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
  ...

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>

Documentation/devicetree/bindings/sound/ak4642.txt

+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>;
+	};
+};

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:
 

Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt

+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>;
+};

Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt

+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>;
+};

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 {

Documentation/devicetree/bindings/sound/renesas,fsi.txt

+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;
+};

Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt

+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>;
+};

Documentation/devicetree/bindings/sound/samsung-i2s.txt

+* 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] */
+};

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:
 

Documentation/devicetree/bindings/sound/wm8962.txt

+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>;
+};

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

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
 ==========

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
+

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

arch/arm/boot/dts/exynos5250-smdk5250.dts

@@ -49,6 +49,11 @@ eeprom@51 {
 			compatible = "samsung,s524ad0xd1";
 			reg = <0x51>;
 		};
+
+		wm8994: wm8994@1a {
+			 compatible = "wlf,wm8994";
+			 reg = <0x1a>;
+		};
 	};
 
 	i2c@121D0000 {
@@ -204,4 +209,25 @@ codec@11000000 {
 		samsung,mfc-r = <0x43000000 0x800000>;
 		samsung,mfc-l = <0x51000000 0x800000>;
 	};
+
+	i2s0: i2s@03830000 {
+		gpios = <&gpz 0 2 0 0>, <&gpz 1 2 0 0>, <&gpz 2 2 0 0>,
+			<&gpz 3 2 0 0>, <&gpz 4 2 0 0>, <&gpz 5 2 0 0>,
+			<&gpz 6 2 0 0>;
+	};
+
+	i2s1: i2s@12D60000 {
+		status = "disabled";
+	};
+
+	i2s2: i2s@12D70000 {
+		status = "disabled";
+	};
+
+	sound {
+		compatible = "samsung,smdk-wm8994";
+
+		samsung,i2s-controller = <&i2s0>;
+		samsung,audio-codec = <&wm8994>;
+	};
 };

arch/arm/boot/dts/exynos5250.dtsi

@@ -211,8 +211,9 @@ spi_0: spi@12d20000 {
 		compatible = "samsung,exynos4210-spi";
 		reg = <0x12d20000 0x100>;
 		interrupts = <0 66 0>;
-		tx-dma-channel = <&pdma0 5>; /* preliminary */
-		rx-dma-channel = <&pdma0 4>; /* preliminary */
+		dmas = <&pdma0 5
+			&pdma0 4>;
+		dma-names = "tx", "rx";
 		#address-cells = <1>;
 		#size-cells = <0>;
 	};
@@ -221,8 +222,9 @@ spi_1: spi@12d30000 {
 		compatible = "samsung,exynos4210-spi";
 		reg = <0x12d30000 0x100>;
 		interrupts = <0 67 0>;
-		tx-dma-channel = <&pdma1 5>; /* preliminary */
-		rx-dma-channel = <&pdma1 4>; /* preliminary */
+		dmas = <&pdma1 5
+			&pdma1 4>;
+		dma-names = "tx", "rx";
 		#address-cells = <1>;
 		#size-cells = <0>;
 	};
@@ -231,8 +233,9 @@ spi_2: spi@12d40000 {
 		compatible = "samsung,exynos4210-spi";
 		reg = <0x12d40000 0x100>;
 		interrupts = <0 68 0>;
-		tx-dma-channel = <&pdma0 7>; /* preliminary */
-		rx-dma-channel = <&pdma0 6>; /* preliminary */
+		dmas = <&pdma0 7
+			&pdma0 6>;
+		dma-names = "tx", "rx";
 		#address-cells = <1>;
 		#size-cells = <0>;
 	};
@@ -269,6 +272,35 @@ dwmmc_3: dwmmc3@12230000 {
 		#size-cells = <0>;
 	};
 
+	i2s0: 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>;
+	};
+
+	i2s1: i2s@12D60000 {
+		compatible = "samsung,i2s-v5";
+		reg = <0x12D60000 0x100>;
+		dmas = <&pdma1 12
+			&pdma1 11>;
+		dma-names = "tx", "rx";
+	};
+
+	i2s2: i2s@12D70000 {
+		compatible = "samsung,i2s-v5";
+		reg = <0x12D70000 0x100>;
+		dmas = <&pdma0 12
+			&pdma0 11>;
+		dma-names = "tx", "rx";
+	};
+
 	amba {
 		#address-cells = <1>;
 		#size-cells = <1>;

arch/arm/mach-exynos/mach-exynos5-dt.c

@@ -104,6 +104,12 @@ static const struct of_dev_auxdata exynos5250_auxdata_lookup[] __initconst = {
 	OF_DEV_AUXDATA("samsung,mfc-v6", 0x11000000, "s5p-mfc-v6", NULL),
 	OF_DEV_AUXDATA("samsung,exynos5250-tmu", 0x10060000,
 				"exynos-tmu", NULL),
+	OF_DEV_AUXDATA("samsung,i2s-v5", 0x03830000,
+				"samsung-i2s.0", NULL),
+	OF_DEV_AUXDATA("samsung,i2s-v5", 0x12D60000,
+				"samsung-i2s.1", NULL),
+	OF_DEV_AUXDATA("samsung,i2s-v5", 0x12D70000,
+				"samsung-i2s.2", NULL),
 	{},
 };
 

arch/arm/mach-pxa/pxa27x.c

@@ -53,17 +53,25 @@ static unsigned long ac97_reset_config[] = {
 	GPIO95_AC97_nRESET,
 };
 
-void pxa27x_assert_ac97reset(int reset_gpio, int on)
+void pxa27x_configure_ac97reset(int reset_gpio, bool to_gpio)
 {
+	/*
+	 * This helper function is used to work around a bug in the pxa27x's
+	 * ac97 controller during a warm reset.  The configuration of the
+	 * reset_gpio is changed as follows:
+	 * to_gpio == true: configured to generic output gpio and driven high
+	 * to_gpio == false: configured to ac97 controller alt fn AC97_nRESET
+	 */
+
 	if (reset_gpio == 113)
-		pxa2xx_mfp_config(on ? &ac97_reset_config[0] :
-				       &ac97_reset_config[1], 1);
+		pxa2xx_mfp_config(to_gpio ? &ac97_reset_config[0] :
+				  &ac97_reset_config[1], 1);
 
 	if (reset_gpio == 95)
-		pxa2xx_mfp_config(on ? &ac97_reset_config[2] :
-				       &ac97_reset_config[3], 1);
+		pxa2xx_mfp_config(to_gpio ? &ac97_reset_config[2] :
+				  &ac97_reset_config[3], 1);
 }
-EXPORT_SYMBOL_GPL(pxa27x_assert_ac97reset);
+EXPORT_SYMBOL_GPL(pxa27x_configure_ac97reset);
 
 /* Crystal clock: 13MHz */
 #define BASE_CLK	13000000

arch/arm/mach-shmobile/board-ap4evb.c

@@ -657,14 +657,8 @@ static struct platform_device lcdc_device = {
 /* FSI */
 #define IRQ_FSI		evt2irq(0x1840)
 static struct sh_fsi_platform_info fsi_info = {
-	.port_a = {
-		.flags		= SH_FSI_BRS_INV,
-	},
 	.port_b = {
-		.flags		= SH_FSI_BRS_INV |
-				  SH_FSI_BRM_INV |
-				  SH_FSI_LRS_INV |
-				  SH_FSI_CLK_CPG |
+		.flags		= SH_FSI_CLK_CPG |
 				  SH_FSI_FMT_SPDIF,
 	},
 };
@@ -692,21 +686,21 @@ static struct platform_device fsi_device = {
 	},
 };
 
-static struct asoc_simple_dai_init_info fsi2_ak4643_init_info = {
-	.fmt		= SND_SOC_DAIFMT_LEFT_J,
-	.codec_daifmt	= SND_SOC_DAIFMT_CBM_CFM,
-	.cpu_daifmt	= SND_SOC_DAIFMT_CBS_CFS,
-	.sysclk		= 11289600,
-};
-
 static struct asoc_simple_card_info fsi2_ak4643_info = {
 	.name		= "AK4643",
 	.card		= "FSI2A-AK4643",
-	.cpu_dai	= "fsia-dai",
 	.codec		= "ak4642-codec.0-0013",
 	.platform	= "sh_fsi2",
-	.codec_dai	= "ak4642-hifi",
-	.init		= &fsi2_ak4643_init_info,
+	.daifmt		= SND_SOC_DAIFMT_LEFT_J,
+	.cpu_dai = {
+		.name	= "fsia-dai",
+		.fmt	= SND_SOC_DAIFMT_CBS_CFS,
+	},
+	.codec_dai = {
+		.name	= "ak4642-hifi",
+		.fmt	= SND_SOC_DAIFMT_CBM_CFM,
+		.sysclk	= 11289600,
+	},
 };
 
 static struct platform_device fsi_ak4643_device = {
@@ -815,18 +809,18 @@ static struct platform_device lcdc1_device = {
 	},
 };
 
-static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = {
-	.cpu_daifmt	= SND_SOC_DAIFMT_CBM_CFM,
-};
-
 static struct asoc_simple_card_info fsi2_hdmi_info = {
 	.name		= "HDMI",
 	.card		= "FSI2B-HDMI",
-	.cpu_dai	= "fsib-dai",
 	.codec		= "sh-mobile-hdmi",
 	.platform	= "sh_fsi2",
-	.codec_dai	= "sh_mobile_hdmi-hifi",
-	.init		= &fsi2_hdmi_init_info,
+	.cpu_dai = {
+		.name	= "fsib-dai",
+		.fmt	= SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF,
+	},
+	.codec_dai = {
+		.name	= "sh_mobile_hdmi-hifi",
+	},
 };
 
 static struct platform_device fsi_hdmi_device = {

arch/arm/mach-shmobile/board-armadillo800eva.c

@@ -806,21 +806,21 @@ static struct platform_device fsi_device = {
 };
 
 /* FSI-WM8978 */
-static struct asoc_simple_dai_init_info fsi_wm8978_init_info = {
-	.fmt		= SND_SOC_DAIFMT_I2S,
-	.codec_daifmt	= SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_NB_NF,
-	.cpu_daifmt	= SND_SOC_DAIFMT_CBS_CFS,
-	.sysclk		= 12288000,
-};
-
 static struct asoc_simple_card_info fsi_wm8978_info = {
 	.name		= "wm8978",
 	.card		= "FSI2A-WM8978",
-	.cpu_dai	= "fsia-dai",
 	.codec		= "wm8978.0-001a",
 	.platform	= "sh_fsi2",
-	.codec_dai	= "wm8978-hifi",
-	.init		= &fsi_wm8978_init_info,
+	.daifmt		= SND_SOC_DAIFMT_I2S,
+	.cpu_dai = {
+		.name	= "fsia-dai",
+		.fmt	= SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF,
+	},
+	.codec_dai = {
+		.name	= "wm8978-hifi",
+		.fmt	= SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_NB_NF,
+		.sysclk	= 12288000,
+	},
 };
 
 static struct platform_device fsi_wm8978_device = {
@@ -832,18 +832,18 @@ static struct platform_device fsi_wm8978_device = {
 };
 
 /* FSI-HDMI */
-static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = {
-	.cpu_daifmt	= SND_SOC_DAIFMT_CBM_CFM,
-};
-
 static struct asoc_simple_card_info fsi2_hdmi_info = {
 	.name		= "HDMI",
 	.card		= "FSI2B-HDMI",
-	.cpu_dai	= "fsib-dai",
 	.codec		= "sh-mobile-hdmi",
 	.platform	= "sh_fsi2",
-	.codec_dai	= "sh_mobile_hdmi-hifi",
-	.init		= &fsi2_hdmi_init_info,
+	.cpu_dai = {
+		.name	= "fsib-dai",
+		.fmt	= SND_SOC_DAIFMT_CBM_CFM,
+	},
+	.codec_dai = {
+		.name = "sh_mobile_hdmi-hifi",
+	},
 };
 
 static struct platform_device fsi_hdmi_device = {

arch/arm/mach-shmobile/board-kzm9g.c

@@ -525,21 +525,21 @@ static struct platform_device fsi_device = {
 	},
 };
 
-static struct asoc_simple_dai_init_info fsi2_ak4648_init_info = {
-	.fmt		= SND_SOC_DAIFMT_LEFT_J,
-	.codec_daifmt	= SND_SOC_DAIFMT_CBM_CFM,
-	.cpu_daifmt	= SND_SOC_DAIFMT_CBS_CFS,
-	.sysclk		= 11289600,
-};
-
 static struct asoc_simple_card_info fsi2_ak4648_info = {
 	.name		= "AK4648",
 	.card		= "FSI2A-AK4648",
-	.cpu_dai	= "fsia-dai",
 	.codec		= "ak4642-codec.0-0012",
 	.platform	= "sh_fsi2",
-	.codec_dai	= "ak4642-hifi",
-	.init		= &fsi2_ak4648_init_info,
+	.daifmt		= SND_SOC_DAIFMT_LEFT_J,
+	.cpu_dai = {
+		.name	= "fsia-dai",
+		.fmt	= SND_SOC_DAIFMT_CBS_CFS,
+	},
+	.codec_dai = {
+		.name	= "ak4642-hifi",
+		.fmt	= SND_SOC_DAIFMT_CBM_CFM,
+		.sysclk	= 11289600,
+	},
 };
 
 static struct platform_device fsi_ak4648_device = {

arch/arm/mach-shmobile/board-mackerel.c

@@ -502,18 +502,18 @@ static struct platform_device hdmi_lcdc_device = {
 	},
 };
 
-static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = {
-	.cpu_daifmt	= SND_SOC_DAIFMT_CBM_CFM,
-};
-
 static struct asoc_simple_card_info fsi2_hdmi_info = {
 	.name		= "HDMI",
 	.card		= "FSI2B-HDMI",
-	.cpu_dai	= "fsib-dai",
 	.codec		= "sh-mobile-hdmi",
 	.platform	= "sh_fsi2",
-	.codec_dai	= "sh_mobile_hdmi-hifi",
-	.init		= &fsi2_hdmi_init_info,
+	.cpu_dai = {
+		.name	= "fsib-dai",
+		.fmt	= SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF,
+	},
+	.codec_dai = {
+		.name	= "sh_mobile_hdmi-hifi",
+	},
 };
 
 static struct platform_device fsi_hdmi_device = {
@@ -858,16 +858,12 @@ static struct platform_device leds_device = {
 #define IRQ_FSI evt2irq(0x1840)
 static struct sh_fsi_platform_info fsi_info = {
 	.port_a = {
-		.flags = SH_FSI_BRS_INV,
 		.tx_id = SHDMA_SLAVE_FSIA_TX,
 		.rx_id = SHDMA_SLAVE_FSIA_RX,
 	},
 	.port_b = {
-		.flags = SH_FSI_BRS_INV	|
-			SH_FSI_BRM_INV	|
-			SH_FSI_LRS_INV	|
-			SH_FSI_CLK_CPG	|
-			SH_FSI_FMT_SPDIF,
+		.flags = SH_FSI_CLK_CPG	|
+			 SH_FSI_FMT_SPDIF,
 	}
 };
 
@@ -896,21 +892,21 @@ static struct platform_device fsi_device = {
 	},
 };
 
-static struct asoc_simple_dai_init_info fsi2_ak4643_init_info = {
-	.fmt		= SND_SOC_DAIFMT_LEFT_J,
-	.codec_daifmt	= SND_SOC_DAIFMT_CBM_CFM,
-	.cpu_daifmt	= SND_SOC_DAIFMT_CBS_CFS,
-	.sysclk		= 11289600,
-};
-
 static struct asoc_simple_card_info fsi2_ak4643_info = {
 	.name		= "AK4643",
 	.card		= "FSI2A-AK4643",
-	.cpu_dai	= "fsia-dai",
 	.codec		= "ak4642-codec.0-0013",
 	.platform	= "sh_fsi2",
-	.codec_dai	= "ak4642-hifi",
-	.init		= &fsi2_ak4643_init_info,
+	.daifmt		= SND_SOC_DAIFMT_LEFT_J,
+	.cpu_dai = {
+		.name	= "fsia-dai",
+		.fmt	= SND_SOC_DAIFMT_CBS_CFS,
+	},
+	.codec_dai = {
+		.name	= "ak4642-hifi",
+		.fmt	= SND_SOC_DAIFMT_CBM_CFM,
+		.sysclk	= 11289600,
+	},
 };
 
 static struct platform_device fsi_ak4643_device = {

arch/arm/plat-samsung/dma-ops.c

@@ -19,7 +19,8 @@
 #include <mach/dma.h>
 
 static unsigned samsung_dmadev_request(enum dma_ch dma_ch,
-				struct samsung_dma_req *param)
+				struct samsung_dma_req *param,
+				struct device *dev, char *ch_name)
 {
 	dma_cap_mask_t mask;
 	void *filter_param;
@@ -33,7 +34,12 @@ static unsigned samsung_dmadev_request(enum dma_ch dma_ch,
 	 */
 	filter_param = (dma_ch == DMACH_DT_PROP) ?
 		(void *)param->dt_dmach_prop : (void *)dma_ch;
-	return (unsigned)dma_request_channel(mask, pl330_filter, filter_param);
+
+	if (dev->of_node)
+		return (unsigned)dma_request_slave_channel(dev, ch_name);
+	else
+		return (unsigned)dma_request_channel(mask, pl330_filter,
+							filter_param);
 }
 
 static int samsung_dmadev_release(unsigned ch, void *param)

arch/arm/plat-samsung/include/plat/dma-ops.h

@@ -39,7 +39,8 @@ struct samsung_dma_config {
 };
 
 struct samsung_dma_ops {
-	unsigned (*request)(enum dma_ch ch, struct samsung_dma_req *param);
+	unsigned (*request)(enum dma_ch ch, struct samsung_dma_req *param,
+				struct device *dev, char *ch_name);
 	int (*release)(unsigned ch, void *param);
 	int (*config)(unsigned ch, struct samsung_dma_config *param);
 	int (*prepare)(unsigned ch, struct samsung_dma_prep *param);

arch/arm/plat-samsung/s3c-dma-ops.c

@@ -36,7 +36,8 @@ static void s3c_dma_cb(struct s3c2410_dma_chan *channel, void *param,
 }
 
 static unsigned s3c_dma_request(enum dma_ch dma_ch,
-					struct samsung_dma_req *param)
+				struct samsung_dma_req *param,
+				struct device *dev, char *ch_name)
 {
 	struct cb_data *data;
 

arch/sh/boards/mach-ecovec24/setup.c

@@ -887,12 +887,6 @@ static struct platform_device camera_devices[] = {
 };
 
 /* FSI */
-static struct sh_fsi_platform_info fsi_info = {
-	.port_b = {
-		.flags = SH_FSI_BRS_INV,
-	},
-};
-
 static struct resource fsi_resources[] = {
 	[0] = {
 		.name	= "FSI",
@@ -911,25 +905,22 @@ static struct platform_device fsi_device = {
 	.id		= 0,
 	.num_resources	= ARRAY_SIZE(fsi_resources),
 	.resource	= fsi_resources,
-	.dev	= {
-		.platform_data	= &fsi_info,
-	},
-};
-
-static struct asoc_simple_dai_init_info fsi_da7210_init_info = {
-	.fmt		= SND_SOC_DAIFMT_I2S,
-	.codec_daifmt	= SND_SOC_DAIFMT_CBM_CFM,
-	.cpu_daifmt	= SND_SOC_DAIFMT_CBS_CFS,
 };
 
 static struct asoc_simple_card_info fsi_da7210_info = {
 	.name		= "DA7210",
 	.card		= "FSIB-DA7210",
-	.cpu_dai	= "fsib-dai",
 	.codec		= "da7210.0-001a",
 	.platform	= "sh_fsi.0",
-	.codec_dai	= "da7210-hifi",
-	.init		= &fsi_da7210_init_info,
+	.daifmt		= SND_SOC_DAIFMT_I2S,
+	.cpu_dai = {
+		.name	= "fsib-dai",
+		.fmt	= SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF,
+	},
+	.codec_dai = {
+		.name	= "da7210-hifi",
+		.fmt	= SND_SOC_DAIFMT_CBM_CFM,
+	},
 };
 
 static struct platform_device fsi_da7210_device = {

arch/sh/boards/mach-se/7724/setup.c

@@ -279,12 +279,6 @@ static struct platform_device ceu1_device = {
 
 /* FSI */
 /* change J20, J21, J22 pin to 1-2 connection to use slave mode */
-static struct sh_fsi_platform_info fsi_info = {
-	.port_a = {
-		.flags = SH_FSI_BRS_INV,
-	},
-};
-
 static struct resource fsi_resources[] = {
 	[0] = {
 		.name	= "FSI",
@@ -303,26 +297,23 @@ static struct platform_device fsi_device = {
 	.id		= 0,
 	.num_resources	= ARRAY_SIZE(fsi_resources),
 	.resource	= fsi_resources,
-	.dev	= {
-		.platform_data	= &fsi_info,
-	},
-};
-
-static struct asoc_simple_dai_init_info fsi2_ak4642_init_info = {
-	.fmt		= SND_SOC_DAIFMT_LEFT_J,
-	.codec_daifmt	= SND_SOC_DAIFMT_CBM_CFM,
-	.cpu_daifmt	= SND_SOC_DAIFMT_CBS_CFS,
-	.sysclk		= 11289600,
 };
 
 static struct asoc_simple_card_info fsi_ak4642_info = {
 	.name		= "AK4642",
 	.card		= "FSIA-AK4642",
-	.cpu_dai	= "fsia-dai",
 	.codec		= "ak4642-codec.0-0012",
 	.platform	= "sh_fsi.0",
-	.codec_dai	= "ak4642-hifi",
-	.init		= &fsi2_ak4642_init_info,
+	.daifmt		= SND_SOC_DAIFMT_LEFT_J,
+	.cpu_dai = {
+		.name	= "fsia-dai",
+		.fmt	= SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF,
+	},
+	.codec_dai = {
+		.name	= "ak4642-hifi",
+		.fmt	= SND_SOC_DAIFMT_CBM_CFM,
+		.sysclk	= 11289600,
+	},
 };
 
 static struct platform_device fsi_ak4642_device = {

drivers/misc/Kconfig

@@ -192,7 +192,7 @@ config ICS932S401
 
 config ATMEL_SSC
 	tristate "Device driver for Atmel SSC peripheral"
-	depends on AVR32 || ARCH_AT91
+	depends on HAS_IOMEM
 	---help---
 	  This option enables device driver support for Atmel Synchronized
 	  Serial Communication peripheral (SSC).

drivers/misc/atmel-ssc.c

@@ -175,7 +175,7 @@ static int ssc_probe(struct platform_device *pdev)
 
 	/* disable all interrupts */
 	clk_enable(ssc->clk);
-	ssc_writel(ssc->regs, IDR, ~0UL);
+	ssc_writel(ssc->regs, IDR, -1);
 	ssc_readl(ssc->regs, SR);
 	clk_disable(ssc->clk);
 

drivers/spi/spi-s3c64xx.c

@@ -134,7 +134,6 @@ struct s3c64xx_spi_dma_data {
 	unsigned		ch;
 	enum dma_transfer_direction direction;
 	enum dma_ch	dmach;
-	struct property		*dma_prop;
 };
 
 /**
@@ -319,16 +318,15 @@ static void prepare_dma(struct s3c64xx_spi_dma_data *dma,
 static int acquire_dma(struct s3c64xx_spi_driver_data *sdd)
 {
 	struct samsung_dma_req req;
+	struct device *dev = &sdd->pdev->dev;
 
 	sdd->ops = samsung_dma_get_ops();
 
 	req.cap = DMA_SLAVE;
 	req.client = &s3c64xx_spi_dma_client;
 
-	req.dt_dmach_prop = sdd->rx_dma.dma_prop;
-	sdd->rx_dma.ch = sdd->ops->request(sdd->rx_dma.dmach, &req);
-	req.dt_dmach_prop = sdd->tx_dma.dma_prop;
-	sdd->tx_dma.ch = sdd->ops->request(sdd->tx_dma.dmach, &req);
+	sdd->rx_dma.ch = sdd->ops->request(sdd->rx_dma.dmach, &req, dev, "rx");
+	sdd->tx_dma.ch = sdd->ops->request(sdd->tx_dma.dmach, &req, dev, "tx");
 
 	return 1;
 }
@@ -1053,49 +1051,6 @@ static void s3c64xx_spi_hwinit(struct s3c64xx_spi_driver_data *sdd, int channel)
 	flush_fifo(sdd);
 }
 
-static int s3c64xx_spi_get_dmares(
-			struct s3c64xx_spi_driver_data *sdd, bool tx)
-{
-	struct platform_device *pdev = sdd->pdev;
-	struct s3c64xx_spi_dma_data *dma_data;
-	struct property *prop;
-	struct resource *res;
-	char prop_name[15], *chan_str;
-
-	if (tx) {
-		dma_data = &sdd->tx_dma;
-		dma_data->direction = DMA_MEM_TO_DEV;
-		chan_str = "tx";
-	} else {
-		dma_data = &sdd->rx_dma;
-		dma_data->direction = DMA_DEV_TO_MEM;
-		chan_str = "rx";
-	}
-
-	if (!sdd->pdev->dev.of_node) {
-		res = platform_get_resource(pdev, IORESOURCE_DMA, tx ? 0 : 1);
-		if (!res) {
-			dev_err(&pdev->dev, "Unable to get SPI-%s dma resource\n",
-				chan_str);
-			return -ENXIO;
-		}
-		dma_data->dmach = res->start;
-		return 0;
-	}
-
-	sprintf(prop_name, "%s-dma-channel", chan_str);
-	prop = of_find_property(pdev->dev.of_node, prop_name, NULL);
-	if (!prop) {
-		dev_err(&pdev->dev, "%s dma channel property not specified\n",
-					chan_str);
-		return -ENXIO;
-	}
-
-	dma_data->dmach = DMACH_DT_PROP;
-	dma_data->dma_prop = prop;
-	return 0;
-}
-
 #ifdef CONFIG_OF
 static int s3c64xx_spi_parse_dt_gpio(struct s3c64xx_spi_driver_data *sdd)
 {
@@ -1194,6 +1149,7 @@ static inline struct s3c64xx_spi_port_config *s3c64xx_spi_get_port_config(
 static int s3c64xx_spi_probe(struct platform_device *pdev)
 {
 	struct resource	*mem_res;
+	struct resource	*res;
 	struct s3c64xx_spi_driver_data *sdd;
 	struct s3c64xx_spi_info *sci = pdev->dev.platform_data;
 	struct spi_master *master;
@@ -1252,13 +1208,26 @@ static int s3c64xx_spi_probe(struct platform_device *pdev)
 
 	sdd->cur_bpw = 8;
 
-	ret = s3c64xx_spi_get_dmares(sdd, true);
-	if (ret)
-		goto err0;
+	if (!sdd->pdev->dev.of_node) {
+		res = platform_get_resource(pdev, IORESOURCE_DMA,  0);
+		if (!res) {
+			dev_err(&pdev->dev, "Unable to get SPI tx dma "
+					"resource\n");
+			return -ENXIO;
+		}
+		sdd->tx_dma.dmach = res->start;
 
-	ret = s3c64xx_spi_get_dmares(sdd, false);
-	if (ret)
-		goto err0;
+		res = platform_get_resource(pdev, IORESOURCE_DMA,  1);
+		if (!res) {
+			dev_err(&pdev->dev, "Unable to get SPI rx dma "
+					"resource\n");
+			return -ENXIO;
+		}
+		sdd->rx_dma.dmach = res->start;
+	}
+
+	sdd->tx_dma.direction = DMA_MEM_TO_DEV;
+	sdd->rx_dma.direction = DMA_DEV_TO_MEM;
 
 	master->dev.of_node = pdev->dev.of_node;
 	master->bus_num = sdd->port_id;

include/linux/mfd/arizona/pdata.h

@@ -62,6 +62,8 @@
 
 #define ARIZONA_MAX_OUTPUT 6
 
+#define ARIZONA_MAX_AIF 3
+
 #define ARIZONA_HAP_ACT_ERM 0
 #define ARIZONA_HAP_ACT_LRA 2
 
@@ -96,6 +98,13 @@ struct arizona_pdata {
 	/** Pin state for GPIO pins */
 	int gpio_defaults[ARIZONA_MAX_GPIO];
 
+	/**
+	 * Maximum number of channels clocks will be generated for,
+	 * useful for systems where and I2S bus with multiple data
+	 * lines is mastered.
+	 */
+	int max_channels_clocked[ARIZONA_MAX_AIF];
+
 	/** GPIO for mic detection polarity */
 	int micd_pol_gpio;
 

include/sound/compress_driver.h

@@ -71,6 +71,8 @@ struct snd_compr_runtime {
  * @runtime: pointer to runtime structure
  * @device: device pointer
  * @direction: stream direction, playback/recording
+ * @metadata_set: metadata set flag, true when set
+ * @next_track: has userspace signall next track transistion, true when set
  * @private_data: pointer to DSP private data
  */
 struct snd_compr_stream {
@@ -79,6 +81,8 @@ struct snd_compr_stream {
 	struct snd_compr_runtime *runtime;
 	struct snd_compr *device;
 	enum snd_compr_direction direction;
+	bool metadata_set;
+	bool next_track;
 	void *private_data;
 };
 
@@ -110,6 +114,10 @@ struct snd_compr_ops {
 			struct snd_compr_params *params);
 	int (*get_params)(struct snd_compr_stream *stream,
 			struct snd_codec *params);
+	int (*set_metadata)(struct snd_compr_stream *stream,
+			struct snd_compr_metadata *metadata);
+	int (*get_metadata)(struct snd_compr_stream *stream,
+			struct snd_compr_metadata *metadata);
 	int (*trigger)(struct snd_compr_stream *stream, int cmd);
 	int (*pointer)(struct snd_compr_stream *stream,
 			struct snd_compr_tstamp *tstamp);

include/sound/core.h

@@ -394,8 +394,11 @@ void __snd_printk(unsigned int level, const char *file, int line,
 
 #else /* !CONFIG_SND_DEBUG */
 
-#define snd_printd(fmt, args...)	do { } while (0)
-#define _snd_printd(level, fmt, args...) do { } while (0)
+__printf(1, 2)
+static inline void snd_printd(const char *format, ...) {}
+__printf(2, 3)
+static inline void _snd_printd(int level, const char *format, ...) {}
+
 #define snd_BUG()			do { } while (0)
 static inline int __snd_bug_on(int cond)
 {
@@ -416,7 +419,8 @@ static inline int __snd_bug_on(int cond)
 #define snd_printdd(format, args...) \
 	__snd_printk(2, __FILE__, __LINE__, format, ##args)
 #else
-#define snd_printdd(format, args...)	do { } while (0)
+__printf(1, 2)
+static inline void snd_printdd(const char *format, ...) {}
 #endif
 
 
@@ -454,6 +458,7 @@ struct snd_pci_quirk {
 #define SND_PCI_QUIRK_MASK(vend, mask, dev, xname, val)			\
 	{_SND_PCI_QUIRK_ID_MASK(vend, mask, dev),			\
 			.value = (val), .name = (xname)}
+#define snd_pci_quirk_name(q)	((q)->name)
 #else
 #define SND_PCI_QUIRK(vend,dev,xname,val) \
 	{_SND_PCI_QUIRK_ID(vend, dev), .value = (val)}
@@ -461,6 +466,7 @@ struct snd_pci_quirk {
 	{_SND_PCI_QUIRK_ID_MASK(vend, mask, dev), .value = (val)}
 #define SND_PCI_QUIRK_VENDOR(vend, xname, val)			\
 	{_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val)}
+#define snd_pci_quirk_name(q)	""
 #endif
 
 const struct snd_pci_quirk *

include/sound/cs4271.h

@@ -20,6 +20,21 @@
 struct cs4271_platform_data {
 	int gpio_nreset;	/* GPIO driving Reset pin, if any */
 	bool amutec_eq_bmutec;	/* flag to enable AMUTEC=BMUTEC */
+
+	/*
+	 * 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 the
+	 * following flag.
+	 *
+	 * Note that this is not needed in case the clocks are stable
+	 * throughout the entire runtime of the codec.
+	 */
+	bool enable_soft_reset;
 };
 
 #endif /* __CS4271_H */

include/sound/da7213.h

+/*
+ * da7213.h - DA7213 ASoC Codec Driver Platform Data
+ *
+ * Copyright (c) 2013 Dialog Semiconductor
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.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 _DA7213_PDATA_H
+#define _DA7213_PDATA_H
+
+enum da7213_micbias_voltage {
+	DA7213_MICBIAS_1_6V = 0,
+	DA7213_MICBIAS_2_2V = 1,
+	DA7213_MICBIAS_2_5V = 2,
+	DA7213_MICBIAS_3_0V = 3,
+};
+
+enum da7213_dmic_data_sel {
+	DA7213_DMIC_DATA_LRISE_RFALL = 0,
+	DA7213_DMIC_DATA_LFALL_RRISE = 1,
+};
+
+enum da7213_dmic_samplephase {
+	DA7213_DMIC_SAMPLE_ON_CLKEDGE = 0,
+	DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE = 1,
+};
+
+enum da7213_dmic_clk_rate {
+	DA7213_DMIC_CLK_3_0MHZ = 0,
+	DA7213_DMIC_CLK_1_5MHZ = 1,
+};
+
+struct da7213_platform_data {
+	/* Mic Bias voltage */
+	enum da7213_micbias_voltage micbias1_lvl;
+	enum da7213_micbias_voltage micbias2_lvl;
+
+	/* DMIC config */
+	enum da7213_dmic_data_sel dmic_data_sel;
+	enum da7213_dmic_samplephase dmic_samplephase;
+	enum da7213_dmic_clk_rate dmic_clk_rate;
+
+	/* MCLK squaring config */
+	bool mclk_squaring;
+};
+
+#endif /* _DA7213_PDATA_H */

include/sound/max98090.h

+/*
+ * Platform data for MAX98090
+ *
+ * Copyright 2011-2012 Maxim Integrated Products
+ *
+ *  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 __SOUND_MAX98090_PDATA_H__
+#define __SOUND_MAX98090_PDATA_H__
+
+/* codec platform data */
+struct max98090_pdata {
+
+	/* Analog/digital microphone configuration:
+	 * 0 = analog microphone input (normal setting)
+	 * 1 = digital microphone input
+	 */
+	unsigned int digmic_left_mode:1;
+	unsigned int digmic_right_mode:1;
+	unsigned int digmic_3_mode:1;
+	unsigned int digmic_4_mode:1;
+};
+
+#endif

include/sound/memalloc.h

@@ -37,7 +37,7 @@ struct snd_dma_device {
 #ifndef snd_dma_pci_data
 #define snd_dma_pci_data(pci)	(&(pci)->dev)
 #define snd_dma_isa_data()	NULL
-#define snd_dma_continuous_data(x)	((struct device *)(unsigned long)(x))
+#define snd_dma_continuous_data(x)	((struct device *)(__force unsigned long)(x))
 #endif
 
 

include/sound/saif.h

-/*
- * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved.
- *
- * 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_SAIF_H__
-#define __SOUND_SAIF_H__
-
-struct mxs_saif_platform_data {
-	bool master_mode;	/* if true use master mode */
-	int master_id;		/* id of the master if in slave mode */
-};
-#endif

include/sound/sh_fsi.h

@@ -11,82 +11,20 @@
  * it under the terms of the GNU General Public License version 2 as
  * published by the Free Software Foundation.
  */
-
-#define FSI_PORT_A	0
-#define FSI_PORT_B	1
-
 #include <linux/clk.h>
 #include <sound/soc.h>
 
 /*
- * flags format
- *
- * 0x00000CBA
- *
- * A:  inversion
- * B:  format mode
- * C:  chip specific
- * D:  clock selecter if master mode
+ * flags
  */
-
-/* A: clock inversion */
-#define SH_FSI_INVERSION_MASK	0x0000000F
-#define SH_FSI_LRM_INV		(1 << 0)
-#define SH_FSI_BRM_INV		(1 << 1)
-#define SH_FSI_LRS_INV		(1 << 2)
-#define SH_FSI_BRS_INV		(1 << 3)
-
-/* B: format mode */
-#define SH_FSI_FMT_MASK		0x000000F0
-#define SH_FSI_FMT_DAI		(0 << 4)
-#define SH_FSI_FMT_SPDIF	(1 << 4)
-
-/* C: chip specific */
-#define SH_FSI_OPTION_MASK	0x00000F00
-#define SH_FSI_ENABLE_STREAM_MODE	(1 << 8) /* for 16bit data */
-
-/* D:  clock selecter if master mode */
-#define SH_FSI_CLK_MASK		0x0000F000
-#define SH_FSI_CLK_EXTERNAL	(0 << 12)
-#define SH_FSI_CLK_CPG		(1 << 12) /* FSIxCK + FSI-DIV */
-
-/*
- * set_rate return value
- *
- * see ACKMD/BPFMD on
- *     ACK_MD (FSI2)
- *     CKG1   (FSI)
- *
- * err		: return value <  0
- * no change	: return value == 0
- * change xMD	: return value >  0
- *
- * 0x-00000AB
- *
- * A:  ACKMD value
- * B:  BPFMD value
- */
-
-#define SH_FSI_ACKMD_MASK	(0xF << 0)
-#define SH_FSI_ACKMD_512	(1 << 0)
-#define SH_FSI_ACKMD_256	(2 << 0)
-#define SH_FSI_ACKMD_128	(3 << 0)
-#define SH_FSI_ACKMD_64		(4 << 0)
-#define SH_FSI_ACKMD_32		(5 << 0)
-
-#define SH_FSI_BPFMD_MASK	(0xF << 4)
-#define SH_FSI_BPFMD_512	(1 << 4)
-#define SH_FSI_BPFMD_256	(2 << 4)
-#define SH_FSI_BPFMD_128	(3 << 4)
-#define SH_FSI_BPFMD_64		(4 << 4)
-#define SH_FSI_BPFMD_32		(5 << 4)
-#define SH_FSI_BPFMD_16		(6 << 4)
+#define SH_FSI_FMT_SPDIF		(1 << 0) /* spdif for HDMI */
+#define SH_FSI_ENABLE_STREAM_MODE	(1 << 1) /* for 16bit data */
+#define SH_FSI_CLK_CPG			(1 << 2) /* FSIxCK + FSI-DIV */
 
 struct sh_fsi_port_info {
 	unsigned long flags;
 	int tx_id;
 	int rx_id;
-	int (*set_rate)(struct device *dev, int rate, int enable);
 };
 
 struct sh_fsi_platform_info {

include/sound/simple_card.h

@@ -14,21 +14,21 @@
 
 #include <sound/soc.h>
 
-struct asoc_simple_dai_init_info {
+struct asoc_simple_dai {
+	const char *name;
 	unsigned int fmt;
-	unsigned int cpu_daifmt;
-	unsigned int codec_daifmt;
 	unsigned int sysclk;
 };
 
 struct asoc_simple_card_info {
 	const char *name;
 	const char *card;
-	const char *cpu_dai;
 	const char *codec;
 	const char *platform;
-	const char *codec_dai;
-	struct asoc_simple_dai_init_info *init; /* for snd_link.init */
+
+	unsigned int daifmt;
+	struct asoc_simple_dai cpu_dai;
+	struct asoc_simple_dai codec_dai;
 
 	/* used in simple-card.c */
 	struct snd_soc_dai_link snd_link;

include/sound/soc-dai.h

@@ -45,7 +45,7 @@ struct snd_compr_stream;
  * sending or receiving PCM data in a frame. This can be used to save power.
  */
 #define SND_SOC_DAIFMT_CONT		(1 << 4) /* continuous clock */
-#define SND_SOC_DAIFMT_GATED		(2 << 4) /* clock is gated */
+#define SND_SOC_DAIFMT_GATED		(0 << 4) /* clock is gated */
 
 /*
  * DAI hardware signal inversions.
@@ -53,7 +53,7 @@ struct snd_compr_stream;
  * Specifies whether the DAI can also support inverted clocks for the specified
  * format.
  */
-#define SND_SOC_DAIFMT_NB_NF		(1 << 8) /* normal bit clock + frame */
+#define SND_SOC_DAIFMT_NB_NF		(0 << 8) /* normal bit clock + frame */
 #define SND_SOC_DAIFMT_NB_IF		(2 << 8) /* normal BCLK + inv FRM */
 #define SND_SOC_DAIFMT_IB_NF		(3 << 8) /* invert BCLK + nor FRM */
 #define SND_SOC_DAIFMT_IB_IF		(4 << 8) /* invert BCLK + FRM */
@@ -126,7 +126,8 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
 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);
+int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
+			     int direction);
 
 struct snd_soc_dai_ops {
 	/*
@@ -157,6 +158,7 @@ struct snd_soc_dai_ops {
 	 * Called by soc-core to minimise any pops.
 	 */
 	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
+	int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
 
 	/*
 	 * ALSA PCM audio operations - all optional.

include/sound/soc.h

@@ -906,8 +906,8 @@ struct snd_soc_dai_link {
 			struct snd_pcm_hw_params *params);
 
 	/* machine stream operations */
-	struct snd_soc_ops *ops;
-	struct snd_soc_compr_ops *compr_ops;
+	const struct snd_soc_ops *ops;
+	const struct snd_soc_compr_ops *compr_ops;
 };
 
 struct snd_soc_codec_conf {
@@ -1171,6 +1171,8 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card,
 			       const char *propname);
 int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 				   const char *propname);
+unsigned int snd_soc_of_parse_daifmt(struct device_node *np,
+				     const char *prefix);
 
 #include <sound/soc-dai.h>
 

include/sound/tlv320aic3x.h

@@ -46,6 +46,13 @@ enum {
 	AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ	= 15
 };
 
+enum aic3x_micbias_voltage {
+	AIC3X_MICBIAS_OFF = 0,
+	AIC3X_MICBIAS_2_0V = 1,
+	AIC3X_MICBIAS_2_5V = 2,
+	AIC3X_MICBIAS_AVDDV = 3,
+};
+
 struct aic3x_setup_data {
 	unsigned int gpio_func[2];
 };
@@ -53,6 +60,9 @@ struct aic3x_setup_data {
 struct aic3x_pdata {
 	int gpio_reset; /* < 0 if not used */
 	struct aic3x_setup_data *setup;
+
+	/* Selects the micbias voltage */
+	enum aic3x_micbias_voltage micbias_vg;
 };
 
 #endif

include/sound/wm2000.h

@@ -15,9 +15,6 @@ struct wm2000_platform_data {
 	/** Filename for system-specific image to download to device. */
 	const char *download_file;
 
-	/** Divide MCLK by 2 for system clock? */
-	unsigned int mclkdiv2:1;
-
 	/** Disable speech clarity enhancement, for use when an
 	 * external algorithm is used. */
 	unsigned int speech_enh_disable:1;

include/sound/wm2200.h

@@ -12,6 +12,7 @@
 #define __LINUX_SND_WM2200_H
 
 #define WM2200_GPIO_SET 0x10000
+#define WM2200_MAX_MICBIAS 2
 
 enum wm2200_in_mode {
 	WM2200_IN_SE = 0,
@@ -25,6 +26,24 @@ enum wm2200_dmic_sup {
 	WM2200_DMIC_SUP_MICBIAS2 = 2,
 };
 
+enum wm2200_mbias_lvl {
+	WM2200_MBIAS_LVL_1V5 = 1,
+	WM2200_MBIAS_LVL_1V8 = 2,
+	WM2200_MBIAS_LVL_1V9 = 3,
+	WM2200_MBIAS_LVL_2V0 = 4,
+	WM2200_MBIAS_LVL_2V2 = 5,
+	WM2200_MBIAS_LVL_2V4 = 6,
+	WM2200_MBIAS_LVL_2V5 = 7,
+	WM2200_MBIAS_LVL_2V6 = 8,
+};
+
+struct wm2200_micbias {
+	enum wm2200_mbias_lvl mb_lvl;      /** Regulated voltage */
+	unsigned int discharge:1;          /** Actively discharge */
+	unsigned int fast_start:1;         /** Enable aggressive startup ramp rate */
+	unsigned int bypass:1;             /** Use bypass mode */
+};
+
 struct wm2200_pdata {
 	int reset;      /** GPIO controlling /RESET, if any */
 	int ldo_ena;    /** GPIO controlling LODENA, if any */
@@ -35,7 +54,8 @@ struct wm2200_pdata {
 	enum wm2200_in_mode in_mode[3];
 	enum wm2200_dmic_sup dmic_sup[3];
 
-	int micbias_cfg[2];  /** Register value to configure MICBIAS */
+	/** MICBIAS configurations */
+	struct wm2200_micbias micbias[WM2200_MAX_MICBIAS];
 };
 
 #endif

include/uapi/linux/usb/audio.h

@@ -384,14 +384,16 @@ static inline __u8 uac_processing_unit_iProcessing(struct uac_processing_unit_de
 						   int protocol)
 {
 	__u8 control_size = uac_processing_unit_bControlSize(desc, protocol);
-	return desc->baSourceID[desc->bNrInPins + control_size];
+	return *(uac_processing_unit_bmControls(desc, protocol)
+			+ control_size);
 }
 
 static inline __u8 *uac_processing_unit_specific(struct uac_processing_unit_descriptor *desc,
 						 int protocol)
 {
 	__u8 control_size = uac_processing_unit_bControlSize(desc, protocol);
-	return &desc->baSourceID[desc->bNrInPins + control_size + 1];
+	return uac_processing_unit_bmControls(desc, protocol)
+			+ control_size + 1;
 }
 
 /* 4.5.2 Class-Specific AS Interface Descriptor */

include/uapi/sound/compress_offload.h

@@ -30,7 +30,7 @@
 #include <sound/compress_params.h>
 
 
-#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0)
+#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1)
 /**
  * struct snd_compressed_buffer: compressed buffer
  * @fragment_size: size of buffer fragment in bytes
@@ -121,6 +121,27 @@ struct snd_compr_codec_caps {
 	struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS];
 };
 
+/**
+ * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the
+ * end of the track
+ * @SNDRV_COMPRESS_ENCODER_DELAY: no of samples inserted by the encoder at the
+ * beginning of the track
+ */
+enum {
+	SNDRV_COMPRESS_ENCODER_PADDING = 1,
+	SNDRV_COMPRESS_ENCODER_DELAY = 2,
+};
+
+/**
+ * struct snd_compr_metadata: compressed stream metadata
+ * @key: key id
+ * @value: key value
+ */
+struct snd_compr_metadata {
+	 __u32 key;
+	 __u32 value[8];
+};
+
 /**
  * compress path ioctl definitions
  * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP
@@ -145,6 +166,10 @@ struct snd_compr_codec_caps {
 						struct snd_compr_codec_caps)
 #define SNDRV_COMPRESS_SET_PARAMS	_IOW('C', 0x12, struct snd_compr_params)
 #define SNDRV_COMPRESS_GET_PARAMS	_IOR('C', 0x13, struct snd_codec)
+#define SNDRV_COMPRESS_SET_METADATA	_IOW('C', 0x14,\
+						 struct snd_compr_metadata)
+#define SNDRV_COMPRESS_GET_METADATA	_IOWR('C', 0x15,\
+						 struct snd_compr_metadata)
 #define SNDRV_COMPRESS_TSTAMP		_IOR('C', 0x20, struct snd_compr_tstamp)
 #define SNDRV_COMPRESS_AVAIL		_IOR('C', 0x21, struct snd_compr_avail)
 #define SNDRV_COMPRESS_PAUSE		_IO('C', 0x30)
@@ -152,10 +177,14 @@ struct snd_compr_codec_caps {
 #define SNDRV_COMPRESS_START		_IO('C', 0x32)
 #define SNDRV_COMPRESS_STOP		_IO('C', 0x33)
 #define SNDRV_COMPRESS_DRAIN		_IO('C', 0x34)
+#define SNDRV_COMPRESS_NEXT_TRACK	_IO('C', 0x35)
+#define SNDRV_COMPRESS_PARTIAL_DRAIN	_IO('C', 0x36)
 /*
  * TODO
  * 1. add mmap support
  *
  */
 #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */
+#define SND_COMPR_TRIGGER_NEXT_TRACK 8
+#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9
 #endif

sound/arm/pxa2xx-ac97-lib.c

@@ -34,7 +34,7 @@ static struct clk *ac97_clk;
 static struct clk *ac97conf_clk;
 static int reset_gpio;
 
-extern void pxa27x_assert_ac97reset(int reset_gpio, int on);
+extern void pxa27x_configure_ac97reset(int reset_gpio, bool to_gpio);
 
 /*
  * Beware PXA27x bugs:
@@ -140,10 +140,10 @@ static inline void pxa_ac97_warm_pxa27x(void)
 	gsr_bits = 0;
 
 	/* warm reset broken on Bulverde, so manually keep AC97 reset high */
-	pxa27x_assert_ac97reset(reset_gpio, 1);
+	pxa27x_configure_ac97reset(reset_gpio, true);
 	udelay(10);
 	GCR |= GCR_WARM_RST;
-	pxa27x_assert_ac97reset(reset_gpio, 0);
+	pxa27x_configure_ac97reset(reset_gpio, false);
 	udelay(500);
 }
 
@@ -358,7 +358,7 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev)
 			       __func__, ret);
 			goto err_conf;
 		}
-		pxa27x_assert_ac97reset(reset_gpio, 0);
+		pxa27x_configure_ac97reset(reset_gpio, false);
 
 		ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK");
 		if (IS_ERR(ac97conf_clk)) {

sound/core/compress_offload.c

@@ -144,16 +144,17 @@ static int snd_compr_free(struct inode *inode, struct file *f)
 	return 0;
 }
 
-static void snd_compr_update_tstamp(struct snd_compr_stream *stream,
+static int snd_compr_update_tstamp(struct snd_compr_stream *stream,
 		struct snd_compr_tstamp *tstamp)
 {
 	if (!stream->ops->pointer)
-		return;
+		return -ENOTSUPP;
 	stream->ops->pointer(stream, tstamp);
 	pr_debug("dsp consumed till %d total %d bytes\n",
 		tstamp->byte_offset, tstamp->copied_total);
 	stream->runtime->hw_pointer = tstamp->byte_offset;
 	stream->runtime->total_bytes_transferred = tstamp->copied_total;
+	return 0;
 }
 
 static size_t snd_compr_calc_avail(struct snd_compr_stream *stream,
@@ -161,7 +162,9 @@ static size_t snd_compr_calc_avail(struct snd_compr_stream *stream,
 {
 	long avail_calc; /*this needs to be signed variable */
 
+	memset(avail, 0, sizeof(*avail));
 	snd_compr_update_tstamp(stream, &avail->tstamp);
+	/* Still need to return avail even if tstamp can't be filled in */
 
 	/* FIXME: This needs to be different for capture stream,
 	   available is # of compressed data, for playback it's
@@ -483,6 +486,8 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
 		if (retval)
 			goto out;
 		stream->runtime->state = SNDRV_PCM_STATE_SETUP;
+		stream->metadata_set = false;
+		stream->next_track = false;
 	} else {
 		return -EPERM;
 	}
@@ -514,14 +519,60 @@ snd_compr_get_params(struct snd_compr_stream *stream, unsigned long arg)
 	return retval;
 }
 
+static int
+snd_compr_get_metadata(struct snd_compr_stream *stream, unsigned long arg)
+{
+	struct snd_compr_metadata metadata;
+	int retval;
+
+	if (!stream->ops->get_metadata)
+		return -ENXIO;
+
+	if (copy_from_user(&metadata, (void __user *)arg, sizeof(metadata)))
+		return -EFAULT;
+
+	retval = stream->ops->get_metadata(stream, &metadata);
+	if (retval != 0)
+		return retval;
+
+	if (copy_to_user((void __user *)arg, &metadata, sizeof(metadata)))
+		return -EFAULT;
+
+	return 0;
+}
+
+static int
+snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg)
+{
+	struct snd_compr_metadata metadata;
+	int retval;
+
+	if (!stream->ops->set_metadata)
+		return -ENXIO;
+	/*
+	* we should allow parameter change only when stream has been
+	* opened not in other cases
+	*/
+	if (copy_from_user(&metadata, (void __user *)arg, sizeof(metadata)))
+		return -EFAULT;
+
+	retval = stream->ops->set_metadata(stream, &metadata);
+	stream->metadata_set = true;
+
+	return retval;
+}
+
 static inline int
 snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg)
 {
-	struct snd_compr_tstamp tstamp;
+	struct snd_compr_tstamp tstamp = {0};
+	int ret;
 
-	snd_compr_update_tstamp(stream, &tstamp);
-	return copy_to_user((struct snd_compr_tstamp __user *)arg,
-		&tstamp, sizeof(tstamp)) ? -EFAULT : 0;
+	ret = snd_compr_update_tstamp(stream, &tstamp);
+	if (ret == 0)
+		ret = copy_to_user((struct snd_compr_tstamp __user *)arg,
+			&tstamp, sizeof(tstamp)) ? -EFAULT : 0;
+	return ret;
 }
 
 static int snd_compr_pause(struct snd_compr_stream *stream)
@@ -594,6 +645,44 @@ static int snd_compr_drain(struct snd_compr_stream *stream)
 	return retval;
 }
 
+static int snd_compr_next_track(struct snd_compr_stream *stream)
+{
+	int retval;
+
+	/* only a running stream can transition to next track */
+	if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING)
+		return -EPERM;
+
+	/* you can signal next track isf this is intended to be a gapless stream
+	 * and current track metadata is set
+	 */
+	if (stream->metadata_set == false)
+		return -EPERM;
+
+	retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK);
+	if (retval != 0)
+		return retval;
+	stream->metadata_set = false;
+	stream->next_track = true;
+	return 0;
+}
+
+static int snd_compr_partial_drain(struct snd_compr_stream *stream)
+{
+	int retval;
+	if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
+			stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+		return -EPERM;
+	/* stream can be drained only when next track has been signalled */
+	if (stream->next_track == false)
+		return -EPERM;
+
+	retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN);
+
+	stream->next_track = false;
+	return retval;
+}
+
 static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg)
 {
 	struct snd_compr_file *data = f->private_data;
@@ -623,6 +712,12 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg)
 	case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS):
 		retval = snd_compr_get_params(stream, arg);
 		break;
+	case _IOC_NR(SNDRV_COMPRESS_SET_METADATA):
+		retval = snd_compr_set_metadata(stream, arg);
+		break;
+	case _IOC_NR(SNDRV_COMPRESS_GET_METADATA):
+		retval = snd_compr_get_metadata(stream, arg);
+		break;
 	case _IOC_NR(SNDRV_COMPRESS_TSTAMP):
 		retval = snd_compr_tstamp(stream, arg);
 		break;
@@ -644,6 +739,13 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg)
 	case _IOC_NR(SNDRV_COMPRESS_DRAIN):
 		retval = snd_compr_drain(stream);
 		break;
+	case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN):
+		retval = snd_compr_partial_drain(stream);
+		break;
+	case _IOC_NR(SNDRV_COMPRESS_NEXT_TRACK):
+		retval = snd_compr_next_track(stream);
+		break;
+
 	}
 	mutex_unlock(&stream->device->lock);
 	return retval;

sound/drivers/aloop.c

@@ -286,12 +286,14 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd)
 			loopback_active_notify(dpcm);
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 		spin_lock(&cable->lock);	
 		cable->pause |= stream;
 		loopback_timer_stop(dpcm);
 		spin_unlock(&cable->lock);
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_RESUME:
 		spin_lock(&cable->lock);
 		dpcm->last_jiffies = jiffies;
 		cable->pause &= ~stream;
@@ -563,7 +565,8 @@ static snd_pcm_uframes_t loopback_pointer(struct snd_pcm_substream *substream)
 static struct snd_pcm_hardware loopback_pcm_hardware =
 {
 	.info =		(SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP |
-			 SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE),
+			 SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE |
+			 SNDRV_PCM_INFO_RESUME),
 	.formats =	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
 			 SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE |
 			 SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE),

sound/drivers/vx/vx_core.c

@@ -52,7 +52,6 @@ MODULE_LICENSE("GPL");
 int snd_vx_check_reg_bit(struct vx_core *chip, int reg, int mask, int bit, int time)
 {
 	unsigned long end_time = jiffies + (time * HZ + 999) / 1000;
-#ifdef CONFIG_SND_DEBUG
 	static char *reg_names[VX_REG_MAX] = {
 		"ICR", "CVR", "ISR", "IVR", "RXH", "RXM", "RXL",
 		"DMA", "CDSP", "RFREQ", "RUER/V2", "DATA", "MEMIRQ",
@@ -60,7 +59,7 @@ int snd_vx_check_reg_bit(struct vx_core *chip, int reg, int mask, int bit, int t
 		"MIC3", "INTCSR", "CNTRL", "GPIOC",
 		"LOFREQ", "HIFREQ", "CSUER", "RUER"
 	};
-#endif
+
 	do {
 		if ((snd_vx_inb(chip, reg) & mask) == bit)
 			return 0;

sound/pci/Kconfig

@@ -678,6 +678,7 @@ config SND_LOLA
 
 config SND_LX6464ES
 	tristate "Digigram LX6464ES"
+	depends on HAS_IOPORT
 	select SND_PCM
 	help
 	  Say Y here to include support for Digigram LX6464ES boards.

sound/pci/ali5451/ali5451.c

@@ -1435,7 +1435,7 @@ static snd_pcm_uframes_t snd_ali_pointer(struct snd_pcm_substream *substream)
 
 	spin_lock(&codec->reg_lock);
 	if (!pvoice->running) {
-		spin_unlock_irq(&codec->reg_lock);
+		spin_unlock(&codec->reg_lock);
 		return 0;
 	}
 	outb(pvoice->number, ALI_REG(codec, ALI_GC_CIR));

sound/pci/atiixp.c

@@ -567,8 +567,9 @@ static int ac97_probing_bugs(struct pci_dev *pci)
 
 	q = snd_pci_quirk_lookup(pci, atiixp_quirks);
 	if (q) {
-		snd_printdd(KERN_INFO "Atiixp quirk for %s.  "
-			    "Forcing codec %d\n", q->name, q->value);
+		snd_printdd(KERN_INFO
+			    "Atiixp quirk for %s.  Forcing codec %d\n",
+			    snd_pci_quirk_name(q), q->value);
 		return q->value;
 	}
 	/* this hardware doesn't need workarounds.  Probe for codec */

sound/pci/au88x0/au88x0_pcm.c

@@ -650,6 +650,29 @@ static int snd_vortex_new_pcm(vortex_t *chip, int idx, int nr)
 					      snd_dma_pci_data(chip->pci_dev),
 					      0x10000, 0x10000);
 
+	switch (VORTEX_PCM_TYPE(pcm)) {
+	case VORTEX_PCM_ADB:
+		err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+					     snd_pcm_std_chmaps,
+					     VORTEX_IS_QUAD(chip) ? 4 : 2,
+					     0, NULL);
+		if (err < 0)
+			return err;
+		err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_CAPTURE,
+					     snd_pcm_std_chmaps, 2, 0, NULL);
+		if (err < 0)
+			return err;
+		break;
+#ifdef CHIP_AU8830
+	case VORTEX_PCM_A3D:
+		err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+					     snd_pcm_std_chmaps, 1, 0, NULL);
+		if (err < 0)
+			return err;
+		break;
+#endif
+	};
+
 	if (VORTEX_PCM_TYPE(pcm) == VORTEX_PCM_SPDIF) {
 		for (i = 0; i < ARRAY_SIZE(snd_vortex_mixer_spdif); i++) {
 			kctl = snd_ctl_new1(&snd_vortex_mixer_spdif[i], chip);

sound/pci/hda/Kconfig

@@ -15,6 +15,9 @@ menuconfig SND_HDA_INTEL
 
 if SND_HDA_INTEL
 
+config SND_HDA_DSP_LOADER
+	bool
+
 config SND_HDA_PREALLOC_SIZE
 	int "Pre-allocated buffer size for HD-audio driver"
 	range 0 32768
@@ -86,6 +89,7 @@ config SND_HDA_PATCH_LOADER
 config SND_HDA_CODEC_REALTEK
 	bool "Build Realtek HD-audio codec support"
 	default y
+	select SND_HDA_GENERIC
 	help
 	  Say Y here to include Realtek HD-audio codec support in
 	  snd-hda-intel driver, such as ALC880.
@@ -98,6 +102,7 @@ config SND_HDA_CODEC_REALTEK
 config SND_HDA_CODEC_ANALOG
 	bool "Build Analog Device HD-audio codec support"
 	default y
+	select SND_HDA_GENERIC
 	help
 	  Say Y here to include Analog Device HD-audio codec support in
 	  snd-hda-intel driver, such as AD1986A.
@@ -110,6 +115,7 @@ config SND_HDA_CODEC_ANALOG
 config SND_HDA_CODEC_SIGMATEL
 	bool "Build IDT/Sigmatel HD-audio codec support"
 	default y
+	select SND_HDA_GENERIC
 	help
 	  Say Y here to include IDT (Sigmatel) HD-audio codec support in
 	  snd-hda-intel driver, such as STAC9200.
@@ -122,6 +128,7 @@ config SND_HDA_CODEC_SIGMATEL
 config SND_HDA_CODEC_VIA
 	bool "Build VIA HD-audio codec support"
 	default y
+	select SND_HDA_GENERIC
 	help
 	  Say Y here to include VIA HD-audio codec support in
 	  snd-hda-intel driver, such as VT1708.
@@ -147,8 +154,8 @@ config SND_HDA_CODEC_HDMI
 
 config SND_HDA_CODEC_CIRRUS
 	bool "Build Cirrus Logic codec support"
-	depends on SND_HDA_INTEL
 	default y
+	select SND_HDA_GENERIC
 	help
 	  Say Y here to include Cirrus Logic codec support in
 	  snd-hda-intel driver, such as CS4206.
@@ -161,6 +168,7 @@ config SND_HDA_CODEC_CIRRUS
 config SND_HDA_CODEC_CONEXANT
 	bool "Build Conexant HD-audio codec support"
 	default y
+	select SND_HDA_GENERIC
 	help
 	  Say Y here to include Conexant HD-audio codec support in
 	  snd-hda-intel driver, such as CX20549.
@@ -172,8 +180,8 @@ config SND_HDA_CODEC_CONEXANT
 
 config SND_HDA_CODEC_CA0110
 	bool "Build Creative CA0110-IBG codec support"
-	depends on SND_HDA_INTEL
 	default y
+	select SND_HDA_GENERIC
 	help
 	  Say Y here to include Creative CA0110-IBG codec support in
 	  snd-hda-intel driver, found on some Creative X-Fi cards.
@@ -185,7 +193,6 @@ config SND_HDA_CODEC_CA0110
 
 config SND_HDA_CODEC_CA0132
 	bool "Build Creative CA0132 codec support"
-	depends on SND_HDA_INTEL
 	default y
 	help
 	  Say Y here to include Creative CA0132 codec support in
@@ -196,9 +203,21 @@ config SND_HDA_CODEC_CA0132
 	  snd-hda-codec-ca0132.
 	  This module is automatically loaded at probing.
 
+config SND_HDA_CODEC_CA0132_DSP
+	bool "Support new DSP code for CA0132 codec"
+	depends on SND_HDA_CODEC_CA0132 && FW_LOADER
+	select SND_HDA_DSP_LOADER
+	help
+	  Say Y here to enable the DSP for Creative CA0132 for extended
+	  features like equalizer or echo cancellation.
+
+	  Note that this option requires the external firmware file
+	  (ctefx.bin).
+
 config SND_HDA_CODEC_CMEDIA
 	bool "Build C-Media HD-audio codec support"
 	default y
+	select SND_HDA_GENERIC
 	help
 	  Say Y here to include C-Media HD-audio codec support in
 	  snd-hda-intel driver, such as CMI9880.

sound/pci/hda/hda_auto_parser.c

@@ -97,6 +97,28 @@ static void reorder_outputs(unsigned int nums, hda_nid_t *pins)
 	}
 }
 
+/* check whether the given pin has a proper pin I/O capability bit */
+static bool check_pincap_validity(struct hda_codec *codec, hda_nid_t pin,
+				  unsigned int dev)
+{
+	unsigned int pincap = snd_hda_query_pin_caps(codec, pin);
+
+	/* some old hardware don't return the proper pincaps */
+	if (!pincap)
+		return true;
+
+	switch (dev) {
+	case AC_JACK_LINE_OUT:
+	case AC_JACK_SPEAKER:
+	case AC_JACK_HP_OUT:
+	case AC_JACK_SPDIF_OUT:
+	case AC_JACK_DIG_OTHER_OUT:
+		return !!(pincap & AC_PINCAP_OUT);
+	default:
+		return !!(pincap & AC_PINCAP_IN);
+	}
+}
+
 /*
  * Parse all pin widgets and store the useful pin nids to cfg
  *
@@ -126,6 +148,9 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 	struct auto_out_pin hp_out[ARRAY_SIZE(cfg->hp_pins)];
 	int i;
 
+	if (!snd_hda_get_int_hint(codec, "parser_flags", &i))
+		cond_flags = i;
+
 	memset(cfg, 0, sizeof(*cfg));
 
 	memset(line_out, 0, sizeof(line_out));
@@ -156,10 +181,14 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 
 		/* workaround for buggy BIOS setups */
 		if (dev == AC_JACK_LINE_OUT) {
-			if (conn == AC_JACK_PORT_FIXED)
+			if (conn == AC_JACK_PORT_FIXED ||
+			    conn == AC_JACK_PORT_BOTH)
 				dev = AC_JACK_SPEAKER;
 		}
 
+		if (!check_pincap_validity(codec, nid, dev))
+			continue;
+
 		switch (dev) {
 		case AC_JACK_LINE_OUT:
 			seq = get_defcfg_sequence(def_conf);
@@ -363,7 +392,7 @@ static const char *hda_get_input_pin_label(struct hda_codec *codec,
 {
 	unsigned int def_conf;
 	static const char * const mic_names[] = {
-		"Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic",
+		"Internal Mic", "Dock Mic", "Mic", "Rear Mic", "Front Mic"
 	};
 	int attr;
 
@@ -394,6 +423,8 @@ static const char *hda_get_input_pin_label(struct hda_codec *codec,
 		return "SPDIF In";
 	case AC_JACK_DIG_OTHER_IN:
 		return "Digital In";
+	case AC_JACK_HP_OUT:
+		return "Headphone Mic";
 	default:
 		return "Misc";
 	}
@@ -552,6 +583,9 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid,
 	return 1;
 }
 
+#define is_hdmi_cfg(conf) \
+	(get_defcfg_location(conf) == AC_JACK_LOC_HDMI)
+
 /**
  * snd_hda_get_pin_label - Get a label for the given I/O pin
  *
@@ -572,6 +606,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
 	unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid);
 	const char *name = NULL;
 	int i;
+	bool hdmi;
 
 	if (indexp)
 		*indexp = 0;
@@ -590,16 +625,18 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
 					   label, maxlen, indexp);
 	case AC_JACK_SPDIF_OUT:
 	case AC_JACK_DIG_OTHER_OUT:
-		if (get_defcfg_location(def_conf) == AC_JACK_LOC_HDMI)
-			name = "HDMI";
-		else
-			name = "SPDIF";
-		if (cfg && indexp) {
-			i = find_idx_in_nid_list(nid, cfg->dig_out_pins,
-						 cfg->dig_outs);
-			if (i >= 0)
-				*indexp = i;
-		}
+		hdmi = is_hdmi_cfg(def_conf);
+		name = hdmi ? "HDMI" : "SPDIF";
+		if (cfg && indexp)
+			for (i = 0; i < cfg->dig_outs; i++) {
+				hda_nid_t pin = cfg->dig_out_pins[i];
+				unsigned int c;
+				if (pin == nid)
+					break;
+				c = snd_hda_codec_get_pincfg(codec, pin);
+				if (hdmi == is_hdmi_cfg(c))
+					(*indexp)++;
+			}
 		break;
 	default:
 		if (cfg) {
@@ -622,28 +659,27 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
 }
 EXPORT_SYMBOL_HDA(snd_hda_get_pin_label);
 
-int snd_hda_gen_add_verbs(struct hda_gen_spec *spec,
-			  const struct hda_verb *list)
+int snd_hda_add_verbs(struct hda_codec *codec,
+		      const struct hda_verb *list)
 {
 	const struct hda_verb **v;
-	v = snd_array_new(&spec->verbs);
+	v = snd_array_new(&codec->verbs);
 	if (!v)
 		return -ENOMEM;
 	*v = list;
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_add_verbs);
+EXPORT_SYMBOL_HDA(snd_hda_add_verbs);
 
-void snd_hda_gen_apply_verbs(struct hda_codec *codec)
+void snd_hda_apply_verbs(struct hda_codec *codec)
 {
-	struct hda_gen_spec *spec = codec->spec;
 	int i;
-	for (i = 0; i < spec->verbs.used; i++) {
-		struct hda_verb **v = snd_array_elem(&spec->verbs, i);
+	for (i = 0; i < codec->verbs.used; i++) {
+		struct hda_verb **v = snd_array_elem(&codec->verbs, i);
 		snd_hda_sequence_write(codec, *v);
 	}
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_apply_verbs);
+EXPORT_SYMBOL_HDA(snd_hda_apply_verbs);
 
 void snd_hda_apply_pincfgs(struct hda_codec *codec,
 			   const struct hda_pintbl *cfg)
@@ -653,20 +689,22 @@ void snd_hda_apply_pincfgs(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_HDA(snd_hda_apply_pincfgs);
 
-void snd_hda_apply_fixup(struct hda_codec *codec, int action)
+static void set_pin_targets(struct hda_codec *codec,
+			    const struct hda_pintbl *cfg)
 {
-	struct hda_gen_spec *spec = codec->spec;
-	int id = spec->fixup_id;
-#ifdef CONFIG_SND_DEBUG_VERBOSE
-	const char *modelname = spec->fixup_name;
-#endif
-	int depth = 0;
+	for (; cfg->nid; cfg++)
+		snd_hda_set_pin_ctl_cache(codec, cfg->nid, cfg->val);
+}
 
-	if (!spec->fixup_list)
-		return;
+static void apply_fixup(struct hda_codec *codec, int id, int action, int depth)
+{
+	const char *modelname = codec->fixup_name;
 
 	while (id >= 0) {
-		const struct hda_fixup *fix = spec->fixup_list + id;
+		const struct hda_fixup *fix = codec->fixup_list + id;
+
+		if (fix->chained_before)
+			apply_fixup(codec, fix->chain_id, action, depth + 1);
 
 		switch (fix->type) {
 		case HDA_FIXUP_PINS:
@@ -683,7 +721,7 @@ void snd_hda_apply_fixup(struct hda_codec *codec, int action)
 			snd_printdd(KERN_INFO SFX
 				    "%s: Apply fix-verbs for %s\n",
 				    codec->chip_name, modelname);
-			snd_hda_gen_add_verbs(codec->spec, fix->v.verbs);
+			snd_hda_add_verbs(codec, fix->v.verbs);
 			break;
 		case HDA_FIXUP_FUNC:
 			if (!fix->v.func)
@@ -693,19 +731,33 @@ void snd_hda_apply_fixup(struct hda_codec *codec, int action)
 				    codec->chip_name, modelname);
 			fix->v.func(codec, fix, action);
 			break;
+		case HDA_FIXUP_PINCTLS:
+			if (action != HDA_FIXUP_ACT_PROBE || !fix->v.pins)
+				break;
+			snd_printdd(KERN_INFO SFX
+				    "%s: Apply pinctl for %s\n",
+				    codec->chip_name, modelname);
+			set_pin_targets(codec, fix->v.pins);
+			break;
 		default:
 			snd_printk(KERN_ERR SFX
 				   "%s: Invalid fixup type %d\n",
 				   codec->chip_name, fix->type);
 			break;
 		}
-		if (!fix->chained)
+		if (!fix->chained || fix->chained_before)
 			break;
 		if (++depth > 10)
 			break;
 		id = fix->chain_id;
 	}
 }
+
+void snd_hda_apply_fixup(struct hda_codec *codec, int action)
+{
+	if (codec->fixup_list)
+		apply_fixup(codec, codec->fixup_id, action, 0);
+}
 EXPORT_SYMBOL_HDA(snd_hda_apply_fixup);
 
 void snd_hda_pick_fixup(struct hda_codec *codec,
@@ -713,15 +765,14 @@ void snd_hda_pick_fixup(struct hda_codec *codec,
 			const struct snd_pci_quirk *quirk,
 			const struct hda_fixup *fixlist)
 {
-	struct hda_gen_spec *spec = codec->spec;
 	const struct snd_pci_quirk *q;
 	int id = -1;
 	const char *name = NULL;
 
 	/* when model=nofixup is given, don't pick up any fixups */
 	if (codec->modelname && !strcmp(codec->modelname, "nofixup")) {
-		spec->fixup_list = NULL;
-		spec->fixup_id = -1;
+		codec->fixup_list = NULL;
+		codec->fixup_id = -1;
 		return;
 	}
 
@@ -759,10 +810,10 @@ void snd_hda_pick_fixup(struct hda_codec *codec,
 		}
 	}
 
-	spec->fixup_id = id;
+	codec->fixup_id = id;
 	if (id >= 0) {
-		spec->fixup_list = fixlist;
-		spec->fixup_name = name;
+		codec->fixup_list = fixlist;
+		codec->fixup_name = name;
 	}
 }
 EXPORT_SYMBOL_HDA(snd_hda_pick_fixup);

sound/pci/hda/hda_auto_parser.h

@@ -51,8 +51,9 @@ enum {
 	INPUT_PIN_ATTR_INT,	/* internal mic/line-in */
 	INPUT_PIN_ATTR_DOCK,	/* docking mic/line-in */
 	INPUT_PIN_ATTR_NORMAL,	/* mic/line-in jack */
-	INPUT_PIN_ATTR_FRONT,	/* mic/line-in jack in front */
 	INPUT_PIN_ATTR_REAR,	/* mic/line-in jack in rear */
+	INPUT_PIN_ATTR_FRONT,	/* mic/line-in jack in front */
+	INPUT_PIN_ATTR_LAST = INPUT_PIN_ATTR_FRONT,
 };
 
 int snd_hda_get_input_pin_attr(unsigned int def_conf);
@@ -89,82 +90,4 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 #define snd_hda_parse_pin_def_config(codec, cfg, ignore) \
 	snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0)
 
-/*
- */
-
-struct hda_gen_spec {
-	/* fix-up list */
-	int fixup_id;
-	const struct hda_fixup *fixup_list;
-	const char *fixup_name;
-
-	/* additional init verbs */
-	struct snd_array verbs;
-};
-
-
-/*
- * Fix-up pin default configurations and add default verbs
- */
-
-struct hda_pintbl {
-	hda_nid_t nid;
-	u32 val;
-};
-
-struct hda_model_fixup {
-	const int id;
-	const char *name;
-};
-
-struct hda_fixup {
-	int type;
-	bool chained;
-	int chain_id;
-	union {
-		const struct hda_pintbl *pins;
-		const struct hda_verb *verbs;
-		void (*func)(struct hda_codec *codec,
-			     const struct hda_fixup *fix,
-			     int action);
-	} v;
-};
-
-/* fixup types */
-enum {
-	HDA_FIXUP_INVALID,
-	HDA_FIXUP_PINS,
-	HDA_FIXUP_VERBS,
-	HDA_FIXUP_FUNC,
-};
-
-/* fixup action definitions */
-enum {
-	HDA_FIXUP_ACT_PRE_PROBE,
-	HDA_FIXUP_ACT_PROBE,
-	HDA_FIXUP_ACT_INIT,
-	HDA_FIXUP_ACT_BUILD,
-};
-
-int snd_hda_gen_add_verbs(struct hda_gen_spec *spec,
-			  const struct hda_verb *list);
-void snd_hda_gen_apply_verbs(struct hda_codec *codec);
-void snd_hda_apply_pincfgs(struct hda_codec *codec,
-			   const struct hda_pintbl *cfg);
-void snd_hda_apply_fixup(struct hda_codec *codec, int action);
-void snd_hda_pick_fixup(struct hda_codec *codec,
-			const struct hda_model_fixup *models,
-			const struct snd_pci_quirk *quirk,
-			const struct hda_fixup *fixlist);
-
-static inline void snd_hda_gen_init(struct hda_gen_spec *spec)
-{
-	snd_array_init(&spec->verbs, sizeof(struct hda_verb *), 8);
-}
-
-static inline void snd_hda_gen_free(struct hda_gen_spec *spec)
-{
-	snd_array_free(&spec->verbs);
-}
-
 #endif /* __SOUND_HDA_AUTO_PARSER_H */

sound/pci/hda/hda_codec.h

@@ -551,9 +551,6 @@ enum {
 	AC_JACK_PORT_BOTH,
 };
 
-/* max. connections to a widget */
-#define HDA_MAX_CONNECTIONS	32
-
 /* max. codec address */
 #define HDA_MAX_CODEC_ADDRESS	0x0f
 
@@ -618,6 +615,17 @@ struct hda_bus_ops {
 	/* notify power-up/down from codec to controller */
 	void (*pm_notify)(struct hda_bus *bus, bool power_up);
 #endif
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+	/* prepare DSP transfer */
+	int (*load_dsp_prepare)(struct hda_bus *bus, unsigned int format,
+				unsigned int byte_size,
+				struct snd_dma_buffer *bufp);
+	/* start/stop DSP transfer */
+	void (*load_dsp_trigger)(struct hda_bus *bus, bool start);
+	/* clean up DSP transfer */
+	void (*load_dsp_cleanup)(struct hda_bus *bus,
+				 struct snd_dma_buffer *dmab);
+#endif
 };
 
 /* template to pass to the bus constructor */
@@ -671,6 +679,8 @@ struct hda_bus {
 	unsigned int response_reset:1;	/* controller was reset */
 	unsigned int in_reset:1;	/* during reset operation */
 	unsigned int power_keep_link_on:1; /* don't power off HDA link */
+
+	int primary_dig_out_type;	/* primary digital out PCM type */
 };
 
 /*
@@ -719,9 +729,10 @@ struct hda_codec_ops {
 
 /* record for amp information cache */
 struct hda_cache_head {
-	u32 key;		/* hash key */
+	u32 key:31;		/* hash key */
+	u32 dirty:1;
 	u16 val;		/* assigned value */
-	u16 next;		/* next link; -1 = terminal */
+	u16 next;
 };
 
 struct hda_amp_info {
@@ -830,20 +841,20 @@ struct hda_codec {
 	struct hda_cache_rec amp_cache;	/* cache for amp access */
 	struct hda_cache_rec cmd_cache;	/* cache for other commands */
 
-	struct snd_array conn_lists;	/* connection-list array */
+	struct list_head conn_list;	/* linked-list of connection-list */
 
 	struct mutex spdif_mutex;
 	struct mutex control_mutex;
 	struct mutex hash_mutex;
 	struct snd_array spdif_out;
 	unsigned int spdif_in_enable;	/* SPDIF input enable? */
-	int primary_dig_out_type;	/* primary digital out PCM type */
 	const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
 	struct snd_array init_pins;	/* initial (BIOS) pin configurations */
 	struct snd_array driver_pins;	/* pin configs set by codec parser */
 	struct snd_array cvt_setups;	/* audio convert setups */
 
 #ifdef CONFIG_SND_HDA_HWDEP
+	struct mutex user_mutex;
 	struct snd_hwdep *hwdep;	/* assigned hwdep device */
 	struct snd_array init_verbs;	/* additional init verbs */
 	struct snd_array hints;		/* additional hints */
@@ -865,8 +876,11 @@ struct hda_codec {
 	unsigned int pins_shutup:1;	/* pins are shut up */
 	unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
 	unsigned int no_jack_detect:1;	/* Machine has no jack-detection */
+	unsigned int inv_eapd:1; /* broken h/w: inverted EAPD control */
+	unsigned int inv_jack_detect:1;	/* broken h/w: inverted detection bit */
 	unsigned int pcm_format_first:1; /* PCM format must be set first */
 	unsigned int epss:1;		/* supporting EPSS? */
+	unsigned int cached_write:1;	/* write only to caches */
 #ifdef CONFIG_PM
 	unsigned int power_on :1;	/* current (global) power-state */
 	unsigned int d3_stop_clk:1;	/* support D3 operation without BCLK */
@@ -881,6 +895,10 @@ struct hda_codec {
 	spinlock_t power_lock;
 #endif
 
+	/* filter the requested power state per nid */
+	unsigned int (*power_filter)(struct hda_codec *codec, hda_nid_t nid,
+				     unsigned int power_state);
+
 	/* codec-specific additional proc output */
 	void (*proc_widget_hook)(struct snd_info_buffer *buffer,
 				 struct hda_codec *codec, hda_nid_t nid);
@@ -894,6 +912,14 @@ struct hda_codec {
 	/* jack detection */
 	struct snd_array jacks;
 #endif
+
+	/* fix-up list */
+	int fixup_id;
+	const struct hda_fixup *fixup_list;
+	const char *fixup_name;
+
+	/* additional init verbs */
+	struct snd_array verbs;
 };
 
 /* direction */
@@ -910,6 +936,7 @@ int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp,
 int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
 		      struct hda_codec **codecp);
 int snd_hda_codec_configure(struct hda_codec *codec);
+int snd_hda_codec_update_widgets(struct hda_codec *codec);
 
 /*
  * low level functions
@@ -930,8 +957,11 @@ snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid)
 {
 	return snd_hda_get_connections(codec, nid, NULL, 0);
 }
+int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid);
 int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
 			    hda_nid_t *conn_list, int max_conns);
+int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
+			  const hda_nid_t **listp);
 int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums,
 			  const hda_nid_t *list);
 int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
@@ -952,7 +982,6 @@ void snd_hda_sequence_write(struct hda_codec *codec,
 int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex);
 
 /* cached write */
-#ifdef CONFIG_PM
 int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
 			      int direct, unsigned int verb, unsigned int parm);
 void snd_hda_sequence_write_cache(struct hda_codec *codec,
@@ -960,17 +989,14 @@ void snd_hda_sequence_write_cache(struct hda_codec *codec,
 int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid,
 			      int direct, unsigned int verb, unsigned int parm);
 void snd_hda_codec_resume_cache(struct hda_codec *codec);
-#else
-#define snd_hda_codec_write_cache	snd_hda_codec_write
-#define snd_hda_codec_update_cache	snd_hda_codec_write
-#define snd_hda_sequence_write_cache	snd_hda_sequence_write
-#endif
+/* both for cmd & amp caches */
+void snd_hda_codec_flush_cache(struct hda_codec *codec);
 
 /* the struct for codec->pin_configs */
 struct hda_pincfg {
 	hda_nid_t nid;
-	unsigned char ctrl;	/* current pin control value */
-	unsigned char pad;	/* reserved */
+	unsigned char ctrl;	/* original pin control value */
+	unsigned char target;	/* target pin control value */
 	unsigned int cfg;	/* default configuration */
 };
 
@@ -1036,8 +1062,7 @@ extern const struct snd_pcm_chmap_elem snd_pcm_2_1_chmaps[];
 void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen);
 void snd_hda_bus_reboot_notify(struct hda_bus *bus);
 void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg,
-				    unsigned int power_state,
-				    bool eapd_workaround);
+				    unsigned int power_state);
 
 int snd_hda_lock_devices(struct hda_bus *bus);
 void snd_hda_unlock_devices(struct hda_bus *bus);
@@ -1136,6 +1161,40 @@ static inline void snd_hda_power_sync(struct hda_codec *codec)
 int snd_hda_load_patch(struct hda_bus *bus, size_t size, const void *buf);
 #endif
 
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+static inline int
+snd_hda_codec_load_dsp_prepare(struct hda_codec *codec, unsigned int format,
+				unsigned int size,
+				struct snd_dma_buffer *bufp)
+{
+	return codec->bus->ops.load_dsp_prepare(codec->bus, format, size, bufp);
+}
+static inline void
+snd_hda_codec_load_dsp_trigger(struct hda_codec *codec, bool start)
+{
+	return codec->bus->ops.load_dsp_trigger(codec->bus, start);
+}
+static inline void
+snd_hda_codec_load_dsp_cleanup(struct hda_codec *codec,
+				struct snd_dma_buffer *dmab)
+{
+	return codec->bus->ops.load_dsp_cleanup(codec->bus, dmab);
+}
+#else
+static inline int
+snd_hda_codec_load_dsp_prepare(struct hda_codec *codec, unsigned int format,
+				unsigned int size,
+				struct snd_dma_buffer *bufp)
+{
+	return -ENOSYS;
+}
+static inline void
+snd_hda_codec_load_dsp_trigger(struct hda_codec *codec, bool start) {}
+static inline void
+snd_hda_codec_load_dsp_cleanup(struct hda_codec *codec,
+				struct snd_dma_buffer *dmab) {}
+#endif
+
 /*
  * Codec modularization
  */

sound/pci/hda/hda_eld.c

@@ -246,8 +246,8 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a,
 /*
  * Be careful, ELD buf could be totally rubbish!
  */
-static int hdmi_update_eld(struct hdmi_eld *e,
-			   const unsigned char *buf, int size)
+int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e,
+			  const unsigned char *buf, int size)
 {
 	int mnl;
 	int i;
@@ -260,7 +260,6 @@ static int hdmi_update_eld(struct hdmi_eld *e,
 		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);
@@ -305,7 +304,6 @@ static int hdmi_update_eld(struct hdmi_eld *e,
 	if (!e->spk_alloc)
 		e->spk_alloc = 0xffff;
 
-	e->eld_valid = true;
 	return 0;
 
 out_fail:
@@ -318,17 +316,16 @@ int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid)
 						 AC_DIPSIZE_ELD_BUF);
 }
 
-int snd_hdmi_get_eld(struct hdmi_eld *eld,
-		     struct hda_codec *codec, hda_nid_t nid)
+int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid,
+		     unsigned char *buf, int *eld_size)
 {
 	int i;
 	int ret;
 	int size;
-	unsigned char *buf;
 
 	/*
 	 * ELD size is initialized to zero in caller function. If no errors and
-	 * ELD is valid, actual eld_size is assigned in hdmi_update_eld()
+	 * ELD is valid, actual eld_size is assigned.
 	 */
 
 	size = snd_hdmi_get_eld_size(codec, nid);
@@ -343,8 +340,6 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
 	}
 
 	/* set ELD buffer */
-	buf = eld->eld_buffer;
-
 	for (i = 0; i < size; i++) {
 		unsigned int val = hdmi_get_eld_data(codec, nid, i);
 		/*
@@ -372,8 +367,7 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
 		buf[i] = val;
 	}
 
-	ret = hdmi_update_eld(eld, buf, size);
-
+	*eld_size = size;
 error:
 	return ret;
 }
@@ -438,7 +432,7 @@ void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen)
 	buf[j] = '\0';	/* necessary when j == 0 */
 }
 
-void snd_hdmi_show_eld(struct hdmi_eld *e)
+void snd_hdmi_show_eld(struct parsed_hdmi_eld *e)
 {
 	int i;
 
@@ -487,10 +481,11 @@ static void hdmi_print_sad_info(int i, struct cea_sad *a,
 static void hdmi_print_eld_info(struct snd_info_entry *entry,
 				struct snd_info_buffer *buffer)
 {
-	struct hdmi_eld *e = entry->private_data;
+	struct hdmi_eld *eld = entry->private_data;
+	struct parsed_hdmi_eld *e = &eld->info;
 	char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE];
 	int i;
-	static char *eld_versoin_names[32] = {
+	static char *eld_version_names[32] = {
 		"reserved",
 		"reserved",
 		"CEA-861D or below",
@@ -505,15 +500,18 @@ static void hdmi_print_eld_info(struct snd_info_entry *entry,
 		[4 ... 7] = "reserved"
 	};
 
-	snd_iprintf(buffer, "monitor_present\t\t%d\n", e->monitor_present);
-	snd_iprintf(buffer, "eld_valid\t\t%d\n", e->eld_valid);
-	if (!e->eld_valid)
+	mutex_lock(&eld->lock);
+	snd_iprintf(buffer, "monitor_present\t\t%d\n", eld->monitor_present);
+	snd_iprintf(buffer, "eld_valid\t\t%d\n", eld->eld_valid);
+	if (!eld->eld_valid) {
+		mutex_unlock(&eld->lock);
 		return;
+	}
 	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]);
+					eld_version_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);
@@ -530,18 +528,21 @@ static void hdmi_print_eld_info(struct snd_info_entry *entry,
 
 	for (i = 0; i < e->sad_count; i++)
 		hdmi_print_sad_info(i, e->sad + i, buffer);
+	mutex_unlock(&eld->lock);
 }
 
 static void hdmi_write_eld_info(struct snd_info_entry *entry,
 				struct snd_info_buffer *buffer)
 {
-	struct hdmi_eld *e = entry->private_data;
+	struct hdmi_eld *eld = entry->private_data;
+	struct parsed_hdmi_eld *e = &eld->info;
 	char line[64];
 	char name[64];
 	char *sname;
 	long long val;
 	unsigned int n;
 
+	mutex_lock(&eld->lock);
 	while (!snd_info_get_line(buffer, line, sizeof(line))) {
 		if (sscanf(line, "%s %llx", name, &val) != 2)
 			continue;
@@ -551,9 +552,9 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry,
 		 * 	eld_version edid_version
 		 */
 		if (!strcmp(name, "monitor_present"))
-			e->monitor_present = val;
+			eld->monitor_present = val;
 		else if (!strcmp(name, "eld_valid"))
-			e->eld_valid = val;
+			eld->eld_valid = val;
 		else if (!strcmp(name, "connection_type"))
 			e->conn_type = val;
 		else if (!strcmp(name, "port_id"))
@@ -593,6 +594,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry,
 				e->sad_count = n + 1;
 		}
 	}
+	mutex_unlock(&eld->lock);
 }
 
 
@@ -627,7 +629,7 @@ void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld)
 #endif /* CONFIG_PROC_FS */
 
 /* update PCM info based on ELD */
-void snd_hdmi_eld_update_pcm_info(struct hdmi_eld *eld,
+void snd_hdmi_eld_update_pcm_info(struct parsed_hdmi_eld *e,
 			      struct hda_pcm_stream *hinfo)
 {
 	u32 rates;
@@ -644,8 +646,8 @@ void snd_hdmi_eld_update_pcm_info(struct hdmi_eld *eld,
 	formats = SNDRV_PCM_FMTBIT_S16_LE;
 	maxbps = 16;
 	channels_max = 2;
-	for (i = 0; i < eld->sad_count; i++) {
-		struct cea_sad *a = &eld->sad[i];
+	for (i = 0; i < e->sad_count; i++) {
+		struct cea_sad *a = &e->sad[i];
 		rates |= a->rates;
 		if (a->channels > channels_max)
 			channels_max = a->channels;

sound/pci/hda/hda_hwdep.c

@@ -148,6 +148,7 @@ int snd_hda_create_hwdep(struct hda_codec *codec)
 	hwdep->ops.ioctl_compat = hda_hwdep_ioctl_compat;
 #endif
 
+	mutex_init(&codec->user_mutex);
 	snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32);
 	snd_array_init(&codec->hints, sizeof(struct hda_hint), 32);
 	snd_array_init(&codec->user_pins, sizeof(struct hda_pincfg), 16);
@@ -346,12 +347,14 @@ static ssize_t init_verbs_show(struct device *dev,
 	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
 	struct hda_codec *codec = hwdep->private_data;
 	int i, len = 0;
+	mutex_lock(&codec->user_mutex);
 	for (i = 0; i < codec->init_verbs.used; i++) {
 		struct hda_verb *v = snd_array_elem(&codec->init_verbs, i);
 		len += snprintf(buf + len, PAGE_SIZE - len,
 				"0x%02x 0x%03x 0x%04x\n",
 				v->nid, v->verb, v->param);
 	}
+	mutex_unlock(&codec->user_mutex);
 	return len;
 }
 
@@ -364,12 +367,16 @@ static int parse_init_verbs(struct hda_codec *codec, const char *buf)
 		return -EINVAL;
 	if (!nid || !verb)
 		return -EINVAL;
+	mutex_lock(&codec->user_mutex);
 	v = snd_array_new(&codec->init_verbs);
-	if (!v)
+	if (!v) {
+		mutex_unlock(&codec->user_mutex);
 		return -ENOMEM;
+	}
 	v->nid = nid;
 	v->verb = verb;
 	v->param = param;
+	mutex_unlock(&codec->user_mutex);
 	return 0;
 }
 
@@ -392,11 +399,13 @@ static ssize_t hints_show(struct device *dev,
 	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
 	struct hda_codec *codec = hwdep->private_data;
 	int i, len = 0;
+	mutex_lock(&codec->user_mutex);
 	for (i = 0; i < codec->hints.used; i++) {
 		struct hda_hint *hint = snd_array_elem(&codec->hints, i);
 		len += snprintf(buf + len, PAGE_SIZE - len,
 				"%s = %s\n", hint->key, hint->val);
 	}
+	mutex_unlock(&codec->user_mutex);
 	return len;
 }
 
@@ -431,6 +440,7 @@ static int parse_hints(struct hda_codec *codec, const char *buf)
 {
 	char *key, *val;
 	struct hda_hint *hint;
+	int err = 0;
 
 	buf = skip_spaces(buf);
 	if (!*buf || *buf == '#' || *buf == '\n')
@@ -450,26 +460,31 @@ static int parse_hints(struct hda_codec *codec, const char *buf)
 	val = skip_spaces(val);
 	remove_trail_spaces(key);
 	remove_trail_spaces(val);
+	mutex_lock(&codec->user_mutex);
 	hint = get_hint(codec, key);
 	if (hint) {
 		/* replace */
 		kfree(hint->key);
 		hint->key = key;
 		hint->val = val;
-		return 0;
+		goto unlock;
 	}
 	/* allocate a new hint entry */
 	if (codec->hints.used >= MAX_HINTS)
 		hint = NULL;
 	else
 		hint = snd_array_new(&codec->hints);
-	if (!hint) {
-		kfree(key);
-		return -ENOMEM;
+	if (hint) {
+		hint->key = key;
+		hint->val = val;
+	} else {
+		err = -ENOMEM;
 	}
-	hint->key = key;
-	hint->val = val;
-	return 0;
+ unlock:
+	mutex_unlock(&codec->user_mutex);
+	if (err)
+		kfree(key);
+	return err;
 }
 
 static ssize_t hints_store(struct device *dev,
@@ -489,11 +504,13 @@ static ssize_t pin_configs_show(struct hda_codec *codec,
 				char *buf)
 {
 	int i, len = 0;
+	mutex_lock(&codec->user_mutex);
 	for (i = 0; i < list->used; i++) {
 		struct hda_pincfg *pin = snd_array_elem(list, i);
 		len += sprintf(buf + len, "0x%02x 0x%08x\n",
 			       pin->nid, pin->cfg);
 	}
+	mutex_unlock(&codec->user_mutex);
 	return len;
 }
 
@@ -528,13 +545,16 @@ static ssize_t driver_pin_configs_show(struct device *dev,
 
 static int parse_user_pin_configs(struct hda_codec *codec, const char *buf)
 {
-	int nid, cfg;
+	int nid, cfg, err;
 
 	if (sscanf(buf, "%i %i", &nid, &cfg) != 2)
 		return -EINVAL;
 	if (!nid)
 		return -EINVAL;
-	return snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg);
+	mutex_lock(&codec->user_mutex);
+	err = snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg);
+	mutex_unlock(&codec->user_mutex);
+	return err;
 }
 
 static ssize_t user_pin_configs_store(struct device *dev,
@@ -600,19 +620,50 @@ EXPORT_SYMBOL_HDA(snd_hda_get_hint);
 
 int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
 {
-	const char *p = snd_hda_get_hint(codec, key);
+	const char *p;
+	int ret;
+
+	mutex_lock(&codec->user_mutex);
+	p = snd_hda_get_hint(codec, key);
 	if (!p || !*p)
-		return -ENOENT;
-	switch (toupper(*p)) {
-	case 'T': /* true */
-	case 'Y': /* yes */
-	case '1':
-		return 1;
+		ret = -ENOENT;
+	else {
+		switch (toupper(*p)) {
+		case 'T': /* true */
+		case 'Y': /* yes */
+		case '1':
+			ret = 1;
+			break;
+		default:
+			ret = 0;
+			break;
+		}
 	}
-	return 0;
+	mutex_unlock(&codec->user_mutex);
+	return ret;
 }
 EXPORT_SYMBOL_HDA(snd_hda_get_bool_hint);
 
+int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp)
+{
+	const char *p;
+	unsigned long val;
+	int ret;
+
+	mutex_lock(&codec->user_mutex);
+	p = snd_hda_get_hint(codec, key);
+	if (!p)
+		ret = -ENOENT;
+	else if (strict_strtoul(p, 0, &val))
+		ret = -EINVAL;
+	else {
+		*valp = val;
+		ret = 0;
+	}
+	mutex_unlock(&codec->user_mutex);
+	return ret;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_int_hint);
 #endif /* CONFIG_SND_HDA_RECONFIG */
 
 #ifdef CONFIG_SND_HDA_PATCH_LOADER

sound/pci/hda/hda_jack.c

@@ -29,7 +29,8 @@ bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
 	if (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
 	     AC_DEFCFG_MISC_NO_PRESENCE)
 		return false;
-	if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP))
+	if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) &&
+	    !codec->jackpoll_interval)
 		return false;
 	return true;
 }
@@ -39,6 +40,7 @@ EXPORT_SYMBOL_HDA(is_jack_detectable);
 static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid)
 {
 	u32 pincap;
+	u32 val;
 
 	if (!codec->no_trigger_sense) {
 		pincap = snd_hda_query_pin_caps(codec, nid);
@@ -46,8 +48,11 @@ static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid)
 			snd_hda_codec_read(codec, nid, 0,
 					AC_VERB_SET_PIN_SENSE, 0);
 	}
-	return snd_hda_codec_read(codec, nid, 0,
+	val = snd_hda_codec_read(codec, nid, 0,
 				  AC_VERB_GET_PIN_SENSE, 0);
+	if (codec->inv_jack_detect)
+		val ^= AC_PINSENSE_PRESENCE;
+	return val;
 }
 
 /**

sound/pci/hda/hda_proc.c

@@ -22,6 +22,7 @@
  */
 
 #include <linux/init.h>
+#include <linux/slab.h>
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
@@ -138,16 +139,17 @@ static void print_amp_vals(struct snd_info_buffer *buffer,
 	dir = dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
 	for (i = 0; i < indices; i++) {
 		snd_iprintf(buffer, " [");
+		val = snd_hda_codec_read(codec, nid, 0,
+					 AC_VERB_GET_AMP_GAIN_MUTE,
+					 AC_AMP_GET_LEFT | dir | i);
+		snd_iprintf(buffer, "0x%02x", val);
 		if (stereo) {
 			val = snd_hda_codec_read(codec, nid, 0,
 						 AC_VERB_GET_AMP_GAIN_MUTE,
-						 AC_AMP_GET_LEFT | dir | i);
-			snd_iprintf(buffer, "0x%02x ", val);
+						 AC_AMP_GET_RIGHT | dir | i);
+			snd_iprintf(buffer, " 0x%02x", val);
 		}
-		val = snd_hda_codec_read(codec, nid, 0,
-					 AC_VERB_GET_AMP_GAIN_MUTE,
-					 AC_AMP_GET_RIGHT | dir | i);
-		snd_iprintf(buffer, "0x%02x]", val);
+		snd_iprintf(buffer, "]");
 	}
 	snd_iprintf(buffer, "\n");
 }
@@ -603,6 +605,8 @@ static void print_codec_info(struct snd_info_entry *entry,
 	print_amp_caps(buffer, codec, codec->afg, HDA_INPUT);
 	snd_iprintf(buffer, "Default Amp-Out caps: ");
 	print_amp_caps(buffer, codec, codec->afg, HDA_OUTPUT);
+	snd_iprintf(buffer, "State of AFG node 0x%02x:\n", codec->afg);
+	print_power_state(buffer, codec, codec->afg);
 
 	nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid);
 	if (! nid || nodes < 0) {
@@ -620,7 +624,7 @@ static void print_codec_info(struct snd_info_entry *entry,
 			snd_hda_param_read(codec, nid,
 					   AC_PAR_AUDIO_WIDGET_CAP);
 		unsigned int wid_type = get_wcaps_type(wid_caps);
-		hda_nid_t conn[HDA_MAX_CONNECTIONS];
+		hda_nid_t *conn = NULL;
 		int conn_len = 0;
 
 		snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid,
@@ -657,9 +661,18 @@ static void print_codec_info(struct snd_info_entry *entry,
 		if (wid_type == AC_WID_VOL_KNB)
 			wid_caps |= AC_WCAP_CONN_LIST;
 
-		if (wid_caps & AC_WCAP_CONN_LIST)
-			conn_len = snd_hda_get_raw_connections(codec, nid, conn,
-							   HDA_MAX_CONNECTIONS);
+		if (wid_caps & AC_WCAP_CONN_LIST) {
+			conn_len = snd_hda_get_num_raw_conns(codec, nid);
+			if (conn_len > 0) {
+				conn = kmalloc(sizeof(hda_nid_t) * conn_len,
+					       GFP_KERNEL);
+				if (!conn)
+					return;
+				if (snd_hda_get_raw_connections(codec, nid, conn,
+								conn_len) < 0)
+					conn_len = 0;
+			}
+		}
 
 		if (wid_caps & AC_WCAP_IN_AMP) {
 			snd_iprintf(buffer, "  Amp-In caps: ");
@@ -732,6 +745,8 @@ static void print_codec_info(struct snd_info_entry *entry,
 
 		if (codec->proc_widget_hook)
 			codec->proc_widget_hook(buffer, codec, nid);
+
+		kfree(conn);
 	}
 	snd_hda_power_down(codec);
 }

sound/pci/ice1712/wm8766.c

@@ -31,7 +31,7 @@
 
 static void snd_wm8766_write(struct snd_wm8766 *wm, u16 addr, u16 data)
 {
-	if (addr < WM8766_REG_RESET)
+	if (addr < WM8766_REG_COUNT)
 		wm->regs[addr] = data;
 	wm->ops.write(wm, addr, data);
 }

sound/pci/intel8x0.c

@@ -3266,11 +3266,13 @@ static int check_default_spdif_aclink(struct pci_dev *pci)
 	w = snd_pci_quirk_lookup(pci, spdif_aclink_defaults);
 	if (w) {
 		if (w->value)
-			snd_printdd(KERN_INFO "intel8x0: Using SPDIF over "
-				    "AC-Link for %s\n", w->name);
+			snd_printdd(KERN_INFO
+				    "intel8x0: Using SPDIF over AC-Link for %s\n",
+				    snd_pci_quirk_name(w));
 		else
-			snd_printdd(KERN_INFO "intel8x0: Using integrated "
-				    "SPDIF DMA for %s\n", w->name);
+			snd_printdd(KERN_INFO
+				    "intel8x0: Using integrated SPDIF DMA for %s\n",
+				    snd_pci_quirk_name(w));
 		return w->value;
 	}
 	return 0;

sound/pci/maestro3.c

@@ -2586,8 +2586,9 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci,
 	else {
 		quirk = snd_pci_quirk_lookup(pci, m3_amp_quirk_list);
 		if (quirk) {
-			snd_printdd(KERN_INFO "maestro3: set amp-gpio "
-				    "for '%s'\n", quirk->name);
+			snd_printdd(KERN_INFO
+				    "maestro3: set amp-gpio for '%s'\n",
+				    snd_pci_quirk_name(quirk));
 			chip->amp_gpio = quirk->value;
 		} else if (chip->allegro_flag)
 			chip->amp_gpio = GPO_EXT_AMP_ALLEGRO;
@@ -2597,8 +2598,9 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci,
 
 	quirk = snd_pci_quirk_lookup(pci, m3_irda_quirk_list);
 	if (quirk) {
-		snd_printdd(KERN_INFO "maestro3: enabled irda workaround "
-			    "for '%s'\n", quirk->name);
+		snd_printdd(KERN_INFO
+			    "maestro3: enabled irda workaround for '%s'\n",
+			    snd_pci_quirk_name(quirk));
 		chip->irda_workaround = 1;
 	}
 	quirk = snd_pci_quirk_lookup(pci, m3_hv_quirk_list);