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-rw-r--r--dts/Bindings/sound/adi,adau1977.txt13
-rw-r--r--dts/Bindings/sound/ak4458.txt2
-rw-r--r--dts/Bindings/sound/audio-graph-scu-card.txt123
-rw-r--r--dts/Bindings/sound/cs35l36.txt168
-rw-r--r--dts/Bindings/sound/cs4341.txt22
-rw-r--r--dts/Bindings/sound/davinci-mcasp-audio.txt17
-rw-r--r--dts/Bindings/sound/fsl,micfil.txt32
-rw-r--r--dts/Bindings/sound/google,cros-ec-codec.txt26
-rw-r--r--dts/Bindings/sound/ingenic,jz4725b-codec.txt20
-rw-r--r--dts/Bindings/sound/ingenic,jz4740-codec.txt20
-rw-r--r--dts/Bindings/sound/mt6358.txt18
-rw-r--r--dts/Bindings/sound/mt8183-afe-pcm.txt36
-rw-r--r--dts/Bindings/sound/mtk-btcvsd-snd.txt24
-rw-r--r--dts/Bindings/sound/nvidia,tegra30-hda.txt5
-rw-r--r--dts/Bindings/sound/qcom,msm8916-wcd-analog.txt7
-rw-r--r--dts/Bindings/sound/qcom,wcd9335.txt10
-rw-r--r--dts/Bindings/sound/rockchip,rk3328-codec.txt23
-rw-r--r--dts/Bindings/sound/sgtl5000.txt9
-rw-r--r--dts/Bindings/sound/simple-scu-card.txt94
-rw-r--r--dts/Bindings/sound/sprd-pcm.txt23
-rw-r--r--dts/Bindings/sound/xlnx,audio-formatter.txt29
-rw-r--r--dts/Bindings/sound/xlnx,spdif.txt28
22 files changed, 520 insertions, 229 deletions
diff --git a/dts/Bindings/sound/adi,adau1977.txt b/dts/Bindings/sound/adi,adau1977.txt
index e79aeef73f..9225472c80 100644
--- a/dts/Bindings/sound/adi,adau1977.txt
+++ b/dts/Bindings/sound/adi,adau1977.txt
@@ -17,12 +17,18 @@ Required properties:
Documentation/devicetree/bindings/regulator/regulator.txt
Optional properties:
- - reset-gpio: the reset pin for the chip, for more details consult
+ - reset-gpios: the reset pin for the chip, for more details consult
Documentation/devicetree/bindings/gpio/gpio.txt
- DVDD-supply: supply voltage for the digital core, please consult
Documentation/devicetree/bindings/regulator/regulator.txt
+- adi,micbias: configures the voltage setting for the MICBIAS pin.
+ Select 0/1/2/3/4/5/6/7/8 to specify MICBIAS voltage
+ 5V/5.5V/6V/6.5V/7V/7.5V/8V/8.5V/9V
+ If not specified the default value will be "7" meaning 8.5 Volts.
+ This property is only valid for the ADAU1977
+
For required properties on SPI, please consult
Documentation/devicetree/bindings/spi/spi-bus.txt
@@ -40,7 +46,8 @@ Examples:
AVDD-supply = <&regulator>;
DVDD-supply = <&regulator_digital>;
- reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>;
+ adi,micbias = <3>;
+ reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>;
};
adau1977_i2c: adau1977@11 {
@@ -50,5 +57,5 @@ Examples:
AVDD-supply = <&regulator>;
DVDD-supply = <&regulator_digital>;
- reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>;
+ reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>;
};
diff --git a/dts/Bindings/sound/ak4458.txt b/dts/Bindings/sound/ak4458.txt
index 7839be7844..e5820235e0 100644
--- a/dts/Bindings/sound/ak4458.txt
+++ b/dts/Bindings/sound/ak4458.txt
@@ -4,7 +4,7 @@ This device supports I2C mode.
Required properties:
-- compatible : "asahi-kasei,ak4458"
+- compatible : "asahi-kasei,ak4458" or "asahi-kasei,ak4497"
- reg : The I2C address of the device for I2C
Optional properties:
diff --git a/dts/Bindings/sound/audio-graph-scu-card.txt b/dts/Bindings/sound/audio-graph-scu-card.txt
deleted file mode 100644
index 62d42768a0..0000000000
--- a/dts/Bindings/sound/audio-graph-scu-card.txt
+++ /dev/null
@@ -1,123 +0,0 @@
-Audio-Graph-SCU-Card:
-
-Audio-Graph-SCU-Card is "Audio-Graph-Card" + "ALSA DPCM".
-
-It is based on common bindings for device graphs.
-see ${LINUX}/Documentation/devicetree/bindings/graph.txt
-
-Basically, Audio-Graph-SCU-Card property is same as
-Simple-Card / Simple-SCU-Card / Audio-Graph-Card.
-see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt
- ${LINUX}/Documentation/devicetree/bindings/sound/simple-scu-card.txt
- ${LINUX}/Documentation/devicetree/bindings/sound/audio-graph-card.txt
-
-Below are same as Simple-Card / Audio-Graph-Card.
-
-- label
-- dai-format
-- frame-master
-- bitclock-master
-- bitclock-inversion
-- frame-inversion
-- dai-tdm-slot-num
-- dai-tdm-slot-width
-- clocks / system-clock-frequency
-
-Below are same as Simple-SCU-Card.
-
-- convert-rate
-- convert-channels
-- prefix
-- routing
-
-Required properties:
-
-- compatible : "audio-graph-scu-card";
-- dais : list of CPU DAI port{s}
-
-Example 1. Sampling Rate Conversion
-
- sound_card {
- compatible = "audio-graph-scu-card";
-
- label = "sound-card";
- prefix = "codec";
- routing = "codec Playback", "DAI0 Playback",
- "DAI0 Capture", "codec Capture";
- convert-rate = <48000>;
-
- dais = <&cpu_port>;
- };
-
- audio-codec {
- ...
-
- port {
- codec_endpoint: endpoint {
- remote-endpoint = <&cpu_endpoint>;
- };
- };
- };
-
- dai-controller {
- ...
- cpu_port: port {
- cpu_endpoint: endpoint {
- remote-endpoint = <&codec_endpoint>;
-
- dai-format = "left_j";
- ...
- };
- };
- };
-
-Example 2. 2 CPU 1 Codec (Mixing)
-
- sound_card {
- compatible = "audio-graph-scu-card";
-
- label = "sound-card";
- routing = "codec Playback", "DAI0 Playback",
- "codec Playback", "DAI1 Playback",
- "DAI0 Capture", "codec Capture";
-
- dais = <&cpu_port0
- &cpu_port1>;
- };
-
- audio-codec {
- ...
-
- audio-graph-card,prefix = "codec";
- audio-graph-card,convert-rate = <48000>;
- port {
- codec_endpoint0: endpoint {
- remote-endpoint = <&cpu_endpoint0>;
- };
- codec_endpoint1: endpoint {
- remote-endpoint = <&cpu_endpoint1>;
- };
- };
- };
-
- dai-controller {
- ...
- ports {
- cpu_port0: port {
- cpu_endpoint0: endpoint {
- remote-endpoint = <&codec_endpoint0>;
-
- dai-format = "left_j";
- ...
- };
- };
- cpu_port1: port {
- cpu_endpoint1: endpoint {
- remote-endpoint = <&codec_endpoint1>;
-
- dai-format = "left_j";
- ...
- };
- };
- };
- };
diff --git a/dts/Bindings/sound/cs35l36.txt b/dts/Bindings/sound/cs35l36.txt
new file mode 100644
index 0000000000..912bd162b4
--- /dev/null
+++ b/dts/Bindings/sound/cs35l36.txt
@@ -0,0 +1,168 @@
+CS35L36 Speaker Amplifier
+
+Required properties:
+
+ - compatible : "cirrus,cs35l36"
+
+ - reg : the I2C address of the device for I2C
+
+ - VA-supply, VP-supply : power supplies for the device,
+ as covered in
+ Documentation/devicetree/bindings/regulator/regulator.txt.
+
+ - cirrus,boost-ctl-millivolt : Boost Voltage Value. Configures the boost
+ converter's output voltage in mV. The range is from 2550mV to 12000mV with
+ increments of 50mV.
+ (Default) VP
+
+ - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA.
+ Configures the peak current by monitoring the current through the boost FET.
+ Range starts at 1600mA and goes to a maximum of 4500mA with increments of
+ 50mA.
+ (Default) 4.50 Amps
+
+ - cirrus,boost-ind-nanohenry : Inductor estimation LBST reference value.
+ Seeds the digital boost converter's inductor estimation block with the initial
+ inductance value to reference.
+
+ 1000 = 1uH (Default)
+ 1200 = 1.2uH
+
+Optional properties:
+ - cirrus,multi-amp-mode : Boolean to determine if there are more than
+ one amplifier in the system. If more than one it is best to Hi-Z the ASP
+ port to prevent bus contention on the output signal
+
+ - cirrus,boost-ctl-select : Boost conerter control source selection.
+ Selects the source of the BST_CTL target VBST voltage for the boost
+ converter to generate.
+ 0x00 - Control Port Value
+ 0x01 - Class H Tracking (Default)
+ 0x10 - MultiDevice Sync Value
+
+ - cirrus,amp-pcm-inv : Boolean to determine Amplifier will invert incoming
+ PCM data
+
+ - cirrus,imon-pol-inv : Boolean to determine Amplifier will invert the
+ polarity of outbound IMON feedback data
+
+ - cirrus,vmon-pol-inv : Boolean to determine Amplifier will invert the
+ polarity of outbound VMON feedback data
+
+ - cirrus,dcm-mode-enable : Boost converter automatic DCM Mode enable.
+ This enables the digital boost converter to operate in a low power
+ (Discontinuous Conduction) mode during low loading conditions.
+
+ - cirrus,weak-fet-disable : Boolean : The strength of the output drivers is
+ reduced when operating in a Weak-FET Drive Mode and must not be used to drive
+ a large load.
+
+ - cirrus,classh-wk-fet-delay : Weak-FET entry delay. Controls the delay
+ (in ms) before the Class H algorithm switches to the weak-FET voltage
+ (after the audio falls and remains below the value specified in WKFET_AMP_THLD).
+
+ 0 = 0ms
+ 1 = 5ms
+ 2 = 10ms
+ 3 = 50ms
+ 4 = 100ms (Default)
+ 5 = 200ms
+ 6 = 500ms
+ 7 = 1000ms
+
+ - cirrus,classh-weak-fet-thld-millivolt : Weak-FET amplifier drive threshold.
+ Configures the signal threshold at which the PWM output stage enters
+ weak-FET operation. The range is 50mV to 700mV in 50mV increments.
+
+ - cirrus,temp-warn-threshold : Amplifier overtemperature warning threshold.
+ Configures the threshold at which the overtemperature warning condition occurs.
+ When the threshold is met, the overtemperature warning attenuation is applied
+ and the TEMP_WARN_EINT interrupt status bit is set.
+ If TEMP_WARN_MASK = 0, INTb is asserted.
+
+ 0 = 105C
+ 1 = 115C
+ 2 = 125C (Default)
+ 3 = 135C
+
+ - cirrus,irq-drive-select : Selects the driver type of the selected interrupt
+ output.
+
+ 0 = Open-drain
+ 1 = Push-pull (Default)
+
+ - cirrus,irq-gpio-select : Selects the pin to serve as the programmable
+ interrupt output.
+
+ 0 = PDM_DATA / SWIRE_SD / INT (Default)
+ 1 = GPIO
+
+Optional properties for the "cirrus,vpbr-config" Sub-node
+
+ - cirrus,vpbr-en : VBST brownout prevention enable. Configures whether the
+ VBST brownout prevention algorithm is enabled or disabled.
+
+ 0 = VBST brownout prevention disabled (default)
+ 1 = VBST brownout prevention enabled
+
+ See Section 7.31.1 VPBR Config for configuration options & further details
+
+ - cirrus,vpbr-thld : Initial VPBR threshold. Configures the VP brownout
+ threshold voltage
+
+ - cirrus,cirrus,vpbr-atk-rate : Attenuation attack step rate. Configures the
+ amount delay between consecutive volume attenuation steps when a brownout
+ condition is present and the VP brownout condition is in an attacking state.
+
+ - cirrus,vpbr-atk-vol : VP brownout prevention step size. Configures the VP
+ brownout prevention attacking attenuation step size when operating in either
+ digital volume or analog gain modes.
+
+ - cirrus,vpbr-max-attn : Maximum attenuation that the VP brownout prevention
+ can apply to the audio signal.
+
+ - cirrus,vpbr-wait : Configures the delay time between a brownout condition
+ no longer being present and the VP brownout prevention entering an attenuation
+ release state.
+
+ - cirrus,vpbr-rel-rate : Attenuation release step rate. Configures the delay
+ between consecutive volume attenuation release steps when a brownout condition
+ is not longer present and the VP brownout is in an attenuation release state.
+
+ - cirrus,vpbr-mute-en : During the attack state, if the vpbr-max-attn value
+ is reached, the error condition still remains, and this bit is set, the audio
+ is muted.
+
+Example:
+
+cs35l36: cs35l36@40 {
+ compatible = "cirrus,cs35l36";
+ reg = <0x40>;
+ VA-supply = <&dummy_vreg>;
+ VP-supply = <&dummy_vreg>;
+ reset-gpios = <&gpio0 54 0>;
+ interrupt-parent = <&gpio8>;
+ interrupts = <3 IRQ_TYPE_LEVEL_LOW>;
+
+ cirrus,boost-ind-nanohenry = <1000>;
+ cirrus,boost-ctl-millivolt = <10000>;
+ cirrus,boost-peak-milliamp = <4500>;
+ cirrus,boost-ctl-select = <0x00>;
+ cirrus,weak-fet-delay = <0x04>;
+ cirrus,weak-fet-thld = <0x01>;
+ cirrus,temp-warn-threshold = <0x01>;
+ cirrus,multi-amp-mode;
+ cirrus,irq-drive-select = <0x01>;
+ cirrus,irq-gpio-select = <0x01>;
+
+ cirrus,vpbr-config {
+ cirrus,vpbr-en = <0x00>;
+ cirrus,vpbr-thld = <0x05>;
+ cirrus,vpbr-atk-rate = <0x02>;
+ cirrus,vpbr-atk-vol = <0x01>;
+ cirrus,vpbr-max-attn = <0x09>;
+ cirrus,vpbr-wait = <0x01>;
+ cirrus,vpbr-rel-rate = <0x05>;
+ cirrus,vpbr-mute-en = <0x00>;
+ };
+};
diff --git a/dts/Bindings/sound/cs4341.txt b/dts/Bindings/sound/cs4341.txt
new file mode 100644
index 0000000000..12b4aa8ef0
--- /dev/null
+++ b/dts/Bindings/sound/cs4341.txt
@@ -0,0 +1,22 @@
+Cirrus Logic CS4341 audio DAC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+ - compatible: "cirrus,cs4341a"
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+For required properties on I2C-bus, please consult
+Documentation/devicetree/bindings/i2c/i2c.txt
+For required properties on SPI-bus, please consult
+Documentation/devicetree/bindings/spi/spi-bus.txt
+
+Example:
+ codec: cs4341@0 {
+ #sound-dai-cells = <0>;
+ compatible = "cirrus,cs4341a";
+ reg = <0>;
+ spi-max-frequency = <6000000>;
+ };
diff --git a/dts/Bindings/sound/davinci-mcasp-audio.txt b/dts/Bindings/sound/davinci-mcasp-audio.txt
index b279b6072b..a58f79f534 100644
--- a/dts/Bindings/sound/davinci-mcasp-audio.txt
+++ b/dts/Bindings/sound/davinci-mcasp-audio.txt
@@ -45,6 +45,23 @@ Optional properties:
- fck_parent : Should contain a valid clock name which will be used as parent
for the McASP fck
+Optional GPIO support:
+If any McASP pin need to be used as GPIO then the McASP node must have:
+...
+ gpio-controller
+ #gpio-cells = <2>;
+...
+
+When requesting a GPIO, the first parameter is the PIN index in McASP_P*
+registers.
+For example to request the AXR2 pin of mcasp8:
+function-gpios = <&mcasp8 2 0>;
+
+Or to request the ACLKR pin of mcasp8:
+function-gpios = <&mcasp8 29 0>;
+
+For generic gpio information, please refer to bindings/gpio/gpio.txt
+
Example:
mcasp0: mcasp0@1d00000 {
diff --git a/dts/Bindings/sound/fsl,micfil.txt b/dts/Bindings/sound/fsl,micfil.txt
new file mode 100644
index 0000000000..53e227b152
--- /dev/null
+++ b/dts/Bindings/sound/fsl,micfil.txt
@@ -0,0 +1,32 @@
+NXP MICFIL Digital Audio Interface (MICFIL).
+
+The MICFIL digital interface provides a 16-bit audio signal from a PDM
+microphone bitstream in a configurable output sampling rate.
+
+Required properties:
+
+ - compatible : Compatible list, contains "fsl,imx8mm-micfil"
+
+ - reg : Offset and length of the register set for the device.
+
+ - interrupts : Contains the micfil interrupts.
+
+ - clocks : Must contain an entry for each entry in clock-names.
+
+ - clock-names : Must include the "ipg_clk" for register access and
+ "ipg_clk_app" for internal micfil clock.
+
+ - dmas : Generic dma devicetree binding as described in
+ Documentation/devicetree/bindings/dma/dma.txt.
+
+Example:
+micfil: micfil@30080000 {
+ compatible = "fsl,imx8mm-micfil";
+ reg = <0x0 0x30080000 0x0 0x10000>;
+ interrupts = <GIC_SPI 109 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 110 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&clk IMX8MM_CLK_PDM_IPG>,
+ <&clk IMX8MM_CLK_PDM_ROOT>;
+ clock-names = "ipg_clk", "ipg_clk_app";
+ dmas = <&sdma2 24 26 0x80000000>;
+};
diff --git a/dts/Bindings/sound/google,cros-ec-codec.txt b/dts/Bindings/sound/google,cros-ec-codec.txt
new file mode 100644
index 0000000000..1084f7f22e
--- /dev/null
+++ b/dts/Bindings/sound/google,cros-ec-codec.txt
@@ -0,0 +1,26 @@
+* Audio codec controlled by ChromeOS EC
+
+Google's ChromeOS EC codec is a digital mic codec provided by the
+Embedded Controller (EC) and is controlled via a host-command interface.
+
+An EC codec node should only be found as a sub-node of the EC node (see
+Documentation/devicetree/bindings/mfd/cros-ec.txt).
+
+Required properties:
+- compatible: Must contain "google,cros-ec-codec"
+- #sound-dai-cells: Should be 1. The cell specifies number of DAIs.
+- max-dmic-gain: A number for maximum gain in dB on digital microphone.
+
+Example:
+
+cros-ec@0 {
+ compatible = "google,cros-ec-spi";
+
+ ...
+
+ cros_ec_codec: ec-codec {
+ compatible = "google,cros-ec-codec";
+ #sound-dai-cells = <1>;
+ max-dmic-gain = <43>;
+ };
+};
diff --git a/dts/Bindings/sound/ingenic,jz4725b-codec.txt b/dts/Bindings/sound/ingenic,jz4725b-codec.txt
new file mode 100644
index 0000000000..05adc0d47b
--- /dev/null
+++ b/dts/Bindings/sound/ingenic,jz4725b-codec.txt
@@ -0,0 +1,20 @@
+Ingenic JZ4725B codec controller
+
+Required properties:
+- compatible : "ingenic,jz4725b-codec"
+- reg : codec registers location and length
+- clocks : phandle to the AIC clock.
+- clock-names: must be set to "aic".
+- #sound-dai-cells: Must be set to 0.
+
+Example:
+
+codec: audio-codec@100200a4 {
+ compatible = "ingenic,jz4725b-codec";
+ reg = <0x100200a4 0x8>;
+
+ #sound-dai-cells = <0>;
+
+ clocks = <&cgu JZ4725B_CLK_AIC>;
+ clock-names = "aic";
+};
diff --git a/dts/Bindings/sound/ingenic,jz4740-codec.txt b/dts/Bindings/sound/ingenic,jz4740-codec.txt
new file mode 100644
index 0000000000..1ffcade87e
--- /dev/null
+++ b/dts/Bindings/sound/ingenic,jz4740-codec.txt
@@ -0,0 +1,20 @@
+Ingenic JZ4740 codec controller
+
+Required properties:
+- compatible : "ingenic,jz4740-codec"
+- reg : codec registers location and length
+- clocks : phandle to the AIC clock.
+- clock-names: must be set to "aic".
+- #sound-dai-cells: Must be set to 0.
+
+Example:
+
+codec: audio-codec@10020080 {
+ compatible = "ingenic,jz4740-codec";
+ reg = <0x10020080 0x8>;
+
+ #sound-dai-cells = <0>;
+
+ clocks = <&cgu JZ4740_CLK_AIC>;
+ clock-names = "aic";
+};
diff --git a/dts/Bindings/sound/mt6358.txt b/dts/Bindings/sound/mt6358.txt
new file mode 100644
index 0000000000..5465730013
--- /dev/null
+++ b/dts/Bindings/sound/mt6358.txt
@@ -0,0 +1,18 @@
+Mediatek MT6358 Audio Codec
+
+The communication between MT6358 and SoC is through Mediatek PMIC wrapper.
+For more detail, please visit Mediatek PMIC wrapper documentation.
+
+Must be a child node of PMIC wrapper.
+
+Required properties:
+
+- compatible : "mediatek,mt6358-sound".
+- Avdd-supply : power source of AVDD
+
+Example:
+
+mt6358_snd {
+ compatible = "mediatek,mt6358-sound";
+ Avdd-supply = <&mt6358_vaud28_reg>;
+};
diff --git a/dts/Bindings/sound/mt8183-afe-pcm.txt b/dts/Bindings/sound/mt8183-afe-pcm.txt
new file mode 100644
index 0000000000..396ba38619
--- /dev/null
+++ b/dts/Bindings/sound/mt8183-afe-pcm.txt
@@ -0,0 +1,36 @@
+Mediatek AFE PCM controller for mt8183
+
+Required properties:
+- compatible = "mediatek,mt68183-audio";
+- reg: register location and size
+- interrupts: should contain AFE interrupt
+- power-domains: should define the power domain
+- clocks: Must contain an entry for each entry in clock-names
+- clock-names: should have these clock names:
+ "infra_sys_audio_clk",
+ "mtkaif_26m_clk",
+ "top_mux_audio",
+ "top_mux_aud_intbus",
+ "top_sys_pll3_d4",
+ "top_clk26m_clk";
+
+Example:
+
+ afe: mt8183-afe-pcm@11220000 {
+ compatible = "mediatek,mt8183-audio";
+ reg = <0 0x11220000 0 0x1000>;
+ interrupts = <GIC_SPI 161 IRQ_TYPE_LEVEL_LOW>;
+ power-domains = <&scpsys MT8183_POWER_DOMAIN_AUDIO>;
+ clocks = <&infrasys CLK_INFRA_AUDIO>,
+ <&infrasys CLK_INFRA_AUDIO_26M_BCLK>,
+ <&topckgen CLK_TOP_MUX_AUDIO>,
+ <&topckgen CLK_TOP_MUX_AUD_INTBUS>,
+ <&topckgen CLK_TOP_SYSPLL_D2_D4>,
+ <&clk26m>;
+ clock-names = "infra_sys_audio_clk",
+ "mtkaif_26m_clk",
+ "top_mux_audio",
+ "top_mux_aud_intbus",
+ "top_sys_pll_d2_d4",
+ "top_clk26m_clk";
+ };
diff --git a/dts/Bindings/sound/mtk-btcvsd-snd.txt b/dts/Bindings/sound/mtk-btcvsd-snd.txt
new file mode 100644
index 0000000000..679e44839b
--- /dev/null
+++ b/dts/Bindings/sound/mtk-btcvsd-snd.txt
@@ -0,0 +1,24 @@
+Mediatek ALSA BT SCO CVSD/MSBC Driver
+
+Required properties:
+- compatible = "mediatek,mtk-btcvsd-snd";
+- reg: register location and size of PKV and SRAM_BANK2
+- interrupts: should contain BTSCO interrupt
+- mediatek,infracfg: the phandles of INFRASYS
+- mediatek,offset: Array contains of register offset and mask
+ infra_misc_offset,
+ infra_conn_bt_cvsd_mask,
+ cvsd_mcu_read_offset,
+ cvsd_mcu_write_offset,
+ cvsd_packet_indicator_offset
+
+Example:
+
+ mtk-btcvsd-snd@18000000 {
+ compatible = "mediatek,mtk-btcvsd-snd";
+ reg=<0 0x18000000 0 0x1000>,
+ <0 0x18080000 0 0x8000>;
+ interrupts = <GIC_SPI 286 IRQ_TYPE_LEVEL_LOW>;
+ mediatek,infracfg = <&infrasys>;
+ mediatek,offset = <0xf00 0x800 0xfd0 0xfd4 0xfd8>;
+ };
diff --git a/dts/Bindings/sound/nvidia,tegra30-hda.txt b/dts/Bindings/sound/nvidia,tegra30-hda.txt
index 44d27456e8..21cd310963 100644
--- a/dts/Bindings/sound/nvidia,tegra30-hda.txt
+++ b/dts/Bindings/sound/nvidia,tegra30-hda.txt
@@ -13,6 +13,10 @@ Required properties:
See ../reset/reset.txt for details.
- reset-names : Must include the following entries: hda, hda2hdmi, hda2codec_2x
+Optional properties:
+- nvidia,model : The user-visible name of this sound complex. Since the property
+ is optional, legacy boards can use default name provided in hda driver.
+
Example:
hda@70030000 {
@@ -27,4 +31,5 @@ hda@70030000 {
<&tegra_car 128>, /* hda2hdmi */
<&tegra_car 111>; /* hda2codec_2x */
reset-names = "hda", "hda2hdmi", "hda2codec_2x";
+ nvidia,model = "jetson-tk1-hda";
};
diff --git a/dts/Bindings/sound/qcom,msm8916-wcd-analog.txt b/dts/Bindings/sound/qcom,msm8916-wcd-analog.txt
index fdcea3d12e..e7d17dda55 100644
--- a/dts/Bindings/sound/qcom,msm8916-wcd-analog.txt
+++ b/dts/Bindings/sound/qcom,msm8916-wcd-analog.txt
@@ -30,6 +30,7 @@ Required properties
- vdd-cdc-io-supply: phandle to VDD_CDC_IO regulator DT node.
- vdd-cdc-tx-rx-cx-supply: phandle to VDD_CDC_TX/RX/CX regulator DT node.
- vdd-micbias-supply: phandle of VDD_MICBIAS supply's regulator DT node.
+
Optional Properties:
- qcom,mbhc-vthreshold-low: Array of 5 threshold voltages in mV for 5 buttons
detection on headset when the mbhc is powered up
@@ -92,9 +93,9 @@ spmi_bus {
"cdc_ear_cnp_int",
"cdc_hphr_cnp_int",
"cdc_hphl_cnp_int";
- VDD-CDC-IO-supply = <&pm8916_l5>;
- VDD-CDC-TX-RX-CX-supply = <&pm8916_l5>;
- VDD-MICBIAS-supply = <&pm8916_l13>;
+ vdd-cdc-io-supply = <&pm8916_l5>;
+ vdd-cdc-tx-rx-cx-supply = <&pm8916_l5>;
+ vdd-micbias-supply = <&pm8916_l13>;
#sound-dai-cells = <1>;
};
};
diff --git a/dts/Bindings/sound/qcom,wcd9335.txt b/dts/Bindings/sound/qcom,wcd9335.txt
index 1d8d49e30a..5d6ea66a86 100644
--- a/dts/Bindings/sound/qcom,wcd9335.txt
+++ b/dts/Bindings/sound/qcom,wcd9335.txt
@@ -34,12 +34,12 @@ Required properties with SLIMbus Interface:
Definition: Interrupt names of WCD INTR1 and INTR2
Should be: "intr1", "intr2"
-- reset-gpio:
+- reset-gpios:
Usage: required
Value type: <String Array>
Definition: Reset gpio line
-- qcom,ifd:
+- slim-ifc-dev:
Usage: required
Value type: <phandle>
Definition: SLIM interface device
@@ -104,13 +104,13 @@ Required properties with SLIMbus Interface:
Value type: <u32>
Definition: Must be 1
-codec@1{
+audio-codec@1{
compatible = "slim217,1a0";
reg = <1 0>;
interrupts = <&msmgpio 54 IRQ_TYPE_LEVEL_HIGH>;
interrupt-names = "intr2"
- reset-gpio = <&msmgpio 64 0>;
- qcom,ifd = <&wc9335_ifd>;
+ reset-gpios = <&msmgpio 64 0>;
+ slim-ifc-dev = <&wc9335_ifd>;
clock-names = "mclk", "native";
clocks = <&rpmcc RPM_SMD_DIV_CLK1>,
<&rpmcc RPM_SMD_BB_CLK1>;
diff --git a/dts/Bindings/sound/rockchip,rk3328-codec.txt b/dts/Bindings/sound/rockchip,rk3328-codec.txt
new file mode 100644
index 0000000000..2469588c7c
--- /dev/null
+++ b/dts/Bindings/sound/rockchip,rk3328-codec.txt
@@ -0,0 +1,23 @@
+* Rockchip Rk3328 internal codec
+
+Required properties:
+
+- compatible: "rockchip,rk3328-codec"
+- reg: physical base address of the controller and length of memory mapped
+ region.
+- rockchip,grf: the phandle of the syscon node for GRF register.
+- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names.
+- clock-names: should be "pclk".
+- spk-depop-time-ms: speak depop time msec.
+
+Example for rk3328 internal codec:
+
+codec: codec@ff410000 {
+ compatible = "rockchip,rk3328-codec";
+ reg = <0x0 0xff410000 0x0 0x1000>;
+ rockchip,grf = <&grf>;
+ clocks = <&cru PCLK_ACODEC>;
+ clock-names = "pclk";
+ spk-depop-time-ms = 100;
+ status = "disabled";
+};
diff --git a/dts/Bindings/sound/sgtl5000.txt b/dts/Bindings/sound/sgtl5000.txt
index 9c58f72439..9d9ff51849 100644
--- a/dts/Bindings/sound/sgtl5000.txt
+++ b/dts/Bindings/sound/sgtl5000.txt
@@ -37,6 +37,15 @@ VDDIO 1.8V 2.5V 3.3V
2 = 3.33 mA 5.74 mA 8.03 mA
3 = 4.99 mA 8.61 mA 12.05 mA
+- sclk-strength: the SCLK pad strength. Possible values are:
+0, 1, 2 and 3 as per the table below:
+
+VDDIO 1.8V 2.5V 3.3V
+0 = Disable
+1 = 1.66 mA 2.87 mA 4.02 mA
+2 = 3.33 mA 5.74 mA 8.03 mA
+3 = 4.99 mA 8.61 mA 12.05 mA
+
Example:
sgtl5000: codec@a {
diff --git a/dts/Bindings/sound/simple-scu-card.txt b/dts/Bindings/sound/simple-scu-card.txt
deleted file mode 100644
index 3a2f71616c..0000000000
--- a/dts/Bindings/sound/simple-scu-card.txt
+++ /dev/null
@@ -1,94 +0,0 @@
-ASoC Simple SCU Sound Card
-
-Simple SCU Sound Card is "Simple Sound Card" + "ALSA DPCM".
-For example, you can use this driver if you want to exchange sampling rate convert,
-Mixing, etc...
-
-Required properties:
-
-- compatible : "simple-scu-audio-card"
- "renesas,rsrc-card"
-Optional properties:
-
-- simple-audio-card,name : see simple-audio-card.txt
-- simple-audio-card,cpu : see simple-audio-card.txt
-- simple-audio-card,codec : see simple-audio-card.txt
-
-Optional subnode properties:
-
-- simple-audio-card,format : see simple-audio-card.txt
-- simple-audio-card,frame-master : see simple-audio-card.txt
-- simple-audio-card,bitclock-master : see simple-audio-card.txt
-- simple-audio-card,bitclock-inversion : see simple-audio-card.txt
-- simple-audio-card,frame-inversion : see simple-audio-card.txt
-- simple-audio-card,convert-rate : platform specified sampling rate convert
-- simple-audio-card,convert-channels : platform specified converted channel size (2 - 8 ch)
-- simple-audio-card,prefix : see routing
-- simple-audio-card,widgets : Please refer to widgets.txt.
-- simple-audio-card,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.
- use audio-prefix if some components is using same sink/sources naming.
- it can be used if compatible was "renesas,rsrc-card";
-
-Required CPU/CODEC subnodes properties:
-
-- sound-dai : see simple-audio-card.txt
-
-Optional CPU/CODEC subnodes properties:
-
-- clocks / system-clock-frequency : see simple-audio-card.txt
-
-Example 1. Sampling Rate Conversion
-
-sound {
- compatible = "simple-scu-audio-card";
-
- simple-audio-card,name = "rsnd-ak4643";
- simple-audio-card,format = "left_j";
- simple-audio-card,bitclock-master = <&sndcodec>;
- simple-audio-card,frame-master = <&sndcodec>;
-
- simple-audio-card,convert-rate = <48000>;
-
- simple-audio-card,prefix = "ak4642";
- simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback",
- "DAI0 Capture", "ak4642 Capture";
-
- sndcpu: simple-audio-card,cpu {
- sound-dai = <&rcar_sound>;
- };
-
- sndcodec: simple-audio-card,codec {
- sound-dai = <&ak4643>;
- system-clock-frequency = <11289600>;
- };
-};
-
-Example 2. 2 CPU 1 Codec (Mixing)
-
-sound {
- compatible = "simple-scu-audio-card";
-
- simple-audio-card,name = "rsnd-ak4643";
- simple-audio-card,format = "left_j";
- simple-audio-card,bitclock-master = <&dpcmcpu>;
- simple-audio-card,frame-master = <&dpcmcpu>;
-
- simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback",
- "ak4642 Playback", "DAI1 Playback";
-
- dpcmcpu: cpu@0 {
- sound-dai = <&rcar_sound 0>;
- };
-
- cpu@1 {
- sound-dai = <&rcar_sound 1>;
- };
-
- codec {
- prefix = "ak4642";
- sound-dai = <&ak4643>;
- clocks = <&audio_clock>;
- };
-};
diff --git a/dts/Bindings/sound/sprd-pcm.txt b/dts/Bindings/sound/sprd-pcm.txt
new file mode 100644
index 0000000000..4b23e84b2e
--- /dev/null
+++ b/dts/Bindings/sound/sprd-pcm.txt
@@ -0,0 +1,23 @@
+* Spreadtrum DMA platfrom bindings
+
+Required properties:
+- compatible: Should be "sprd,pcm-platform".
+- dmas: Specify the 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.
+
+Example:
+
+ audio_platform:platform@0 {
+ compatible = "sprd,pcm-platform";
+ dmas = <&agcp_dma 1 1>, <&agcp_dma 2 2>,
+ <&agcp_dma 3 3>, <&agcp_dma 4 4>,
+ <&agcp_dma 5 5>, <&agcp_dma 6 6>,
+ <&agcp_dma 7 7>, <&agcp_dma 8 8>,
+ <&agcp_dma 9 9>, <&agcp_dma 10 10>;
+ dma-names = "normal_p_l", "normal_p_r",
+ "normal_c_l", "normal_c_r",
+ "voice_c", "fast_p",
+ "loop_c", "loop_p",
+ "voip_c", "voip_p";
+ };
diff --git a/dts/Bindings/sound/xlnx,audio-formatter.txt b/dts/Bindings/sound/xlnx,audio-formatter.txt
new file mode 100644
index 0000000000..cbc93c8f49
--- /dev/null
+++ b/dts/Bindings/sound/xlnx,audio-formatter.txt
@@ -0,0 +1,29 @@
+Device-Tree bindings for Xilinx PL audio formatter
+
+The IP core supports DMA, data formatting(AES<->PCM conversion)
+of audio samples.
+
+Required properties:
+ - compatible: "xlnx,audio-formatter-1.0"
+ - interrupt-names: Names specified to list of interrupts in same
+ order mentioned under "interrupts".
+ List of supported interrupt names are:
+ "irq_mm2s" : interrupt from MM2S block
+ "irq_s2mm" : interrupt from S2MM block
+ - interrupts-parent: Phandle for interrupt controller.
+ - interrupts: List of Interrupt numbers.
+ - reg: Base address and size of the IP core instance.
+ - clock-names: List of input clocks.
+ Required elements: "s_axi_lite_aclk", "aud_mclk"
+ - clocks: Input clock specifier. Refer to common clock bindings.
+
+Example:
+ audio_ss_0_audio_formatter_0: audio_formatter@80010000 {
+ compatible = "xlnx,audio-formatter-1.0";
+ interrupt-names = "irq_mm2s", "irq_s2mm";
+ interrupt-parent = <&gic>;
+ interrupts = <0 104 4>, <0 105 4>;
+ reg = <0x0 0x80010000 0x0 0x1000>;
+ clock-names = "s_axi_lite_aclk", "aud_mclk";
+ clocks = <&clk 71>, <&clk_wiz_1 0>;
+ };
diff --git a/dts/Bindings/sound/xlnx,spdif.txt b/dts/Bindings/sound/xlnx,spdif.txt
new file mode 100644
index 0000000000..15c2d64d24
--- /dev/null
+++ b/dts/Bindings/sound/xlnx,spdif.txt
@@ -0,0 +1,28 @@
+Device-Tree bindings for Xilinx SPDIF IP
+
+The IP supports playback and capture of SPDIF audio
+
+Required properties:
+ - compatible: "xlnx,spdif-2.0"
+ - clock-names: List of input clocks.
+ Required elements: "s_axi_aclk", "aud_clk_i"
+ - clocks: Input clock specifier. Refer to common clock bindings.
+ - reg: Base address and address length of the IP core instance.
+ - interrupts-parent: Phandle for interrupt controller.
+ - interrupts: List of Interrupt numbers.
+ - xlnx,spdif-mode: 0 :- receiver mode
+ 1 :- transmitter mode
+ - xlnx,aud_clk_i: input audio clock value.
+
+Example:
+ spdif_0: spdif@80010000 {
+ clock-names = "aud_clk_i", "s_axi_aclk";
+ clocks = <&misc_clk_0>, <&clk 71>;
+ compatible = "xlnx,spdif-2.0";
+ interrupt-names = "spdif_interrupt";
+ interrupt-parent = <&gic>;
+ interrupts = <0 91 4>;
+ reg = <0x0 0x80010000 0x0 0x10000>;
+ xlnx,spdif-mode = <1>;
+ xlnx,aud_clk_i = <49152913>;
+ };