summaryrefslogtreecommitdiffstats
path: root/Documentation
diff options
context:
space:
mode:
authorLinus Torvalds <torvalds@linux-foundation.org>2012-03-22 13:00:13 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2012-03-22 13:00:13 -0700
commitb2094ef840697bc8ca5d17a83b7e30fad5f1e9fa (patch)
tree64e5f7253b6a85b6d5d36f95c0d3c67c1798918d /Documentation
parent424a6f6ef990b7e9f56f6627bfc6c46b493faeb4 (diff)
parent6681bc0deba495fad0d6fb349e40524abd1b1732 (diff)
downloadlinux-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.txt24
-rw-r--r--Documentation/devicetree/bindings/sound/imx-audmux.txt13
-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.txt59
-rw-r--r--Documentation/devicetree/bindings/vendor-prefixes.txt1
-rw-r--r--Documentation/sound/alsa/ALSA-Configuration.txt8
-rw-r--r--Documentation/sound/alsa/HD-Audio-Models.txt79
-rw-r--r--Documentation/sound/alsa/HD-Audio.txt7
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.