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-rw-r--r--Documentation/DocBook/writing-an-alsa-driver.tmpl28
-rw-r--r--Documentation/devicetree/bindings/sound/adi,ssm2602.txt19
-rw-r--r--Documentation/devicetree/bindings/sound/cs35l32.txt62
-rw-r--r--Documentation/devicetree/bindings/sound/es8328.txt38
-rw-r--r--Documentation/devicetree/bindings/sound/fsl,esai.txt3
-rw-r--r--Documentation/devicetree/bindings/sound/fsl,ssi.txt8
-rw-r--r--Documentation/devicetree/bindings/sound/fsl-asoc-card.txt82
-rw-r--r--Documentation/devicetree/bindings/sound/fsl-sai.txt30
-rw-r--r--Documentation/devicetree/bindings/sound/imx-audio-es8328.txt60
-rw-r--r--Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt1
-rw-r--r--Documentation/devicetree/bindings/sound/rt5677.txt59
-rw-r--r--Documentation/devicetree/bindings/sound/simple-card.txt4
-rw-r--r--Documentation/devicetree/bindings/sound/ssm4567.txt15
-rw-r--r--Documentation/devicetree/bindings/vendor-prefixes.txt1
14 files changed, 389 insertions, 21 deletions
diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl
index 6f639d9530b5..784793df81ed 100644
--- a/Documentation/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl
@@ -2742,7 +2742,9 @@ struct _snd_pcm_runtime {
<para>
Another note is that this callback is non-atomic
- (schedulable). This is important, because the
+ (schedulable) as default, i.e. when no
+ <structfield>nonatomic</structfield> flag set.
+ This is important, because the
<structfield>trigger</structfield> callback
is atomic (non-schedulable). That is, mutexes or any
schedule-related functions are not available in
@@ -2900,8 +2902,9 @@ struct _snd_pcm_runtime {
</para>
<para>
- As mentioned, this callback is atomic. You cannot call
- functions which may sleep.
+ As mentioned, this callback is atomic as default unless
+ <structfield>nonatomic</structfield> flag set, and
+ you cannot call functions which may sleep.
The trigger callback should be as minimal as possible,
just really triggering the DMA. The other stuff should be
initialized hw_params and prepare callbacks properly
@@ -2936,7 +2939,7 @@ struct _snd_pcm_runtime {
</para>
<para>
- This callback is also atomic.
+ This callback is also atomic as default.
</para>
</section>
@@ -2972,7 +2975,7 @@ struct _snd_pcm_runtime {
is useful only for such a purpose.
</para>
<para>
- This callback is atomic.
+ This callback is atomic as default.
</para>
</section>
@@ -3175,6 +3178,21 @@ struct _snd_pcm_runtime {
called with local interrupts disabled.
</para>
+ <para>
+ The recent changes in PCM core code, however, allow all PCM
+ operations to be non-atomic. This assumes that the all caller
+ sides are in non-atomic contexts. For example, the function
+ <function>snd_pcm_period_elapsed()</function> is called
+ typically from the interrupt handler. But, if you set up the
+ driver to use a threaded interrupt handler, this call can be in
+ non-atomic context, too. In such a case, you can set
+ <structfield>nonatomic</structfield> filed of
+ <structname>snd_pcm</structname> object after creating it.
+ When this flag is set, mutex and rwsem are used internally in
+ the PCM core instead of spin and rwlocks, so that you can call
+ all PCM functions safely in a non-atomic context.
+ </para>
+
</section>
<section id="pcm-interface-constraints">
<title>Constraints</title>
diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2602.txt b/Documentation/devicetree/bindings/sound/adi,ssm2602.txt
new file mode 100644
index 000000000000..3b3302fe399b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,ssm2602.txt
@@ -0,0 +1,19 @@
+Analog Devices SSM2602, SSM2603 and SSM2604 I2S audio CODEC devices
+
+SSM2602 support both I2C and SPI as the configuration interface,
+the selection is made by the MODE strap-in pin.
+SSM2603 and SSM2604 only support I2C as the configuration interface.
+
+Required properties:
+
+ - compatible : One of "adi,ssm2602", "adi,ssm2603" or "adi,ssm2604"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+ Example:
+
+ ssm2602: ssm2602@1a {
+ compatible = "adi,ssm2602";
+ reg = <0x1a>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/cs35l32.txt b/Documentation/devicetree/bindings/sound/cs35l32.txt
new file mode 100644
index 000000000000..1417d3f5cc22
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs35l32.txt
@@ -0,0 +1,62 @@
+CS35L32 audio CODEC
+
+Required properties:
+
+ - compatible : "cirrus,cs35l32"
+
+ - reg : the I2C address of the device for I2C. Address is determined by the level
+ of the AD0 pin. Level 0 is 0x40 while Level 1 is 0x41.
+
+ - VA-supply, VP-supply : power supplies for the device,
+ as covered in Documentation/devicetree/bindings/regulator/regulator.txt.
+
+Optional properties:
+
+ - reset-gpios : a GPIO spec for the reset pin. If specified, it will be
+ deasserted before communication to the codec starts.
+
+ - cirrus,boost-manager : Boost voltage control.
+ 0 = Automatically managed. Boost-converter output voltage is the higher
+ of the two: Class G or adaptive LED voltage.
+ 1 = Automatically managed irrespective of audio, adapting for low-power
+ dissipation when LEDs are ON, and operating in Fixed-Boost Bypass Mode
+ if LEDs are OFF (VBST = VP).
+ 2 = (Default) Boost voltage fixed in Bypass Mode (VBST = VP).
+ 3 = Boost voltage fixed at 5 V.
+
+ - cirrus,sdout-datacfg : Data configuration for dual CS35L32 applications only.
+ Determines the data packed in a two-CS35L32 configuration.
+ 0 = Left/right channels VMON[11:0], IMON[11:0], VPMON[7:0].
+ 1 = Left/right channels VMON[11:0], IMON[11:0], STATUS.
+ 2 = (Default) left/right channels VMON[15:0], IMON [15:0].
+ 3 = Left/right channels VPMON[7:0], STATUS.
+
+ - cirrus,sdout-share : SDOUT sharing. Determines whether one or two CS35L32
+ devices are on board sharing SDOUT.
+ 0 = (Default) One IC.
+ 1 = Two IC's.
+
+ - cirrus,battery-recovery : Low battery nominal recovery threshold, rising VP.
+ 0 = 3.1V
+ 1 = 3.2V
+ 2 = 3.3V (Default)
+ 3 = 3.4V
+
+ - cirrus,battery-threshold : Low battery nominal threshold, falling VP.
+ 0 = 3.1V
+ 1 = 3.2V
+ 2 = 3.3V
+ 3 = 3.4V (Default)
+ 4 = 3.5V
+ 5 = 3.6V
+
+Example:
+
+codec: codec@40 {
+ compatible = "cirrus,cs35l32";
+ reg = <0x40>;
+ reset-gpios = <&gpio 10 0>;
+ cirrus,boost-manager = <0x03>;
+ cirrus,sdout-datacfg = <0x02>;
+ VA-supply = <&reg_audio>;
+};
diff --git a/Documentation/devicetree/bindings/sound/es8328.txt b/Documentation/devicetree/bindings/sound/es8328.txt
new file mode 100644
index 000000000000..30ea8a318ae9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/es8328.txt
@@ -0,0 +1,38 @@
+Everest ES8328 audio CODEC
+
+This device supports both I2C and SPI.
+
+Required properties:
+
+ - compatible : "everest,es8328"
+ - DVDD-supply : Regulator providing digital core supply voltage 1.8 - 3.6V
+ - AVDD-supply : Regulator providing analog supply voltage 3.3V
+ - PVDD-supply : Regulator providing digital IO supply voltage 1.8 - 3.6V
+ - IPVDD-supply : Regulator providing analog output voltage 3.3V
+ - clocks : A 22.5792 or 11.2896 MHz clock
+ - reg : the I2C address of the device for I2C, the chip select number for SPI
+
+Pins on the device (for linking into audio routes):
+
+ * LOUT1
+ * LOUT2
+ * ROUT1
+ * ROUT2
+ * LINPUT1
+ * RINPUT1
+ * LINPUT2
+ * RINPUT2
+ * Mic Bias
+
+
+Example:
+
+codec: es8328@11 {
+ compatible = "everest,es8328";
+ DVDD-supply = <&reg_3p3v>;
+ AVDD-supply = <&reg_3p3v>;
+ PVDD-supply = <&reg_3p3v>;
+ HPVDD-supply = <&reg_3p3v>;
+ clocks = <&clks 169>;
+ reg = <0x11>;
+};
diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt
index aeb8c4a0b88d..52f5b6bf3e8e 100644
--- a/Documentation/devicetree/bindings/sound/fsl,esai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt
@@ -7,7 +7,8 @@ other DSPs. It has up to six transmitters and four receivers.
Required properties:
- - compatible : Compatible list, must contain "fsl,imx35-esai".
+ - compatible : Compatible list, must contain "fsl,imx35-esai" or
+ "fsl,vf610-esai"
- reg : Offset and length of the register set for the device.
diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt
index 3aa4a8f528f4..5b76be45d18b 100644
--- a/Documentation/devicetree/bindings/sound/fsl,ssi.txt
+++ b/Documentation/devicetree/bindings/sound/fsl,ssi.txt
@@ -58,13 +58,7 @@ Optional properties:
Documentation/devicetree/bindings/dma/dma.txt.
- dma-names: Two dmas have to be defined, "tx" and "rx", if fsl,imx-fiq
is not defined.
-- fsl,mode: The operating mode for the SSI interface.
- "i2s-slave" - I2S mode, SSI is clock slave
- "i2s-master" - I2S mode, SSI is clock master
- "lj-slave" - left-justified mode, SSI is clock slave
- "lj-master" - l.j. mode, SSI is clock master
- "rj-slave" - right-justified mode, SSI is clock slave
- "rj-master" - r.j., SSI is clock master
+- fsl,mode: The operating mode for the AC97 interface only.
"ac97-slave" - AC97 mode, SSI is clock slave
"ac97-master" - AC97 mode, SSI is clock master
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
new file mode 100644
index 000000000000..a96774c194c8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
@@ -0,0 +1,82 @@
+Freescale Generic ASoC Sound Card with ASRC support
+
+The Freescale Generic ASoC Sound Card can be used, ideally, for all Freescale
+SoCs connecting with external CODECs.
+
+The idea of this generic sound card is a bit like ASoC Simple Card. However,
+for Freescale SoCs (especially those released in recent years), most of them
+have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And
+this is a specific feature that might be painstakingly controlled and merged
+into the Simple Card.
+
+So having this generic sound card allows all Freescale SoC users to benefit
+from the simplification of a new card support and the capability of the wide
+sample rates support through ASRC.
+
+Note: The card is initially designed for those sound cards who use I2S and
+ PCM DAI formats. However, it'll be also possible to support those non
+ I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long
+ as the driver has been properly upgraded.
+
+
+The compatible list for this generic sound card currently:
+ "fsl,imx-audio-cs42888"
+
+ "fsl,imx-audio-wm8962"
+ (compatible with Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt)
+
+ "fsl,imx-audio-sgtl5000"
+ (compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt)
+
+Required properties:
+
+ - compatible : Contains one of entries in the compatible list.
+
+ - model : The user-visible name of this sound complex
+
+ - audio-cpu : The phandle of an CPU DAI controller
+
+ - audio-codec : The phandle of an audio codec
+
+ - 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. There're a few pre-designed board connectors:
+ * Line Out Jack
+ * Line In Jack
+ * Headphone Jack
+ * Mic Jack
+ * Ext Spk
+ * AMIC (stands for Analog Microphone Jack)
+ * DMIC (stands for Digital Microphone Jack)
+
+ Note: The "Mic Jack" and "AMIC" are redundant while
+ coexsiting in order to support the old bindings
+ of wm8962 and sgtl5000.
+
+Optional properties:
+
+ - audio-asrc : The phandle of ASRC. It can be absent if there's no
+ need to add ASRC support via DPCM.
+
+Example:
+sound-cs42888 {
+ compatible = "fsl,imx-audio-cs42888";
+ model = "cs42888-audio";
+ audio-cpu = <&esai>;
+ audio-asrc = <&asrc>;
+ audio-codec = <&cs42888>;
+ audio-routing =
+ "Line Out Jack", "AOUT1L",
+ "Line Out Jack", "AOUT1R",
+ "Line Out Jack", "AOUT2L",
+ "Line Out Jack", "AOUT2R",
+ "Line Out Jack", "AOUT3L",
+ "Line Out Jack", "AOUT3R",
+ "Line Out Jack", "AOUT4L",
+ "Line Out Jack", "AOUT4R",
+ "AIN1L", "Line In Jack",
+ "AIN1R", "Line In Jack",
+ "AIN2L", "Line In Jack",
+ "AIN2R", "Line In Jack";
+};
diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt
index 0f4e23828190..4956b14d4b06 100644
--- a/Documentation/devicetree/bindings/sound/fsl-sai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -18,12 +18,26 @@ Required properties:
- pinctrl-names: Must contain a "default" entry.
- pinctrl-NNN: One property must exist for each entry in pinctrl-names.
See ../pinctrl/pinctrl-bindings.txt for details of the property values.
-- big-endian-regs: If this property is absent, the little endian mode will
- be in use as default, or the big endian mode will be in use for all the
- device registers.
-- big-endian-data: If this property is absent, the little endian mode will
- be in use as default, or the big endian mode will be in use for all the
- fifo data.
+- big-endian: Boolean property, required if all the FTM_PWM registers
+ are big-endian rather than little-endian.
+- lsb-first: Configures whether the LSB or the MSB is transmitted first for
+ the fifo data. If this property is absent, the MSB is transmitted first as
+ default, or the LSB is transmitted first.
+- fsl,sai-synchronous-rx: This is a boolean property. If present, indicating
+ that SAI will work in the synchronous mode (sync Tx with Rx) which means
+ both the transimitter and receiver will send and receive data by following
+ receiver's bit clocks and frame sync clocks.
+- fsl,sai-asynchronous: This is a boolean property. If present, indicating
+ that SAI will work in the asynchronous mode, which means both transimitter
+ and receiver will send and receive data by following their own bit clocks
+ and frame sync clocks separately.
+
+Note:
+- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the
+ default synchronous mode (sync Rx with Tx) will be used, which means both
+ transimitter and receiver will send and receive data by following clocks
+ of transimitter.
+- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive.
Example:
sai2: sai@40031000 {
@@ -38,6 +52,6 @@ sai2: sai@40031000 {
dma-names = "tx", "rx";
dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>,
<&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>;
- big-endian-regs;
- big-endian-data;
+ big-endian;
+ lsb-first;
};
diff --git a/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt b/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt
new file mode 100644
index 000000000000..07b68ab206fb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt
@@ -0,0 +1,60 @@
+Freescale i.MX audio complex with ES8328 codec
+
+Required properties:
+- compatible : "fsl,imx-audio-es8328"
+- model : The user-visible name of this sound complex
+- ssi-controller : The phandle of the i.MX SSI controller
+- jack-gpio : Optional GPIO for headphone jack
+- audio-amp-supply : Power regulator for speaker amps
+- audio-codec : The phandle of the ES8328 audio codec
+- 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 could be power supplies, ES8328
+ pins, and the jacks on the board:
+
+ Power supplies:
+ * audio-amp
+
+ ES8328 pins:
+ * LOUT1
+ * LOUT2
+ * ROUT1
+ * ROUT2
+ * LINPUT1
+ * LINPUT2
+ * RINPUT1
+ * RINPUT2
+ * Mic PGA
+
+ Board connectors:
+ * Headphone
+ * Speaker
+ * Mic Jack
+- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX)
+- mux-ext-port : The external port of the i.MX audio muxer (AUDMIX)
+
+Note: The AUDMUX port numbering should start at 1, which is consistent with
+hardware manual.
+
+Example:
+
+sound {
+ compatible = "fsl,imx-audio-es8328";
+ model = "imx-audio-es8328";
+ ssi-controller = <&ssi1>;
+ audio-codec = <&codec>;
+ jack-gpio = <&gpio5 15 0>;
+ audio-amp-supply = <&reg_audio_amp>;
+ audio-routing =
+ "Speaker", "LOUT2",
+ "Speaker", "ROUT2",
+ "Speaker", "audio-amp",
+ "Headphone", "ROUT1",
+ "Headphone", "LOUT1",
+ "LINPUT1", "Mic Jack",
+ "RINPUT1", "Mic Jack",
+ "Mic Jack", "Mic Bias";
+ mux-int-port = <1>;
+ mux-ext-port = <3>;
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt
index 9c7c55c71370..c949abc2992f 100644
--- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt
@@ -25,6 +25,7 @@ Required properties:
Optional properties:
- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in
+- nvidia,mic-det-gpios : The GPIO that detect microphones are plugged in
Example:
diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt
new file mode 100644
index 000000000000..0701b834fc73
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5677.txt
@@ -0,0 +1,59 @@
+RT5677 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5677".
+
+- reg : The I2C address of the device.
+
+- interrupts : The CODEC's interrupt output.
+
+- gpio-controller : Indicates this device is a GPIO controller.
+
+- #gpio-cells : Should be two. The first cell is the pin number and the
+ second cell is used to specify optional parameters (currently unused).
+
+Optional properties:
+
+- realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin.
+
+- realtek,in1-differential
+- realtek,in2-differential
+- realtek,lout1-differential
+- realtek,lout2-differential
+- realtek,lout3-differential
+ Boolean. Indicate MIC1/2 input and LOUT1/2/3 outputs are differential,
+ rather than single-ended.
+
+Pins on the device (for linking into audio routes):
+
+ * IN1P
+ * IN1N
+ * IN2P
+ * IN2N
+ * MICBIAS1
+ * DMIC1
+ * DMIC2
+ * DMIC3
+ * DMIC4
+ * LOUT1
+ * LOUT2
+ * LOUT3
+
+Example:
+
+rt5677 {
+ compatible = "realtek,rt5677";
+ reg = <0x2c>;
+ interrupt-parent = <&gpio>;
+ interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>;
+
+ gpio-controller;
+ #gpio-cells = <2>;
+
+ realtek,pow-ldo2-gpio =
+ <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>;
+ realtek,in1-differential = "true";
+};
diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt
index c2e9841dfce4..c3cba600bf11 100644
--- a/Documentation/devicetree/bindings/sound/simple-card.txt
+++ b/Documentation/devicetree/bindings/sound/simple-card.txt
@@ -17,6 +17,10 @@ Optional properties:
source.
- simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec
mclk.
+- simple-audio-card,hp-det-gpio : Reference to GPIO that signals when
+ headphones are attached.
+- simple-audio-card,mic-det-gpio : Reference to GPIO that signals when
+ a microphone is attached.
Optional subnodes:
diff --git a/Documentation/devicetree/bindings/sound/ssm4567.txt b/Documentation/devicetree/bindings/sound/ssm4567.txt
new file mode 100644
index 000000000000..ec3d9e7004b5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ssm4567.txt
@@ -0,0 +1,15 @@
+Analog Devices SSM4567 audio amplifier
+
+This device supports I2C only.
+
+Required properties:
+ - compatible : Must be "adi,ssm4567"
+ - reg : the I2C address of the device. This will either be 0x34 (LR_SEL/ADDR connected to AGND),
+ 0x35 (LR_SEL/ADDR connected to IOVDD) or 0x36 (LR_SEL/ADDR open).
+
+Example:
+
+ ssm4567: ssm4567@34 {
+ compatible = "adi,ssm4567";
+ reg = <0x34>;
+ };
diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt
index 653beaa392dc..71259df60e96 100644
--- a/Documentation/devicetree/bindings/vendor-prefixes.txt
+++ b/Documentation/devicetree/bindings/vendor-prefixes.txt
@@ -51,6 +51,7 @@ epfl Ecole Polytechnique Fédérale de Lausanne
epson Seiko Epson Corp.
est ESTeem Wireless Modems
eukrea Eukréa Electromatique
+everest Everest Semiconductor Co. Ltd.
excito Excito
fcs Fairchild Semiconductor
fsl Freescale Semiconductor