Merge tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai: "There have been major modernization with the standard bus: in ALSA sequencer core and HD-audio. Also, HD-audio receives the regmap support replacing the in-house cache register cache code. These changes shouldn't impact the existing behavior, but rather refactoring. In addition, HD-audio got the code split to a core library part and the "legacy" driver parts. This is a preliminary work for adapting the upcoming ASoC HD-audio driver, and the whole transition is still work in progress, likely finished in 4.1. Along with them, there are many updates in ASoC area as usual, too: lots of cleanups, Intel code shuffling, etc. Here are some highlights: ALSA core: - PCM: the audio timestamp / wallclock enhancement - PCM: fixes in DPCM management - Fixes / cleanups of user-space control element management - Sequencer: modernization using the standard bus HD-audio: - Modernization using the standard bus - Regmap support - Use standard runtime PM for codec power saving - Widget-path based power-saving for IDT, VIA and Realtek codecs - Reorganized sysfs entries for each codec object - More Dell headset support ASoC: - Move of jack registration to the card level - Lots of ASoC cleanups, mainly moving things from the CODEC level to the card level - Support for DAPM routes specified by both the machine driver and DT - Continuing improvements to rcar - pcm512x enhacements - Intel platforms updates - rt5670 updates / fixes - New platforms / devices: some non-DSP Qualcomm platforms, Google's Storm platform, Maxmim MAX98925 CODECs and the Ingenic JZ4780 SoC Misc: - ice1724: Improved ESI W192M support - emu10k1: Emu 1010 fixes/enhancement" * tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (411 commits) ALSA: hda - set GET bit when adding a vendor verb to the codec regmap ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T450 ALSA: hda - Fix another race in runtime PM refcounting ALSA: hda - Expose codec type sysfs ALSA: ctl: fix to handle several elements added by one operation for userspace element ASoC: Intel: fix array_size.cocci warnings ASoC: n810: Automatically disconnect non-connected pins ASoC: n810: Consistently pass the card DAPM context to n810_ext_control() ASoC: davinci-evm: Use card DAPM context to access widgets ASoC: mop500_ab8500: Use card DAPM context to access widgets ASoC: wm1133-ev1: Use card DAPM context to access widgets ASoC: atmel: Improve machine driver compile test coverage ASoC: atmel: Add dependency to SND_SOC_I2C_AND_SPI where necessary ALSA: control: Fix a typo of SNDRV_CTL_ELEM_ACCESS_TLV_* with SNDRV_CTL_TLV_OP_* ALSA: usb-audio: Don't attempt to get Microsoft Lifecam Cinema sample rate ASoC: rnsd: fix build regression without CONFIG_OF ALSA: emu10k1: add toggles for E-mu 1010 optical ports ALSA: ctl: fill identical information to return value when adding userspace elements ALSA: ctl: fix a bug to return no identical information in info operation for userspace controls ALSA: ctl: confirm to return all identical information in 'activate' event ...
2026-09-21 14:15:21 -07:00 · 2015-04-15 15:41:41 -07:00
parent 6d50ff91d9 d6eb9e3ec7
commit d0a3997c0c
315 changed files with 15198 additions and 9045 deletions
@@ -1,7 +1,7 @@
 Ingenic JZ4740 I2S controller

 Required properties:
- compatible : "ingenic,jz4740-i2s"
+- compatible : "ingenic,jz4740-i2s" or "ingenic,jz4780-i2s"
 - reg : I2S registers location and length
 - clocks : AIC and I2S PLL clock specifiers.
 - clock-names: "aic" and "i2s"
@@ -0,0 +1,22 @@
+max98925 audio CODEC
+
+This device supports I2C.
+
+Required properties:
+
+  - compatible : "maxim,max98925"
+
+  - vmon-slot-no : slot number used to send voltage information
+
+  - imon-slot-no : slot number used to send current information
+
+  - reg : the I2C address of the device for I2C
+
+Example:
+
+codec: max98925@1a {
+	compatible = "maxim,max98925";
+	vmon-slot-no = <0>;
+	imon-slot-no = <2>;
+	reg = <0x1a>;
+};
@@ -18,6 +18,7 @@ Required properties:
  * Headphones
  * Speakers
  * Mic Jack
+  * Int Mic

 - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
  connected to the CODEC.
@@ -0,0 +1,43 @@
+* Qualcomm Technologies LPASS CPU DAI
+
+This node models the Qualcomm Technologies Low-Power Audio SubSystem (LPASS).
+
+Required properties:
+
+- compatible		: "qcom,lpass-cpu"
+- clocks		: Must contain an entry for each entry in clock-names.
+- clock-names		: A list which must include the following entries:
+				* "ahbix-clk"
+				* "mi2s-osr-clk"
+				* "mi2s-bit-clk"
+- interrupts		: Must contain an entry for each entry in
+			  interrupt-names.
+- interrupt-names	: A list which must include the following entries:
+				* "lpass-irq-lpaif"
+- pinctrl-N		: One property must exist for each entry in
+			  pinctrl-names.  See ../pinctrl/pinctrl-bindings.txt
+			  for details of the property values.
+- pinctrl-names		: Must contain a "default" entry.
+- reg			: Must contain an address for each entry in reg-names.
+- reg-names		: A list which must include the following entries:
+				* "lpass-lpaif"
+
+Optional properties:
+
+- qcom,adsp		: Phandle for the audio DSP node
+
+Example:
+
+lpass@28100000 {
+	compatible = "qcom,lpass-cpu";
+	clocks = <&lcc AHBIX_CLK>, <&lcc MI2S_OSR_CLK>, <&lcc MI2S_BIT_CLK>;
+	clock-names = "ahbix-clk", "mi2s-osr-clk", "mi2s-bit-clk";
+	interrupts = <0 85 1>;
+	interrupt-names = "lpass-irq-lpaif";
+	pinctrl-names = "default", "idle";
+	pinctrl-0 = <&mi2s_default>;
+	pinctrl-1 = <&mi2s_idle>;
+	reg = <0x28100000 0x10000>;
+	reg-names = "lpass-lpaif";
+	qcom,adsp = <&adsp>;
+};
@@ -29,9 +29,17 @@ SSI subnode properties:
 - shared-pin			: if shared clock pin
 - pio-transfer			: use PIO transfer mode
 - no-busif			: BUSIF is not ussed when [mem -> SSI] via DMA case
+- dma				: Should contain Audio DMAC entry
+- dma-names			: SSI  case "rx"  (=playback), "tx"  (=capture)
+				  SSIU case "rxu" (=playback), "txu" (=capture)

 SRC subnode properties:
-no properties at this point
+- dma				: Should contain Audio DMAC entry
+- dma-names			: "rx" (=playback), "tx" (=capture)
+
+DVC subnode properties:
+- dma				: Should contain Audio DMAC entry
+- dma-names			: "tx" (=playback/capture)

 DAI subnode properties:
 - playback			: list of playback modules
@@ -45,56 +53,145 @@ rcar_sound: rcar_sound@ec500000 {
 	reg =	<0 0xec500000 0 0x1000>, /* SCU */
 		<0 0xec5a0000 0 0x100>,  /* ADG */
 		<0 0xec540000 0 0x1000>, /* SSIU */
-		<0 0xec541000 0 0x1280>; /* SSI */
+		<0 0xec541000 0 0x1280>, /* SSI */
+		<0 0xec740000 0 0x200>;  /* Audio DMAC peri peri*/
+	reg-names = "scu", "adg", "ssiu", "ssi", "audmapp";
+
+	clocks = <&mstp10_clks R8A7790_CLK_SSI_ALL>,
+		<&mstp10_clks R8A7790_CLK_SSI9>, <&mstp10_clks R8A7790_CLK_SSI8>,
+		<&mstp10_clks R8A7790_CLK_SSI7>, <&mstp10_clks R8A7790_CLK_SSI6>,
+		<&mstp10_clks R8A7790_CLK_SSI5>, <&mstp10_clks R8A7790_CLK_SSI4>,
+		<&mstp10_clks R8A7790_CLK_SSI3>, <&mstp10_clks R8A7790_CLK_SSI2>,
+		<&mstp10_clks R8A7790_CLK_SSI1>, <&mstp10_clks R8A7790_CLK_SSI0>,
+		<&mstp10_clks R8A7790_CLK_SCU_SRC9>, <&mstp10_clks R8A7790_CLK_SCU_SRC8>,
+		<&mstp10_clks R8A7790_CLK_SCU_SRC7>, <&mstp10_clks R8A7790_CLK_SCU_SRC6>,
+		<&mstp10_clks R8A7790_CLK_SCU_SRC5>, <&mstp10_clks R8A7790_CLK_SCU_SRC4>,
+		<&mstp10_clks R8A7790_CLK_SCU_SRC3>, <&mstp10_clks R8A7790_CLK_SCU_SRC2>,
+		<&mstp10_clks R8A7790_CLK_SCU_SRC1>, <&mstp10_clks R8A7790_CLK_SCU_SRC0>,
+		<&mstp10_clks R8A7790_CLK_SCU_DVC0>, <&mstp10_clks R8A7790_CLK_SCU_DVC1>,
+		<&audio_clk_a>, <&audio_clk_b>, <&audio_clk_c>, <&m2_clk>;
+	clock-names = "ssi-all",
+			"ssi.9", "ssi.8", "ssi.7", "ssi.6", "ssi.5",
+			"ssi.4", "ssi.3", "ssi.2", "ssi.1", "ssi.0",
+			"src.9", "src.8", "src.7", "src.6", "src.5",
+			"src.4", "src.3", "src.2", "src.1", "src.0",
+			"dvc.0", "dvc.1",
+			"clk_a", "clk_b", "clk_c", "clk_i";

 	rcar_sound,dvc {
-		dvc0: dvc@0 { };
-		dvc1: dvc@1 { };
+		dvc0: dvc@0 {
+			dmas = <&audma0 0xbc>;
+			dma-names = "tx";
+		};
+		dvc1: dvc@1 {
+			dmas = <&audma0 0xbe>;
+			dma-names = "tx";
+		};
 	};

 	rcar_sound,src {
-		src0: src@0 { };
-		src1: src@1 { };
-		src2: src@2 { };
-		src3: src@3 { };
-		src4: src@4 { };
-		src5: src@5 { };
-		src6: src@6 { };
-		src7: src@7 { };
-		src8: src@8 { };
-		src9: src@9 { };
+		src0: src@0 {
+			interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x85>, <&audma1 0x9a>;
+			dma-names = "rx", "tx";
+		};
+		src1: src@1 {
+			interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x87>, <&audma1 0x9c>;
+			dma-names = "rx", "tx";
+		};
+		src2: src@2 {
+			interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x89>, <&audma1 0x9e>;
+			dma-names = "rx", "tx";
+		};
+		src3: src@3 {
+			interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x8b>, <&audma1 0xa0>;
+			dma-names = "rx", "tx";
+		};
+		src4: src@4 {
+			interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x8d>, <&audma1 0xb0>;
+			dma-names = "rx", "tx";
+		};
+		src5: src@5 {
+			interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x8f>, <&audma1 0xb2>;
+			dma-names = "rx", "tx";
+		};
+		src6: src@6 {
+			interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x91>, <&audma1 0xb4>;
+			dma-names = "rx", "tx";
+		};
+		src7: src@7 {
+			interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x93>, <&audma1 0xb6>;
+			dma-names = "rx", "tx";
+		};
+		src8: src@8 {
+			interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x95>, <&audma1 0xb8>;
+			dma-names = "rx", "tx";
+		};
+		src9: src@9 {
+			interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x97>, <&audma1 0xba>;
+			dma-names = "rx", "tx";
+		};
 	};

 	rcar_sound,ssi {
 		ssi0: ssi@0 {
 			interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x01>, <&audma1 0x02>, <&audma0 0x15>, <&audma1 0x16>;
+			dma-names = "rx", "tx", "rxu", "txu";
 		};
 		ssi1: ssi@1 {
 			interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x03>, <&audma1 0x04>, <&audma0 0x49>, <&audma1 0x4a>;
+			dma-names = "rx", "tx", "rxu", "txu";
 		};
 		ssi2: ssi@2 {
 			interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x05>, <&audma1 0x06>, <&audma0 0x63>, <&audma1 0x64>;
+			dma-names = "rx", "tx", "rxu", "txu";
 		};
 		ssi3: ssi@3 {
 			interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x07>, <&audma1 0x08>, <&audma0 0x6f>, <&audma1 0x70>;
+			dma-names = "rx", "tx", "rxu", "txu";
 		};
 		ssi4: ssi@4 {
 			interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x09>, <&audma1 0x0a>, <&audma0 0x71>, <&audma1 0x72>;
+			dma-names = "rx", "tx", "rxu", "txu";
 		};
 		ssi5: ssi@5 {
 			interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x0b>, <&audma1 0x0c>, <&audma0 0x73>, <&audma1 0x74>;
+			dma-names = "rx", "tx", "rxu", "txu";
 		};
 		ssi6: ssi@6 {
 			interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x0d>, <&audma1 0x0e>, <&audma0 0x75>, <&audma1 0x76>;
+			dma-names = "rx", "tx", "rxu", "txu";
 		};
 		ssi7: ssi@7 {
 			interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x0f>, <&audma1 0x10>, <&audma0 0x79>, <&audma1 0x7a>;
+			dma-names = "rx", "tx", "rxu", "txu";
 		};
 		ssi8: ssi@8 {
 			interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x11>, <&audma1 0x12>, <&audma0 0x7b>, <&audma1 0x7c>;
+			dma-names = "rx", "tx", "rxu", "txu";
 		};
 		ssi9: ssi@9 {
 			interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x13>, <&audma1 0x14>, <&audma0 0x7d>, <&audma1 0x7e>;
+			dma-names = "rx", "tx", "rxu", "txu";
 		};
 	};

@@ -0,0 +1,67 @@
+Renesas Sampling Rate Convert Sound Card:
+
+Renesas Sampling Rate Convert Sound Card specifies audio DAI connections of SoC <-> codec.
+
+Required properties:
+
+- compatible				: "renesas,rsrc-card,<board>"
+					  Examples with soctypes are:
+					    - "renesas,rsrc-card,lager"
+					    - "renesas,rsrc-card,koelsch"
+Optional properties:
+
+- card_name				: User specified audio sound card name, one string
+					  property.
+- cpu					: CPU   sub-node
+- codec					: CODEC sub-node
+
+Optional subnode properties:
+
+- format				: CPU/CODEC common audio format.
+					  "i2s", "right_j", "left_j" , "dsp_a"
+					  "dsp_b", "ac97", "pdm", "msb", "lsb"
+- frame-master				: Indicates dai-link frame master.
+					  phandle to a cpu or codec subnode.
+- bitclock-master			: Indicates dai-link bit clock master.
+					  phandle to a cpu or codec subnode.
+- bitclock-inversion			: bool property. Add this if the
+					  dai-link uses bit clock inversion.
+- frame-inversion			: bool property. Add this if the
+					  dai-link uses frame clock inversion.
+- convert-rate				: platform specified sampling rate convert
+
+Required CPU/CODEC subnodes properties:
+
+- sound-dai				: phandle and port of CPU/CODEC
+
+Optional CPU/CODEC subnodes properties:
+
+- clocks / system-clock-frequency	: specify subnode's clock if needed.
+					  it can be specified via "clocks" if system has
+					  clock node (= common clock), or "system-clock-frequency"
+					  (if system doens't support common clock)
+					  If a clock is specified, it is
+					  enabled with clk_prepare_enable()
+					  in dai startup() and disabled with
+					  clk_disable_unprepare() in dai
+					  shutdown().
+
+Example
+
+sound {
+	compatible = "renesas,rsrc-card,lager";
+
+	card-name = "rsnd-ak4643";
+	format = "left_j";
+	bitclock-master = <&sndcodec>;
+	frame-master = <&sndcodec>;
+
+	sndcpu: cpu {
+		sound-dai = <&rcar_sound>;
+	};
+
+	sndcodec: codec {
+		sound-dai = <&ak4643>;
+		system-clock-frequency = <11289600>;
+	};
+};
@@ -0,0 +1,23 @@
+* Sound complex for Storm boards
+
+Models a soundcard for Storm boards with the Qualcomm Technologies IPQ806x SOC
+connected to a MAX98357A DAC via I2S.
+
+Required properties:
+
+- compatible	: "google,storm-audio"
+- cpu		: Phandle of the CPU DAI
+- codec		: Phandle of the codec DAI
+
+Optional properties:
+
+- qcom,model	: The user-visible name of this sound card.
+
+Example:
+
+sound {
+	compatible = "google,storm-audio";
+	qcom,model = "ipq806x-storm";
+	cpu = <&lpass_cpu>;
+	codec = <&max98357a>;
+};
@@ -10,6 +10,13 @@ Required properties:
  - reg : the I2C address of the device for I2C, the chip select
          number for SPI.

+  - PVDD-supply, DVDD-supply : Power supplies for the device, as covered
+    in Documentation/devicetree/bindings/regulator/regulator.txt
+
+Optional properties:
+
+  - wlf,reset-gpio: A GPIO specifier for the GPIO controlling the reset pin
+
 Example:

 codec: wm8804@1a {
@@ -71,11 +71,11 @@ SOURCE:
  HDMI/DP	(either HDMI or DisplayPort)

 Exceptions (deprecated):
-  [Digital] Capture Source
-  [Digital] Capture Switch	(aka input gain switch)
-  [Digital] Capture Volume	(aka input gain volume)
-  [Digital] Playback Switch	(aka output gain switch)
-  [Digital] Playback Volume	(aka output gain volume)
+  [Analogue|Digital] Capture Source
+  [Analogue|Digital] Capture Switch	(aka input gain switch)
+  [Analogue|Digital] Capture Volume	(aka input gain volume)
+  [Analogue|Digital] Playback Switch	(aka output gain switch)
+  [Analogue|Digital] Playback Volume	(aka output gain volume)
  Tone Control - Switch
  Tone Control - Bass
  Tone Control - Treble
@@ -466,7 +466,11 @@ The generic parser supports the following hints:
 - 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
+- power_save_node (bool): advanced power management for each widget,
+  controlling the power sate (D0/D3) of each widget node depending on
+  the actual pin and stream states
+- power_down_unused (bool): power down the unused widgets, a subset of
+  power_save_node, and will be dropped in future
 - 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
@@ -0,0 +1,200 @@
+The ALSA API can provide two different system timestamps:
+
+- Trigger_tstamp is the system time snapshot taken when the .trigger
+callback is invoked. This snapshot is taken by the ALSA core in the
+general case, but specific hardware may have synchronization
+capabilities or conversely may only be able to provide a correct
+estimate with a delay. In the latter two cases, the low-level driver
+is responsible for updating the trigger_tstamp at the most appropriate
+and precise moment. Applications should not rely solely on the first
+trigger_tstamp but update their internal calculations if the driver
+provides a refined estimate with a delay.
+
+- tstamp is the current system timestamp updated during the last
+event or application query.
+The difference (tstamp - trigger_tstamp) defines the elapsed time.
+
+The ALSA API provides reports two basic pieces of information, avail
+and delay, which combined with the trigger and current system
+timestamps allow for applications to keep track of the 'fullness' of
+the ring buffer and the amount of queued samples.
+
+The use of these different pointers and time information depends on
+the application needs:
+
+- 'avail' reports how much can be written in the ring buffer
+- 'delay' reports the time it will take to hear a new sample after all
+queued samples have been played out.
+
+When timestamps are enabled, the avail/delay information is reported
+along with a snapshot of system time. Applications can select from
+CLOCK_REALTIME (NTP corrections including going backwards),
+CLOCK_MONOTONIC (NTP corrections but never going backwards),
+CLOCK_MONOTIC_RAW (without NTP corrections) and change the mode
+dynamically with sw_params
+
+
+The ALSA API also provide an audio_tstamp which reflects the passage
+of time as measured by different components of audio hardware.  In
+ascii-art, this could be represented as follows (for the playback
+case):
+
+
+--------------------------------------------------------------> time
+  ^               ^              ^                ^           ^
+  |               |              |                |           |
+ analog         link            dma              app       FullBuffer
+ time           time           time              time        time
+  |               |              |                |           |
+  |< codec delay >|<--hw delay-->|<queued samples>|<---avail->|
+  |<----------------- delay---------------------->|           |
+			         |<----ring buffer length---->|
+
+The analog time is taken at the last stage of the playback, as close
+as possible to the actual transducer
+
+The link time is taken at the output of the SOC/chipset as the samples
+are pushed on a link. The link time can be directly measured if
+supported in hardware by sample counters or wallclocks (e.g. with
+HDAudio 24MHz or PTP clock for networked solutions) or indirectly
+estimated (e.g. with the frame counter in USB).
+
+The DMA time is measured using counters - typically the least reliable
+of all measurements due to the bursty natured of DMA transfers.
+
+The app time corresponds to the time tracked by an application after
+writing in the ring buffer.
+
+The application can query what the hardware supports, define which
+audio time it wants reported by selecting the relevant settings in
+audio_tstamp_config fields, get an estimate of the timestamp
+accuracy. It can also request the delay-to-analog be included in the
+measurement. Direct access to the link time is very interesting on
+platforms that provide an embedded DSP; measuring directly the link
+time with dedicated hardware, possibly synchronized with system time,
+removes the need to keep track of internal DSP processing times and
+latency.
+
+In case the application requests an audio tstamp that is not supported
+in hardware/low-level driver, the type is overridden as DEFAULT and the
+timestamp will report the DMA time based on the hw_pointer value.
+
+For backwards compatibility with previous implementations that did not
+provide timestamp selection, with a zero-valued COMPAT timestamp type
+the results will default to the HDAudio wall clock for playback
+streams and to the DMA time (hw_ptr) in all other cases.
+
+The audio timestamp accuracy can be returned to user-space, so that
+appropriate decisions are made:
+
+- for dma time (default), the granularity of the transfers can be
+  inferred from the steps between updates and in turn provide
+  information on how much the application pointer can be rewound
+  safely.
+
+- the link time can be used to track long-term drifts between audio
+  and system time using the (tstamp-trigger_tstamp)/audio_tstamp
+  ratio, the precision helps define how much smoothing/low-pass
+  filtering is required. The link time can be either reset on startup
+  or reported as is (the latter being useful to compare progress of
+  different streams - but may require the wallclock to be always
+  running and not wrap-around during idle periods). If supported in
+  hardware, the absolute link time could also be used to define a
+  precise start time (patches WIP)
+
+- including the delay in the audio timestamp may
+  counter-intuitively not increase the precision of timestamps, e.g. if a
+  codec includes variable-latency DSP processing or a chain of
+  hardware components the delay is typically not known with precision.
+
+The accuracy is reported in nanosecond units (using an unsigned 32-bit
+word), which gives a max precision of 4.29s, more than enough for
+audio applications...
+
+Due to the varied nature of timestamping needs, even for a single
+application, the audio_tstamp_config can be changed dynamically. In
+the STATUS ioctl, the parameters are read-only and do not allow for
+any application selection. To work around this limitation without
+impacting legacy applications, a new STATUS_EXT ioctl is introduced
+with read/write parameters. ALSA-lib will be modified to make use of
+STATUS_EXT and effectively deprecate STATUS.
+
+The ALSA API only allows for a single audio timestamp to be reported
+at a time. This is a conscious design decision, reading the audio
+timestamps from hardware registers or from IPC takes time, the more
+timestamps are read the more imprecise the combined measurements
+are. To avoid any interpretation issues, a single (system, audio)
+timestamp is reported. Applications that need different timestamps
+will be required to issue multiple queries and perform an
+interpolation of the results
+
+In some hardware-specific configuration, the system timestamp is
+latched by a low-level audio subsytem, and the information provided
+back to the driver. Due to potential delays in the communication with
+the hardware, there is a risk of misalignment with the avail and delay
+information. To make sure applications are not confused, a
+driver_timestamp field is added in the snd_pcm_status structure; this
+timestamp shows when the information is put together by the driver
+before returning from the STATUS and STATUS_EXT ioctl. in most cases
+this driver_timestamp will be identical to the regular system tstamp.
+
+Examples of typestamping with HDaudio:
+
+1. DMA timestamp, no compensation for DMA+analog delay
+$ ./audio_time  -p --ts_type=1
+playback: systime: 341121338 nsec, audio time 342000000 nsec, 	systime delta -878662
+playback: systime: 426236663 nsec, audio time 427187500 nsec, 	systime delta -950837
+playback: systime: 597080580 nsec, audio time 598000000 nsec, 	systime delta -919420
+playback: systime: 682059782 nsec, audio time 683020833 nsec, 	systime delta -961051
+playback: systime: 852896415 nsec, audio time 853854166 nsec, 	systime delta -957751
+playback: systime: 937903344 nsec, audio time 938854166 nsec, 	systime delta -950822
+
+2. DMA timestamp, compensation for DMA+analog delay
+$ ./audio_time  -p --ts_type=1 -d
+playback: systime: 341053347 nsec, audio time 341062500 nsec, 	systime delta -9153
+playback: systime: 426072447 nsec, audio time 426062500 nsec, 	systime delta 9947
+playback: systime: 596899518 nsec, audio time 596895833 nsec, 	systime delta 3685
+playback: systime: 681915317 nsec, audio time 681916666 nsec, 	systime delta -1349
+playback: systime: 852741306 nsec, audio time 852750000 nsec, 	systime delta -8694
+
+3. link timestamp, compensation for DMA+analog delay
+$ ./audio_time  -p --ts_type=2 -d
+playback: systime: 341060004 nsec, audio time 341062791 nsec, 	systime delta -2787
+playback: systime: 426242074 nsec, audio time 426244875 nsec, 	systime delta -2801
+playback: systime: 597080992 nsec, audio time 597084583 nsec, 	systime delta -3591
+playback: systime: 682084512 nsec, audio time 682088291 nsec, 	systime delta -3779
+playback: systime: 852936229 nsec, audio time 852940916 nsec, 	systime delta -4687
+playback: systime: 938107562 nsec, audio time 938112708 nsec, 	systime delta -5146
+
+Example 1 shows that the timestamp at the DMA level is close to 1ms
+ahead of the actual playback time (as a side time this sort of
+measurement can help define rewind safeguards). Compensating for the
+DMA-link delay in example 2 helps remove the hardware buffering abut
+the information is still very jittery, with up to one sample of
+error. In example 3 where the timestamps are measured with the link
+wallclock, the timestamps show a monotonic behavior and a lower
+dispersion.
+
+Example 3 and 4 are with USB audio class. Example 3 shows a high
+offset between audio time and system time due to buffering. Example 4
+shows how compensating for the delay exposes a 1ms accuracy (due to
+the use of the frame counter by the driver)
+
+Example 3: DMA timestamp, no compensation for delay, delta of ~5ms
+$ ./audio_time -p -Dhw:1 -t1
+playback: systime: 120174019 nsec, audio time 125000000 nsec, 	systime delta -4825981
+playback: systime: 245041136 nsec, audio time 250000000 nsec, 	systime delta -4958864
+playback: systime: 370106088 nsec, audio time 375000000 nsec, 	systime delta -4893912
+playback: systime: 495040065 nsec, audio time 500000000 nsec, 	systime delta -4959935
+playback: systime: 620038179 nsec, audio time 625000000 nsec, 	systime delta -4961821
+playback: systime: 745087741 nsec, audio time 750000000 nsec, 	systime delta -4912259
+playback: systime: 870037336 nsec, audio time 875000000 nsec, 	systime delta -4962664
+
+Example 4: DMA timestamp, compensation for delay, delay of ~1ms
+$ ./audio_time -p -Dhw:1 -t1 -d
+playback: systime: 120190520 nsec, audio time 120000000 nsec, 	systime delta 190520
+playback: systime: 245036740 nsec, audio time 244000000 nsec, 	systime delta 1036740
+playback: systime: 370034081 nsec, audio time 369000000 nsec, 	systime delta 1034081
+playback: systime: 495159907 nsec, audio time 494000000 nsec, 	systime delta 1159907
+playback: systime: 620098824 nsec, audio time 619000000 nsec, 	systime delta 1098824
+playback: systime: 745031847 nsec, audio time 744000000 nsec, 	systime delta 1031847
@@ -1764,7 +1764,7 @@ S:	Supported
 F:	drivers/tty/serial/atmel_serial.c

 ATMEL Audio ALSA driver
-M:	Bo Shen <voice.shen@atmel.com>
+M:	Nicolas Ferre <nicolas.ferre@atmel.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Supported
 F:	sound/soc/atmel
@@ -5301,6 +5301,13 @@ F:	drivers/char/ipmi/
 F:	include/linux/ipmi*
 F:	include/uapi/linux/ipmi*

+QCOM AUDIO (ASoC) DRIVERS
+M:	Patrick Lai <plai@codeaurora.org>
+M:	Banajit Goswami <bgoswami@codeaurora.org>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Supported
+F:	sound/soc/qcom/
+
 IPS SCSI RAID DRIVER
 M:	Adaptec OEM Raid Solutions <aacraid@adaptec.com>
 L:	linux-scsi@vger.kernel.org
@@ -1015,7 +1015,6 @@ static struct asoc_simple_card_info fsi_wm8978_info = {
 	.platform	= "sh_fsi2",
 	.daifmt		= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM,
 	.cpu_dai = {
-		.fmt	= SND_SOC_DAIFMT_IB_NF,
 		.name	= "fsia-dai",
 	},
 	.codec_dai = {
@@ -1040,9 +1039,9 @@ static struct asoc_simple_card_info fsi2_hdmi_info = {
 	.card		= "FSI2B-HDMI",
 	.codec		= "sh-mobile-hdmi",
 	.platform	= "sh_fsi2",
+	.daifmt		= SND_SOC_DAIFMT_CBS_CFS,
 	.cpu_dai = {
 		.name	= "fsib-dai",
-		.fmt	= SND_SOC_DAIFMT_CBS_CFS,
 	},
 	.codec_dai = {
 		.name = "sh_mobile_hdmi-hifi",
@@ -194,6 +194,7 @@ struct dma_chan *of_dma_request_slave_channel(struct device_node *np,

 	return ERR_PTR(ret_no_channel);
 }
+EXPORT_SYMBOL_GPL(of_dma_request_slave_channel);

 /**
 * of_dma_simple_xlate - Simple DMA engine translation function
@@ -608,7 +608,9 @@ struct ac97_quirk {
 	int type;		/* quirk type above */
 };

-int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, const char *override);
+int snd_ac97_tune_hardware(struct snd_ac97 *ac97,
+			   const struct ac97_quirk *quirk,
+			   const char *override);
 int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate);

 /*
@@ -70,7 +70,7 @@ struct snd_compr_runtime {
 * @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
+ * @next_track: has userspace signal next track transition, true when set
 * @private_data: pointer to DSP private data
 */
 struct snd_compr_stream {
@@ -95,7 +95,7 @@ struct snd_compr_stream {
 * and the stream properties
 * @get_params: retrieve the codec parameters, mandatory
 * @set_metadata: Set the metadata values for a stream
- * @get_metadata: retreives the requested metadata values from stream
+ * @get_metadata: retrieves the requested metadata values from stream
 * @trigger: Trigger operations like start, pause, resume, drain, stop.
 * This callback is mandatory
 * @pointer: Retrieve current h/w pointer information. Mandatory
@@ -227,7 +227,7 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
 * Add a virtual slave control to the given master.
 * Unlike snd_ctl_add_slave(), the element added via this function
 * is supposed to have volatile values, and get callback is called
- * at each time quried from the master.
+ * at each time queried from the master.
 *
 * When the control peeks the hardware values directly and the value
 * can be changed by other means than the put callback of the element,
@@ -278,7 +278,8 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type,
 		   void *device_data, struct snd_device_ops *ops);
 int snd_device_register(struct snd_card *card, void *device_data);
 int snd_device_register_all(struct snd_card *card);
-int snd_device_disconnect_all(struct snd_card *card);
+void snd_device_disconnect(struct snd_card *card, void *device_data);
+void snd_device_disconnect_all(struct snd_card *card);
 void snd_device_free(struct snd_card *card, void *device_data);
 void snd_device_free_all(struct snd_card *card);

@@ -0,0 +1,217 @@
+/*
+ * HD-audio regmap helpers
+ */
+
+#ifndef __SOUND_HDA_REGMAP_H
+#define __SOUND_HDA_REGMAP_H
+
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/hdaudio.h>
+
+int snd_hdac_regmap_init(struct hdac_device *codec);
+void snd_hdac_regmap_exit(struct hdac_device *codec);
+int snd_hdac_regmap_add_vendor_verb(struct hdac_device *codec,
+				    unsigned int verb);
+int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg,
+			     unsigned int *val);
+int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg,
+			      unsigned int val);
+int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg,
+			       unsigned int mask, unsigned int val);
+
+/**
+ * snd_hdac_regmap_encode_verb - encode the verb to a pseudo register
+ * @nid: widget NID
+ * @verb: codec verb
+ *
+ * Returns an encoded pseudo register.
+ */
+#define snd_hdac_regmap_encode_verb(nid, verb)		\
+	(((verb) << 8) | 0x80000 | ((unsigned int)(nid) << 20))
+
+/**
+ * snd_hdac_regmap_encode_amp - encode the AMP verb to a pseudo register
+ * @nid: widget NID
+ * @ch: channel (left = 0, right = 1)
+ * @dir: direction (#HDA_INPUT, #HDA_OUTPUT)
+ * @idx: input index value
+ *
+ * Returns an encoded pseudo register.
+ */
+#define snd_hdac_regmap_encode_amp(nid, ch, dir, idx)			\
+	(snd_hdac_regmap_encode_verb(nid, AC_VERB_GET_AMP_GAIN_MUTE) |	\
+	 ((ch) ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT) |			\
+	 ((dir) == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT) | \
+	 (idx))
+
+/**
+ * snd_hdac_regmap_encode_amp_stereo - encode a pseudo register for stereo AMPs
+ * @nid: widget NID
+ * @dir: direction (#HDA_INPUT, #HDA_OUTPUT)
+ * @idx: input index value
+ *
+ * Returns an encoded pseudo register.
+ */
+#define snd_hdac_regmap_encode_amp_stereo(nid, dir, idx)		\
+	(snd_hdac_regmap_encode_verb(nid, AC_VERB_GET_AMP_GAIN_MUTE) |	\
+	 AC_AMP_SET_LEFT | AC_AMP_SET_RIGHT | /* both bits set! */	\
+	 ((dir) == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT) | \
+	 (idx))
+
+/**
+ * snd_hdac_regmap_write - Write a verb with caching
+ * @nid: codec NID
+ * @reg: verb to write
+ * @val: value to write
+ *
+ * For writing an amp value, use snd_hda_regmap_amp_update().
+ */
+static inline int
+snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid,
+		      unsigned int verb, unsigned int val)
+{
+	unsigned int cmd = snd_hdac_regmap_encode_verb(nid, verb);
+
+	return snd_hdac_regmap_write_raw(codec, cmd, val);
+}
+
+/**
+ * snd_hda_regmap_update - Update a verb value with caching
+ * @nid: codec NID
+ * @verb: verb to update
+ * @mask: bit mask to update
+ * @val: value to update
+ *
+ * For updating an amp value, use snd_hda_regmap_amp_update().
+ */
+static inline int
+snd_hdac_regmap_update(struct hdac_device *codec, hda_nid_t nid,
+		       unsigned int verb, unsigned int mask,
+		       unsigned int val)
+{
+	unsigned int cmd = snd_hdac_regmap_encode_verb(nid, verb);
+
+	return snd_hdac_regmap_update_raw(codec, cmd, mask, val);
+}
+
+/**
+ * snd_hda_regmap_read - Read a verb with caching
+ * @nid: codec NID
+ * @verb: verb to read
+ * @val: pointer to store the value
+ *
+ * For reading an amp value, use snd_hda_regmap_get_amp().
+ */
+static inline int
+snd_hdac_regmap_read(struct hdac_device *codec, hda_nid_t nid,
+		     unsigned int verb, unsigned int *val)
+{
+	unsigned int cmd = snd_hdac_regmap_encode_verb(nid, verb);
+
+	return snd_hdac_regmap_read_raw(codec, cmd, val);
+}
+
+/**
+ * snd_hdac_regmap_get_amp - Read AMP value
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @ch: channel (left=0 or right=1)
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @index: the index value (only for input direction)
+ * @val: the pointer to store the value
+ *
+ * Read AMP value.  The volume is between 0 to 0x7f, 0x80 = mute bit.
+ * Returns the value or a negative error.
+ */
+static inline int
+snd_hdac_regmap_get_amp(struct hdac_device *codec, hda_nid_t nid,
+			int ch, int dir, int idx)
+{
+	unsigned int cmd = snd_hdac_regmap_encode_amp(nid, ch, dir, idx);
+	int err, val;
+
+	err = snd_hdac_regmap_read_raw(codec, cmd, &val);
+	return err < 0 ? err : val;
+}
+
+/**
+ * snd_hdac_regmap_update_amp - update the AMP value
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @ch: channel (left=0 or right=1)
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @idx: the index value (only for input direction)
+ * @mask: bit mask to set
+ * @val: the bits value to set
+ *
+ * Update the AMP value with a bit mask.
+ * Returns 0 if the value is unchanged, 1 if changed, or a negative error.
+ */
+static inline int
+snd_hdac_regmap_update_amp(struct hdac_device *codec, hda_nid_t nid,
+			   int ch, int dir, int idx, int mask, int val)
+{
+	unsigned int cmd = snd_hdac_regmap_encode_amp(nid, ch, dir, idx);
+
+	return snd_hdac_regmap_update_raw(codec, cmd, mask, val);
+}
+
+/**
+ * snd_hdac_regmap_get_amp_stereo - Read stereo AMP values
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @ch: channel (left=0 or right=1)
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @index: the index value (only for input direction)
+ * @val: the pointer to store the value
+ *
+ * Read stereo AMP values.  The lower byte is left, the upper byte is right.
+ * Returns the value or a negative error.
+ */
+static inline int
+snd_hdac_regmap_get_amp_stereo(struct hdac_device *codec, hda_nid_t nid,
+			       int dir, int idx)
+{
+	unsigned int cmd = snd_hdac_regmap_encode_amp_stereo(nid, dir, idx);
+	int err, val;
+
+	err = snd_hdac_regmap_read_raw(codec, cmd, &val);
+	return err < 0 ? err : val;
+}
+
+/**
+ * snd_hdac_regmap_update_amp_stereo - update the stereo AMP value
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @idx: the index value (only for input direction)
+ * @mask: bit mask to set
+ * @val: the bits value to set
+ *
+ * Update the stereo AMP value with a bit mask.
+ * The lower byte is left, the upper byte is right.
+ * Returns 0 if the value is unchanged, 1 if changed, or a negative error.
+ */
+static inline int
+snd_hdac_regmap_update_amp_stereo(struct hdac_device *codec, hda_nid_t nid,
+				  int dir, int idx, int mask, int val)
+{
+	unsigned int cmd = snd_hdac_regmap_encode_amp_stereo(nid, dir, idx);
+
+	return snd_hdac_regmap_update_raw(codec, cmd, mask, val);
+}
+
+/**
+ * snd_hdac_regmap_sync_node - sync the widget node attributes
+ * @codec: HD-audio codec
+ * @nid: NID to sync
+ */
+static inline void
+snd_hdac_regmap_sync_node(struct hdac_device *codec, hda_nid_t nid)
+{
+	regcache_mark_dirty(codec->regmap);
+	regcache_sync_region(codec->regmap, nid << 20, ((nid + 1) << 20) - 1);
+}
+
+#endif /* __SOUND_HDA_REGMAP_H */
@@ -0,0 +1,247 @@
+/*
+ * HD-audio core stuff
+ */
+
+#ifndef __SOUND_HDAUDIO_H
+#define __SOUND_HDAUDIO_H
+
+#include <linux/device.h>
+#include <sound/hda_verbs.h>
+
+/* codec node id */
+typedef u16 hda_nid_t;
+
+struct hdac_bus;
+struct hdac_device;
+struct hdac_driver;
+struct hdac_widget_tree;
+
+/*
+ * exported bus type
+ */
+extern struct bus_type snd_hda_bus_type;
+
+/*
+ * generic arrays
+ */
+struct snd_array {
+	unsigned int used;
+	unsigned int alloced;
+	unsigned int elem_size;
+	unsigned int alloc_align;
+	void *list;
+};
+
+/*
+ * HD-audio codec base device
+ */
+struct hdac_device {
+	struct device dev;
+	int type;
+	struct hdac_bus *bus;
+	unsigned int addr;		/* codec address */
+	struct list_head list;		/* list point for bus codec_list */
+
+	hda_nid_t afg;			/* AFG node id */
+	hda_nid_t mfg;			/* MFG node id */
+
+	/* ids */
+	unsigned int vendor_id;
+	unsigned int subsystem_id;
+	unsigned int revision_id;
+	unsigned int afg_function_id;
+	unsigned int mfg_function_id;
+	unsigned int afg_unsol:1;
+	unsigned int mfg_unsol:1;
+
+	unsigned int power_caps;	/* FG power caps */
+
+	const char *vendor_name;	/* codec vendor name */
+	const char *chip_name;		/* codec chip name */
+
+	/* verb exec op override */
+	int (*exec_verb)(struct hdac_device *dev, unsigned int cmd,
+			 unsigned int flags, unsigned int *res);
+
+	/* widgets */
+	unsigned int num_nodes;
+	hda_nid_t start_nid, end_nid;
+
+	/* misc flags */
+	atomic_t in_pm;		/* suspend/resume being performed */
+
+	/* sysfs */
+	struct hdac_widget_tree *widgets;
+
+	/* regmap */
+	struct regmap *regmap;
+	struct snd_array vendor_verbs;
+	bool lazy_cache:1;	/* don't wake up for writes */
+	bool caps_overwriting:1; /* caps overwrite being in process */
+	bool cache_coef:1;	/* cache COEF read/write too */
+};
+
+/* device/driver type used for matching */
+enum {
+	HDA_DEV_CORE,
+	HDA_DEV_LEGACY,
+};
+
+/* direction */
+enum {
+	HDA_INPUT, HDA_OUTPUT
+};
+
+#define dev_to_hdac_dev(_dev)	container_of(_dev, struct hdac_device, dev)
+
+int snd_hdac_device_init(struct hdac_device *dev, struct hdac_bus *bus,
+			 const char *name, unsigned int addr);
+void snd_hdac_device_exit(struct hdac_device *dev);
+int snd_hdac_device_register(struct hdac_device *codec);
+void snd_hdac_device_unregister(struct hdac_device *codec);
+
+int snd_hdac_refresh_widgets(struct hdac_device *codec);
+
+unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid,
+			       unsigned int verb, unsigned int parm);
+int snd_hdac_exec_verb(struct hdac_device *codec, unsigned int cmd,
+		       unsigned int flags, unsigned int *res);
+int snd_hdac_read(struct hdac_device *codec, hda_nid_t nid,
+		  unsigned int verb, unsigned int parm, unsigned int *res);
+int _snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm,
+			unsigned int *res);
+int snd_hdac_read_parm_uncached(struct hdac_device *codec, hda_nid_t nid,
+				int parm);
+int snd_hdac_override_parm(struct hdac_device *codec, hda_nid_t nid,
+			   unsigned int parm, unsigned int val);
+int snd_hdac_get_connections(struct hdac_device *codec, hda_nid_t nid,
+			     hda_nid_t *conn_list, int max_conns);
+int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid,
+			   hda_nid_t *start_id);
+
+/**
+ * snd_hdac_read_parm - read a codec parameter
+ * @codec: the codec object
+ * @nid: NID to read a parameter
+ * @parm: parameter to read
+ *
+ * Returns -1 for error.  If you need to distinguish the error more
+ * strictly, use _snd_hdac_read_parm() directly.
+ */
+static inline int snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid,
+				     int parm)
+{
+	unsigned int val;
+
+	return _snd_hdac_read_parm(codec, nid, parm, &val) < 0 ? -1 : val;
+}
+
+#ifdef CONFIG_PM
+void snd_hdac_power_up(struct hdac_device *codec);
+void snd_hdac_power_down(struct hdac_device *codec);
+void snd_hdac_power_up_pm(struct hdac_device *codec);
+void snd_hdac_power_down_pm(struct hdac_device *codec);
+#else
+static inline void snd_hdac_power_up(struct hdac_device *codec) {}
+static inline void snd_hdac_power_down(struct hdac_device *codec) {}
+static inline void snd_hdac_power_up_pm(struct hdac_device *codec) {}
+static inline void snd_hdac_power_down_pm(struct hdac_device *codec) {}
+#endif
+
+/*
+ * HD-audio codec base driver
+ */
+struct hdac_driver {
+	struct device_driver driver;
+	int type;
+	int (*match)(struct hdac_device *dev, struct hdac_driver *drv);
+	void (*unsol_event)(struct hdac_device *dev, unsigned int event);
+};
+
+#define drv_to_hdac_driver(_drv) container_of(_drv, struct hdac_driver, driver)
+
+/*
+ * HD-audio bus base driver
+ */
+struct hdac_bus_ops {
+	/* send a single command */
+	int (*command)(struct hdac_bus *bus, unsigned int cmd);
+	/* get a response from the last command */
+	int (*get_response)(struct hdac_bus *bus, unsigned int addr,
+			    unsigned int *res);
+};
+
+#define HDA_UNSOL_QUEUE_SIZE	64
+
+struct hdac_bus {
+	struct device *dev;
+	const struct hdac_bus_ops *ops;
+
+	/* codec linked list */
+	struct list_head codec_list;
+	unsigned int num_codecs;
+
+	/* link caddr -> codec */
+	struct hdac_device *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1];
+
+	/* unsolicited event queue */
+	u32 unsol_queue[HDA_UNSOL_QUEUE_SIZE * 2]; /* ring buffer */
+	unsigned int unsol_rp, unsol_wp;
+	struct work_struct unsol_work;
+
+	/* bit flags of powered codecs */
+	unsigned long codec_powered;
+
+	/* flags */
+	bool sync_write:1;		/* sync after verb write */
+
+	/* locks */
+	struct mutex cmd_mutex;
+};
+
+int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,
+		      const struct hdac_bus_ops *ops);
+void snd_hdac_bus_exit(struct hdac_bus *bus);
+int snd_hdac_bus_exec_verb(struct hdac_bus *bus, unsigned int addr,
+			   unsigned int cmd, unsigned int *res);
+int snd_hdac_bus_exec_verb_unlocked(struct hdac_bus *bus, unsigned int addr,
+				    unsigned int cmd, unsigned int *res);
+void snd_hdac_bus_queue_event(struct hdac_bus *bus, u32 res, u32 res_ex);
+
+int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec);
+void snd_hdac_bus_remove_device(struct hdac_bus *bus,
+				struct hdac_device *codec);
+
+static inline void snd_hdac_codec_link_up(struct hdac_device *codec)
+{
+	set_bit(codec->addr, &codec->bus->codec_powered);
+}
+
+static inline void snd_hdac_codec_link_down(struct hdac_device *codec)
+{
+	clear_bit(codec->addr, &codec->bus->codec_powered);
+}
+
+/*
+ * generic array helpers
+ */
+void *snd_array_new(struct snd_array *array);
+void snd_array_free(struct snd_array *array);
+static inline void snd_array_init(struct snd_array *array, unsigned int size,
+				  unsigned int align)
+{
+	array->elem_size = size;
+	array->alloc_align = align;
+}
+
+static inline void *snd_array_elem(struct snd_array *array, unsigned int idx)
+{
+	return array->list + idx * array->elem_size;
+}
+
+static inline unsigned int snd_array_index(struct snd_array *array, void *ptr)
+{
+	return (unsigned long)(ptr - array->list) / array->elem_size;
+}
+
+#endif /* __SOUND_HDAUDIO_H */
--- a/Show More
+++ b/Show More