Merge tag 'sound-4.15-rc1' of ssh://gitolite.kernel.org/pub/scm/linux…

…/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There are no big surprising changes in this cycle, yet not too boring,
  either. The biggest change from diffstat POV is the removal of the
  legacy OSS driver codes that have been already disabled for a long
  time. This will bring a few trivial merge conflicts.

  As new features in ASoC side, there are two things: a new AC97 bus
  implementation and AMD Stony platform support. Both include the
  relevant changes shared with other subsystems, e.g. AC97 MFD changes
  and DRM AMD changes.

  Some other highlighted topics are:

   - A bunch of USB-audio drivers got the hardening against the
     malicious device accesses with a new helper code for endpoint
     sanity check

   - Lots of cleanups for ASoC Intel platform code, including support
     for their open source audio firmware

   - Continued ASoC core componentization works

   - Support for scaling MCLK with sample rate in ASoC simple-card

   - Stabler PCM hot-unplug capability, especially for ASoC usages"

* tag 'sound-4.15-rc1' of ssh://gitolite.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (302 commits)
  Documentation: sound: hd-audio: notes.rst
  ASoC: bcm2835: Support left/right justified and DSP modes
  ASoC: bcm2835: Enforce full symmetry
  ASoC: bcm2835: Support additional samplerates up to 384kHz
  ASoC: bcm2835: Add support for TDM modes
  ASoC: add mclk-fs support to audio graph card
  ASoC: add mclk-fs to audio graph card binding
  ASoC: rt5514: work around link error
  ASoC: rt5514: mark PM functions as __maybe_unused
  ASoC: rt5663: Check the JD status in the button pushing
  ASoC: amd: Modified DMA transfer Mechanism for Playback
  ASoC: rt5645: Wait for 400msec before concluding on value of RT5645_VENDOR_ID2
  ASoC: sun4i-codec: fixed 32bit audio capture support for H3/H2+
  ASoC: da7213: add support for DSP modes
  ASoC: sun8i-codec: Add a comment on the LRCK inversion
  ASoC: sun8i-codec: Set the BCLK divider
  ASoC: rt5663: Delay and retry reading rt5663 ID register
  ASoC: amd: use do_div rather than 64 bit division to fix 32 bit builds
  ASoC: cs42l56: Fix reset GPIO name in example DT binding
  ASoC: rt5514-spi: check irq status to schedule data copy in resume function
  ...

Documentation/devicetree/bindings/mfd/arizona.txt

@@ -65,45 +65,6 @@ Optional properties:
     a value that is out of range for a 16 bit register then the chip default
     will be used. If present exactly five values must be specified.
 
-  - wlf,inmode : A list of INn_MODE register values, where n is the number
-    of input signals. Valid values are 0 (Differential), 1 (Single-ended) and
-    2 (Digital Microphone). If absent, INn_MODE registers set to 0 by default.
-    If present, values must be specified less than or equal to the number of
-    input signals. If values less than the number of input signals, elements
-    that have not been specified are set to 0 by default. Entries are:
-    <IN1, IN2, IN3, IN4> (wm5102, wm5110, wm8280, wm8997)
-    <IN1A, IN2A, IN1B, IN2B> (wm8998, wm1814)
-  - wlf,out-mono : A list of boolean values indicating whether each output is
-    mono or stereo. Position within the list indicates the output affected
-    (eg. First entry in the list corresponds to output 1). A non-zero value
-    indicates a mono output. If present, the number of values should be less
-    than or equal to the number of outputs, if less values are supplied the
-    additional outputs will be treated as stereo.
-
-  - wlf,dmic-ref : DMIC reference voltage source for each input, can be
-    selected from either MICVDD or one of the MICBIAS's, defines
-    (ARIZONA_DMIC_xxxx) are provided in <dt-bindings/mfd/arizona.txt>. If
-    present, the number of values should be less than or equal to the
-    number of inputs, unspecified inputs will use the chip default.
-
-  - wlf,max-channels-clocked : The maximum number of channels to be clocked on
-    each AIF, useful for I2S systems with multiple data lines being mastered.
-    Specify one cell for each AIF to be configured, specify zero for AIFs that
-    should be handled normally.
-    If present, number of cells must be less than or equal to the number of
-    AIFs. If less than the number of AIFs, for cells that have not been
-    specified the corresponding AIFs will be treated as default setting.
-
-  - wlf,spk-fmt : PDM speaker data format, must contain 2 cells (OUT5 and OUT6).
-    See the datasheet for values.
-    The second cell is ignored for codecs that do not have OUT6 (wm5102, wm8997,
-      wm8998, wm1814)
-
-  - wlf,spk-mute : PDM speaker mute setting, must contain 2 cells (OUT5 and OUT6).
-    See the datasheet for values.
-    The second cell is ignored for codecs that do not have OUT6 (wm5102, wm8997,
-    wm8998, wm1814)
-
   - DCVDD-supply, MICVDD-supply : Power supplies, only need to be specified if
     they are being externally supplied. As covered in
     Documentation/devicetree/bindings/regulator/regulator.txt
@@ -112,6 +73,7 @@ Optional properties:
 Also see child specific device properties:
   Regulator - ../regulator/arizona-regulator.txt
   Extcon    - ../extcon/extcon-arizona.txt
+  Sound     - ../sound/arizona.txt
 
 Example:
 

Documentation/devicetree/bindings/sound/audio-graph-card.txt

@@ -17,6 +17,7 @@ Below are same as Simple-Card.
 - bitclock-master
 - bitclock-inversion
 - frame-inversion
+- mclk-fs
 - dai-tdm-slot-num
 - dai-tdm-slot-width
 - clocks / system-clock-frequency

Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt

@@ -43,7 +43,7 @@ Example 1. Sampling Rate Conversion
 		label = "sound-card";
 		prefix = "codec";
 		routing = "codec Playback", "DAI0 Playback",
-			"codec Playback", "DAI1 Playback";
+			  "DAI0 Capture",   "codec Capture";
 		convert-rate = <48000>;
 
 		dais = <&cpu_port>;
@@ -79,7 +79,8 @@ Example 2. 2 CPU 1 Codec (Mixing)
 		label = "sound-card";
 		prefix = "codec";
 		routing = "codec Playback", "DAI0 Playback",
-			"codec Playback", "DAI1 Playback";
+			  "codec Playback", "DAI1 Playback",
+			  "DAI0 Capture",   "codec Capture";
 		convert-rate = <48000>;
 
 		dais = <&cpu_port0

Documentation/devicetree/bindings/sound/cs42l56.txt

@@ -55,7 +55,7 @@ Example:
 codec: codec@4b {
 	compatible = "cirrus,cs42l56";
 	reg = <0x4b>;
-	gpio-reset = <&gpio 10 0>;
+	cirrus,gpio-nreset = <&gpio 10 0>;
 	cirrus,chgfreq-divisor = <0x05>;
 	cirrus.ain1_ref_cfg;
 	cirrus,micbias-lvl = <5>;

Documentation/devicetree/bindings/sound/rt5514.txt

@@ -1,22 +1,27 @@
 RT5514 audio CODEC
 
-This device supports I2C only.
+This device supports both I2C and SPI.
 
 Required properties:
 
 - compatible : "realtek,rt5514".
 
-- reg : The I2C address of the device.
+- reg : the I2C address of the device for I2C, the chip select
+        number for SPI.
 
 Optional properties:
 
 - clocks: The phandle of the master clock to the CODEC
 - clock-names: Should be "mclk"
 
+- interrupt-parent: The phandle for the interrupt controller.
+- interrupts: The interrupt number to the cpu. The interrupt specifier format
+	      depends on the interrupt controller.
+
 - realtek,dmic-init-delay-ms
-  Set the DMIC initial delay (ms) to wait it ready.
+  Set the DMIC initial delay (ms) to wait it ready for I2C.
 
-Pins on the device (for linking into audio routes) for RT5514:
+Pins on the device (for linking into audio routes) for I2C:
 
   * DMIC1L
   * DMIC1R

Documentation/devicetree/bindings/sound/rt5663.txt

@@ -19,6 +19,22 @@ Optional properties:
   Based on the different PCB layout, add the manual offset value to
   compensate the DC offset for each L and R channel, and they are different
   between headphone and headset.
+- "realtek,impedance_sensing_num"
+  The matrix row number of the impedance sensing table.
+  If the value is 0, it means the impedance sensing is not supported.
+- "realtek,impedance_sensing_table"
+  The matrix rows of the impedance sensing table are consisted by impedance
+  minimum, impedance maximun, volume, DC offset w/o and w/ mic of each L and
+  R channel accordingly. Example is shown as following.
+  <   0    300  7  0xffd160  0xffd1c0  0xff8a10  0xff8ab0
+    301  65535  4  0xffe470  0xffe470  0xffb8e0  0xffb8e0>
+  The first and second column are defined for the impedance range. If the
+  detected impedance value is in the range, then the volume value of the
+  third column will be set to codec. In our codec design, each volume value
+  should compensate different DC offset to avoid the pop sound, and it is
+  also different between headphone and headset. In the example, the
+  "realtek,impedance_sensing_num" is 2. It means that there are 2 ranges of
+  impedance in the impedance sensing function.
 
 Pins on the device (for linking into audio routes) for RT5663:
 

Documentation/devicetree/bindings/sound/sgtl5000.txt

@@ -37,7 +37,7 @@ VDDIO		1.8V		2.5V		3.3V
 
 Example:
 
-codec: sgtl5000@0a {
+codec: sgtl5000@a {
 	compatible = "fsl,sgtl5000";
 	reg = <0x0a>;
 	clocks = <&clks 150>;

Documentation/devicetree/bindings/sound/st,stm32-sai.txt

@@ -10,13 +10,21 @@ Required properties:
   - reg: Base address and size of SAI common register set.
   - clocks: Must contain phandle and clock specifier pairs for each entry
 	in clock-names.
-  - clock-names: Must contain "x8k" and "x11k"
+  - clock-names: Must contain "pclk" "x8k" and "x11k"
+	"pclk": Clock which feeds the peripheral bus interface.
+	        Mandatory for "st,stm32h7-sai" compatible.
+	        Not used for "st,stm32f4-sai" compatible.
 	"x8k": SAI parent clock for sampling rates multiple of 8kHz.
 	"x11k": SAI parent clock for sampling rates multiple of 11.025kHz.
   - interrupts: cpu DAI interrupt line shared by SAI sub-blocks
 
 Optional properties:
   - resets: Reference to a reset controller asserting the SAI
+  - st,sync: specify synchronization mode.
+	By default SAI sub-block is in asynchronous mode.
+	This property sets SAI sub-block as slave of another SAI sub-block.
+	Must contain the phandle and index of the sai sub-block providing
+	the synchronization.
 
 SAI subnodes:
 Two subnodes corresponding to SAI sub-block instances A et B can be defined.
@@ -52,8 +60,8 @@ sai1: sai1@40015800 {
 	#size-cells = <1>;
 	ranges = <0 0x40015800 0x400>;
 	reg = <0x40015800 0x4>;
-	clocks = <&rcc PLL1_Q>, <&rcc PLL2_P>;
-	clock-names = "x8k", "x11k";
+	clocks = <&rcc SAI1_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>;
+	clock-names = "pclk", "x8k", "x11k";
 	interrupts = <87>;
 
 	sai1a: audio-controller@40015804 {

Documentation/devicetree/bindings/sound/tfa9879.txt

@@ -0,0 +1,23 @@
+NXP TFA9879 class-D audio amplifier
+
+Required properties:
+
+- compatible : "nxp,tfa9879"
+
+- reg : the I2C address of the device
+
+Example:
+
+&i2c1 {
+	clock-frequency = <100000>;
+	pinctrl-names = "default";
+	pinctrl-0 = <&pinctrl_i2c1>;
+	status = "okay";
+
+	codec: tfa9879@6c {
+		#sound-dai-cells = <0>;
+		compatible = "nxp,tfa9879";
+		reg = <0x6c>;
+        };
+};
+

Documentation/devicetree/bindings/sound/wlf,arizona.txt

@@ -0,0 +1,53 @@
+Cirrus Logic Arizona class audio SoCs
+
+These devices are audio SoCs with extensive digital capabilities and a range
+of analogue I/O.
+
+This document lists sound specific bindings, see the primary binding
+document:
+  ../mfd/arizona.txt
+
+Optional properties:
+
+  - wlf,inmode : A list of INn_MODE register values, where n is the number
+    of input signals. Valid values are 0 (Differential), 1 (Single-ended) and
+    2 (Digital Microphone). If absent, INn_MODE registers set to 0 by default.
+    If present, values must be specified less than or equal to the number of
+    input signals. If values less than the number of input signals, elements
+    that have not been specified are set to 0 by default. Entries are:
+    <IN1, IN2, IN3, IN4> (wm5102, wm5110, wm8280, wm8997)
+    <IN1A, IN2A, IN1B, IN2B> (wm8998, wm1814)
+  - wlf,out-mono : A list of boolean values indicating whether each output is
+    mono or stereo. Position within the list indicates the output affected
+    (eg. First entry in the list corresponds to output 1). A non-zero value
+    indicates a mono output. If present, the number of values should be less
+    than or equal to the number of outputs, if less values are supplied the
+    additional outputs will be treated as stereo.
+
+  - wlf,dmic-ref : DMIC reference voltage source for each input, can be
+    selected from either MICVDD or one of the MICBIAS's, defines
+    (ARIZONA_DMIC_xxxx) are provided in <dt-bindings/mfd/arizona.txt>. If
+    present, the number of values should be less than or equal to the
+    number of inputs, unspecified inputs will use the chip default.
+
+  - wlf,max-channels-clocked : The maximum number of channels to be clocked on
+    each AIF, useful for I2S systems with multiple data lines being mastered.
+    Specify one cell for each AIF to be configured, specify zero for AIFs that
+    should be handled normally.
+    If present, number of cells must be less than or equal to the number of
+    AIFs. If less than the number of AIFs, for cells that have not been
+    specified the corresponding AIFs will be treated as default setting.
+
+  - wlf,spk-fmt : PDM speaker data format, must contain 2 cells (OUT5 and OUT6).
+    See the datasheet for values.
+    The second cell is ignored for codecs that do not have OUT6 (wm5102, wm8997,
+      wm8998, wm1814)
+
+  - wlf,spk-mute : PDM speaker mute setting, must contain 2 cells (OUT5 and OUT6).
+    See the datasheet for values.
+    The second cell is ignored for codecs that do not have OUT6 (wm5102, wm8997,
+    wm8998, wm1814)
+
+  - wlf,out-volume-limit : The volume limit value that should be applied to each
+    output channel. See the datasheet for exact values. Channels are specified
+    in the order OUT1L, OUT1R, OUT2L, OUT2R, etc.

Documentation/sound/hd-audio/models.rst

@@ -82,6 +82,8 @@ tpt460
     Lenovo Thinkpad T460/560 setup
 dual-codecs
     Lenovo laptops with dual codecs
+alc700-ref
+    Intel reference board with ALC700 codec
 
 ALC66x/67x/892
 ==============