Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Mostly many small changes spread as seen in diffstat in sound/*
  directory by this update.  A significant change in the subsystem level
  is the introduction of snd_soc_component, which will help more generic
  handling of SoC and off-SoC components.

  Also, snd_BUG_ON() macro is enabled unconditionally now due to its
  misuses, so people might hit kernel warnings (it's a good thing for
  us).

   - compress-offload: support for capture by Charles Keepax
   - HD-audio: codec delay support by Dylan Reid
   - HD-audio: improvements/fixes in generic parser: better headphone
     mic and headset mic support, jack_modes hint consolidation, proper
     beep attach/detachment, generalized power filter controls by David
     Henningsson, et al
   - HD-audio: Improved management of HDMI codec pins/converters
   - HD-audio: Better pin/DAC assignment for VIA codecs
   - HD-audio: Haswell HDMI workarounds
   - HD-audio: ALC268 codec support, a few new quirks for Chromebooks
   - USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
     fix by Clemens Ladisch
   - USB: support for DSD formats by Daniel Mack
   - USB: A few UAC2 device endian/cock fixes by Eldad Zack
   - USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
     Yamaha THRxx devices
   - HDSPM: updates for TCO controls by Adrian Knoth
   - ASoC: Add a snd_soc_component object type for generic handling of
     SoC and off-SoC components by Kuninori Morimoto,
   - dmaengine: a large set of cleanups and conversions by Lars-Peter
     Clausen
   - ASoC DAPM: performance optimizations from Ryo Tsutsui
   - ASoC DAPM: support for mixer control sharing by Stephen Warren
   - ASoC: multiplatform ARM cleanups from Arnd Bergmann
   - ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"

* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
  ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
  ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
  ALSA: pcm_format_to_bits strong-typed conversion
  ALSA: compress: fix the states to check for allowing read
  ALSA: hda - Move Thinkpad X220 to use auto parser
  ALSA: USB: adjust for changed 3.8 USB API
  ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
  sound: oss/dmabuf: use dma_map_single
  ALSA: ali5451: use mdelay instead of large udelay constants
  ALSA: hda - Add the support for ALC286 codec
  ALSA: usb-audio: USB quirk for Yamaha THR10C
  ALSA: usb-audio: USB quirk for Yamaha THR5A
  ALSA: usb-audio: USB quirk for Yamaha THR10
  ALSA: usb-audio: Fix autopm error during probing
  ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
  ALSA: sound kconfig typo
  ALSA: emu10k1: Fix dock firmware loading
  ASoC: ux500: forward declare msp_i2s_platform_data
  ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
  ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
  ...

5 files changed, 83 insertions, 16 deletions

diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl
index bd6fee22c4d..06741e92598 100644
--- a/Documentation/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl
@@ -6164,14 +6164,12 @@ struct _snd_pcm_runtime {
 
       <para>
         The macro takes an conditional expression to evaluate.
-	When <constant>CONFIG_SND_DEBUG</constant>, is set, the
-	expression is actually evaluated. If it's non-zero, it shows
-	the warning message such as
+	When <constant>CONFIG_SND_DEBUG</constant>, is set, if the
+	expression is non-zero, it shows the warning message such as
 	<computeroutput>BUG? (xxx)</computeroutput>
-	normally followed by stack trace.  It returns the evaluated
-	value.
-	When no <constant>CONFIG_SND_DEBUG</constant> is set, this
-	macro always returns zero.
+	normally followed by stack trace.
+
+	In both cases it returns the evaluated value.
       </para>
 
     </section>
diff --git a/Documentation/devicetree/bindings/sound/ak5386.txt b/Documentation/devicetree/bindings/sound/ak5386.txt
new file mode 100644
index 00000000000..dc3914fe6ce
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ak5386.txt
@@ -0,0 +1,19 @@
+AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC
+
+This device has no control interface.
+
+Required properties:
+
+  - compatible : "asahi-kasei,ak5386"
+
+Optional properties:
+
+  - reset-gpio : a GPIO spec for the reset/power down pin.
+		 If specified, it will be deasserted at probe time.
+
+Example:
+
+spdif: ak5386@0 {
+	compatible = "asahi-kasei,ak5386";
+	reset-gpio = <&gpio0 23>;
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt
index 1ac7b164218..0e5c12c6652 100644
--- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt
@@ -1,12 +1,22 @@
 NVIDIA Tegra30 AHUB (Audio Hub)
 
 Required properties:
-- compatible : "nvidia,tegra30-ahub"
+- compatible : "nvidia,tegra30-ahub", "nvidia,tegra114-ahub", etc.
 - reg : Should contain the register physical address and length for each of
-  the AHUB's APBIF registers and the AHUB's own registers.
+  the AHUB's register blocks.
+  - Tegra30 requires 2 entries, for the APBIF and AHUB/AUDIO register blocks.
+  - Tegra114 requires an additional entry, for the APBIF2 register block.
 - interrupts : Should contain AHUB interrupt
-- nvidia,dma-request-selector : The Tegra DMA controller's phandle and
-  request selector for the first APBIF channel.
+- nvidia,dma-request-selector : A list of the DMA channel specifiers. Each
+  entry contains the Tegra DMA controller's phandle and request selector.
+  If a single entry is present, the request selectors for the channels are
+  assumed to be contiguous, and increment from this value.
+  If multiple values are given, one value must be given per channel.
+- clocks : Must contain an entry for each required entry in clock-names.
+- clock-names : Must include the following entries:
+  - Tegra30: Requires d_audio, apbif, i2s0, i2s1, i2s2, i2s3, i2s4, dam0,
+    dam1, dam2, spdif_in.
+  - Tegra114: Additionally requires amx, adx.
 - ranges : The bus address mapping for the configlink register bus.
   Can be empty since the mapping is 1:1.
 - #address-cells : For the configlink bus. Should be <1>;
@@ -25,7 +35,13 @@ ahub@70080000 {
 	reg = <0x70080000 0x200 0x70080200 0x100>;
 	interrupts = < 0 103 0x04 >;
 	nvidia,dma-request-selector = <&apbdma 1>;
-
+	clocks = <&tegra_car 106>, <&tegra_car 107>, <&tegra_car 30>,
+		<&tegra_car 11>, <&tegra_car 18>, <&tegra_car 101>,
+		<&tegra_car 102>, <&tegra_car 108>, <&tegra_car 109>,
+		<&tegra_car 110>, <&tegra_car 162>;
+	clock-names = "d_audio", "apbif", "i2s0", "i2s1", "i2s2",
+		"i2s3", "i2s4", "dam0", "dam1", "dam2",
+		"spdif_in";
 	ranges;
 	#address-cells = <1>;
 	#size-cells = <1>;
diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt
new file mode 100644
index 00000000000..8ea4f5b4818
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt
@@ -0,0 +1,32 @@
+Texas Instruments TAS5086 6-channel PWM Processor
+
+Required properties:
+
+ - compatible:		Should contain "ti,tas5086".
+ - reg:			The i2c address. Should contain <0x1b>.
+
+Optional properties:
+
+ - reset-gpio: 		A GPIO spec to define which pin is connected to the
+			chip's !RESET pin. If specified, the driver will
+			assert a hardware reset at probe time.
+
+ - ti,charge-period:	This property should contain the time in microseconds
+			that closely matches the external single-ended
+			split-capacitor charge period. The hardware chip
+			waits for this period of time before starting the
+			PWM signals. This helps reduce pops and clicks.
+
+			When not specified, the hardware default of 1300ms
+			is retained.
+
+Examples:
+
+	i2c_bus {
+		tas5086@1b {
+			compatible = "ti,tas5086";
+			reg = <0x1b>;
+			reset-gpio = <&gpio 23 0>;
+			ti,charge-period = <156000>;
+		};
+	};
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index d4faa63ff35..c3c912d023c 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -461,11 +461,13 @@ The generic parser supports the following hints:
   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
+- add_jack_modes (bool): add "xxx Jack Mode" enum controls to each
+  I/O jack for allowing to change the headphone amp and mic bias VREF
+  capabilities
 - power_down_unused (bool): power down the unused widgets
+- add_hp_mic (bool): add the headphone to capture source if possible
+- hp_mic_detect (bool): enable/disable the hp/mic shared input for a
+  single built-in mic case; default true
 - mixer_nid (int): specifies the widget NID of the analog-loopback
   mixer