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index df7bc345c320..eab31838badf 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -236,7 +236,8 @@ struct snd_ac97_bus_ops soc_ac97_ops = { EXPORT_SYMBOL_GPL(soc_ac97_ops); static int hac_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id]; @@ -270,7 +271,7 @@ struct snd_soc_dai sh4_hac_dai[] = { { .name = "HAC0", .id = 0, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .playback = { .rates = AC97_RATES, .formats = AC97_FMTS, @@ -290,8 +291,8 @@ struct snd_soc_dai sh4_hac_dai[] = { #ifdef CONFIG_CPU_SUBTYPE_SH7760 { .name = "HAC1", + .ac97_control = 1, .id = 1, - .type = SND_SOC_DAI_AC97, .playback = { .rates = AC97_RATES, .formats = AC97_FMTS, @@ -313,6 +314,18 @@ struct snd_soc_dai sh4_hac_dai[] = { }; EXPORT_SYMBOL_GPL(sh4_hac_dai); +static int __init sh4_hac_init(void) +{ + return snd_soc_register_dais(sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai)); +} +module_init(sh4_hac_init); + +static void __exit sh4_hac_exit(void) +{ + snd_soc_unregister_dais(sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai)); +} +module_exit(sh4_hac_exit); + MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver"); MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 92bfaf4774a7..ce7f95b59de3 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -38,15 +38,15 @@ static struct snd_soc_dai_link sh7760_ac97_dai = { .ops = NULL, }; -static struct snd_soc_machine sh7760_ac97_soc_machine = { +static struct snd_soc_card sh7760_ac97_soc_machine = { .name = "SH7760 AC97", + .platform = &sh7760_soc_platform, .dai_link = &sh7760_ac97_dai, .num_links = 1, }; static struct snd_soc_device sh7760_ac97_snd_devdata = { - .machine = &sh7760_ac97_soc_machine, - .platform = &sh7760_soc_platform, + .card = &sh7760_ac97_soc_machine, .codec_dev = &soc_codec_dev_ac97, }; diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 55c3464163ab..d1e5390fddeb 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -89,7 +89,8 @@ struct ssi_priv { * track usage of the SSI; it is simplex-only so prevent attempts of * concurrent playback + capture. FIXME: any locking required? */ -static int ssi_startup(struct snd_pcm_substream *substream) +static int ssi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; @@ -101,7 +102,8 @@ static int ssi_startup(struct snd_pcm_substream *substream) return 0; } -static void ssi_shutdown(struct snd_pcm_substream *substream) +static void ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; @@ -109,7 +111,8 @@ static void ssi_shutdown(struct snd_pcm_substream *substream) ssi->inuse = 0; } -static int ssi_trigger(struct snd_pcm_substream *substream, int cmd) +static int ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; @@ -129,7 +132,8 @@ static int ssi_trigger(struct snd_pcm_substream *substream, int cmd) } static int ssi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; @@ -336,7 +340,6 @@ struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI0", .id = 0, - .type = SND_SOC_DAI_I2S, .playback = { .rates = SSI_RATES, .formats = SSI_FMTS, @@ -354,8 +357,6 @@ struct snd_soc_dai sh4_ssi_dai[] = { .shutdown = ssi_shutdown, .trigger = ssi_trigger, .hw_params = ssi_hw_params, - }, - .dai_ops = { .set_sysclk = ssi_set_sysclk, .set_clkdiv = ssi_set_clkdiv, .set_fmt = ssi_set_fmt, @@ -365,7 +366,6 @@ struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI1", .id = 1, - .type = SND_SOC_DAI_I2S, .playback = { .rates = SSI_RATES, .formats = SSI_FMTS, @@ -383,8 +383,6 @@ struct snd_soc_dai sh4_ssi_dai[] = { .shutdown = ssi_shutdown, .trigger = ssi_trigger, .hw_params = ssi_hw_params, - }, - .dai_ops = { .set_sysclk = ssi_set_sysclk, .set_clkdiv = ssi_set_clkdiv, .set_fmt = ssi_set_fmt, @@ -394,6 +392,18 @@ struct snd_soc_dai sh4_ssi_dai[] = { }; EXPORT_SYMBOL_GPL(sh4_ssi_dai); +static int __init sh4_ssi_init(void) +{ + return snd_soc_register_dais(sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai)); +} +module_init(sh4_ssi_init); + +static void __exit sh4_ssi_exit(void) +{ + snd_soc_unregister_dais(sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai)); +} +module_exit(sh4_ssi_exit); + MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver"); MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16c7453f4946..b098c0b4c584 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -26,6 +26,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/bitops.h> +#include <linux/debugfs.h> #include <linux/platform_device.h> #include <sound/core.h> #include <sound/pcm.h> @@ -34,18 +35,23 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> -/* debug */ -#define SOC_DEBUG 0 -#if SOC_DEBUG -#define dbg(format, arg...) printk(format, ## arg) -#else -#define dbg(format, arg...) -#endif - static DEFINE_MUTEX(pcm_mutex); static DEFINE_MUTEX(io_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); +#ifdef CONFIG_DEBUG_FS +static struct dentry *debugfs_root; +#endif + +static DEFINE_MUTEX(client_mutex); +static LIST_HEAD(card_list); +static LIST_HEAD(dai_list); +static LIST_HEAD(platform_list); +static LIST_HEAD(codec_list); + +static int snd_soc_register_card(struct snd_soc_card *card); +static int snd_soc_unregister_card(struct snd_soc_card *card); + /* * This is a timeout to do a DAPM powerdown after a stream is closed(). * It can be used to eliminate pops between different playback streams, e.g. @@ -107,20 +113,6 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif -static inline const char *get_dai_name(int type) -{ - switch (type) { - case SND_SOC_DAI_AC97_BUS: - case SND_SOC_DAI_AC97: - return "AC97"; - case SND_SOC_DAI_I2S: - return "I2S"; - case SND_SOC_DAI_PCM: - return "PCM"; - } - return NULL; -} - /* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls @@ -130,9 +122,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_card *card = socdev->card; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *cpu_dai = machine->cpu_dai; struct snd_soc_dai *codec_dai = machine->codec_dai; int ret = 0; @@ -141,7 +134,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) /* startup the audio subsystem */ if (cpu_dai->ops.startup) { - ret = cpu_dai->ops.startup(substream); + ret = cpu_dai->ops.startup(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open interface %s\n", cpu_dai->name); @@ -158,7 +151,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } if (codec_dai->ops.startup) { - ret = codec_dai->ops.startup(substream); + ret = codec_dai->ops.startup(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open codec %s\n", codec_dai->name); @@ -228,12 +221,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto machine_err; } - dbg("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); - dbg("asoc: rate mask 0x%x\n", runtime->hw.rates); - dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, - runtime->hw.channels_max); - dbg("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, - runtime->hw.rate_max); + pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); + pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); + pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, + runtime->hw.channels_max); + pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, + runtime->hw.rate_max); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->playback.active = codec_dai->playback.active = 1; @@ -255,7 +248,7 @@ codec_dai_err: platform_err: if (cpu_dai->ops.shutdown) - cpu_dai->ops.shutdown(substream); + cpu_dai->ops.shutdown(substream, cpu_dai); out: mutex_unlock(&pcm_mutex); return ret; @@ -268,8 +261,9 @@ out: */ static void close_delayed_work(struct work_struct *work) { - struct snd_soc_device *socdev = - container_of(work, struct snd_soc_device, delayed_work.work); + struct snd_soc_card *card = container_of(work, struct snd_soc_card, + delayed_work.work); + struct snd_soc_device *socdev = card->socdev; struct snd_soc_codec *codec = socdev->codec; struct snd_soc_dai *codec_dai; int i; @@ -278,18 +272,18 @@ static void close_delayed_work(struct work_struct *work) for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; - dbg("pop wq checking: %s status: %s waiting: %s\n", - codec_dai->playback.stream_name, - codec_dai->playback.active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); + pr_debug("pop wq checking: %s status: %s waiting: %s\n", + codec_dai->playback.stream_name, + codec_dai->playback.active ? "active" : "inactive", + codec_dai->pop_wait ? "yes" : "no"); /* are we waiting on this codec DAI stream */ if (codec_dai->pop_wait == 1) { /* Reduce power if no longer active */ if (codec->active == 0) { - dbg("pop wq D1 %s %s\n", codec->name, - codec_dai->playback.stream_name); + pr_debug("pop wq D1 %s %s\n", codec->name, + codec_dai->playback.stream_name); snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE); } @@ -301,8 +295,8 @@ static void close_delayed_work(struct work_struct *work) /* Fall into standby if no longer active */ if (codec->active == 0) { - dbg("pop wq D3 %s %s\n", codec->name, - codec_dai->playback.stream_name); + pr_debug("pop wq D3 %s %s\n", codec->name, + codec_dai->playback.stream_name); snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY); } @@ -320,8 +314,9 @@ static int soc_codec_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_card *card = socdev->card; struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *cpu_dai = machine->cpu_dai; struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; @@ -346,10 +341,10 @@ static int soc_codec_close(struct snd_pcm_substream *substream) snd_soc_dai_digital_mute(codec_dai, 1); if (cpu_dai->ops.shutdown) - cpu_dai->ops.shutdown(substream); + cpu_dai->ops.shutdown(substream, cpu_dai); if (codec_dai->ops.shutdown) - codec_dai->ops.shutdown(substream); + codec_dai->ops.shutdown(substream, codec_dai); if (machine->ops && machine->ops->shutdown) machine->ops->shutdown(substream); @@ -361,7 +356,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* start delayed pop wq here for playback streams */ codec_dai->pop_wait = 1; - schedule_delayed_work(&socdev->delayed_work, + schedule_delayed_work(&card->delayed_work, msecs_to_jiffies(pmdown_time)); } else { /* capture streams can be powered down now */ @@ -387,8 +382,9 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_card *card = socdev->card; struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *cpu_dai = machine->cpu_dai; struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; @@ -413,7 +409,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } if (codec_dai->ops.prepare) { - ret = codec_dai->ops.prepare(substream); + ret = codec_dai->ops.prepare(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: codec DAI prepare error\n"); goto out; @@ -421,58 +417,49 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } if (cpu_dai->ops.prepare) { - ret = cpu_dai->ops.prepare(substream); + ret = cpu_dai->ops.prepare(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: cpu DAI prepare error\n"); goto out; } } - /* we only want to start a DAPM playback stream if we are not waiting - * on an existing one stopping */ - if (codec_dai->pop_wait) { - /* we are waiting for the delayed work to start */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - snd_soc_dapm_stream_event(socdev->codec, - codec_dai->capture.stream_name, - SND_SOC_DAPM_STREAM_START); - else { - codec_dai->pop_wait = 0; - cancel_delayed_work(&socdev->delayed_work); - snd_soc_dai_digital_mute(codec_dai, 0); - } - } else { - /* no delayed work - do we need to power up codec */ - if (codec->bias_level != SND_SOC_BIAS_ON) { + /* cancel any delayed stream shutdown that is pending */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + codec_dai->pop_wait) { + codec_dai->pop_wait = 0; + cancel_delayed_work(&card->delayed_work); + } - snd_soc_dapm_set_bias_level(socdev, - SND_SOC_BIAS_PREPARE); + /* do we need to power up codec */ + if (codec->bias_level != SND_SOC_BIAS_ON) { + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_PREPARE); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dapm_stream_event(codec, + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name, SND_SOC_DAPM_STREAM_START); - else - snd_soc_dapm_stream_event(codec, + else + snd_soc_dapm_stream_event(codec, codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); - snd_soc_dai_digital_mute(codec_dai, 0); + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); + snd_soc_dai_digital_mute(codec_dai, 0); - } else { - /* codec already powered - power on widgets */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dapm_stream_event(codec, + } else { + /* codec already powered - power on widgets */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name, SND_SOC_DAPM_STREAM_START); - else - snd_soc_dapm_stream_event(codec, + else + snd_soc_dapm_stream_event(codec, codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - snd_soc_dai_digital_mute(codec_dai, 0); - } + snd_soc_dai_digital_mute(codec_dai, 0); } out: @@ -491,7 +478,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_card *card = socdev->card; + struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *cpu_dai = machine->cpu_dai; struct snd_soc_dai *codec_dai = machine->codec_dai; int ret = 0; @@ -507,7 +495,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } if (codec_dai->ops.hw_params) { - ret = codec_dai->ops.hw_params(substream, params); + ret = codec_dai->ops.hw_params(substream, params, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't set codec %s hw params\n", codec_dai->name); @@ -516,7 +504,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } if (cpu_dai->ops.hw_params) { - ret = cpu_dai->ops.hw_params(substream, params); + ret = cpu_dai->ops.hw_params(substream, params, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: interface %s hw params failed\n", cpu_dai->name); @@ -539,11 +527,11 @@ out: platform_err: if (cpu_dai->ops.hw_free) - cpu_dai->ops.hw_free(substream); + cpu_dai->ops.hw_free(substream, cpu_dai); interface_err: if (codec_dai->ops.hw_free) - codec_dai->ops.hw_free(substream); + codec_dai->ops.hw_free(substream, codec_dai); codec_err: if (machine->ops && machine->ops->hw_free) @@ -561,7 +549,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_card *card = socdev->card; + struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *cpu_dai = machine->cpu_dai; struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; @@ -582,10 +571,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) /* now free hw params for the DAI's */ if (codec_dai->ops.hw_free) - codec_dai->ops.hw_free(substream); + codec_dai->ops.hw_free(substream, codec_dai); if (cpu_dai->ops.hw_free) - cpu_dai->ops.hw_free(substream); + cpu_dai->ops.hw_free(substream, cpu_dai); mutex_unlock(&pcm_mutex); return 0; @@ -595,14 +584,15 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_card *card= socdev->card; struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *cpu_dai = machine->cpu_dai; struct snd_soc_dai *codec_dai = machine->codec_dai; int ret; if (codec_dai->ops.trigger) { - ret = codec_dai->ops.trigger(substream, cmd); + ret = codec_dai->ops.trigger(substream, cmd, codec_dai); if (ret < 0) return ret; } @@ -614,7 +604,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } if (cpu_dai->ops.trigger) { - ret = cpu_dai->ops.trigger(substream, cmd); + ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; } @@ -636,8 +626,8 @@ static struct snd_pcm_ops soc_pcm_ops = { static int soc_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_card *card = socdev->card; + struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; struct snd_soc_codec *codec = socdev->codec; int i; @@ -653,29 +643,29 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot); /* mute any active DAC's */ - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; - if (dai->dai_ops.digital_mute && dai->playback.active) - dai->dai_ops.digital_mute(dai, 1); + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *dai = card->dai_link[i].codec_dai; + if (dai->ops.digital_mute && dai->playback.active) + dai->ops.digital_mute(dai, 1); } /* suspend all pcms */ - for (i = 0; i < machine->num_links; i++) - snd_pcm_suspend_all(machine->dai_link[i].pcm); + for (i = 0; i < card->num_links; i++) + snd_pcm_suspend_all(card->dai_link[i].pcm); - if (machine->suspend_pre) - machine->suspend_pre(pdev, state); + if (card->suspend_pre) + card->suspend_pre(pdev, state); - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; - if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97) - cpu_dai->suspend(pdev, cpu_dai); + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + if (cpu_dai->suspend && !cpu_dai->ac97_control) + cpu_dai->suspend(cpu_dai); if (platform->suspend) - platform->suspend(pdev, cpu_dai); + platform->suspend(cpu_dai); } /* close any waiting streams and save state */ - run_delayed_work(&socdev->delayed_work); + run_delayed_work(&card->delayed_work); codec->suspend_bias_level = codec->bias_level; for (i = 0; i < codec->num_dai; i++) { @@ -692,14 +682,14 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) if (codec_dev->suspend) codec_dev->suspend(pdev, state); - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; - if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97) - cpu_dai->suspend(pdev, cpu_dai); + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + if (cpu_dai->suspend && cpu_dai->ac97_control) + cpu_dai->suspend(cpu_dai); } - if (machine->suspend_post) - machine->suspend_post(pdev, state); + if (card->suspend_post) + card->suspend_post(pdev, state); return 0; } @@ -709,11 +699,11 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) */ static void soc_resume_deferred(struct work_struct *work) { - struct snd_soc_device *socdev = container_of(work, - struct snd_soc_device, - deferred_resume_work); - struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_card *card = container_of(work, + struct snd_soc_card, + deferred_resume_work); + struct snd_soc_device *socdev = card->socdev; + struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; struct snd_soc_codec *codec = socdev->codec; struct platform_device *pdev = to_platform_device(socdev->dev); @@ -723,15 +713,15 @@ static void soc_resume_deferred(struct work_struct *work) * so userspace apps are blocked from touching us */ - dev_info(socdev->dev, "starting resume work\n"); + dev_dbg(socdev->dev, "starting resume work\n"); - if (machine->resume_pre) - machine->resume_pre(pdev); + if (card->resume_pre) + card->resume_pre(pdev); - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; - if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97) - cpu_dai->resume(pdev, cpu_dai); + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + if (cpu_dai->resume && cpu_dai->ac97_control) + cpu_dai->resume(cpu_dai); } if (codec_dev->resume) @@ -749,24 +739,24 @@ static void soc_resume_deferred(struct work_struct *work) } /* unmute any active DACs */ - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; - if (dai->dai_ops.digital_mute && dai->playback.active) - dai->dai_ops.digital_mute(dai, 0); + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *dai = card->dai_link[i].codec_dai; + if (dai->ops.digital_mute && dai->playback.active) + dai->ops.digital_mute(dai, 0); } - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; - if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97) - cpu_dai->resume(pdev, cpu_dai); + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + if (cpu_dai->resume && !cpu_dai->ac97_control) + cpu_dai->resume(cpu_dai); if (platform->resume) - platform->resume(pdev, cpu_dai); + platform->resume(cpu_dai); } - if (machine->resume_post) - machine->resume_post(pdev); + if (card->resume_post) + card->resume_post(pdev); - dev_info(socdev->dev, "resume work completed\n"); + dev_dbg(socdev->dev, "resume work completed\n"); /* userspace can access us now we are back as we were before */ snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0); @@ -776,11 +766,12 @@ static void soc_resume_deferred(struct work_struct *work) static int soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; - dev_info(socdev->dev, "scheduling resume work\n"); + dev_dbg(socdev->dev, "scheduling resume work\n"); - if (!schedule_work(&socdev->deferred_resume_work)) - dev_err(socdev->dev, "work item may be lost\n"); + if (!schedule_work(&card->deferred_resume_work)) + dev_err(socdev->dev, "resume work item may be lost\n"); return 0; } @@ -790,23 +781,83 @@ static int soc_resume(struct platform_device *pdev) #define soc_resume NULL #endif -/* probes a new socdev */ -static int soc_probe(struct platform_device *pdev) +static void snd_soc_instantiate_card(struct snd_soc_card *card) { - int ret = 0, i; - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_codec_device *codec_dev = socdev->codec_dev; + struct platform_device *pdev = container_of(card->dev, + struct platform_device, + dev); + struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev; + struct snd_soc_platform *platform; + struct snd_soc_dai *dai; + int i, found, ret, ac97; + + if (card->instantiated) + return; + + found = 0; + list_for_each_entry(platform, &platform_list, list) + if (card->platform == platform) { + found = 1; + break; + } + if (!found) { + dev_dbg(card->dev, "Platform %s not registered\n", + card->platform->name); + return; + } + + ac97 = 0; + for (i = 0; i < card->num_links; i++) { + found = 0; + list_for_each_entry(dai, &dai_list, list) + if (card->dai_link[i].cpu_dai == dai) { + found = 1; + break; + } + if (!found) { + dev_dbg(card->dev, "DAI %s not registered\n", + card->dai_link[i].cpu_dai->name); + return; + } - if (machine->probe) { - ret = machine->probe(pdev); + if (card->dai_link[i].cpu_dai->ac97_control) + ac97 = 1; + } + + /* If we have AC97 in the system then don't wait for the + * codec. This will need revisiting if we have to handle + * systems with mixed AC97 and non-AC97 parts. Only check for + * DAIs currently; we can't do this per link since some AC97 + * codecs have non-AC97 DAIs. + */ + if (!ac97) + for (i = 0; i < card->num_links; i++) { + found = 0; + list_for_each_entry(dai, &dai_list, list) + if (card->dai_link[i].codec_dai == dai) { + found = 1; + break; + } + if (!found) { + dev_dbg(card->dev, "DAI %s not registered\n", + card->dai_link[i].codec_dai->name); + return; + } + } + + /* Note that we do not current check for codec components */ + + dev_dbg(card->dev, "All components present, instantiating\n"); + + /* Found everything, bring it up */ + if (card->probe) { + ret = card->probe(pdev); if (ret < 0) - return ret; + return; } - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; if (cpu_dai->probe) { ret = cpu_dai->probe(pdev, cpu_dai); if (ret < 0) @@ -827,13 +878,15 @@ static int soc_probe(struct platform_device *pdev) } /* DAPM stream work */ - INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work); + INIT_DELAYED_WORK(&card->delayed_work, close_delayed_work); #ifdef CONFIG_PM /* deferred resume work */ - INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred); + INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); #endif - return 0; + card->instantiated = 1; + + return; platform_err: if (codec_dev->remove) @@ -841,15 +894,45 @@ platform_err: cpu_dai_err: for (i--; i >= 0; i--) { - struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; if (cpu_dai->remove) cpu_dai->remove(pdev, cpu_dai); } - if (machine->remove) - machine->remove(pdev); + if (card->remove) + card->remove(pdev); +} - return ret; +/* + * Attempt to initialise any uninitalised cards. Must be called with + * client_mutex. + */ +static void snd_soc_instantiate_cards(void) +{ + struct snd_soc_card *card; + list_for_each_entry(card, &card_list, list) + snd_soc_instantiate_card(card); +} + +/* probes a new socdev */ +static int soc_probe(struct platform_device *pdev) +{ + int ret = 0; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + + /* Bodge while we push things out of socdev */ + card->socdev = socdev; + + /* Bodge while we unpick instantiation */ + card->dev = &pdev->dev; + ret = snd_soc_register_card(card); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to register card\n"); + return ret; + } + + return 0; } /* removes a socdev */ @@ -857,11 +940,11 @@ static int soc_remove(struct platform_device *pdev) { int i; struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_card *card = socdev->card; + struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - run_delayed_work(&socdev->delayed_work); + run_delayed_work(&card->delayed_work); if (platform->remove) platform->remove(pdev); @@ -869,14 +952,16 @@ static int soc_remove(struct platform_device *pdev) if (codec_dev->remove) codec_dev->remove(pdev); - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; if (cpu_dai->remove) cpu_dai->remove(pdev, cpu_dai); } - if (machine->remove) - machine->remove(pdev); + if (card->remove) + card->remove(pdev); + + snd_soc_unregister_card(card); return 0; } @@ -898,6 +983,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, struct snd_soc_dai_link *dai_link, int num) { struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_card *card = socdev->card; + struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *codec_dai = dai_link->codec_dai; struct snd_soc_dai *cpu_dai = dai_link->cpu_dai; struct snd_soc_pcm_runtime *rtd; @@ -914,8 +1001,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, codec_dai->codec = socdev->codec; /* check client and interface hw capabilities */ - sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name, - get_dai_name(cpu_dai->type), num); + sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name, + num); if (codec_dai->playback.channels_min) playback = 1; @@ -933,13 +1020,13 @@ static int soc_new_pcm(struct snd_soc_device *socdev, dai_link->pcm = pcm; pcm->private_data = rtd; - soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap; - soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer; - soc_pcm_ops.ioctl = socdev->platform->pcm_ops->ioctl; - soc_pcm_ops.copy = socdev->platform->pcm_ops->copy; - soc_pcm_ops.silence = socdev->platform->pcm_ops->silence; - soc_pcm_ops.ack = socdev->platform->pcm_ops->ack; - soc_pcm_ops.page = socdev->platform->pcm_ops->page; + soc_pcm_ops.mmap = platform->pcm_ops->mmap; + soc_pcm_ops.pointer = platform->pcm_ops->pointer; + soc_pcm_ops.ioctl = platform->pcm_ops->ioctl; + soc_pcm_ops.copy = platform->pcm_ops->copy; + soc_pcm_ops.silence = platform->pcm_ops->silence; + soc_pcm_ops.ack = platform->pcm_ops->ack; + soc_pcm_ops.page = platform->pcm_ops->page; if (playback) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); @@ -947,24 +1034,22 @@ static int soc_new_pcm(struct snd_soc_device *socdev, if (capture) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); - ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm); + ret = platform->pcm_new(codec->card, codec_dai, pcm); if (ret < 0) { printk(KERN_ERR "asoc: platform pcm constructor failed\n"); kfree(rtd); return ret; } - pcm->private_free = socdev->platform->pcm_free; + pcm->private_free = platform->pcm_free; printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; } /* codec register dump */ -static ssize_t codec_reg_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf) { - struct snd_soc_device *devdata = dev_get_drvdata(dev); struct snd_soc_codec *codec = devdata->codec; int i, step = 1, count = 0; @@ -1001,8 +1086,110 @@ static ssize_t codec_reg_show(struct device *dev, return count; } +static ssize_t codec_reg_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_device *devdata = dev_get_drvdata(dev); + return soc_codec_reg_show(devdata, buf); +} + static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); +#ifdef CONFIG_DEBUG_FS +static int codec_reg_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + ssize_t ret; + struct snd_soc_codec *codec = file->private_data; + struct device *card_dev = codec->card->dev; + struct snd_soc_device *devdata = card_dev->driver_data; + char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ret = soc_codec_reg_show(devdata, buf); + if (ret >= 0) + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + kfree(buf); + return ret; +} + +static ssize_t codec_reg_write_file(struct file *file, + const char __user *user_buf, size_t count, loff_t *ppos) +{ + char buf[32]; + int buf_size; + char *start = buf; + unsigned long reg, value; + int step = 1; + struct snd_soc_codec *codec = file->private_data; + + buf_size = min(count, (sizeof(buf)-1)); + if (copy_from_user(buf, user_buf, buf_size)) + return -EFAULT; + buf[buf_size] = 0; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + while (*start == ' ') + start++; + reg = simple_strtoul(start, &start, 16); + if ((reg >= codec->reg_cache_size) || (reg % step)) + return -EINVAL; + while (*start == ' ') + start++; + if (strict_strtoul(start, 16, &value)) + return -EINVAL; + codec->write(codec, reg, value); + return buf_size; +} + +static const struct file_operations codec_reg_fops = { + .open = codec_reg_open_file, + .read = codec_reg_read_file, + .write = codec_reg_write_file, +}; + +static void soc_init_codec_debugfs(struct snd_soc_codec *codec) +{ + codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, + debugfs_root, codec, + &codec_reg_fops); + if (!codec->debugfs_reg) + printk(KERN_WARNING + "ASoC: Failed to create codec register debugfs file\n"); + + codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744, + debugfs_root, + &codec->pop_time); + if (!codec->debugfs_pop_time) + printk(KERN_WARNING + "Failed to create pop time debugfs file\n"); +} + +static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) +{ + debugfs_remove(codec->debugfs_pop_time); + debugfs_remove(codec->debugfs_reg); +} + +#else + +static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) +{ +} + +static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) +{ +} +#endif + /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec @@ -1121,7 +1308,7 @@ EXPORT_SYMBOL_GPL(snd_soc_test_bits); int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) { struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_card *card = socdev->card; int ret = 0, i; mutex_lock(&codec->mutex); @@ -1140,11 +1327,11 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver)); /* create the pcms */ - for (i = 0; i < machine->num_links; i++) { - ret = soc_new_pcm(socdev, &machine->dai_link[i], i); + for (i = 0; i < card->num_links; i++) { + ret = soc_new_pcm(socdev, &card->dai_link[i], i); if (ret < 0) { printk(KERN_ERR "asoc: can't create pcm %s\n", - machine->dai_link[i].stream_name); + card->dai_link[i].stream_name); mutex_unlock(&codec->mutex); return ret; } @@ -1156,7 +1343,7 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) EXPORT_SYMBOL_GPL(snd_soc_new_pcms); /** - * snd_soc_register_card - register sound card + * snd_soc_init_card - register sound card * @socdev: the SoC audio device * * Register a SoC sound card. Also registers an AC97 device if the @@ -1164,29 +1351,28 @@ EXPORT_SYMBOL_GPL(snd_soc_new_pcms); * * Returns 0 for success, else error. */ -int snd_soc_register_card(struct snd_soc_device *socdev) +int snd_soc_init_card(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_card *card = socdev->card; int ret = 0, i, ac97 = 0, err = 0; - for (i = 0; i < machine->num_links; i++) { - if (socdev->machine->dai_link[i].init) { - err = socdev->machine->dai_link[i].init(codec); + for (i = 0; i < card->num_links; i++) { + if (card->dai_link[i].init) { + err = card->dai_link[i].init(codec); if (err < 0) { printk(KERN_ERR "asoc: failed to init %s\n", - socdev->machine->dai_link[i].stream_name); + card->dai_link[i].stream_name); continue; } } - if (socdev->machine->dai_link[i].codec_dai->type == - SND_SOC_DAI_AC97_BUS) + if (card->dai_link[i].codec_dai->ac97_control) ac97 = 1; } snprintf(codec->card->shortname, sizeof(codec->card->shortname), - "%s", machine->name); + "%s", card->name); snprintf(codec->card->longname, sizeof(codec->card->longname), - "%s (%s)", machine->name, codec->name); + "%s (%s)", card->name, codec->name); ret = snd_card_register(codec->card); if (ret < 0) { @@ -1216,12 +1402,13 @@ int snd_soc_register_card(struct snd_soc_device *socdev) if (err < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + soc_init_codec_debugfs(socdev->codec); mutex_unlock(&codec->mutex); out: return ret; } -EXPORT_SYMBOL_GPL(snd_soc_register_card); +EXPORT_SYMBOL_GPL(snd_soc_init_card); /** * snd_soc_free_pcms - free sound card and pcms @@ -1239,10 +1426,11 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) #endif mutex_lock(&codec->mutex); + soc_cleanup_codec_debugfs(socdev->codec); #ifdef CONFIG_SND_SOC_AC97_BUS for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; - if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) { + if (codec_dai->ac97_control && codec->ac97) { soc_ac97_dev_unregister(codec); goto free_card; } @@ -1756,8 +1944,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - if (dai->dai_ops.set_sysclk) - return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir); + if (dai->ops.set_sysclk) + return dai->ops.set_sysclk(dai, clk_id, freq, dir); else return -EINVAL; } @@ -1776,8 +1964,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - if (dai->dai_ops.set_clkdiv) - return dai->dai_ops.set_clkdiv(dai, div_id, div); + if (dai->ops.set_clkdiv) + return dai->ops.set_clkdiv(dai, div_id, div); else return -EINVAL; } @@ -1795,8 +1983,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { - if (dai->dai_ops.set_pll) - return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out); + if (dai->ops.set_pll) + return dai->ops.set_pll(dai, pll_id, freq_in, freq_out); else return -EINVAL; } @@ -1805,15 +1993,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); /** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI - * @clk_id: DAI specific clock ID * @fmt: SND_SOC_DAIFMT_ format value. * * Configures the DAI hardware format and clocking. */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->dai_ops.set_fmt) - return dai->dai_ops.set_fmt(dai, fmt); + if (dai->ops.set_fmt) + return dai->ops.set_fmt(dai, fmt); else return -EINVAL; } @@ -1831,8 +2018,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int mask, int slots) { - if (dai->dai_ops.set_sysclk) - return dai->dai_ops.set_tdm_slot(dai, mask, slots); + if (dai->ops.set_sysclk) + return dai->ops.set_tdm_slot(dai, mask, slots); else return -EINVAL; } @@ -1847,8 +2034,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); */ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) { - if (dai->dai_ops.set_sysclk) - return dai->dai_ops.set_tristate(dai, tristate); + if (dai->ops.set_sysclk) + return dai->ops.set_tristate(dai, tristate); else return -EINVAL; } @@ -1863,21 +2050,242 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) { - if (dai->dai_ops.digital_mute) - return dai->dai_ops.digital_mute(dai, mute); + if (dai->ops.digital_mute) + return dai->ops.digital_mute(dai, mute); else return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); -static int __devinit snd_soc_init(void) +/** + * snd_soc_register_card - Register a card with the ASoC core + * + * @param card Card to register + * + * Note that currently this is an internal only function: it will be + * exposed to machine drivers after further backporting of ASoC v2 + * registration APIs. + */ +static int snd_soc_register_card(struct snd_soc_card *card) +{ + if (!card->name || !card->dev) + return -EINVAL; + + INIT_LIST_HEAD(&card->list); + card->instantiated = 0; + + mutex_lock(&client_mutex); + list_add(&card->list, &card_list); + snd_soc_instantiate_cards(); + mutex_unlock(&client_mutex); + + dev_dbg(card->dev, "Registered card '%s'\n", card->name); + + return 0; +} + +/** + * snd_soc_unregister_card - Unregister a card with the ASoC core + * + * @param card Card to unregister + * + * Note that currently this is an internal only function: it will be + * exposed to machine drivers after further backporting of ASoC v2 + * registration APIs. + */ +static int snd_soc_unregister_card(struct snd_soc_card *card) +{ + mutex_lock(&client_mutex); + list_del(&card->list); + mutex_unlock(&client_mutex); + + dev_dbg(card->dev, "Unregistered card '%s'\n", card->name); + + return 0; +} + +/** + * snd_soc_register_dai - Register a DAI with the ASoC core + * + * @param dai DAI to register + */ +int snd_soc_register_dai(struct snd_soc_dai *dai) +{ + if (!dai->name) + return -EINVAL; + + /* The device should become mandatory over time */ + if (!dai->dev) + printk(KERN_WARNING "No device for DAI %s\n", dai->name); + + INIT_LIST_HEAD(&dai->list); + + mutex_lock(&client_mutex); + list_add(&dai->list, &dai_list); + snd_soc_instantiate_cards(); + mutex_unlock(&client_mutex); + + pr_debug("Registered DAI '%s'\n", dai->name); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_register_dai); + +/** + * snd_soc_unregister_dai - Unregister a DAI from the ASoC core + * + * @param dai DAI to unregister + */ +void snd_soc_unregister_dai(struct snd_soc_dai *dai) +{ + mutex_lock(&client_mutex); + list_del(&dai->list); + mutex_unlock(&client_mutex); + + pr_debug("Unregistered DAI '%s'\n", dai->name); +} +EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); + +/** + * snd_soc_register_dais - Register multiple DAIs with the ASoC core + * + * @param dai Array of DAIs to register + * @param count Number of DAIs + */ +int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count) +{ + int i, ret; + + for (i = 0; i < count; i++) { + ret = snd_soc_register_dai(&dai[i]); + if (ret != 0) + goto err; + } + + return 0; + +err: + for (i--; i >= 0; i--) + snd_soc_unregister_dai(&dai[i]); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_register_dais); + +/** + * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core + * + * @param dai Array of DAIs to unregister + * @param count Number of DAIs + */ +void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count) +{ + int i; + + for (i = 0; i < count; i++) + snd_soc_unregister_dai(&dai[i]); +} +EXPORT_SYMBOL_GPL(snd_soc_unregister_dais); + +/** + * snd_soc_register_platform - Register a platform with the ASoC core + * + * @param platform platform to register + */ +int snd_soc_register_platform(struct snd_soc_platform *platform) +{ + if (!platform->name) + return -EINVAL; + + INIT_LIST_HEAD(&platform->list); + + mutex_lock(&client_mutex); + list_add(&platform->list, &platform_list); + snd_soc_instantiate_cards(); + mutex_unlock(&client_mutex); + + pr_debug("Registered platform '%s'\n", platform->name); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_register_platform); + +/** + * snd_soc_unregister_platform - Unregister a platform from the ASoC core + * + * @param platform platform to unregister + */ +void snd_soc_unregister_platform(struct snd_soc_platform *platform) { - printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION); + mutex_lock(&client_mutex); + list_del(&platform->list); + mutex_unlock(&client_mutex); + + pr_debug("Unregistered platform '%s'\n", platform->name); +} +EXPORT_SYMBOL_GPL(snd_soc_unregister_platform); + +/** + * snd_soc_register_codec - Register a codec with the ASoC core + * + * @param codec codec to register + */ +int snd_soc_register_codec(struct snd_soc_codec *codec) +{ + if (!codec->name) + return -EINVAL; + + /* The device should become mandatory over time */ + if (!codec->dev) + printk(KERN_WARNING "No device for codec %s\n", codec->name); + + INIT_LIST_HEAD(&codec->list); + + mutex_lock(&client_mutex); + list_add(&codec->list, &codec_list); + snd_soc_instantiate_cards(); + mutex_unlock(&client_mutex); + + pr_debug("Registered codec '%s'\n", codec->name); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_register_codec); + +/** + * snd_soc_unregister_codec - Unregister a codec from the ASoC core + * + * @param codec codec to unregister + */ +void snd_soc_unregister_codec(struct snd_soc_codec *codec) +{ + mutex_lock(&client_mutex); + list_del(&codec->list); + mutex_unlock(&client_mutex); + + pr_debug("Unregistered codec '%s'\n", codec->name); +} +EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); + +static int __init snd_soc_init(void) +{ +#ifdef CONFIG_DEBUG_FS + debugfs_root = debugfs_create_dir("asoc", NULL); + if (IS_ERR(debugfs_root) || !debugfs_root) { + printk(KERN_WARNING + "ASoC: Failed to create debugfs directory\n"); + debugfs_root = NULL; + } +#endif + return platform_driver_register(&soc_driver); } -static void snd_soc_exit(void) +static void __exit snd_soc_exit(void) { +#ifdef CONFIG_DEBUG_FS + debugfs_remove_recursive(debugfs_root); +#endif platform_driver_unregister(&soc_driver); } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7351db9606e4..8863eddbac02 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -37,7 +37,6 @@ #include <linux/bitops.h> #include <linux/platform_device.h> #include <linux/jiffies.h> -#include <linux/debugfs.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -67,17 +66,13 @@ static int dapm_status = 1; module_param(dapm_status, int, 0); MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries"); -static struct dentry *asoc_debugfs; - -static u32 pop_time; - -static void pop_wait(void) +static void pop_wait(u32 pop_time) { if (pop_time) schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time)); } -static void pop_dbg(const char *fmt, ...) +static void pop_dbg(u32 pop_time, const char *fmt, ...) { va_list args; @@ -85,7 +80,7 @@ static void pop_dbg(const char *fmt, ...) if (pop_time) { vprintk(fmt, args); - pop_wait(); + pop_wait(pop_time); } va_end(args); @@ -230,10 +225,11 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) change = old != new; if (change) { - pop_dbg("pop test %s : %s in %d ms\n", widget->name, - widget->power ? "on" : "off", pop_time); + pop_dbg(codec->pop_time, "pop test %s : %s in %d ms\n", + widget->name, widget->power ? "on" : "off", + codec->pop_time); snd_soc_write(codec, widget->reg, new); - pop_wait(); + pop_wait(codec->pop_time); } pr_debug("reg %x old %x new %x change %d\n", widget->reg, old, new, change); @@ -293,7 +289,7 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, struct snd_soc_dapm_widget *w) { int i, ret = 0; - char name[32]; + size_t name_len; struct snd_soc_dapm_path *path; /* add kcontrol */ @@ -307,11 +303,16 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, continue; /* add dapm control with long name */ - snprintf(name, 32, "%s %s", w->name, w->kcontrols[i].name); - path->long_name = kstrdup (name, GFP_KERNEL); + name_len = 2 + strlen(w->name) + + strlen(w->kcontrols[i].name); + path->long_name = kmalloc(name_len, GFP_KERNEL); if (path->long_name == NULL) return -ENOMEM; + snprintf(path->long_name, name_len, "%s %s", + w->name, w->kcontrols[i].name); + path->long_name[name_len - 1] = '\0'; + path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, path->long_name); ret = snd_ctl_add(codec->card, path->kcontrol); @@ -821,23 +822,9 @@ static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); int snd_soc_dapm_sys_add(struct device *dev) { - int ret = 0; - if (!dapm_status) return 0; - - ret = device_create_file(dev, &dev_attr_dapm_widget); - if (ret != 0) - return ret; - - asoc_debugfs = debugfs_create_dir("asoc", NULL); - if (!IS_ERR(asoc_debugfs) && asoc_debugfs) - debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs, - &pop_time); - else - asoc_debugfs = NULL; - - return 0; + return device_create_file(dev, &dev_attr_dapm_widget); } static void snd_soc_dapm_sys_remove(struct device *dev) @@ -845,9 +832,6 @@ static void snd_soc_dapm_sys_remove(struct device *dev) if (dapm_status) { device_remove_file(dev, &dev_attr_dapm_widget); } - - if (asoc_debugfs) - debugfs_remove_recursive(asoc_debugfs); } /* free all dapm widgets and resources */ @@ -1007,28 +991,6 @@ err: } /** - * snd_soc_dapm_connect_input - connect dapm widgets - * @codec: audio codec - * @sink: name of target widget - * @control: mixer control name - * @source: name of source name - * - * Connects 2 dapm widgets together via a named audio path. The sink is - * the widget receiving the audio signal, whilst the source is the sender - * of the audio signal. - * - * This function has been deprecated in favour of snd_soc_dapm_add_routes(). - * - * Returns 0 for success else error. - */ -int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink, - const char *control, const char *source) -{ - return snd_soc_dapm_add_route(codec, sink, control, source); -} -EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input); - -/** * snd_soc_dapm_add_routes - Add routes between DAPM widgets * @codec: codec * @route: audio routes @@ -1358,8 +1320,12 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, for (i = 0; i < num; i++) { ret = snd_soc_dapm_new_control(codec, widget); - if (ret < 0) + if (ret < 0) { + printk(KERN_ERR + "ASoC: Failed to create DAPM control %s: %d\n", + widget->name, ret); return ret; + } widget++; } return 0; @@ -1440,11 +1406,11 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, enum snd_soc_bias_level level) { struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_card *card = socdev->card; int ret = 0; - if (machine->set_bias_level) - ret = machine->set_bias_level(machine, level); + if (card->set_bias_level) + ret = card->set_bias_level(card, level); if (ret == 0 && codec->set_bias_level) ret = codec->set_bias_level(codec, level); diff --git a/sound/sound_core.c b/sound/sound_core.c index 10ba4218161b..2b302bbffe73 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -10,6 +10,7 @@ #include <linux/module.h> #include <linux/device.h> #include <linux/err.h> +#include <sound/core.h> #ifdef CONFIG_SOUND_OSS_CORE static int __init init_oss_soundcore(void); diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c index 798ca124da58..ccd763dd7167 100644 --- a/sound/usb/caiaq/caiaq-control.c +++ b/sound/usb/caiaq/caiaq-control.c @@ -247,69 +247,56 @@ static struct caiaq_controller a8dj_controller[] = { { "Software lock", 40 } }; -int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) +static int __devinit add_controls(struct caiaq_controller *c, int num, + struct snd_usb_caiaqdev *dev) { - int i; + int i, ret; struct snd_kcontrol *kc; + for (i = 0; i < num; i++, c++) { + kcontrol_template.name = c->name; + kcontrol_template.private_value = c->index; + kc = snd_ctl_new1(&kcontrol_template, dev); + ret = snd_ctl_add(dev->chip.card, kc); + if (ret < 0) + return ret; + } + + return 0; +} + +int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) +{ + int ret = 0; + switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): - for (i = 0; i < ARRAY_SIZE(ak1_controller); i++) { - struct caiaq_controller *c = ak1_controller + i; - kcontrol_template.name = c->name; - kcontrol_template.private_value = c->index; - kc = snd_ctl_new1(&kcontrol_template, dev); - snd_ctl_add(dev->chip.card, kc); - } - + ret = add_controls(ak1_controller, + ARRAY_SIZE(ak1_controller), dev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): - for (i = 0; i < ARRAY_SIZE(rk2_controller); i++) { - struct caiaq_controller *c = rk2_controller + i; - kcontrol_template.name = c->name; - kcontrol_template.private_value = c->index; - kc = snd_ctl_new1(&kcontrol_template, dev); - snd_ctl_add(dev->chip.card, kc); - } - + ret = add_controls(rk2_controller, + ARRAY_SIZE(rk2_controller), dev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): - for (i = 0; i < ARRAY_SIZE(rk3_controller); i++) { - struct caiaq_controller *c = rk3_controller + i; - kcontrol_template.name = c->name; - kcontrol_template.private_value = c->index; - kc = snd_ctl_new1(&kcontrol_template, dev); - snd_ctl_add(dev->chip.card, kc); - } - + ret = add_controls(rk3_controller, + ARRAY_SIZE(rk3_controller), dev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): - for (i = 0; i < ARRAY_SIZE(kore_controller); i++) { - struct caiaq_controller *c = kore_controller + i; - kcontrol_template.name = c->name; - kcontrol_template.private_value = c->index; - kc = snd_ctl_new1(&kcontrol_template, dev); - snd_ctl_add(dev->chip.card, kc); - } - + ret = add_controls(kore_controller, + ARRAY_SIZE(kore_controller), dev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): - for (i = 0; i < ARRAY_SIZE(a8dj_controller); i++) { - struct caiaq_controller *c = a8dj_controller + i; - kcontrol_template.name = c->name; - kcontrol_template.private_value = c->index; - kc = snd_ctl_new1(&kcontrol_template, dev); - snd_ctl_add(dev->chip.card, kc); - } - + ret = add_controls(a8dj_controller, + ARRAY_SIZE(a8dj_controller), dev); break; } - return 0; + return ret; } diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 83175083e50f..b143ef7152f7 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,7 +42,7 @@ #endif MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.8"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.9"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 5962e4b84423..6d9f9b135c62 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -880,7 +880,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream, snd_rawmidi_transmit_ack(substream, 1); return; } - tasklet_hi_schedule(&port->ep->tasklet); + tasklet_schedule(&port->ep->tasklet); } } diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index ff23cc1ce3b9..70b96355ca4c 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -276,7 +276,8 @@ static void subs_set_complete(struct urb **urbs, void (*complete)(struct urb *)) } } -int usb_stream_prepare_playback(struct usb_stream_kernel *sk, struct urb *inurb) +static int usb_stream_prepare_playback(struct usb_stream_kernel *sk, + struct urb *inurb) { struct usb_stream *s = sk->s; struct urb *io; |