Merge tag 'sound-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "The biggest change in this update is the unification of HD-audio codec
  parsers.  Now the HD-audio codec is parsed in a generic parser code
  which is invoked by each HD-audio codec driver.

  Some background information is found in David Henningsson's blog
  entry:

      http://voices.canonical.com/david.henningsson/2013/01/18/upcoming-changes-to-the-intel-hda-drivers/

  Other than that, some random updates/fixes like USB-audio and a bunch
  of small AoC updates as usual.

  Highlights:

   - Unification of HD-audio parser code (aka generic parser)

   - Support of new Intel HD-audio controller, new IDT codecs

   - Fixes for HD-audio HDMI audio hotplug

   - Haswell HDMI audio fixup

   - Support of Creative CA0132 DSP code

   - A few fixes of HDSP driver

   - USB-audio fix for Roland A-PRO, M-Audio FT C600

   - Support PM for aloop driver (and fixes Oops)

   - Compress API updates for gapless playback support

  For ASoC part:

   - Support for a wider range of hardware in the compressed stream code

   - The ability to mute capture streams as well as playback streams
     while inactive

   - DT support for AK4642, FSI, Samsung I2S and WM8962

   - AC'97 support for Tegra

   - New driver for max98090, replacing the stub which was there

   - A new driver from Dialog

  Note that due to dependencies, DTification of DMA support for Samsung
  platforms (used only by the and I2S driver and SPI) is merged here as
  well."

Fix up trivial conflict in drivers/spi/spi-s3c64xx.c due to removed code
being changed.

* tag 'sound-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (453 commits)
  ALSA: usb: Fix Processing Unit Descriptor parsers
  ALSA: hda - hdmi: Notify userspace when ELD control changes
  ALSA: hda - hdmi: Protect ELD buffer
  ALSA: hda - hdmi: Refactor hdmi_eld into parsed_hdmi_eld
  ALSA: hda - hdmi: Do not expose eld data when eld is invalid
  ALSA: hda - hdmi: ELD shouldn't be valid after unplug
  ALSA: hda - Fix the silent speaker output on Fujitsu S7020 laptop
  ALSA: hda - add quirks for mute LED on two HP machines
  ALSA: usb/quirks, fix out-of-bounds access
  ASoC: codecs: Add da7213 codec
  ALSA: au88x0 - Define channel map for au88x0
  ALSA: compress: add support for gapless playback
  ALSA: hda - Remove speaker clicks on CX20549
  ALSA: hda - Disable runtime PM for Intel 5 Series/3400
  ALSA: hda - Increase badness for missing multi-io
  ASoC: arizona: Automatically manage input mutes
  ALSA: hda - Fix broken workaround for HDMI/SPDIF conflicts
  ALSA: hda/ca0132 - Add missing \n to debug prints
  ALSA: hda/ca0132 - Fix type of INVALID_CHIP_ADDRESS
  ALSA: hda - update documentation for no-primary-hp fixup
  ...

15 files changed, 453 insertions, 57 deletions

diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl
index fb32aead5a0..bd6fee22c4d 100644
--- a/Documentation/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl
@@ -871,9 +871,8 @@
       <para>
       This function itself doesn't allocate the data space. The data
       must be allocated manually beforehand, and its pointer is passed
-      as the argument. This pointer is used as the
-      (<parameter>chip</parameter> identifier in the above example)
-      for the instance. 
+      as the argument. This pointer (<parameter>chip</parameter> in the
+      above example) is used as the identifier for the instance.
       </para>
 
       <para>
@@ -2304,7 +2303,7 @@ struct _snd_pcm_runtime {
         <constant>SNDRV_PCM_INFO_XXX</constant>. Here, at least, you
         have to specify whether the mmap is supported and which
         interleaved format is supported.
-        When the is supported, add the
+        When the hardware supports mmap, add the
         <constant>SNDRV_PCM_INFO_MMAP</constant> flag here. When the
         hardware supports the interleaved or the non-interleaved
         formats, <constant>SNDRV_PCM_INFO_INTERLEAVED</constant> or
@@ -2898,7 +2897,7 @@ struct _snd_pcm_runtime {
 
         <para>
           When the pcm supports the pause operation (given in the info
-        field of the hardware table), the <constant>PAUSE_PUSE</constant>
+        field of the hardware table), the <constant>PAUSE_PUSH</constant>
         and <constant>PAUSE_RELEASE</constant> commands must be
         handled here, too. The former is the command to pause the pcm,
         and the latter to restart the pcm again. 
@@ -3085,7 +3084,7 @@ struct _snd_pcm_runtime {
       <section id="pcm-interface-interrupt-handler-timer">
         <title>High frequency timer interrupts</title>
         <para>
-	This happense when the hardware doesn't generate interrupts
+	This happens when the hardware doesn't generate interrupts
         at the period boundary but issues timer interrupts at a fixed
         timer rate (e.g. es1968 or ymfpci drivers). 
         In this case, you need to check the current hardware
@@ -3251,18 +3250,19 @@ struct _snd_pcm_runtime {
 	  <title>Example of Hardware Constraints for Channels</title>
 	  <programlisting>
 <![CDATA[
-  static int hw_rule_format_by_channels(struct snd_pcm_hw_params *params,
+  static int hw_rule_channels_by_format(struct snd_pcm_hw_params *params,
                                         struct snd_pcm_hw_rule *rule)
   {
           struct snd_interval *c = hw_param_interval(params,
-                SNDRV_PCM_HW_PARAM_CHANNELS);
+                        SNDRV_PCM_HW_PARAM_CHANNELS);
           struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
-          struct snd_mask fmt;
+          struct snd_interval ch;
 
-          snd_mask_any(&fmt);    /* Init the struct */
-          if (c->min < 2) {
-                  fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE;
-                  return snd_mask_refine(f, &fmt);
+          snd_interval_any(&ch);
+          if (f->bits[0] == SNDRV_PCM_FMTBIT_S16_LE) {
+                  ch.min = ch.max = 1;
+                  ch.integer = 1;
+                  return snd_interval_refine(c, &ch);
           }
           return 0;
   }
@@ -3278,35 +3278,35 @@ struct _snd_pcm_runtime {
 	 <programlisting>
 <![CDATA[
   snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-                      hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT,
-                      -1);
+                      hw_rule_channels_by_format, NULL,
+                      SNDRV_PCM_HW_PARAM_FORMAT, -1);
 ]]>
           </programlisting>
         </informalexample>
       </para>
 
       <para>
-        The rule function is called when an application sets the number of
-        channels. But an application can set the format before the number of
-        channels. Thus you also need to define the inverse rule:
+        The rule function is called when an application sets the PCM
+	format, and it refines the number of channels accordingly.
+        But an application may set the number of channels before
+	setting the format. Thus you also need to define the inverse rule:
 
        <example>
-	 <title>Example of Hardware Constraints for Channels</title>
+	 <title>Example of Hardware Constraints for Formats</title>
 	 <programlisting>
 <![CDATA[
-  static int hw_rule_channels_by_format(struct snd_pcm_hw_params *params,
+  static int hw_rule_format_by_channels(struct snd_pcm_hw_params *params,
                                         struct snd_pcm_hw_rule *rule)
   {
           struct snd_interval *c = hw_param_interval(params,
-                        SNDRV_PCM_HW_PARAM_CHANNELS);
+                SNDRV_PCM_HW_PARAM_CHANNELS);
           struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
-          struct snd_interval ch;
+          struct snd_mask fmt;
 
-          snd_interval_any(&ch);
-          if (f->bits[0] == SNDRV_PCM_FMTBIT_S16_LE) {
-                  ch.min = ch.max = 1;
-                  ch.integer = 1;
-                  return snd_interval_refine(c, &ch);
+          snd_mask_any(&fmt);    /* Init the struct */
+          if (c->min < 2) {
+                  fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE;
+                  return snd_mask_refine(f, &fmt);
           }
           return 0;
   }
@@ -3321,8 +3321,8 @@ struct _snd_pcm_runtime {
 	 <programlisting>
 <![CDATA[
   snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
-                      hw_rule_format_by_channels, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-                      -1);
+                      hw_rule_format_by_channels, NULL,
+                      SNDRV_PCM_HW_PARAM_CHANNELS, -1);
 ]]>
           </programlisting>
         </informalexample>
diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt
new file mode 100644
index 00000000000..623d4e70ae1
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ak4642.txt
@@ -0,0 +1,17 @@
+AK4642 I2C transmitter
+
+This device supports I2C mode only.
+
+Required properties:
+
+  - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648"
+  - reg : The chip select number on the I2C bus
+
+Example:
+
+&i2c {
+	ak4648: ak4648@0x12 {
+		compatible = "asahi-kasei,ak4642";
+		reg = <0x12>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt
index a850fb9c88e..e2cd1d7539e 100644
--- a/Documentation/devicetree/bindings/sound/cs4271.txt
+++ b/Documentation/devicetree/bindings/sound/cs4271.txt
@@ -20,6 +20,18 @@ Optional properties:
 		!RESET pin
  - cirrus,amuteb-eq-bmutec:	When given, the Codec's AMUTEB=BMUTEC flag
 				is enabled.
+ - cirrus,enable-soft-reset:
+	The CS4271 requires its LRCLK and MCLK to be stable before its RESET
+	line is de-asserted. That also means that clocks cannot be changed
+	without putting the chip back into hardware reset, which also requires
+	a complete re-initialization of all registers.
+
+	One (undocumented) workaround is to assert and de-assert the PDN bit
+	in the MODE2 register. This workaround can be enabled with this DT
+	property.
+
+	Note that this is not needed in case the clocks are stable
+	throughout the entire runtime of the codec.
 
 Examples:
 
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt
new file mode 100644
index 00000000000..be35d34e8b2
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt
@@ -0,0 +1,51 @@
+NVIDIA Tegra audio complex
+
+Required properties:
+- compatible : "nvidia,tegra-audio-wm9712"
+- 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 WM9712's pins, and the jacks on the board:
+
+  WM9712 pins:
+
+  * MONOOUT
+  * HPOUTL
+  * HPOUTR
+  * LOUT2
+  * ROUT2
+  * OUT3
+  * LINEINL
+  * LINEINR
+  * PHONE
+  * PCBEEP
+  * MIC1
+  * MIC2
+  * Mic Bias
+
+  Board connectors:
+
+  * Headphone
+  * LineIn
+  * Mic
+
+- nvidia,ac97-controller : The phandle of the Tegra AC97 controller
+
+
+Example:
+
+sound {
+	compatible = "nvidia,tegra-audio-wm9712-colibri_t20",
+		         "nvidia,tegra-audio-wm9712";
+	nvidia,model = "Toradex Colibri T20";
+
+	nvidia,audio-routing =
+		"Headphone", "HPOUTL",
+		"Headphone", "HPOUTR",
+		"LineIn", "LINEINL",
+		"LineIn", "LINEINR",
+		"Mic", "MIC1";
+
+	nvidia,ac97-controller = <&ac97>;
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt
new file mode 100644
index 00000000000..c1454979c1e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt
@@ -0,0 +1,22 @@
+NVIDIA Tegra 20 AC97 controller
+
+Required properties:
+- compatible : "nvidia,tegra20-ac97"
+- reg : Should contain AC97 controller registers location and length
+- interrupts : Should contain AC97 interrupt
+- nvidia,dma-request-selector : The Tegra DMA controller's phandle and
+  request selector for the AC97 controller
+- nvidia,codec-reset-gpio : The Tegra GPIO controller's phandle and the number
+  of the GPIO used to reset the external AC97 codec
+- nvidia,codec-sync-gpio : The Tegra GPIO controller's phandle and the number
+  of the GPIO corresponding with the AC97 DAP _FS line
+Example:
+
+ac97@70002000 {
+	compatible = "nvidia,tegra20-ac97";
+	reg = <0x70002000 0x200>;
+	interrupts = <0 81 0x04>;
+	nvidia,dma-request-selector = <&apbdma 12>;
+	nvidia,codec-reset-gpio = <&gpio 170 0>;
+	nvidia,codec-sync-gpio = <&gpio 120 0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/omap-twl4030.txt b/Documentation/devicetree/bindings/sound/omap-twl4030.txt
index 6fae51c7f76..1ab6bc8404d 100644
--- a/Documentation/devicetree/bindings/sound/omap-twl4030.txt
+++ b/Documentation/devicetree/bindings/sound/omap-twl4030.txt
@@ -6,6 +6,52 @@ Required properties:
 - ti,mcbsp: phandle for the McBSP node
 - ti,codec: phandle for the twl4030 audio node
 
+Optional properties:
+- ti,mcbsp-voice: phandle for the McBSP node connected to the voice port of twl
+- ti, jack-det-gpio: Jack detect GPIO
+- ti,audio-routing: List of 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.
+  If the routing is not provided all possible connection will be available
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headset Stereophone
+ * Earpiece Spk
+ * Handsfree Spk
+ * Ext Spk
+ * Main Mic
+ * Sub Mic
+ * Headset Mic
+ * Carkit Mic
+ * Digital0 Mic
+ * Digital1 Mic
+ * Line In
+
+twl4030 pins:
+ * HSOL
+ * HSOR
+ * EARPIECE
+ * HFL
+ * HFR
+ * PREDRIVEL
+ * PREDRIVER
+ * CARKITL
+ * CARKITR
+ * MAINMIC
+ * SUBMIC
+ * HSMIC
+ * DIGIMIC0
+ * DIGIMIC1
+ * CARKITMIC
+ * AUXL
+ * AUXR
+
+ * Headset Mic Bias
+ * Mic Bias 1 /* Used for Main Mic or Digimic0 */
+ * Mic Bias 2 /* Used for Sub Mic or Digimic1 */
+
 Example:
 
 sound {
diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.txt b/Documentation/devicetree/bindings/sound/renesas,fsi.txt
new file mode 100644
index 00000000000..c5be003f413
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/renesas,fsi.txt
@@ -0,0 +1,26 @@
+Renesas FSI
+
+Required properties:
+- compatible			: "renesas,sh_fsi2" or "renesas,sh_fsi"
+- reg				: Should contain the register physical address and length
+- interrupts			: Should contain FSI interrupt
+
+- fsia,spdif-connection		: FSI is connected by S/PDFI
+- fsia,stream-mode-support	: FSI supports 16bit stream mode.
+- fsia,use-internal-clock	: FSI uses internal clock when master mode.
+
+- fsib,spdif-connection		: same as fsia
+- fsib,stream-mode-support	: same as fsia
+- fsib,use-internal-clock	: same as fsia
+
+Example:
+
+sh_fsi2: sh_fsi2@0xec230000 {
+	compatible = "renesas,sh_fsi2";
+	reg = <0xec230000 0x400>;
+	interrupts = <0 146 0x4>;
+
+	fsia,spdif-connection;
+	fsia,stream-mode-support;
+	fsia,use-internal-clock;
+};
diff --git a/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt b/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt
new file mode 100644
index 00000000000..4686646fb12
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt
@@ -0,0 +1,14 @@
+Samsung SMDK audio complex
+
+Required properties:
+- compatible : "samsung,smdk-wm8994"
+- samsung,i2s-controller: The phandle of the Samsung I2S0 controller
+- samsung,audio-codec: The phandle of the WM8994 audio codec
+Example:
+
+sound {
+		compatible = "samsung,smdk-wm8994";
+
+		samsung,i2s-controller = <&i2s0>;
+		samsung,audio-codec = <&wm8994>;
+};
diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt
new file mode 100644
index 00000000000..3070046da2e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt
@@ -0,0 +1,63 @@
+* Samsung I2S controller
+
+Required SoC Specific Properties:
+
+- compatible : "samsung,i2s-v5"
+- reg: physical base address of the controller and length of memory mapped
+  region.
+- dmas: list of DMA controller phandle and DMA request line ordered pairs.
+- dma-names: identifier string for each DMA request line in the dmas property.
+  These strings correspond 1:1 with the ordered pairs in dmas.
+
+Optional SoC Specific Properties:
+
+- samsung,supports-6ch: If the I2S Primary sound source has 5.1 Channel
+  support, this flag is enabled.
+- samsung,supports-rstclr: This flag should be set if I2S software reset bit
+  control is required. When this flag is set I2S software reset bit will be
+  enabled or disabled based on need.
+- samsung,supports-secdai:If I2S block has a secondary FIFO and internal DMA,
+  then this flag is enabled.
+- samsung,idma-addr: Internal DMA register base address of the audio
+  sub system(used in secondary sound source).
+
+Required Board Specific Properties:
+
+- gpios: The gpio specifier for data out,data in, LRCLK, CDCLK and SCLK
+  interface lines. The format of the gpio specifier depends on the gpio
+  controller.
+  The syntax of samsung gpio specifier is
+	<[phandle of the gpio controller node]
+	 [pin number within the gpio controller]
+	 [mux function]
+	 [flags and pull up/down]
+	 [drive strength]>
+
+Example:
+
+- SoC Specific Portion:
+
+i2s@03830000 {
+	compatible = "samsung,i2s-v5";
+	reg = <0x03830000 0x100>;
+	dmas = <&pdma0 10
+		&pdma0 9
+		&pdma0 8>;
+	dma-names = "tx", "rx", "tx-sec";
+	samsung,supports-6ch;
+	samsung,supports-rstclr;
+	samsung,supports-secdai;
+	samsung,idma-addr = <0x03000000>;
+};
+
+- Board Specific Portion:
+
+i2s@03830000 {
+	gpios = <&gpz 0 2 0 0>, /* I2S_0_SCLK */
+		<&gpz 1 2 0 0>, /* I2S_0_CDCLK */
+		<&gpz 2 2 0 0>, /* I2S_0_LRCK */
+		<&gpz 3 2 0 0>, /* I2S_0_SDI */
+		<&gpz 4 2 0 0>, /* I2S_0_SDO[1] */
+		<&gpz 5 2 0 0>, /* I2S_0_SDO[2] */
+		<&gpz 6 2 0 0>; /* I2S_0_SDO[3] */
+};
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
index e7b98f41fa5..f47c3f589fd 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
@@ -11,6 +11,12 @@ Optional properties:
 
 - gpio-reset - gpio pin number used for codec reset
 - ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality
+- ai3x-micbias-vg - MicBias Voltage required.
+	1 - MICBIAS output is powered to 2.0V,
+	2 - MICBIAS output is powered to 2.5V,
+	3 - MICBIAS output is connected to AVDD,
+	If this node is not mentioned or if the value is incorrect, then MicBias
+	is powered down.
 
 Example:
 
diff --git a/Documentation/devicetree/bindings/sound/wm8962.txt b/Documentation/devicetree/bindings/sound/wm8962.txt
new file mode 100644
index 00000000000..dceb3b1c2bb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8962.txt
@@ -0,0 +1,16 @@
+WM8962 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "wlf,wm8962"
+
+  - reg : the I2C address of the device.
+
+Example:
+
+codec: wm8962@1a {
+	compatible = "wlf,wm8962";
+	reg = <0x1a>;
+};
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index b9cfd339a6f..ce6581c8ca2 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -890,8 +890,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     enable_msi	- Enable Message Signaled Interrupt (MSI) (default = off)
     power_save	- Automatic power-saving timeout (in second, 0 =
 		disable)
-    power_save_controller - Reset HD-audio controller in power-saving mode
-		(default = on)
+    power_save_controller - Support runtime D3 of HD-audio controller
+		(-1 = on for supported chip (default), false = off,
+		 true = force to on even for unsupported hardware)
     align_buffer_size - Force rounding of buffer/period sizes to multiples
     		      of 128 bytes. This is more efficient in terms of memory
 		      access but isn't required by the HDA spec and prevents
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 16dfe57f173..bb8b0dc532b 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -53,7 +53,7 @@ ALC882/883/885/888/889
   acer-aspire-8930g	Acer Aspire 8330G/6935G
   acer-aspire		Acer Aspire others
   inv-dmic	Inverted internal mic workaround
-  no-primary-hp		VAIO Z workaround (for fixed speaker DAC)
+  no-primary-hp		VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC)
 
 ALC861/660
 ==========
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index 7813c06a5c7..d4faa63ff35 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -176,14 +176,14 @@ 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.
+The preset model (or recently called as "fix-up") 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 with the correct pin setup, etc.
+Thus, if you have a newer machine with a slightly different PCI SSID
+(or codec 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 (and codec-) 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"
@@ -199,17 +199,12 @@ non-working HD-audio hardware is to check HD-audio codec and several
 different `model` option values.  If you have any 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.
+There are a few special model option values:
+- when 'nofixup' is passed, the device-specific fixups in the codec
+  parser are skipped.
+- when `generic` is passed, the codec-specific parser is skipped and
+  only the generic parser is used.
 
-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
 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~
@@ -387,9 +382,8 @@ init_verbs::
   (separated with a space).
 hints::
   Shows / stores hint strings for codec parsers for any use.
-  Its format is `key = value`.  For example, passing `hp_detect = yes`
-  to IDT/STAC codec parser will result in the disablement of the
-  headphone detection.
+  Its format is `key = value`.  For example, passing `jack_detect = no`
+  will disable the jack detection of the machine completely.
 init_pin_configs::
   Shows the initial pin default config values set by BIOS.
 driver_pin_configs::
@@ -421,6 +415,61 @@ re-configure based on that state, run like below:
 ------------------------------------------------------------------------
 
 
+Hint Strings
+~~~~~~~~~~~~
+The codec parser have several switches and adjustment knobs for
+matching better with the actual codec or device behavior.  Many of
+them can be adjusted dynamically via "hints" strings as mentioned in
+the section above.  For example, by passing `jack_detect = no` string
+via sysfs or a patch file, you can disable the jack detection, thus
+the codec parser will skip the features like auto-mute or mic
+auto-switch.  As a boolean value, either `yes`, `no`, `true`, `false`,
+`1` or `0` can be passed.
+
+The generic parser supports the following hints:
+
+- jack_detect (bool): specify whether the jack detection is available
+  at all on this machine; default true
+- inv_jack_detect (bool): indicates that the jack detection logic is
+  inverted
+- trigger_sense (bool): indicates that the jack detection needs the
+  explicit call of AC_VERB_SET_PIN_SENSE verb
+- inv_eapd (bool): indicates that the EAPD is implemented in the
+  inverted logic
+- pcm_format_first (bool): sets the PCM format before the stream tag
+  and channel ID
+- sticky_stream (bool): keep the PCM format, stream tag and ID as long
+  as possible; default true
+- spdif_status_reset (bool): reset the SPDIF status bits at each time
+  the SPDIF stream is set up
+-  pin_amp_workaround (bool): the output pin may have multiple amp
+  values
+- single_adc_amp (bool): ADCs can have only single input amps
+- auto_mute (bool): enable/disable the headphone auto-mute feature;
+  default true
+- auto_mic (bool): enable/disable the mic auto-switch feature; default
+  true
+- line_in_auto_switch (bool): enable/disable the line-in auto-switch
+  feature; default false
+- need_dac_fix (bool): limits the DACs depending on the channel count
+- primary_hp (bool): probe headphone jacks as the primary outputs;
+  default true
+- multi_cap_vol (bool): provide multiple capture volumes
+- inv_dmic_split (bool): provide split internal mic volume/switch for
+  phase-inverted digital mics
+- indep_hp (bool): provide the independent headphone PCM stream and
+  the corresponding mixer control, if available
+- add_stereo_mix_input (bool): add the stereo mix (analog-loopback
+  mix) to the input mux if available
+- add_out_jack_modes (bool): add "xxx Jack Mode" enum controls to each
+  output jack for allowing to change the headphone amp capability
+- add_in_jack_modes (bool): add "xxx Jack Mode" enum controls to each
+  input jack for allowing to change the mic bias vref
+- power_down_unused (bool): power down the unused widgets
+- mixer_nid (int): specifies the widget NID of the analog-loopback
+  mixer
+
+
 Early Patching
 ~~~~~~~~~~~~~~
 When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a
@@ -445,7 +494,7 @@ A patch file is a plain text file which looks like below:
   0x20 0x400 0xff
 
   [hint]
-  hp_detect = yes
+  jack_detect = no
 ------------------------------------------------------------------------
 
 The file needs to have a line `[codec]`.  The next line should contain
@@ -531,6 +580,13 @@ 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.
 
+The recent kernel supports the runtime PM for the HD-audio controller
+chip, too.  It means that the HD-audio controller is also powered up /
+down dynamically.  The feature is enabled only for certain controller
+chips like Intel LynxPoint.  You can enable/disable this feature
+forcibly by setting `power_save_controller` option, which is also
+available at /sys/module/snd_hda_intel/parameters directory.
+
 
 Tracepoints
 ~~~~~~~~~~~
@@ -587,8 +643,9 @@ The latest development codes for HD-audio are found on sound git tree:
 - git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.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.
+development branches in general while the development for the current
+and next kernels are found in for-linus and for-next branches,
+respectively.
 
 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
@@ -699,7 +756,11 @@ 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:
+The hda-verb program is included now in alsa-tools:
+
+- git://git.alsa-project.org/alsa-tools.git
+
+Also, the old stand-alone package is found in the ftp directory:
 
 - ftp://ftp.suse.com/pub/people/tiwai/misc/
 
@@ -777,3 +838,18 @@ A git repository is available:
 
 See README file in the tarball for more details about hda-emu
 program.
+
+
+hda-jack-retask
+~~~~~~~~~~~~~~~
+hda-jack-retask is a user-friendly GUI program to manipulate the
+HD-audio pin control for jack retasking.  If you have a problem about
+the jack assignment, try this program and check whether you can get
+useful results.  Once when you figure out the proper pin assignment,
+it can be fixed either in the driver code statically or via passing a
+firmware patch file (see "Early Patching" section).
+
+The program is included in alsa-tools now:
+
+- git://git.alsa-project.org/alsa-tools.git
+
diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt
index 90e9b3a11ab..0bcc5515591 100644
--- a/Documentation/sound/alsa/compress_offload.txt
+++ b/Documentation/sound/alsa/compress_offload.txt
@@ -145,6 +145,52 @@ Modifications include:
 - Addition of encoding options when required (derived from OpenMAX IL)
 - Addition of rateControlSupported (missing in OpenMAX AL)
 
+Gapless Playback
+================
+When playing thru an album, the decoders have the ability to skip the encoder
+delay and padding and directly move from one track content to another. The end
+user can perceive this as gapless playback as we dont have silence while
+switching from one track to another
+
+Also, there might be low-intensity noises due to encoding. Perfect gapless is
+difficult to reach with all types of compressed data, but works fine with most
+music content. The decoder needs to know the encoder delay and encoder padding.
+So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers
+and are not present by default in the bitstream, hence the need for a new
+interface to pass this information to the DSP. Also DSP and userspace needs to
+switch from one track to another and start using data for second track.
+
+The main additions are:
+
+- set_metadata
+This routine sets the encoder delay and encoder padding. This can be used by
+decoder to strip the silence. This needs to be set before the data in the track
+is written.
+
+- set_next_track
+This routine tells DSP that metadata and write operation sent after this would
+correspond to subsequent track
+
+- partial drain
+This is called when end of file is reached. The userspace can inform DSP that
+EOF is reached and now DSP can start skipping padding delay. Also next write
+data would belong to next track
+
+Sequence flow for gapless would be:
+- Open
+- Get caps / codec caps
+- Set params
+- Set metadata of the first track
+- Fill data of the first track
+- Trigger start
+- User-space finished sending all,
+- Indicaite next track data by sending set_next_track
+- Set metadata of the next track
+- then call partial_drain to flush most of buffer in DSP
+- Fill data of the next track
+- DSP switches to second track
+(note: order for partial_drain and write for next track can be reversed as well)
+
 Not supported:
 
 - Support for VoIP/circuit-switched calls is not the target of this