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author | Linus Torvalds <torvalds@linux-foundation.org> | 2012-03-22 13:00:13 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2012-03-22 13:00:13 -0700 |
commit | b2094ef840697bc8ca5d17a83b7e30fad5f1e9fa (patch) | |
tree | 64e5f7253b6a85b6d5d36f95c0d3c67c1798918d /Documentation | |
parent | 424a6f6ef990b7e9f56f6627bfc6c46b493faeb4 (diff) | |
parent | 6681bc0deba495fad0d6fb349e40524abd1b1732 (diff) | |
download | linux-b2094ef840697bc8ca5d17a83b7e30fad5f1e9fa.tar.bz2 |
Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull updates of sound stuff from Takashi Iwai:
"Here is the first big update chunk of sound stuff for 3.4-rc1.
In the common sound infrastructure, there are a few changes for
dynamic PCM support (used in ASoC) and a few clean-ups. Majority of
changes are found, as usual, in HD-audio and ASoC.
Some highlights of HD-audio changes:
- All the long-standing static quirk codes for Realtek codec were
finally removed by fixing and extending the Realtek auto-parser.
- The mute-LED control is standardized over all HD-audio codec
drivers using the extended vmaster hook.
- The vmaster slave mixer elements are initialized to 0dB as default
so that the user won't be annoyed by the silent output after
updates, e.g. due to the additions of new elements.
- Other many fix-ups for the misc HD-audio devices.
In the ASoC side, this is a very active release, including a quite a
few framework enhancements. Some highlights:
- Support for widgets not associated with a CODEC, an important part
of the dynamic PCM framework.
- A library factoring out the common code shared by dmaengine based
DMA drivers contributed by Lars-Peter Clausen. This will save a
lot of code and make it much easier to deploy enhancements to
dmaengine.
- Support for binary controls, used for providing runtime
configuration of algorithm coefficients.
- A new DAPM widget type for regulator supplies allowing drivers for
devices that can power down unused supplies while active to do
without any per-driver code.
- DAPM widgets for DAIs, initially giving a speed boost for playback
startup and shutdown and also the basis for CODEC<->CODEC DAI link
support.
- Support for specifying the number of significant bits on audio
interfaces, useful for allowing applications to know how much
effort to put into generating data for a larger sample format.
- Conversion of the FSI driver used on some SH processors to
DMAEngine.
- Conversion of EP93xx drivers to DMAEngine.
- New CODEC drivers for Maxim MAX9768 and Wolfson Microelectronics
WM2200.
- Move audmux driver from arc/arm to sound/soc
- McBSP move from arch/ to sound/ and updates
Also, a few small updates and fixes for other drivers like au88x0,
ymfpci, USB 6fire, USB usx2yaudio are included."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (446 commits)
ASoC: wm8994: Provide VMID mode control and fix default sequence
ASoC: wm8994: Add missing break in resume
ASoC: wm_hubs: Don't actively manage LINEOUT_VMID_BUF
ASoC: pxa-ssp: atomically set stream active masks
ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master
ASoC: Samsung: Added to support mono recording
ALSA: hda - Fix build with CONFIG_PM=n
ALSA: au88x0 - Avoid possible Oops at unbinding
ALSA: usb-audio - Fix build error by consitification of rate list
ASoC: core: Fix obscure leak of runtime array
ALSA: pcm - Avoid GFP_ATOMIC in snd_pcm_link()
ALSA: pcm: Constify the list in snd_pcm_hw_constraint_list
ASoC: wm8996: Add 44.1kHz support
ALSA: hda - Fix build of patch_sigmatel.c without CONFIG_SND_HDA_POWER_SAVE
ASoC: mx27vis-aic32x4: Convert it to platform driver
ALSA: hda - fix printing of high HDMI sample rates
ALSA: ymfpci - Fix legacy registers on S3/S4 resume
ALSA: control - Fixe a trailing white space error
ALSA: hda - Add expose_enum_ctl flag to snd_hda_add_vmaster_hook()
ALSA: hda - Add "Mute-LED Mode" enum control
...
Diffstat (limited to 'Documentation')
-rw-r--r-- | Documentation/devicetree/bindings/sound/alc5632.txt | 24 | ||||
-rw-r--r-- | Documentation/devicetree/bindings/sound/imx-audmux.txt | 13 | ||||
-rw-r--r-- | Documentation/devicetree/bindings/sound/sgtl5000.txt (renamed from Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt) | 0 | ||||
-rw-r--r-- | Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt | 59 | ||||
-rw-r--r-- | Documentation/devicetree/bindings/vendor-prefixes.txt | 1 | ||||
-rw-r--r-- | Documentation/sound/alsa/ALSA-Configuration.txt | 8 | ||||
-rw-r--r-- | Documentation/sound/alsa/HD-Audio-Models.txt | 79 | ||||
-rw-r--r-- | Documentation/sound/alsa/HD-Audio.txt | 7 |
8 files changed, 112 insertions, 79 deletions
diff --git a/Documentation/devicetree/bindings/sound/alc5632.txt b/Documentation/devicetree/bindings/sound/alc5632.txt new file mode 100644 index 000000000000..8608f747dcfe --- /dev/null +++ b/Documentation/devicetree/bindings/sound/alc5632.txt @@ -0,0 +1,24 @@ +ALC5632 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "realtek,alc5632" + + - reg : the I2C address of the device. + + - gpio-controller : Indicates this device is a GPIO controller. + + - #gpio-cells : Should be two. The first cell is the pin number and the + second cell is used to specify optional parameters (currently unused). + +Example: + +alc5632: alc5632@1e { + compatible = "realtek,alc5632"; + reg = <0x1a>; + + gpio-controller; + #gpio-cells = <2>; +}; diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.txt b/Documentation/devicetree/bindings/sound/imx-audmux.txt new file mode 100644 index 000000000000..215aa9817213 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audmux.txt @@ -0,0 +1,13 @@ +Freescale Digital Audio Mux (AUDMUX) device + +Required properties: +- compatible : "fsl,imx21-audmux" for AUDMUX version firstly used on i.MX21, + or "fsl,imx31-audmux" for the version firstly used on i.MX31. +- reg : Should contain AUDMUX registers location and length + +Example: + +audmux@021d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; +}; diff --git a/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 2c3cd413f042..2c3cd413f042 100644 --- a/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt diff --git a/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt b/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt new file mode 100644 index 000000000000..b77a97c9101e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt @@ -0,0 +1,59 @@ +NVIDIA Tegra audio complex + +Required properties: +- compatible : "nvidia,tegra-audio-alc5632" +- nvidia,model : The user-visible name of this sound complex. +- nvidia,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the ALC5632's pins: + + ALC5632 pins: + + * SPK_OUTP + * SPK_OUTN + * HP_OUT_L + * HP_OUT_R + * AUX_OUT_P + * AUX_OUT_N + * LINE_IN_L + * LINE_IN_R + * PHONE_P + * PHONE_N + * MIC1_P + * MIC1_N + * MIC2_P + * MIC2_N + * MICBIAS1 + * DMICDAT + + Board connectors: + + * Headset Stereophone + * Int Spk + * Headset Mic + * Digital Mic + +- nvidia,i2s-controller : The phandle of the Tegra I2S controller +- nvidia,audio-codec : The phandle of the ALC5632 audio codec + +Example: + +sound { + compatible = "nvidia,tegra-audio-alc5632-paz00", + "nvidia,tegra-audio-alc5632"; + + nvidia,model = "Compal PAZ00"; + + nvidia,audio-routing = + "Int Spk", "SPK_OUTP", + "Int Spk", "SPK_OUTN", + "Headset Mic","MICBIAS1", + "MIC1_N", "Headset Mic", + "MIC1_P", "Headset Mic", + "Headset Stereophone", "HP_OUT_R", + "Headset Stereophone", "HP_OUT_L"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&alc5632>; +}; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index a20008ab319a..82ac057a24a9 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -34,6 +34,7 @@ picochip Picochip Ltd powervr Imagination Technologies qcom Qualcomm, Inc. ramtron Ramtron International +realtek Realtek Semiconductor Corp. samsung Samsung Semiconductor sbs Smart Battery System schindler Schindler diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 12e3a0fb9bec..6f75ba3b8a39 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -860,7 +860,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. [Multiple options for each card instance] model - force the model name - position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF) + position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF, + 3 = VIACOMBO, 4 = COMBO) probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) When the bit 8 (0x100) is set, the lower 8 bits are used as the "fixed" codec slots; i.e. the driver probes the @@ -925,6 +926,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. (Usually SD_LPIB register is more accurate than the position buffer.) + position_fix=3 is specific to VIA devices. The position + of the capture stream is checked from both LPIB and POSBUF + values. position_fix=4 is a combination mode, using LPIB + for playback and POSBUF for capture. + NB: If you get many "azx_get_response timeout" messages at loading, it's likely a problem of interrupts (e.g. ACPI irq routing). Try to boot with options like "pci=noacpi". Also, you diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index c8c54544abc5..d97d992ced14 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -8,37 +8,10 @@ ALC880 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 ====== - fujitsu Fujitsu S7020 - acer Acer TravelMate - will Will laptops (PB V7900) - replacer Replacer 672V - favorit100 Maxdata Favorit 100XS - 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) + N/A ALC262 ====== @@ -70,55 +43,7 @@ ALC680 ALC882/883/885/888/889 ====================== - 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 - mb5 Macbook 5,1 - macmini3 Macmini 3,1 - mba21 Macbook Air 2,1 - mbp3 Macbook Pro rev3 - imac24 iMac 24'' with jack detection - imac91 iMac 9,1 - w2jc ASUS W2JC - 3stack-2ch-dig 3-jack with SPDIF I/O (ALC883) - alc883-6stack-dig 6-jack digital with SPDIF I/O (ALC883) - 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 - acer-aspire-6530g Acer Aspire 6530G - acer-aspire-7730g Acer Aspire 7730G - acer-aspire-8930g Acer Aspire 8930G - medion Medion Laptops - targa-dig Targa/MSI - targa-2ch-dig Targa/MSI with 2-channel - targa-8ch-dig Targa/MSI with 8-channel (MSI GX620) - 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-m540r Clevo M540R (6ch + digital) - clevo-m720 Clevo M720 laptop series - fujitsu-pi2515 Fujitsu AMILO Pi2515 - fujitsu-xa3530 Fujitsu AMILO XA3530 - 3stack-6ch-intel Intel DG33* boards - intel-alc889a Intel IbexPeak with ALC889A - intel-x58 Intel DX58 with ALC889 - asus-p5q ASUS P5Q-EM boards - mb31 MacBook 3,1 - sony-vaio-tt Sony VAIO TT - auto auto-config reading BIOS (default) + N/A ALC861/660 ========== diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 91fee3b45fb8..7813c06a5c71 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -59,7 +59,12 @@ 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. `position_fix=3` means to use a combination of both methods, needed -for some VIA and ATI controllers. 0 is the default value for all other +for some VIA controllers. The capture stream position is corrected +by comparing both LPIB and position-buffer values. +`position_fix=4` is another combination available for all controllers, +and uses LPIB for the playback and the position-buffer for the capture +streams. +0 is the default value for all other controllers, the automatic check and fallback to LPIB as described in the above. If you get a problem of repeated sounds, this option might help. |