Merge remote-tracking branch 'asoc/topic/core' into tmp
diff --git a/Documentation/devicetree/bindings/misc/atmel-ssc.txt b/Documentation/devicetree/bindings/misc/atmel-ssc.txt
index 38e51ad..a45ae08 100644
--- a/Documentation/devicetree/bindings/misc/atmel-ssc.txt
+++ b/Documentation/devicetree/bindings/misc/atmel-ssc.txt
@@ -7,9 +7,30 @@
- reg: Should contain SSC registers location and length
- interrupts: Should contain SSC interrupt
-Example:
+
+Required properties for devices compatible with "atmel,at91sam9g45-ssc":
+- dmas: DMA specifier, consisting of a phandle to DMA controller node,
+ the memory interface and SSC DMA channel ID (for tx and rx).
+ See Documentation/devicetree/bindings/dma/atmel-dma.txt for details.
+- dma-names: Must be "tx", "rx".
+
+Examples:
+- PDC transfer:
ssc0: ssc@fffbc000 {
compatible = "atmel,at91rm9200-ssc";
reg = <0xfffbc000 0x4000>;
interrupts = <14 4 5>;
};
+
+- DMA transfer:
+ssc0: ssc@f0010000 {
+ compatible = "atmel,at91sam9g45-ssc";
+ reg = <0xf0010000 0x4000>;
+ interrupts = <28 4 5>;
+ dmas = <&dma0 1 13>,
+ <&dma0 1 14>;
+ dma-names = "tx", "rx";
+ pinctrl-names = "default";
+ pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>;
+ status = "disabled";
+};
diff --git a/Documentation/devicetree/bindings/serial/mrvl,pxa-ssp.txt b/Documentation/devicetree/bindings/serial/mrvl,pxa-ssp.txt
new file mode 100644
index 0000000..669b814
--- /dev/null
+++ b/Documentation/devicetree/bindings/serial/mrvl,pxa-ssp.txt
@@ -0,0 +1,65 @@
+Device tree bindings for Marvell PXA SSP ports
+
+Required properties:
+
+ - compatible: Must be one of
+ mrvl,pxa25x-ssp
+ mvrl,pxa25x-nssp
+ mrvl,pxa27x-ssp
+ mrvl,pxa3xx-ssp
+ mvrl,pxa168-ssp
+ mrvl,pxa910-ssp
+ mrvl,ce4100-ssp
+ mrvl,lpss-ssp
+
+ - reg: The memory base
+ - dmas: Two dma phandles, one for rx, one for tx
+ - dma-names: Must be "rx", "tx"
+
+
+Example for PXA3xx:
+
+ ssp0: ssp@41000000 {
+ compatible = "mrvl,pxa3xx-ssp";
+ reg = <0x41000000 0x40>;
+ ssp-id = <1>;
+ interrupts = <24>;
+ clock-names = "pxa27x-ssp.0";
+ dmas = <&dma 13
+ &dma 14>;
+ dma-names = "rx", "tx";
+ };
+
+ ssp1: ssp@41700000 {
+ compatible = "mrvl,pxa3xx-ssp";
+ reg = <0x41700000 0x40>;
+ ssp-id = <2>;
+ interrupts = <16>;
+ clock-names = "pxa27x-ssp.1";
+ dmas = <&dma 15
+ &dma 16>;
+ dma-names = "rx", "tx";
+ };
+
+ ssp2: ssp@41900000 {
+ compatibl3 = "mrvl,pxa3xx-ssp";
+ reg = <0x41900000 0x40>;
+ ssp-id = <3>;
+ interrupts = <0>;
+ clock-names = "pxa27x-ssp.2";
+ dmas = <&dma 66
+ &dma 67>;
+ dma-names = "rx", "tx";
+ };
+
+ ssp3: ssp@41a00000 {
+ compatible = "mrvl,pxa3xx-ssp";
+ reg = <0x41a00000 0x40>;
+ ssp-id = <4>;
+ interrupts = <13>;
+ clock-names = "pxa27x-ssp.3";
+ dmas = <&dma 2
+ &dma 3>;
+ dma-names = "rx", "tx";
+ };
+
diff --git a/Documentation/devicetree/bindings/sound/ak4554.c b/Documentation/devicetree/bindings/sound/ak4554.c
new file mode 100644
index 0000000..934fa02
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ak4554.c
@@ -0,0 +1,11 @@
+AK4554 ADC/DAC
+
+Required properties:
+
+ - compatible : "asahi-kasei,ak4554"
+
+Example:
+
+ak4554-adc-dac {
+ compatible = "asahi-kasei,ak4554";
+};
diff --git a/Documentation/devicetree/bindings/sound/alc5632.txt b/Documentation/devicetree/bindings/sound/alc5632.txt
index 8608f74..ffd886d 100644
--- a/Documentation/devicetree/bindings/sound/alc5632.txt
+++ b/Documentation/devicetree/bindings/sound/alc5632.txt
@@ -13,6 +13,25 @@
- #gpio-cells : Should be two. The first cell is the pin number and the
second cell is used to specify optional parameters (currently unused).
+Pins on the device (for linking into audio routes):
+
+ * SPK_OUTP
+ * SPK_OUTN
+ * HP_OUT_L
+ * HP_OUT_R
+ * AUX_OUT_P
+ * AUX_OUT_N
+ * LINE_IN_L
+ * LINE_IN_R
+ * PHONE_P
+ * PHONE_N
+ * MIC1_P
+ * MIC1_N
+ * MIC2_P
+ * MIC2_N
+ * MICBIAS1
+ * DMICDAT
+
Example:
alc5632: alc5632@1e {
diff --git a/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt b/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt
new file mode 100644
index 0000000..0720857
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt
@@ -0,0 +1,35 @@
+* Atmel at91sam9x5ek wm8731 audio complex
+
+Required properties:
+ - compatible: "atmel,sam9x5-wm8731-audio"
+ - atmel,model: The user-visible name of this sound complex.
+ - atmel,ssc-controller: The phandle of the SSC controller
+ - atmel,audio-codec: The phandle of the WM8731 audio codec
+ - atmel,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.
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headphone Jack
+ * Line In Jack
+
+wm8731 pins:
+cf Documentation/devicetree/bindings/sound/wm8731.txt
+
+Example:
+sound {
+ compatible = "atmel,sam9x5-wm8731-audio";
+
+ atmel,model = "wm8731 @ AT91SAM9X5EK";
+
+ atmel,audio-routing =
+ "Headphone Jack", "RHPOUT",
+ "Headphone Jack", "LHPOUT",
+ "LLINEIN", "Line In Jack",
+ "RLINEIN", "Line In Jack";
+
+ atmel,ssc-controller = <&ssc0>;
+ atmel,audio-codec = <&wm8731>;
+};
diff --git a/Documentation/devicetree/bindings/sound/atmel-wm8904.txt b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt
new file mode 100644
index 0000000..8bbe50c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt
@@ -0,0 +1,55 @@
+Atmel ASoC driver with wm8904 audio codec complex
+
+Required properties:
+ - compatible: "atmel,asoc-wm8904"
+ - atmel,model: The user-visible name of this sound complex.
+ - atmel,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 for sources and
+ sinks are the WM8904's pins, and the jacks on the board:
+
+ WM8904 pins:
+
+ * IN1L
+ * IN1R
+ * IN2L
+ * IN2R
+ * IN3L
+ * IN3R
+ * HPOUTL
+ * HPOUTR
+ * LINEOUTL
+ * LINEOUTR
+ * MICBIAS
+
+ Board connectors:
+
+ * Headphone Jack
+ * Line In Jack
+ * Mic
+
+ - atmel,ssc-controller: The phandle of the SSC controller
+ - atmel,audio-codec: The phandle of the WM8904 audio codec
+
+Optional properties:
+ - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt
+
+Example:
+sound {
+ compatible = "atmel,asoc-wm8904";
+ pinctrl-names = "default";
+ pinctrl-0 = <&pinctrl_pck0_as_mck>;
+
+ atmel,model = "wm8904 @ AT91SAM9N12EK";
+
+ atmel,audio-routing =
+ "Headphone Jack", "HPOUTL",
+ "Headphone Jack", "HPOUTR",
+ "IN2L", "Line In Jack",
+ "IN2R", "Line In Jack",
+ "Mic", "MICBIAS",
+ "IN1L", "Mic";
+
+ atmel,ssc-controller = <&ssc0>;
+ atmel,audio-codec = <&wm8904>;
+};
diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt
new file mode 100644
index 0000000..f2ae335
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt
@@ -0,0 +1,54 @@
+Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller
+
+The Freescale S/PDIF audio block is a stereo transceiver that allows the
+processor to receive and transmit digital audio via an coaxial cable or
+a fibre cable.
+
+Required properties:
+
+ - compatible : Compatible list, must contain "fsl,imx35-spdif".
+
+ - reg : Offset and length of the register set for the device.
+
+ - interrupts : Contains the spdif interrupt.
+
+ - dmas : Generic dma devicetree binding as described in
+ Documentation/devicetree/bindings/dma/dma.txt.
+
+ - dma-names : Two dmas have to be defined, "tx" and "rx".
+
+ - clocks : Contains an entry for each entry in clock-names.
+
+ - clock-names : Includes the following entries:
+ "core" The core clock of spdif controller
+ "rxtx<0-7>" Clock source list for tx and rx clock.
+ This clock list should be identical to
+ the source list connecting to the spdif
+ clock mux in "SPDIF Transceiver Clock
+ Diagram" of SoC reference manual. It
+ can also be referred to TxClk_Source
+ bit of register SPDIF_STC.
+
+Example:
+
+spdif: spdif@02004000 {
+ compatible = "fsl,imx35-spdif";
+ reg = <0x02004000 0x4000>;
+ interrupts = <0 52 0x04>;
+ dmas = <&sdma 14 18 0>,
+ <&sdma 15 18 0>;
+ dma-names = "rx", "tx";
+
+ clocks = <&clks 197>, <&clks 3>,
+ <&clks 197>, <&clks 107>,
+ <&clks 0>, <&clks 118>,
+ <&clks 62>, <&clks 139>,
+ <&clks 0>;
+ clock-names = "core", "rxtx0",
+ "rxtx1", "rxtx2",
+ "rxtx3", "rxtx4",
+ "rxtx5", "rxtx6",
+ "rxtx7";
+
+ status = "okay";
+};
diff --git a/Documentation/devicetree/bindings/powerpc/fsl/ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt
similarity index 87%
rename from Documentation/devicetree/bindings/powerpc/fsl/ssi.txt
rename to Documentation/devicetree/bindings/sound/fsl,ssi.txt
index 5ff76c9..4303b6a 100644
--- a/Documentation/devicetree/bindings/powerpc/fsl/ssi.txt
+++ b/Documentation/devicetree/bindings/sound/fsl,ssi.txt
@@ -43,10 +43,22 @@
together. This would still allow different sample sizes,
but not different sample rates.
+Required are also ac97 link bindings if ac97 is used. See
+Documentation/devicetree/bindings/sound/soc-ac97link.txt for the necessary
+bindings.
+
Optional properties:
- codec-handle: Phandle to a 'codec' node that defines an audio
codec connected to this SSI. This node is typically
a child of an I2C or other control node.
+- fsl,fiq-stream-filter: Bool property. Disabled DMA and use FIQ instead to
+ filter the codec stream. This is necessary for some boards
+ where an incompatible codec is connected to this SSI, e.g.
+ on pca100 and pcm043.
+- dmas: Generic dma devicetree binding as described in
+ Documentation/devicetree/bindings/dma/dma.txt.
+- dma-names: Two dmas have to be defined, "tx" and "rx", if fsl,imx-fiq
+ is not defined.
Child 'codec' node required properties:
- compatible: Compatible list, contains the name of the codec
diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
new file mode 100644
index 0000000..7d13479
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
@@ -0,0 +1,34 @@
+Freescale i.MX audio complex with S/PDIF transceiver
+
+Required properties:
+
+ - compatible : "fsl,imx-audio-spdif"
+
+ - model : The user-visible name of this sound complex
+
+ - spdif-controller : The phandle of the i.MX S/PDIF controller
+
+
+Optional properties:
+
+ - spdif-out : This is a boolean property. If present, the transmitting
+ function of S/PDIF will be enabled, indicating there's a physical
+ S/PDIF out connector/jack on the board or it's connecting to some
+ other IP block, such as an HDMI encoder/display-controller.
+
+ - spdif-in : This is a boolean property. If present, the receiving
+ function of S/PDIF will be enabled, indicating there's a physical
+ S/PDIF in connector/jack on the board.
+
+* Note: At least one of these two properties should be set in the DT binding.
+
+
+Example:
+
+sound-spdif {
+ compatible = "fsl,imx-audio-spdif";
+ model = "imx-spdif";
+ spdif-controller = <&spdif>;
+ spdif-out;
+ spdif-in;
+};
diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.txt b/Documentation/devicetree/bindings/sound/imx-audmux.txt
index 215aa98..f88a00e 100644
--- a/Documentation/devicetree/bindings/sound/imx-audmux.txt
+++ b/Documentation/devicetree/bindings/sound/imx-audmux.txt
@@ -5,6 +5,15 @@
or "fsl,imx31-audmux" for the version firstly used on i.MX31.
- reg : Should contain AUDMUX registers location and length
+An initial configuration can be setup using child nodes.
+
+Required properties of optional child nodes:
+- fsl,audmux-port : Integer of the audmux port that is configured by this
+ child node.
+- fsl,port-config : List of configuration options for the specific port. For
+ imx31-audmux and above, it is a list of tuples <ptcr pdcr>. For
+ imx21-audmux it is a list of pcr values.
+
Example:
audmux@021d8000 {
diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt
new file mode 100644
index 0000000..74c9ba6
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt
@@ -0,0 +1,28 @@
+Marvell PXA SSP CPU DAI bindings
+
+Required properties:
+
+ compatible Must be "mrvl,pxa-ssp-dai"
+ port A phandle reference to a PXA ssp upstream device
+
+Example:
+
+ /* upstream device */
+
+ ssp0: ssp@41000000 {
+ compatible = "mrvl,pxa3xx-ssp";
+ reg = <0x41000000 0x40>;
+ interrupts = <24>;
+ clock-names = "pxa27x-ssp.0";
+ dmas = <&dma 13
+ &dma 14>;
+ dma-names = "rx", "tx";
+ };
+
+ /* DAI as user */
+
+ ssp_dai0: ssp_dai@0 {
+ compatible = "mrvl,pxa-ssp-dai";
+ port = <&ssp0>;
+ };
+
diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt
new file mode 100644
index 0000000..551fbb8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt
@@ -0,0 +1,15 @@
+DT bindings for ARM PXA2xx PCM platform driver
+
+This is just a dummy driver that registers the PXA ASoC platform driver.
+It does not have any resources assigned.
+
+Required properties:
+
+ - compatible 'mrvl,pxa-pcm-audio'
+
+Example:
+
+ pxa_pcm_audio: snd_soc_pxa_audio {
+ compatible = "mrvl,pxa-pcm-audio";
+ };
+
diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt
new file mode 100644
index 0000000..7e5fd37
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt
@@ -0,0 +1,29 @@
+* mvebu (Kirkwood, Dove, Armada 370) audio controller
+
+Required properties:
+
+- compatible: "marvell,mvebu-audio"
+
+- reg: physical base address of the controller and length of memory mapped
+ region.
+
+- interrupts: list of two irq numbers.
+ The first irq is used for data flow and the second one is used for errors.
+
+- clocks: one or two phandles.
+ The first one is mandatory and defines the internal clock.
+ The second one is optional and defines an external clock.
+
+- clock-names: names associated to the clocks:
+ "internal" for the internal clock
+ "extclk" for the external clock
+
+Example:
+
+i2s1: audio-controller@b4000 {
+ compatible = "marvell,mvebu-audio";
+ reg = <0xb4000 0x2210>;
+ interrupts = <21>, <22>;
+ clocks = <&gate_clk 13>;
+ clock-names = "internal";
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt
index 05ffecb..8b8903e 100644
--- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt
@@ -11,28 +11,8 @@
- nvidia,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 for sources and
- sinks are the ALC5632's pins:
-
- ALC5632 pins:
-
- * SPK_OUTP
- * SPK_OUTN
- * HP_OUT_L
- * HP_OUT_R
- * AUX_OUT_P
- * AUX_OUT_N
- * LINE_IN_L
- * LINE_IN_R
- * PHONE_P
- * PHONE_N
- * MIC1_P
- * MIC1_N
- * MIC2_P
- * MIC2_N
- * MICBIAS1
- * DMICDAT
-
- Board connectors:
+ sinks are the ALC5632's pins as documented in the binding for the device
+ and:
* Headset Stereophone
* Int Spk
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt
index d130818..dc62249 100644
--- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt
@@ -11,32 +11,12 @@
- nvidia,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 for sources and
- sinks are the RT5640's pins, and the jacks on the board:
-
- RT5640 pins:
-
- * DMIC1
- * DMIC2
- * MICBIAS1
- * IN1P
- * IN1R
- * IN2P
- * IN2R
- * HPOL
- * HPOR
- * LOUTL
- * LOUTR
- * MONOP
- * MONON
- * SPOLP
- * SPOLN
- * SPORP
- * SPORN
-
- Board connectors:
+ sinks are the RT5640's pins (as documented in its binding), and the jacks
+ on the board:
* Headphones
* Speakers
+ * Mic Jack
- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
connected to the CODEC.
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt
index d145106..aab6ce0 100644
--- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt
@@ -11,31 +11,8 @@
- nvidia,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 for sources and
- sinks are the WM8753's pins, and the jacks on the board:
-
- WM8753 pins:
-
- * LOUT1
- * LOUT2
- * ROUT1
- * ROUT2
- * MONO1
- * MONO2
- * OUT3
- * OUT4
- * LINE1
- * LINE2
- * RXP
- * RXN
- * ACIN
- * ACOP
- * MIC1N
- * MIC1
- * MIC2N
- * MIC2
- * Mic Bias
-
- Board connectors:
+ sinks are the WM8753's pins as documented in the binding for the WM8753,
+ and the jacks on the board:
* Headphone Jack
* Mic Jack
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt
index 3bf722d..4b44dfb 100644
--- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt
@@ -11,28 +11,8 @@
- nvidia,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 for sources and
- sinks are the WM8903's pins, and the jacks on the board:
-
- WM8903 pins:
-
- * IN1L
- * IN1R
- * IN2L
- * IN2R
- * IN3L
- * IN3R
- * DMICDAT
- * HPOUTL
- * HPOUTR
- * LINEOUTL
- * LINEOUTR
- * LOP
- * LON
- * ROP
- * RON
- * MICBIAS
-
- Board connectors:
+ sinks are the WM8903's pins (documented in the WM8903 binding document),
+ and the jacks on the board:
* Headphone Jack
* Int Spk
diff --git a/Documentation/devicetree/bindings/sound/pcm1792a.txt b/Documentation/devicetree/bindings/sound/pcm1792a.txt
new file mode 100644
index 0000000..970ba1e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/pcm1792a.txt
@@ -0,0 +1,18 @@
+Texas Instruments pcm1792a DT bindings
+
+This driver supports the SPI bus.
+
+Required properties:
+
+ - compatible: "ti,pcm1792a"
+
+For required properties on SPI, please consult
+Documentation/devicetree/bindings/spi/spi-bus.txt
+
+Examples:
+
+ codec_spi: 1792a@0 {
+ compatible = "ti,pcm1792a";
+ spi-max-frequency = <600000>;
+ };
+
diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt
index 005bcb2..068a114 100644
--- a/Documentation/devicetree/bindings/sound/rt5640.txt
+++ b/Documentation/devicetree/bindings/sound/rt5640.txt
@@ -18,6 +18,26 @@
- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
+Pins on the device (for linking into audio routes):
+
+ * DMIC1
+ * DMIC2
+ * MICBIAS1
+ * IN1P
+ * IN1R
+ * IN2P
+ * IN2R
+ * HPOL
+ * HPOR
+ * LOUTL
+ * LOUTR
+ * MONOP
+ * MONON
+ * SPOLP
+ * SPOLN
+ * SPORP
+ * SPORN
+
Example:
rt5640 {
diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt
index 025e66b..7386d44 100644
--- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt
+++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt
@@ -2,7 +2,15 @@
Required SoC Specific Properties:
-- compatible : "samsung,i2s-v5"
+- compatible : should be one of the following.
+ - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S.
+ - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with
+ secondary fifo, s/w reset control and internal mux for root clk src.
+ - samsung,exynos5420-i2s: for 8/16/24bit multichannel(7.1) I2S with
+ secondary fifo, s/w reset control, internal mux for root clk src and
+ TDM support. TDM (Time division multiplexing) is to allow transfer of
+ multiple channel audio data on single data line.
+
- reg: physical base address of the controller and length of memory mapped
region.
- dmas: list of DMA controller phandle and DMA request line ordered pairs.
@@ -21,13 +29,6 @@
Optional SoC Specific Properties:
-- samsung,supports-6ch: If the I2S Primary sound source has 5.1 Channel
- support, this flag is enabled.
-- samsung,supports-rstclr: This flag should be set if I2S software reset bit
- control is required. When this flag is set I2S software reset bit will be
- enabled or disabled based on need.
-- samsung,supports-secdai:If I2S block has a secondary FIFO and internal DMA,
- then this flag is enabled.
- samsung,idma-addr: Internal DMA register base address of the audio
sub system(used in secondary sound source).
- pinctrl-0: Should specify pin control groups used for this controller.
@@ -36,7 +37,7 @@
Example:
i2s0: i2s@03830000 {
- compatible = "samsung,i2s-v5";
+ compatible = "samsung,s5pv210-i2s";
reg = <0x03830000 0x100>;
dmas = <&pdma0 10
&pdma0 9
@@ -46,9 +47,6 @@
<&clock_audss EXYNOS_I2S_BUS>,
<&clock_audss EXYNOS_SCLK_I2S>;
clock-names = "iis", "i2s_opclk0", "i2s_opclk1";
- samsung,supports-6ch;
- samsung,supports-rstclr;
- samsung,supports-secdai;
samsung,idma-addr = <0x03000000>;
pinctrl-names = "default";
pinctrl-0 = <&i2s0_bus>;
diff --git a/Documentation/devicetree/bindings/sound/soc-ac97link.txt b/Documentation/devicetree/bindings/sound/soc-ac97link.txt
new file mode 100644
index 0000000..80152a8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/soc-ac97link.txt
@@ -0,0 +1,28 @@
+AC97 link bindings
+
+These bindings can be included within any other device node.
+
+Required properties:
+ - pinctrl-names: Has to contain following states to setup the correct
+ pinmuxing for the used gpios:
+ "ac97-running": AC97-link is active
+ "ac97-reset": AC97-link reset state
+ "ac97-warm-reset": AC97-link warm reset state
+ - ac97-gpios: List of gpio phandles with args in the order ac97-sync,
+ ac97-sdata, ac97-reset
+
+
+Example:
+
+ssi {
+ ...
+
+ pinctrl-names = "default", "ac97-running", "ac97-reset", "ac97-warm-reset";
+ pinctrl-0 = <&ac97link_running>;
+ pinctrl-1 = <&ac97link_running>;
+ pinctrl-2 = <&ac97link_reset>;
+ pinctrl-3 = <&ac97link_warm_reset>;
+ ac97-gpios = <&gpio3 20 0 &gpio3 22 0 &gpio3 28 0>;
+
+ ...
+};
diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt
new file mode 100644
index 0000000..4df1718
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt
@@ -0,0 +1,15 @@
+Texas Instruments PCM1681 8-channel PWM Processor
+
+Required properties:
+
+ - compatible: Should contain "ti,pcm1681".
+ - reg: The i2c address. Should contain <0x4c>.
+
+Examples:
+
+ i2c_bus {
+ pcm1681@4c {
+ compatible = "ti,pcm1681";
+ reg = <0x4c>;
+ };
+ };
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
index f47c3f5..705a6b1 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
@@ -3,7 +3,14 @@
The tlv320aic3x serial control bus communicates through I2C protocols
Required properties:
-- compatible - "string" - "ti,tlv320aic3x"
+
+- compatible - "string" - One of:
+ "ti,tlv320aic3x" - Generic TLV320AIC3x device
+ "ti,tlv320aic33" - TLV320AIC33
+ "ti,tlv320aic3007" - TLV320AIC3007
+ "ti,tlv320aic3106" - TLV320AIC3106
+
+
- reg - <int> - I2C slave address
diff --git a/Documentation/devicetree/bindings/sound/wm8731.txt b/Documentation/devicetree/bindings/sound/wm8731.txt
index 15f7004..236690e 100644
--- a/Documentation/devicetree/bindings/sound/wm8731.txt
+++ b/Documentation/devicetree/bindings/sound/wm8731.txt
@@ -16,3 +16,12 @@
compatible = "wlf,wm8731";
reg = <0x1a>;
};
+
+Available audio endpoints for an audio-routing table:
+ * LOUT: Left Channel Line Output
+ * ROUT: Right Channel Line Output
+ * LHPOUT: Left Channel Headphone Output
+ * RHPOUT: Right Channel Headphone Output
+ * LLINEIN: Left Channel Line Input
+ * RLINEIN: Right Channel Line Input
+ * MICIN: Microphone Input
diff --git a/Documentation/devicetree/bindings/sound/wm8753.txt b/Documentation/devicetree/bindings/sound/wm8753.txt
index e65277a..8eee612 100644
--- a/Documentation/devicetree/bindings/sound/wm8753.txt
+++ b/Documentation/devicetree/bindings/sound/wm8753.txt
@@ -10,9 +10,31 @@
- 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
+ * MONO1
+ * MONO2
+ * OUT3
+ * OUT4
+ * LINE1
+ * LINE2
+ * RXP
+ * RXN
+ * ACIN
+ * ACOP
+ * MIC1N
+ * MIC1
+ * MIC2N
+ * MIC2
+ * Mic Bias
+
Example:
-codec: wm8737@1a {
+codec: wm8753@1a {
compatible = "wlf,wm8753";
reg = <0x1a>;
};
diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt
index f102cbc..94ec32c 100644
--- a/Documentation/devicetree/bindings/sound/wm8903.txt
+++ b/Documentation/devicetree/bindings/sound/wm8903.txt
@@ -28,6 +28,25 @@
performed. If any entry has the value 0xffffffff, that GPIO's
configuration will not be modified.
+Pins on the device (for linking into audio routes):
+
+ * IN1L
+ * IN1R
+ * IN2L
+ * IN2R
+ * IN3L
+ * IN3R
+ * DMICDAT
+ * HPOUTL
+ * HPOUTR
+ * LINEOUTL
+ * LINEOUTR
+ * LOP
+ * LON
+ * ROP
+ * RON
+ * MICBIAS
+
Example:
codec: wm8903@1a {
diff --git a/Documentation/devicetree/bindings/sound/wm8994.txt b/Documentation/devicetree/bindings/sound/wm8994.txt
index f2f3e80..e045e90 100644
--- a/Documentation/devicetree/bindings/sound/wm8994.txt
+++ b/Documentation/devicetree/bindings/sound/wm8994.txt
@@ -32,6 +32,10 @@
The second cell is the flags, encoded as the trigger masks from
Documentation/devicetree/bindings/interrupts.txt
+ - clocks : A list of up to two phandle and clock specifier pairs
+ - clock-names : A list of clock names sorted in the same order as clocks.
+ Valid clock names are "MCLK1" and "MCLK2".
+
- wlf,gpio-cfg : A list of GPIO configuration register values. If absent,
no configuration of these registers is performed. If any value is
over 0xffff then the register will be left as default. If present 11
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 809d72b..a46ddb8 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -244,6 +244,7 @@
5stack-no-fp D965 5stack without front panel
dell-3stack Dell Dimension E520
dell-bios Fixes with Dell BIOS setup
+ dell-bios-amic Fixes with Dell BIOS setup including analog mic
volknob Fixes with volume-knob widget 0x24
auto BIOS setup (default)
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index c3c912d..42a0a39 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -454,6 +454,8 @@
- need_dac_fix (bool): limits the DACs depending on the channel count
- primary_hp (bool): probe headphone jacks as the primary outputs;
default true
+- multi_io (bool): try probing multi-I/O config (e.g. shared
+ line-in/surround, mic/clfe jacks)
- multi_cap_vol (bool): provide multiple capture volumes
- inv_dmic_split (bool): provide split internal mic volume/switch for
phase-inverted digital mics
diff --git a/MAINTAINERS b/MAINTAINERS
index 6881db5..ee738a6 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -595,6 +595,7 @@
F: sound/soc/codecs/adau*
F: sound/soc/codecs/adav*
F: sound/soc/codecs/ad1*
+F: sound/soc/codecs/ad7*
F: sound/soc/codecs/ssm*
F: sound/soc/codecs/sigmadsp.*
@@ -7675,6 +7676,17 @@
F: include/uapi/sound/
F: sound/
+SOUND - COMPRESSED AUDIO
+M: Vinod Koul <vinod.koul@intel.com>
+L: alsa-devel@alsa-project.org (moderated for non-subscribers)
+T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
+S: Supported
+F: Documentation/sound/alsa/compress_offload.txt
+F: include/sound/compress_driver.h
+F: include/uapi/sound/compress_*
+F: sound/core/compress_offload.c
+F: sound/soc/soc-compress.c
+
SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC)
M: Liam Girdwood <lgirdwood@gmail.com>
M: Mark Brown <broonie@kernel.org>
diff --git a/arch/arm/boot/dts/exynos5250.dtsi b/arch/arm/boot/dts/exynos5250.dtsi
index ef57277..376090f 100644
--- a/arch/arm/boot/dts/exynos5250.dtsi
+++ b/arch/arm/boot/dts/exynos5250.dtsi
@@ -405,7 +405,7 @@
};
i2s0: i2s@03830000 {
- compatible = "samsung,i2s-v5";
+ compatible = "samsung,s5pv210-i2s";
reg = <0x03830000 0x100>;
dmas = <&pdma0 10
&pdma0 9
@@ -415,16 +415,13 @@
<&clock_audss EXYNOS_I2S_BUS>,
<&clock_audss EXYNOS_SCLK_I2S>;
clock-names = "iis", "i2s_opclk0", "i2s_opclk1";
- samsung,supports-6ch;
- samsung,supports-rstclr;
- samsung,supports-secdai;
samsung,idma-addr = <0x03000000>;
pinctrl-names = "default";
pinctrl-0 = <&i2s0_bus>;
};
i2s1: i2s@12D60000 {
- compatible = "samsung,i2s-v5";
+ compatible = "samsung,s3c6410-i2s";
reg = <0x12D60000 0x100>;
dmas = <&pdma1 12
&pdma1 11>;
@@ -436,7 +433,7 @@
};
i2s2: i2s@12D70000 {
- compatible = "samsung,i2s-v5";
+ compatible = "samsung,s3c6410-i2s";
reg = <0x12D70000 0x100>;
dmas = <&pdma0 12
&pdma0 11>;
diff --git a/arch/arm/mach-dove/common.c b/arch/arm/mach-dove/common.c
index 00247c7..304f069 100644
--- a/arch/arm/mach-dove/common.c
+++ b/arch/arm/mach-dove/common.c
@@ -108,8 +108,8 @@
orion_clkdev_add(NULL, "sdhci-dove.1", sdio1);
orion_clkdev_add(NULL, "orion_nand", nand);
orion_clkdev_add(NULL, "cafe1000-ccic.0", camera);
- orion_clkdev_add(NULL, "kirkwood-i2s.0", i2s0);
- orion_clkdev_add(NULL, "kirkwood-i2s.1", i2s1);
+ orion_clkdev_add(NULL, "mvebu-audio.0", i2s0);
+ orion_clkdev_add(NULL, "mvebu-audio.1", i2s1);
orion_clkdev_add(NULL, "mv_crypto", crypto);
orion_clkdev_add(NULL, "dove-ac97", ac97);
orion_clkdev_add(NULL, "dove-pdma", pdma);
diff --git a/arch/arm/mach-kirkwood/common.c b/arch/arm/mach-kirkwood/common.c
index e9238b5..1663de0 100644
--- a/arch/arm/mach-kirkwood/common.c
+++ b/arch/arm/mach-kirkwood/common.c
@@ -264,7 +264,7 @@
orion_clkdev_add(NULL, MV_XOR_NAME ".1", xor1);
orion_clkdev_add("0", "pcie", pex0);
orion_clkdev_add("1", "pcie", pex1);
- orion_clkdev_add(NULL, "kirkwood-i2s", audio);
+ orion_clkdev_add(NULL, "mvebu-audio", audio);
orion_clkdev_add(NULL, MV64XXX_I2C_CTLR_NAME ".0", runit);
orion_clkdev_add(NULL, MV64XXX_I2C_CTLR_NAME ".1", runit);
@@ -560,7 +560,7 @@
/*****************************************************************************
* Audio
****************************************************************************/
-static struct resource kirkwood_i2s_resources[] = {
+static struct resource kirkwood_audio_resources[] = {
[0] = {
.start = AUDIO_PHYS_BASE,
.end = AUDIO_PHYS_BASE + SZ_16K - 1,
@@ -573,29 +573,23 @@
},
};
-static struct kirkwood_asoc_platform_data kirkwood_i2s_data = {
+static struct kirkwood_asoc_platform_data kirkwood_audio_data = {
.burst = 128,
};
-static struct platform_device kirkwood_i2s_device = {
- .name = "kirkwood-i2s",
+static struct platform_device kirkwood_audio_device = {
+ .name = "mvebu-audio",
.id = -1,
- .num_resources = ARRAY_SIZE(kirkwood_i2s_resources),
- .resource = kirkwood_i2s_resources,
+ .num_resources = ARRAY_SIZE(kirkwood_audio_resources),
+ .resource = kirkwood_audio_resources,
.dev = {
- .platform_data = &kirkwood_i2s_data,
+ .platform_data = &kirkwood_audio_data,
},
};
-static struct platform_device kirkwood_pcm_device = {
- .name = "kirkwood-pcm-audio",
- .id = -1,
-};
-
void __init kirkwood_audio_init(void)
{
- platform_device_register(&kirkwood_i2s_device);
- platform_device_register(&kirkwood_pcm_device);
+ platform_device_register(&kirkwood_audio_device);
}
/*****************************************************************************
diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c
index 8e11e96..c83f27b 100644
--- a/arch/arm/plat-pxa/ssp.c
+++ b/arch/arm/plat-pxa/ssp.c
@@ -30,6 +30,8 @@
#include <linux/platform_device.h>
#include <linux/spi/pxa2xx_spi.h>
#include <linux/io.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
#include <asm/irq.h>
#include <mach/hardware.h>
@@ -60,6 +62,30 @@
}
EXPORT_SYMBOL(pxa_ssp_request);
+struct ssp_device *pxa_ssp_request_of(const struct device_node *of_node,
+ const char *label)
+{
+ struct ssp_device *ssp = NULL;
+
+ mutex_lock(&ssp_lock);
+
+ list_for_each_entry(ssp, &ssp_list, node) {
+ if (ssp->of_node == of_node && ssp->use_count == 0) {
+ ssp->use_count++;
+ ssp->label = label;
+ break;
+ }
+ }
+
+ mutex_unlock(&ssp_lock);
+
+ if (&ssp->node == &ssp_list)
+ return NULL;
+
+ return ssp;
+}
+EXPORT_SYMBOL(pxa_ssp_request_of);
+
void pxa_ssp_free(struct ssp_device *ssp)
{
mutex_lock(&ssp_lock);
@@ -72,96 +98,126 @@
}
EXPORT_SYMBOL(pxa_ssp_free);
+#ifdef CONFIG_OF
+static const struct of_device_id pxa_ssp_of_ids[] = {
+ { .compatible = "mrvl,pxa25x-ssp", .data = (void *) PXA25x_SSP },
+ { .compatible = "mvrl,pxa25x-nssp", .data = (void *) PXA25x_NSSP },
+ { .compatible = "mrvl,pxa27x-ssp", .data = (void *) PXA27x_SSP },
+ { .compatible = "mrvl,pxa3xx-ssp", .data = (void *) PXA3xx_SSP },
+ { .compatible = "mvrl,pxa168-ssp", .data = (void *) PXA168_SSP },
+ { .compatible = "mrvl,pxa910-ssp", .data = (void *) PXA910_SSP },
+ { .compatible = "mrvl,ce4100-ssp", .data = (void *) CE4100_SSP },
+ { .compatible = "mrvl,lpss-ssp", .data = (void *) LPSS_SSP },
+ { },
+};
+MODULE_DEVICE_TABLE(of, pxa_ssp_of_ids);
+#endif
+
static int pxa_ssp_probe(struct platform_device *pdev)
{
- const struct platform_device_id *id = platform_get_device_id(pdev);
struct resource *res;
struct ssp_device *ssp;
- int ret = 0;
+ struct device *dev = &pdev->dev;
- ssp = kzalloc(sizeof(struct ssp_device), GFP_KERNEL);
- if (ssp == NULL) {
- dev_err(&pdev->dev, "failed to allocate memory");
+ ssp = devm_kzalloc(dev, sizeof(struct ssp_device), GFP_KERNEL);
+ if (ssp == NULL)
return -ENOMEM;
- }
+
ssp->pdev = pdev;
- ssp->clk = clk_get(&pdev->dev, NULL);
- if (IS_ERR(ssp->clk)) {
- ret = PTR_ERR(ssp->clk);
- goto err_free;
- }
+ ssp->clk = devm_clk_get(dev, NULL);
+ if (IS_ERR(ssp->clk))
+ return PTR_ERR(ssp->clk);
- res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
- if (res == NULL) {
- dev_err(&pdev->dev, "no SSP RX DRCMR defined\n");
- ret = -ENODEV;
- goto err_free_clk;
- }
- ssp->drcmr_rx = res->start;
+ if (dev->of_node) {
+ struct of_phandle_args dma_spec;
+ struct device_node *np = dev->of_node;
- res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
- if (res == NULL) {
- dev_err(&pdev->dev, "no SSP TX DRCMR defined\n");
- ret = -ENODEV;
- goto err_free_clk;
+ /*
+ * FIXME: we should allocate the DMA channel from this
+ * context and pass the channel down to the ssp users.
+ * For now, we lookup the rx and tx indices manually
+ */
+
+ /* rx */
+ of_parse_phandle_with_args(np, "dmas", "#dma-cells",
+ 0, &dma_spec);
+ ssp->drcmr_rx = dma_spec.args[0];
+ of_node_put(dma_spec.np);
+
+ /* tx */
+ of_parse_phandle_with_args(np, "dmas", "#dma-cells",
+ 1, &dma_spec);
+ ssp->drcmr_tx = dma_spec.args[0];
+ of_node_put(dma_spec.np);
+ } else {
+ res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (res == NULL) {
+ dev_err(dev, "no SSP RX DRCMR defined\n");
+ return -ENODEV;
+ }
+ ssp->drcmr_rx = res->start;
+
+ res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (res == NULL) {
+ dev_err(dev, "no SSP TX DRCMR defined\n");
+ return -ENODEV;
+ }
+ ssp->drcmr_tx = res->start;
}
- ssp->drcmr_tx = res->start;
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (res == NULL) {
- dev_err(&pdev->dev, "no memory resource defined\n");
- ret = -ENODEV;
- goto err_free_clk;
+ dev_err(dev, "no memory resource defined\n");
+ return -ENODEV;
}
- res = request_mem_region(res->start, resource_size(res),
- pdev->name);
+ res = devm_request_mem_region(dev, res->start, resource_size(res),
+ pdev->name);
if (res == NULL) {
- dev_err(&pdev->dev, "failed to request memory resource\n");
- ret = -EBUSY;
- goto err_free_clk;
+ dev_err(dev, "failed to request memory resource\n");
+ return -EBUSY;
}
ssp->phys_base = res->start;
- ssp->mmio_base = ioremap(res->start, resource_size(res));
+ ssp->mmio_base = devm_ioremap(dev, res->start, resource_size(res));
if (ssp->mmio_base == NULL) {
- dev_err(&pdev->dev, "failed to ioremap() registers\n");
- ret = -ENODEV;
- goto err_free_mem;
+ dev_err(dev, "failed to ioremap() registers\n");
+ return -ENODEV;
}
ssp->irq = platform_get_irq(pdev, 0);
if (ssp->irq < 0) {
- dev_err(&pdev->dev, "no IRQ resource defined\n");
- ret = -ENODEV;
- goto err_free_io;
+ dev_err(dev, "no IRQ resource defined\n");
+ return -ENODEV;
}
- /* PXA2xx/3xx SSP ports starts from 1 and the internal pdev->id
- * starts from 0, do a translation here
- */
- ssp->port_id = pdev->id + 1;
+ if (dev->of_node) {
+ const struct of_device_id *id =
+ of_match_device(of_match_ptr(pxa_ssp_of_ids), dev);
+ ssp->type = (int) id->data;
+ } else {
+ const struct platform_device_id *id =
+ platform_get_device_id(pdev);
+ ssp->type = (int) id->driver_data;
+
+ /* PXA2xx/3xx SSP ports starts from 1 and the internal pdev->id
+ * starts from 0, do a translation here
+ */
+ ssp->port_id = pdev->id + 1;
+ }
+
ssp->use_count = 0;
- ssp->type = (int)id->driver_data;
+ ssp->of_node = dev->of_node;
mutex_lock(&ssp_lock);
list_add(&ssp->node, &ssp_list);
mutex_unlock(&ssp_lock);
platform_set_drvdata(pdev, ssp);
- return 0;
-err_free_io:
- iounmap(ssp->mmio_base);
-err_free_mem:
- release_mem_region(res->start, resource_size(res));
-err_free_clk:
- clk_put(ssp->clk);
-err_free:
- kfree(ssp);
- return ret;
+ return 0;
}
static int pxa_ssp_remove(struct platform_device *pdev)
@@ -201,8 +257,9 @@
.probe = pxa_ssp_probe,
.remove = pxa_ssp_remove,
.driver = {
- .owner = THIS_MODULE,
- .name = "pxa2xx-ssp",
+ .owner = THIS_MODULE,
+ .name = "pxa2xx-ssp",
+ .of_match_table = of_match_ptr(pxa_ssp_of_ids),
},
.id_table = ssp_id_table,
};
diff --git a/arch/arm/plat-samsung/s3c-dma-ops.c b/arch/arm/plat-samsung/s3c-dma-ops.c
index 0cc40ae..98b10ba 100644
--- a/arch/arm/plat-samsung/s3c-dma-ops.c
+++ b/arch/arm/plat-samsung/s3c-dma-ops.c
@@ -82,7 +82,8 @@
static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep *param)
{
struct cb_data *data;
- int len = (param->cap == DMA_CYCLIC) ? param->period : param->len;
+ dma_addr_t pos = param->buf;
+ dma_addr_t end = param->buf + param->len;
list_for_each_entry(data, &dma_list, node)
if (data->ch == ch)
@@ -94,7 +95,15 @@
data->fp_param = param->fp_param;
}
- s3c2410_dma_enqueue(ch, (void *)data, param->buf, len);
+ if (param->cap != DMA_CYCLIC) {
+ s3c2410_dma_enqueue(ch, (void *)data, param->buf, param->len);
+ return 0;
+ }
+
+ while (pos < end) {
+ s3c2410_dma_enqueue(ch, (void *)data, pos, param->period);
+ pos += param->period;
+ }
return 0;
}
diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h
new file mode 100644
index 0000000..50b09e9
--- /dev/null
+++ b/include/dt-bindings/sound/fsl-imx-audmux.h
@@ -0,0 +1,56 @@
+#ifndef __DT_FSL_IMX_AUDMUX_H
+#define __DT_FSL_IMX_AUDMUX_H
+
+#define MX27_AUDMUX_HPCR1_SSI0 0
+#define MX27_AUDMUX_HPCR2_SSI1 1
+#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2
+#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3
+#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4
+#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5
+
+#define MX31_AUDMUX_PORT1_SSI0 0
+#define MX31_AUDMUX_PORT2_SSI1 1
+#define MX31_AUDMUX_PORT3_SSI_PINS_3 2
+#define MX31_AUDMUX_PORT4_SSI_PINS_4 3
+#define MX31_AUDMUX_PORT5_SSI_PINS_5 4
+#define MX31_AUDMUX_PORT6_SSI_PINS_6 5
+#define MX31_AUDMUX_PORT7_SSI_PINS_7 6
+
+#define MX51_AUDMUX_PORT1_SSI0 0
+#define MX51_AUDMUX_PORT2_SSI1 1
+#define MX51_AUDMUX_PORT3 2
+#define MX51_AUDMUX_PORT4 3
+#define MX51_AUDMUX_PORT5 4
+#define MX51_AUDMUX_PORT6 5
+#define MX51_AUDMUX_PORT7 6
+
+/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */
+#define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff)
+#define IMX_AUDMUX_V1_PCR_INMEN (1 << 8)
+#define IMX_AUDMUX_V1_PCR_TXRXEN (1 << 10)
+#define IMX_AUDMUX_V1_PCR_SYN (1 << 12)
+#define IMX_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13)
+#define IMX_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20)
+#define IMX_AUDMUX_V1_PCR_RCLKDIR (1 << 24)
+#define IMX_AUDMUX_V1_PCR_RFSDIR (1 << 25)
+#define IMX_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26)
+#define IMX_AUDMUX_V1_PCR_TCLKDIR (1 << 30)
+#define IMX_AUDMUX_V1_PCR_TFSDIR (1 << 31)
+
+/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */
+#define IMX_AUDMUX_V2_PTCR_TFSDIR (1 << 31)
+#define IMX_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27)
+#define IMX_AUDMUX_V2_PTCR_TCLKDIR (1 << 26)
+#define IMX_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22)
+#define IMX_AUDMUX_V2_PTCR_RFSDIR (1 << 21)
+#define IMX_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17)
+#define IMX_AUDMUX_V2_PTCR_RCLKDIR (1 << 16)
+#define IMX_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12)
+#define IMX_AUDMUX_V2_PTCR_SYN (1 << 11)
+
+#define IMX_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13)
+#define IMX_AUDMUX_V2_PDCR_TXRXEN (1 << 12)
+#define IMX_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8)
+#define IMX_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff)
+
+#endif /* __DT_FSL_IMX_AUDMUX_H */
diff --git a/include/linux/atmel-ssc.h b/include/linux/atmel-ssc.h
index deb0ae5..66a0e53 100644
--- a/include/linux/atmel-ssc.h
+++ b/include/linux/atmel-ssc.h
@@ -11,7 +11,7 @@
struct ssc_device {
struct list_head list;
- resource_size_t phybase;
+ dma_addr_t phybase;
void __iomem *regs;
struct platform_device *pdev;
struct atmel_ssc_platform_data *pdata;
diff --git a/include/linux/mfd/arizona/gpio.h b/include/linux/mfd/arizona/gpio.h
new file mode 100644
index 0000000..d2146bb
--- /dev/null
+++ b/include/linux/mfd/arizona/gpio.h
@@ -0,0 +1,96 @@
+/*
+ * GPIO configuration for Arizona devices
+ *
+ * Copyright 2013 Wolfson Microelectronics. PLC.
+ *
+ * Author: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _ARIZONA_GPIO_H
+#define _ARIZONA_GPIO_H
+
+#define ARIZONA_GP_FN_TXLRCLK 0x00
+#define ARIZONA_GP_FN_GPIO 0x01
+#define ARIZONA_GP_FN_IRQ1 0x02
+#define ARIZONA_GP_FN_IRQ2 0x03
+#define ARIZONA_GP_FN_OPCLK 0x04
+#define ARIZONA_GP_FN_FLL1_OUT 0x05
+#define ARIZONA_GP_FN_FLL2_OUT 0x06
+#define ARIZONA_GP_FN_PWM1 0x08
+#define ARIZONA_GP_FN_PWM2 0x09
+#define ARIZONA_GP_FN_SYSCLK_UNDERCLOCKED 0x0A
+#define ARIZONA_GP_FN_ASYNCCLK_UNDERCLOCKED 0x0B
+#define ARIZONA_GP_FN_FLL1_LOCK 0x0C
+#define ARIZONA_GP_FN_FLL2_LOCK 0x0D
+#define ARIZONA_GP_FN_FLL1_CLOCK_OK 0x0F
+#define ARIZONA_GP_FN_FLL2_CLOCK_OK 0x10
+#define ARIZONA_GP_FN_HEADPHONE_DET 0x12
+#define ARIZONA_GP_FN_MIC_DET 0x13
+#define ARIZONA_GP_FN_WSEQ_STATUS 0x15
+#define ARIZONA_GP_FN_CIF_ADDRESS_ERROR 0x16
+#define ARIZONA_GP_FN_ASRC1_LOCK 0x1A
+#define ARIZONA_GP_FN_ASRC2_LOCK 0x1B
+#define ARIZONA_GP_FN_ASRC_CONFIG_ERROR 0x1C
+#define ARIZONA_GP_FN_DRC1_SIGNAL_DETECT 0x1D
+#define ARIZONA_GP_FN_DRC1_ANTICLIP 0x1E
+#define ARIZONA_GP_FN_DRC1_DECAY 0x1F
+#define ARIZONA_GP_FN_DRC1_NOISE 0x20
+#define ARIZONA_GP_FN_DRC1_QUICK_RELEASE 0x21
+#define ARIZONA_GP_FN_DRC2_SIGNAL_DETECT 0x22
+#define ARIZONA_GP_FN_DRC2_ANTICLIP 0x23
+#define ARIZONA_GP_FN_DRC2_DECAY 0x24
+#define ARIZONA_GP_FN_DRC2_NOISE 0x25
+#define ARIZONA_GP_FN_DRC2_QUICK_RELEASE 0x26
+#define ARIZONA_GP_FN_MIXER_DROPPED_SAMPLE 0x27
+#define ARIZONA_GP_FN_AIF1_CONFIG_ERROR 0x28
+#define ARIZONA_GP_FN_AIF2_CONFIG_ERROR 0x29
+#define ARIZONA_GP_FN_AIF3_CONFIG_ERROR 0x2A
+#define ARIZONA_GP_FN_SPK_TEMP_SHUTDOWN 0x2B
+#define ARIZONA_GP_FN_SPK_TEMP_WARNING 0x2C
+#define ARIZONA_GP_FN_UNDERCLOCKED 0x2D
+#define ARIZONA_GP_FN_OVERCLOCKED 0x2E
+#define ARIZONA_GP_FN_DSP_IRQ1 0x35
+#define ARIZONA_GP_FN_DSP_IRQ2 0x36
+#define ARIZONA_GP_FN_ASYNC_OPCLK 0x3D
+#define ARIZONA_GP_FN_BOOT_DONE 0x44
+#define ARIZONA_GP_FN_DSP1_RAM_READY 0x45
+#define ARIZONA_GP_FN_SYSCLK_ENA_STATUS 0x4B
+#define ARIZONA_GP_FN_ASYNCCLK_ENA_STATUS 0x4C
+
+#define ARIZONA_GPN_DIR 0x8000 /* GPN_DIR */
+#define ARIZONA_GPN_DIR_MASK 0x8000 /* GPN_DIR */
+#define ARIZONA_GPN_DIR_SHIFT 15 /* GPN_DIR */
+#define ARIZONA_GPN_DIR_WIDTH 1 /* GPN_DIR */
+#define ARIZONA_GPN_PU 0x4000 /* GPN_PU */
+#define ARIZONA_GPN_PU_MASK 0x4000 /* GPN_PU */
+#define ARIZONA_GPN_PU_SHIFT 14 /* GPN_PU */
+#define ARIZONA_GPN_PU_WIDTH 1 /* GPN_PU */
+#define ARIZONA_GPN_PD 0x2000 /* GPN_PD */
+#define ARIZONA_GPN_PD_MASK 0x2000 /* GPN_PD */
+#define ARIZONA_GPN_PD_SHIFT 13 /* GPN_PD */
+#define ARIZONA_GPN_PD_WIDTH 1 /* GPN_PD */
+#define ARIZONA_GPN_LVL 0x0800 /* GPN_LVL */
+#define ARIZONA_GPN_LVL_MASK 0x0800 /* GPN_LVL */
+#define ARIZONA_GPN_LVL_SHIFT 11 /* GPN_LVL */
+#define ARIZONA_GPN_LVL_WIDTH 1 /* GPN_LVL */
+#define ARIZONA_GPN_POL 0x0400 /* GPN_POL */
+#define ARIZONA_GPN_POL_MASK 0x0400 /* GPN_POL */
+#define ARIZONA_GPN_POL_SHIFT 10 /* GPN_POL */
+#define ARIZONA_GPN_POL_WIDTH 1 /* GPN_POL */
+#define ARIZONA_GPN_OP_CFG 0x0200 /* GPN_OP_CFG */
+#define ARIZONA_GPN_OP_CFG_MASK 0x0200 /* GPN_OP_CFG */
+#define ARIZONA_GPN_OP_CFG_SHIFT 9 /* GPN_OP_CFG */
+#define ARIZONA_GPN_OP_CFG_WIDTH 1 /* GPN_OP_CFG */
+#define ARIZONA_GPN_DB 0x0100 /* GPN_DB */
+#define ARIZONA_GPN_DB_MASK 0x0100 /* GPN_DB */
+#define ARIZONA_GPN_DB_SHIFT 8 /* GPN_DB */
+#define ARIZONA_GPN_DB_WIDTH 1 /* GPN_DB */
+#define ARIZONA_GPN_FN_MASK 0x007F /* GPN_DB */
+#define ARIZONA_GPN_FN_SHIFT 0 /* GPN_DB */
+#define ARIZONA_GPN_FN_WIDTH 7 /* GPN_DB */
+
+#endif
diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h
index 8827259..9efc04d 100644
--- a/include/linux/platform_data/asoc-s3c.h
+++ b/include/linux/platform_data/asoc-s3c.h
@@ -36,6 +36,7 @@
*/
#define QUIRK_NO_MUXPSR (1 << 2)
#define QUIRK_NEED_RSTCLR (1 << 3)
+#define QUIRK_SUPPORTS_TDM (1 << 4)
/* Quirks of the I2S controller */
u32 quirks;
dma_addr_t idma_addr;
diff --git a/include/linux/platform_data/omap-abe-twl6040.h b/include/linux/platform_data/omap-abe-twl6040.h
deleted file mode 100644
index 5d298ac..0000000
--- a/include/linux/platform_data/omap-abe-twl6040.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/**
- * omap-abe-twl6040.h - ASoC machine driver OMAP4+ devices, header.
- *
- * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com
- * All rights reserved.
- *
- * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- */
-
-#ifndef _OMAP_ABE_TWL6040_H_
-#define _OMAP_ABE_TWL6040_H_
-
-/* To select if only one channel is connected in a stereo port */
-#define ABE_TWL6040_LEFT (1 << 0)
-#define ABE_TWL6040_RIGHT (1 << 1)
-
-struct omap_abe_twl6040_data {
- char *card_name;
- /* Feature flags for connected audio pins */
- u8 has_hs;
- u8 has_hf;
- bool has_ep;
- u8 has_aux;
- u8 has_vibra;
- bool has_dmic;
- bool has_hsmic;
- bool has_mainmic;
- bool has_submic;
- u8 has_afm;
- /* Other features */
- bool jack_detection; /* board can detect jack events */
- int mclk_freq; /* MCLK frequency speed for twl6040 */
-};
-
-#endif /* _OMAP_ABE_TWL6040_H_ */
diff --git a/include/linux/pxa2xx_ssp.h b/include/linux/pxa2xx_ssp.h
index 467cc63..4944420 100644
--- a/include/linux/pxa2xx_ssp.h
+++ b/include/linux/pxa2xx_ssp.h
@@ -21,6 +21,8 @@
#include <linux/list.h>
#include <linux/io.h>
+#include <linux/of.h>
+
/*
* SSP Serial Port Registers
@@ -190,6 +192,8 @@
int irq;
int drcmr_rx;
int drcmr_tx;
+
+ struct device_node *of_node;
};
/**
@@ -218,11 +222,18 @@
#ifdef CONFIG_ARCH_PXA
struct ssp_device *pxa_ssp_request(int port, const char *label);
void pxa_ssp_free(struct ssp_device *);
+struct ssp_device *pxa_ssp_request_of(const struct device_node *of_node,
+ const char *label);
#else
static inline struct ssp_device *pxa_ssp_request(int port, const char *label)
{
return NULL;
}
+static inline struct ssp_device *pxa_ssp_request_of(const struct device_node *n,
+ const char *name)
+{
+ return NULL;
+}
static inline void pxa_ssp_free(struct ssp_device *ssp) {}
#endif
diff --git a/include/sound/core.h b/include/sound/core.h
index c586617..2a14f1f 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -27,6 +27,7 @@
#include <linux/rwsem.h> /* struct rw_semaphore */
#include <linux/pm.h> /* pm_message_t */
#include <linux/stringify.h>
+#include <linux/printk.h>
/* number of supported soundcards */
#ifdef CONFIG_SND_DYNAMIC_MINORS
@@ -376,6 +377,11 @@
#define snd_BUG() WARN(1, "BUG?\n")
/**
+ * Suppress high rates of output when CONFIG_SND_DEBUG is enabled.
+ */
+#define snd_printd_ratelimit() printk_ratelimit()
+
+/**
* snd_BUG_ON - debugging check macro
* @cond: condition to evaluate
*
@@ -398,6 +404,8 @@
unlikely(__ret_warn_on); \
})
+static inline bool snd_printd_ratelimit(void) { return false; }
+
#endif /* CONFIG_SND_DEBUG */
#ifdef CONFIG_SND_DEBUG_VERBOSE
diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h
index 2fd3d25..56e818e 100644
--- a/include/sound/pxa2xx-lib.h
+++ b/include/sound/pxa2xx-lib.h
@@ -6,13 +6,6 @@
/* PCM */
-struct pxa2xx_pcm_dma_params {
- char *name; /* stream identifier */
- u32 dcmd; /* DMA descriptor dcmd field */
- volatile u32 *drcmr; /* the DMA request channel to use */
- u32 dev_addr; /* device physical address for DMA */
-};
-
extern int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params);
extern int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream);
diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h
new file mode 100644
index 0000000..d35412a
--- /dev/null
+++ b/include/sound/rcar_snd.h
@@ -0,0 +1,84 @@
+/*
+ * Renesas R-Car SRU/SCU/SSIU/SSI support
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef RCAR_SND_H
+#define RCAR_SND_H
+
+#include <linux/sh_clk.h>
+
+#define RSND_GEN1_SRU 0
+#define RSND_GEN1_ADG 1
+#define RSND_GEN1_SSI 2
+
+#define RSND_GEN2_SRU 0
+#define RSND_GEN2_ADG 1
+#define RSND_GEN2_SSIU 2
+#define RSND_GEN2_SSI 3
+
+#define RSND_BASE_MAX 4
+
+/*
+ * flags
+ *
+ * 0xAB000000
+ *
+ * A : clock sharing settings
+ * B : SSI direction
+ */
+#define RSND_SSI_CLK_PIN_SHARE (1 << 31)
+#define RSND_SSI_CLK_FROM_ADG (1 << 30) /* clock parent is master */
+#define RSND_SSI_SYNC (1 << 29) /* SSI34_sync etc */
+#define RSND_SSI_DEPENDENT (1 << 28) /* SSI needs SRU/SCU */
+
+#define RSND_SSI_PLAY (1 << 24)
+
+#define RSND_SSI_SET(_dai_id, _dma_id, _pio_irq, _flags) \
+{ .dai_id = _dai_id, .dma_id = _dma_id, .pio_irq = _pio_irq, .flags = _flags }
+#define RSND_SSI_UNUSED \
+{ .dai_id = -1, .dma_id = -1, .pio_irq = -1, .flags = 0 }
+
+struct rsnd_ssi_platform_info {
+ int dai_id;
+ int dma_id;
+ int pio_irq;
+ u32 flags;
+};
+
+/*
+ * flags
+ */
+#define RSND_SCU_USB_HPBIF (1 << 31) /* it needs RSND_SSI_DEPENDENT */
+
+struct rsnd_scu_platform_info {
+ u32 flags;
+};
+
+/*
+ * flags
+ *
+ * 0x0000000A
+ *
+ * A : generation
+ */
+#define RSND_GEN1 (1 << 0) /* fixme */
+#define RSND_GEN2 (2 << 0) /* fixme */
+
+struct rcar_snd_info {
+ u32 flags;
+ struct rsnd_ssi_platform_info *ssi_info;
+ int ssi_info_nr;
+ struct rsnd_scu_platform_info *scu_info;
+ int scu_info_nr;
+ int (*start)(int id);
+ int (*stop)(int id);
+};
+
+#endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 26e0df0..7184805 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -468,6 +468,8 @@
void snd_soc_free_ac97_codec(struct snd_soc_codec *codec);
int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops);
+int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops,
+ struct platform_device *pdev);
/*
*Controls
@@ -475,6 +477,8 @@
struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
void *data, const char *long_name,
const char *prefix);
+struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card,
+ const char *name);
int snd_soc_add_codec_controls(struct snd_soc_codec *codec,
const struct snd_kcontrol_new *controls, int num_controls);
int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h
index 1f59ea2..d956c35 100644
--- a/include/uapi/sound/hdspm.h
+++ b/include/uapi/sound/hdspm.h
@@ -111,7 +111,7 @@
enum hdspm_ltc_input_format input_format;
};
-#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_mixer_ioctl)
+#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_ltc)
/**
* The status data reflects the device's current state
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index ce431e6..5066a37 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -14,12 +14,14 @@
#include <linux/io.h>
#include <linux/module.h>
#include <linux/platform_device.h>
+#include <linux/dmaengine.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/initval.h>
#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
#include <mach/regs-ac97.h>
#include <mach/audio.h>
@@ -41,20 +43,20 @@
.reset = pxa2xx_ac97_reset,
};
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_out = {
- .name = "AC97 PCM out",
- .dev_addr = __PREG(PCDR),
- .drcmr = &DRCMR(12),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_ac97_pcm_out_req = 12;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = {
+ .addr = __PREG(PCDR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .maxburst = 32,
+ .filter_data = &pxa2xx_ac97_pcm_out_req,
};
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_in = {
- .name = "AC97 PCM in",
- .dev_addr = __PREG(PCDR),
- .drcmr = &DRCMR(11),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_ac97_pcm_in_req = 11;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = {
+ .addr = __PREG(PCDR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .maxburst = 32,
+ .filter_data = &pxa2xx_ac97_pcm_in_req,
};
static struct snd_pcm *pxa2xx_ac97_pcm;
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 823359e..a61d7a9 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -7,11 +7,13 @@
#include <linux/slab.h>
#include <linux/module.h>
#include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
#include <mach/dma.h>
@@ -43,6 +45,35 @@
size_t period = params_period_bytes(params);
pxa_dma_desc *dma_desc;
dma_addr_t dma_buff_phys, next_desc_phys;
+ u32 dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG;
+
+ /* temporary transition hack */
+ switch (rtd->params->addr_width) {
+ case DMA_SLAVE_BUSWIDTH_1_BYTE:
+ dcmd |= DCMD_WIDTH1;
+ break;
+ case DMA_SLAVE_BUSWIDTH_2_BYTES:
+ dcmd |= DCMD_WIDTH2;
+ break;
+ case DMA_SLAVE_BUSWIDTH_4_BYTES:
+ dcmd |= DCMD_WIDTH4;
+ break;
+ default:
+ /* can't happen */
+ break;
+ }
+
+ switch (rtd->params->maxburst) {
+ case 8:
+ dcmd |= DCMD_BURST8;
+ break;
+ case 16:
+ dcmd |= DCMD_BURST16;
+ break;
+ case 32:
+ dcmd |= DCMD_BURST32;
+ break;
+ }
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
runtime->dma_bytes = totsize;
@@ -55,14 +86,14 @@
dma_desc->ddadr = next_desc_phys;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
dma_desc->dsadr = dma_buff_phys;
- dma_desc->dtadr = rtd->params->dev_addr;
+ dma_desc->dtadr = rtd->params->addr;
} else {
- dma_desc->dsadr = rtd->params->dev_addr;
+ dma_desc->dsadr = rtd->params->addr;
dma_desc->dtadr = dma_buff_phys;
}
if (period > totsize)
period = totsize;
- dma_desc->dcmd = rtd->params->dcmd | period | DCMD_ENDIRQEN;
+ dma_desc->dcmd = dcmd | period | DCMD_ENDIRQEN;
dma_desc++;
dma_buff_phys += period;
} while (totsize -= period);
@@ -76,8 +107,10 @@
{
struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
- if (rtd && rtd->params && rtd->params->drcmr)
- *rtd->params->drcmr = 0;
+ if (rtd && rtd->params && rtd->params->filter_data) {
+ unsigned long req = *(unsigned long *) rtd->params->filter_data;
+ DRCMR(req) = 0;
+ }
snd_pcm_set_runtime_buffer(substream, NULL);
return 0;
@@ -136,6 +169,7 @@
int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
{
struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+ unsigned long req;
if (!prtd || !prtd->params)
return 0;
@@ -146,7 +180,8 @@
DCSR(prtd->dma_ch) &= ~DCSR_RUN;
DCSR(prtd->dma_ch) = 0;
DCMD(prtd->dma_ch) = 0;
- *prtd->params->drcmr = prtd->dma_ch | DRCMR_MAPVLD;
+ req = *(unsigned long *) prtd->params->filter_data;
+ DRCMR(req) = prtd->dma_ch | DRCMR_MAPVLD;
return 0;
}
@@ -155,7 +190,6 @@
void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
{
struct snd_pcm_substream *substream = dev_id;
- struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
int dcsr;
dcsr = DCSR(dma_ch);
@@ -164,8 +198,8 @@
if (dcsr & DCSR_ENDINTR) {
snd_pcm_period_elapsed(substream);
} else {
- printk(KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n",
- rtd->params->name, dma_ch, dcsr);
+ printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n",
+ dma_ch, dcsr);
snd_pcm_stream_lock(substream);
snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
snd_pcm_stream_unlock(substream);
diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c
index 26422a3..69a2455 100644
--- a/sound/arm/pxa2xx-pcm.c
+++ b/sound/arm/pxa2xx-pcm.c
@@ -11,8 +11,11 @@
*/
#include <linux/module.h>
+#include <linux/dmaengine.h>
+
#include <sound/core.h>
#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
#include "pxa2xx-pcm.h"
@@ -40,7 +43,7 @@
rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
client->playback_params : client->capture_params;
- ret = pxa_request_dma(rtd->params->name, DMA_PRIO_LOW,
+ ret = pxa_request_dma("dma", DMA_PRIO_LOW,
pxa2xx_pcm_dma_irq, substream);
if (ret < 0)
goto err2;
diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h
index 65f86b5..2a8fc08 100644
--- a/sound/arm/pxa2xx-pcm.h
+++ b/sound/arm/pxa2xx-pcm.h
@@ -13,14 +13,14 @@
struct pxa2xx_runtime_data {
int dma_ch;
- struct pxa2xx_pcm_dma_params *params;
+ struct snd_dmaengine_dai_dma_data *params;
pxa_dma_desc *dma_desc_array;
dma_addr_t dma_desc_array_phys;
};
struct pxa2xx_pcm_client {
- struct pxa2xx_pcm_dma_params *playback_params;
- struct pxa2xx_pcm_dma_params *capture_params;
+ struct snd_dmaengine_dai_dma_data *playback_params;
+ struct snd_dmaengine_dai_dma_data *capture_params;
int (*startup)(struct snd_pcm_substream *);
void (*shutdown)(struct snd_pcm_substream *);
int (*prepare)(struct snd_pcm_substream *);
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index c0c2f57..313f22e 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -6,6 +6,9 @@
tristate
select SND_TIMER
+config SND_DMAENGINE_PCM
+ tristate
+
config SND_HWDEP
tristate
diff --git a/sound/core/Makefile b/sound/core/Makefile
index 43d4117..5e890cf 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -13,6 +13,8 @@
snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \
pcm_memory.o
+snd-pcm-dmaengine-objs := pcm_dmaengine.o
+
snd-page-alloc-y := memalloc.o
snd-page-alloc-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o
@@ -30,6 +32,7 @@
obj-$(CONFIG_SND_HRTIMER) += snd-hrtimer.o
obj-$(CONFIG_SND_RTCTIMER) += snd-rtctimer.o
obj-$(CONFIG_SND_PCM) += snd-pcm.o snd-page-alloc.o
+obj-$(CONFIG_SND_DMAENGINE_PCM) += snd-pcm-dmaengine.o
obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o
obj-$(CONFIG_SND_OSSEMUL) += oss/
diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/core/pcm_dmaengine.c
similarity index 100%
rename from sound/soc/soc-dmaengine-pcm.c
rename to sound/core/pcm_dmaengine.c
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 82bb029..6e03b46 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -184,7 +184,7 @@
do { \
if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \
xrun_log_show(substream); \
- if (printk_ratelimit()) { \
+ if (snd_printd_ratelimit()) { \
snd_printd("PCM: " fmt, ##args); \
} \
dump_stack_on_xrun(substream); \
@@ -342,7 +342,7 @@
return -EPIPE;
}
if (pos >= runtime->buffer_size) {
- if (printk_ratelimit()) {
+ if (snd_printd_ratelimit()) {
char name[16];
snd_pcm_debug_name(substream, name, sizeof(name));
xrun_log_show(substream);
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 11048cc..915b4d7 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -1022,7 +1022,7 @@
if (i >= ARRAY_SIZE(fields))
continue;
snd_info_get_str(item, ptr, sizeof(item));
- if (strict_strtoull(item, 0, &val))
+ if (kstrtoull(item, 0, &val))
continue;
if (fields[i].size == sizeof(int))
*get_dummy_int_ptr(dummy, fields[i].offset) = val;
diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c
index 2c63865..fe9e6e2 100644
--- a/sound/firewire/speakers.c
+++ b/sound/firewire/speakers.c
@@ -49,7 +49,6 @@
struct snd_card *card;
struct fw_unit *unit;
const struct device_info *device_info;
- struct snd_pcm_substream *pcm;
struct mutex mutex;
struct cmp_connection connection;
struct amdtp_out_stream stream;
@@ -363,8 +362,7 @@
return err;
pcm->private_data = fwspk;
strcpy(pcm->name, fwspk->device_info->short_name);
- fwspk->pcm = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
- fwspk->pcm->ops = &ops;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops);
return 0;
}
diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c
index 9942691..afef0d7 100644
--- a/sound/isa/gus/interwave.c
+++ b/sound/isa/gus/interwave.c
@@ -443,8 +443,7 @@
for (i = 0; i < 8; ++i)
iwave[i] = snd_gf1_peek(gus, bank_pos + i);
#ifdef CONFIG_SND_DEBUG_ROM
- printk(KERN_DEBUG "ROM at 0x%06x = %*phC\n", bank_pos,
- 8, iwave);
+ printk(KERN_DEBUG "ROM at 0x%06x = %8phC\n", bank_pos, iwave);
#endif
if (strncmp(iwave, "INTRWAVE", 8))
continue; /* first check */
diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c
index a59c888..461d94c 100644
--- a/sound/oss/dmabuf.c
+++ b/sound/oss/dmabuf.c
@@ -557,7 +557,6 @@
unsigned long flags;
int err = 0, n = 0;
struct dma_buffparms *dmap = adev->dmap_in;
- int go;
if (!(adev->open_mode & OPEN_READ))
return -EIO;
@@ -584,7 +583,7 @@
spin_unlock_irqrestore(&dmap->lock,flags);
return -EAGAIN;
}
- if ((go = adev->go))
+ if (adev->go)
timeout = dmabuf_timeout(dmap);
spin_unlock_irqrestore(&dmap->lock,flags);
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 59c5e9c..8de66cc 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -152,14 +152,9 @@
This module is automatically loaded at probing.
config SND_HDA_I915
- bool "Build Display HD-audio controller/codec power well support for i915 cards"
+ bool
+ default y
depends on DRM_I915
- help
- Say Y here to include full HDMI and DisplayPort HD-audio controller/codec
- power-well support for Intel Haswell graphics cards based on the i915 driver.
-
- Note that this option must be enabled for Intel Haswell C+ stepping machines, otherwise
- the GPU audio controller/codecs will not be initialized or damaged when exit from S3 mode.
config SND_HDA_CODEC_CIRRUS
bool "Build Cirrus Logic codec support"
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8a005f0..5b6c4e3 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -666,6 +666,64 @@
}
EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
+
+/* return DEVLIST_LEN parameter of the given widget */
+static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int wcaps = get_wcaps(codec, nid);
+ unsigned int parm;
+
+ if (!codec->dp_mst || !(wcaps & AC_WCAP_DIGITAL) ||
+ get_wcaps_type(wcaps) != AC_WID_PIN)
+ return 0;
+
+ parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN);
+ if (parm == -1 && codec->bus->rirb_error)
+ parm = 0;
+ return parm & AC_DEV_LIST_LEN_MASK;
+}
+
+/**
+ * snd_hda_get_devices - copy device list without cache
+ * @codec: the HDA codec
+ * @nid: NID of the pin to parse
+ * @dev_list: device list array
+ * @max_devices: max. number of devices to store
+ *
+ * Copy the device list. This info is dynamic and so not cached.
+ * Currently called only from hda_proc.c, so not exported.
+ */
+int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
+ u8 *dev_list, int max_devices)
+{
+ unsigned int parm;
+ int i, dev_len, devices;
+
+ parm = get_num_devices(codec, nid);
+ if (!parm) /* not multi-stream capable */
+ return 0;
+
+ dev_len = parm + 1;
+ dev_len = dev_len < max_devices ? dev_len : max_devices;
+
+ devices = 0;
+ while (devices < dev_len) {
+ parm = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_DEVICE_LIST, devices);
+ if (parm == -1 && codec->bus->rirb_error)
+ break;
+
+ for (i = 0; i < 8; i++) {
+ dev_list[devices] = (u8)parm;
+ parm >>= 4;
+ devices++;
+ if (devices >= dev_len)
+ break;
+ }
+ }
+ return devices;
+}
+
/**
* snd_hda_queue_unsol_event - add an unsolicited event to queue
* @bus: the BUS
@@ -1216,11 +1274,13 @@
{
struct hda_codec *codec =
container_of(work, struct hda_codec, jackpoll_work.work);
- if (!codec->jackpoll_interval)
- return;
snd_hda_jack_set_dirty_all(codec);
snd_hda_jack_poll_all(codec);
+
+ if (!codec->jackpoll_interval)
+ return;
+
queue_delayed_work(codec->bus->workq, &codec->jackpoll_work,
codec->jackpoll_interval);
}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 701c2e0..7aa9870 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -94,6 +94,8 @@
#define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32
#define AC_VERB_GET_HDMI_CP_CTRL 0x0f33
#define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34
+#define AC_VERB_GET_DEVICE_SEL 0xf35
+#define AC_VERB_GET_DEVICE_LIST 0xf36
/*
* SET verbs
@@ -133,6 +135,7 @@
#define AC_VERB_SET_HDMI_DIP_XMIT 0x732
#define AC_VERB_SET_HDMI_CP_CTRL 0x733
#define AC_VERB_SET_HDMI_CHAN_SLOT 0x734
+#define AC_VERB_SET_DEVICE_SEL 0x735
/*
* Parameter IDs
@@ -154,6 +157,7 @@
#define AC_PAR_GPIO_CAP 0x11
#define AC_PAR_AMP_OUT_CAP 0x12
#define AC_PAR_VOL_KNB_CAP 0x13
+#define AC_PAR_DEVLIST_LEN 0x15
#define AC_PAR_HDMI_LPCM_CAP 0x20
/*
@@ -251,6 +255,11 @@
#define AC_UNSOL_RES_TAG_SHIFT 26
#define AC_UNSOL_RES_SUBTAG (0x1f<<21)
#define AC_UNSOL_RES_SUBTAG_SHIFT 21
+#define AC_UNSOL_RES_DE (0x3f<<15) /* Device Entry
+ * (for DP1.2 MST)
+ */
+#define AC_UNSOL_RES_DE_SHIFT 15
+#define AC_UNSOL_RES_IA (1<<2) /* Inactive (for DP1.2 MST) */
#define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */
#define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */
#define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */
@@ -352,6 +361,10 @@
#define AC_LPCMCAP_44K (1<<30) /* 44.1kHz support */
#define AC_LPCMCAP_44K_MS (1<<31) /* 44.1kHz-multiplies support */
+/* Display pin's device list length */
+#define AC_DEV_LIST_LEN_MASK 0x3f
+#define AC_MAX_DEV_LIST_LEN 64
+
/*
* Control Parameters
*/
@@ -460,6 +473,11 @@
#define AC_DEFCFG_PORT_CONN (0x3<<30)
#define AC_DEFCFG_PORT_CONN_SHIFT 30
+/* Display pin's device list entry */
+#define AC_DE_PD (1<<0)
+#define AC_DE_ELDV (1<<1)
+#define AC_DE_IA (1<<2)
+
/* device device types (0x0-0xf) */
enum {
AC_JACK_LINE_OUT,
@@ -885,6 +903,7 @@
unsigned int pcm_format_first:1; /* PCM format must be set first */
unsigned int epss:1; /* supporting EPSS? */
unsigned int cached_write:1; /* write only to caches */
+ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */
#ifdef CONFIG_PM
unsigned int power_on :1; /* current (global) power-state */
unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */
@@ -972,6 +991,8 @@
const hda_nid_t *list);
int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
hda_nid_t nid, int recursive);
+int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
+ u8 *dev_list, int max_devices);
int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index e3c7ba8..ac41e9c 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -142,6 +142,9 @@
val = snd_hda_get_bool_hint(codec, "primary_hp");
if (val >= 0)
spec->no_primary_hp = !val;
+ val = snd_hda_get_bool_hint(codec, "multi_io");
+ if (val >= 0)
+ spec->no_multi_io = !val;
val = snd_hda_get_bool_hint(codec, "multi_cap_vol");
if (val >= 0)
spec->multi_cap_vol = !!val;
@@ -813,6 +816,8 @@
static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
+static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
enum {
HDA_CTL_WIDGET_VOL,
@@ -830,7 +835,13 @@
.put = hda_gen_mixer_mute_put, /* replaced */
.private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0),
},
- HDA_BIND_MUTE(NULL, 0, 0, 0),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_bind_switch_get,
+ .put = hda_gen_bind_mute_put, /* replaced */
+ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0),
+ },
};
/* add dynamic controls from template */
@@ -937,8 +948,8 @@
}
/* playback mute control with the software mute bit check */
-static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+static void sync_auto_mute_bits(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct hda_gen_spec *spec = codec->spec;
@@ -949,10 +960,22 @@
ucontrol->value.integer.value[0] &= enabled;
ucontrol->value.integer.value[1] &= enabled;
}
+}
+static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ sync_auto_mute_bits(kcontrol, ucontrol);
return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
}
+static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ sync_auto_mute_bits(kcontrol, ucontrol);
+ return snd_hda_mixer_bind_switch_put(kcontrol, ucontrol);
+}
+
/* any ctl assigned to the path with the given index? */
static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type)
{
@@ -1541,7 +1564,8 @@
cfg->speaker_pins,
spec->multiout.extra_out_nid,
spec->speaker_paths);
- if (fill_mio_first && cfg->line_outs == 1 &&
+ if (!spec->no_multi_io &&
+ fill_mio_first && cfg->line_outs == 1 &&
cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
err = fill_multi_ios(codec, cfg->line_out_pins[0], true);
if (!err)
@@ -1554,7 +1578,7 @@
spec->private_dac_nids, spec->out_paths,
spec->main_out_badness);
- if (fill_mio_first &&
+ if (!spec->no_multi_io && fill_mio_first &&
cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
/* try to fill multi-io first */
err = fill_multi_ios(codec, cfg->line_out_pins[0], false);
@@ -1582,7 +1606,8 @@
return err;
badness += err;
}
- if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
+ if (!spec->no_multi_io &&
+ cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
err = fill_multi_ios(codec, cfg->line_out_pins[0], false);
if (err < 0)
return err;
@@ -1600,7 +1625,8 @@
check_aamix_out_path(codec, spec->speaker_paths[0]);
}
- if (cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+ if (!spec->no_multi_io &&
+ cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
if (count_multiio_pins(codec, cfg->hp_pins[0]) >= 2)
spec->multi_ios = 1; /* give badness */
@@ -3724,7 +3750,8 @@
/* check each pin in the given array; returns true if any of them is plugged */
static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
{
- int i, present = 0;
+ int i;
+ bool present = false;
for (i = 0; i < num_pins; i++) {
hda_nid_t nid = pins[i];
@@ -3733,14 +3760,15 @@
/* don't detect pins retasked as inputs */
if (snd_hda_codec_get_pin_target(codec, nid) & AC_PINCTL_IN_EN)
continue;
- present |= snd_hda_jack_detect(codec, nid);
+ if (snd_hda_jack_detect_state(codec, nid) == HDA_JACK_PRESENT)
+ present = true;
}
return present;
}
/* standard HP/line-out auto-mute helper */
static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
- bool mute)
+ int *paths, bool mute)
{
struct hda_gen_spec *spec = codec->spec;
int i;
@@ -3752,10 +3780,19 @@
break;
if (spec->auto_mute_via_amp) {
+ struct nid_path *path;
+ hda_nid_t mute_nid;
+
+ path = snd_hda_get_path_from_idx(codec, paths[i]);
+ if (!path)
+ continue;
+ mute_nid = get_amp_nid_(path->ctls[NID_PATH_MUTE_CTL]);
+ if (!mute_nid)
+ continue;
if (mute)
- spec->mute_bits |= (1ULL << nid);
+ spec->mute_bits |= (1ULL << mute_nid);
else
- spec->mute_bits &= ~(1ULL << nid);
+ spec->mute_bits &= ~(1ULL << mute_nid);
set_pin_eapd(codec, nid, !mute);
continue;
}
@@ -3786,14 +3823,19 @@
void snd_hda_gen_update_outputs(struct hda_codec *codec)
{
struct hda_gen_spec *spec = codec->spec;
+ int *paths;
int on;
/* Control HP pins/amps depending on master_mute state;
* in general, HP pins/amps control should be enabled in all cases,
* but currently set only for master_mute, just to be safe
*/
+ if (spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)
+ paths = spec->out_paths;
+ else
+ paths = spec->hp_paths;
do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
- spec->autocfg.hp_pins, spec->master_mute);
+ spec->autocfg.hp_pins, paths, spec->master_mute);
if (!spec->automute_speaker)
on = 0;
@@ -3801,8 +3843,12 @@
on = spec->hp_jack_present | spec->line_jack_present;
on |= spec->master_mute;
spec->speaker_muted = on;
+ if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
+ paths = spec->out_paths;
+ else
+ paths = spec->speaker_paths;
do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins),
- spec->autocfg.speaker_pins, on);
+ spec->autocfg.speaker_pins, paths, on);
/* toggle line-out mutes if needed, too */
/* if LO is a copy of either HP or Speaker, don't need to handle it */
@@ -3815,8 +3861,9 @@
on = spec->hp_jack_present;
on |= spec->master_mute;
spec->line_out_muted = on;
+ paths = spec->out_paths;
do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
- spec->autocfg.line_out_pins, on);
+ spec->autocfg.line_out_pins, paths, on);
}
EXPORT_SYMBOL_HDA(snd_hda_gen_update_outputs);
@@ -3887,7 +3934,7 @@
/* don't detect pins retasked as outputs */
if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN)
continue;
- if (snd_hda_jack_detect(codec, pin)) {
+ if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) {
mux_select(codec, 0, spec->am_entry[i].idx);
return;
}
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index e199a85..48d4402 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -220,6 +220,7 @@
unsigned int hp_mic:1; /* Allow HP as a mic-in */
unsigned int suppress_hp_mic_detect:1; /* Don't detect HP/mic */
unsigned int no_primary_hp:1; /* Don't prefer HP pins to speaker pins */
+ unsigned int no_multi_io:1; /* Don't try multi I/O config */
unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */
unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */
unsigned int own_eapd_ctl:1; /* set EAPD by own function */
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index ce67608..fe0bda1 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -295,7 +295,7 @@
struct snd_hwdep *hwdep = dev_get_drvdata(dev); \
struct hda_codec *codec = hwdep->private_data; \
unsigned long val; \
- int err = strict_strtoul(buf, 0, &val); \
+ int err = kstrtoul(buf, 0, &val); \
if (err < 0) \
return err; \
codec->type = val; \
@@ -654,7 +654,7 @@
p = snd_hda_get_hint(codec, key);
if (!p)
ret = -ENOENT;
- else if (strict_strtoul(p, 0, &val))
+ else if (kstrtoul(p, 0, &val))
ret = -EINVAL;
else {
*valp = val;
@@ -751,7 +751,7 @@
struct hda_codec **codecp) \
{ \
unsigned long val; \
- if (!strict_strtoul(buf, 0, &val)) \
+ if (!kstrtoul(buf, 0, &val)) \
(*codecp)->name = val; \
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 8860dd52..c6c9829 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1160,7 +1160,7 @@
goto __skip;
/* clear STATESTS */
- azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+ azx_writew(chip, STATESTS, STATESTS_INT_MASK);
/* reset controller */
azx_enter_link_reset(chip);
@@ -1242,7 +1242,7 @@
}
/* clear STATESTS */
- azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+ azx_writew(chip, STATESTS, STATESTS_INT_MASK);
/* clear rirb status */
azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
@@ -1451,8 +1451,8 @@
#if 0
/* clear state status int */
- if (azx_readb(chip, STATESTS) & 0x04)
- azx_writeb(chip, STATESTS, 0x04);
+ if (azx_readw(chip, STATESTS) & 0x04)
+ azx_writew(chip, STATESTS, 0x04);
#endif
spin_unlock(&chip->reg_lock);
@@ -2971,6 +2971,10 @@
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip = card->private_data;
+ /* enable controller wake up event */
+ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
+ STATESTS_INT_MASK);
+
azx_stop_chip(chip);
azx_enter_link_reset(chip);
azx_clear_irq_pending(chip);
@@ -2983,11 +2987,31 @@
{
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip = card->private_data;
+ struct hda_bus *bus;
+ struct hda_codec *codec;
+ int status;
if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
hda_display_power(true);
+
+ /* Read STATESTS before controller reset */
+ status = azx_readw(chip, STATESTS);
+
azx_init_pci(chip);
azx_init_chip(chip, 1);
+
+ bus = chip->bus;
+ if (status && bus) {
+ list_for_each_entry(codec, &bus->codec_list, list)
+ if (status & (1 << codec->addr))
+ queue_delayed_work(codec->bus->workq,
+ &codec->jackpoll_work, codec->jackpoll_interval);
+ }
+
+ /* disable controller Wake Up event*/
+ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
+ ~STATESTS_INT_MASK);
+
return 0;
}
@@ -3831,11 +3855,13 @@
/* Request power well for Haswell HDA controller and codec */
if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) {
+#ifdef CONFIG_SND_HDA_I915
err = hda_i915_init();
if (err < 0) {
snd_printk(KERN_ERR SFX "Error request power-well from i915\n");
goto out_free;
}
+#endif
hda_display_power(true);
}
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index 3fd2973..05b3e3e 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -194,18 +194,24 @@
EXPORT_SYMBOL_HDA(snd_hda_pin_sense);
/**
- * snd_hda_jack_detect - query pin Presence Detect status
+ * snd_hda_jack_detect_state - query pin Presence Detect status
* @codec: the CODEC to sense
* @nid: the pin NID to sense
*
- * Query and return the pin's Presence Detect status.
+ * Query and return the pin's Presence Detect status, as either
+ * HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT or HDA_JACK_PHANTOM.
*/
-int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
+int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid)
{
- u32 sense = snd_hda_pin_sense(codec, nid);
- return get_jack_plug_state(sense);
+ struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid);
+ if (jack && jack->phantom_jack)
+ return HDA_JACK_PHANTOM;
+ else if (snd_hda_pin_sense(codec, nid) & AC_PINSENSE_PRESENCE)
+ return HDA_JACK_PRESENT;
+ else
+ return HDA_JACK_NOT_PRESENT;
}
-EXPORT_SYMBOL_HDA(snd_hda_jack_detect);
+EXPORT_SYMBOL_HDA(snd_hda_jack_detect_state);
/**
* snd_hda_jack_detect_enable - enable the jack-detection
@@ -247,8 +253,8 @@
int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid,
hda_nid_t gating_nid)
{
- struct hda_jack_tbl *gated = snd_hda_jack_tbl_get(codec, gated_nid);
- struct hda_jack_tbl *gating = snd_hda_jack_tbl_get(codec, gating_nid);
+ struct hda_jack_tbl *gated = snd_hda_jack_tbl_new(codec, gated_nid);
+ struct hda_jack_tbl *gating = snd_hda_jack_tbl_new(codec, gating_nid);
if (!gated || !gating)
return -EINVAL;
diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index ec12abd..379420c 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -75,7 +75,18 @@
hda_nid_t gating_nid);
u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid);
-int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
+
+/* the jack state returned from snd_hda_jack_detect_state() */
+enum {
+ HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT, HDA_JACK_PHANTOM,
+};
+
+int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid);
+
+static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
+{
+ return snd_hda_jack_detect_state(codec, nid) != HDA_JACK_NOT_PRESENT;
+}
bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid);
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 9760f00..a8cb22e 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -582,6 +582,36 @@
print_nid_array(buffer, codec, nid, &codec->nids);
}
+static void print_device_list(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid)
+{
+ int i, curr = -1;
+ u8 dev_list[AC_MAX_DEV_LIST_LEN];
+ int devlist_len;
+
+ devlist_len = snd_hda_get_devices(codec, nid, dev_list,
+ AC_MAX_DEV_LIST_LEN);
+ snd_iprintf(buffer, " Devices: %d\n", devlist_len);
+ if (devlist_len <= 0)
+ return;
+
+ curr = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_DEVICE_SEL, 0);
+
+ for (i = 0; i < devlist_len; i++) {
+ if (i == curr)
+ snd_iprintf(buffer, " *");
+ else
+ snd_iprintf(buffer, " ");
+
+ snd_iprintf(buffer,
+ "Dev %02d: PD = %d, ELDV = %d, IA = %d\n", i,
+ !!(dev_list[i] & AC_DE_PD),
+ !!(dev_list[i] & AC_DE_ELDV),
+ !!(dev_list[i] & AC_DE_IA));
+ }
+}
+
static void print_codec_info(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
@@ -751,6 +781,9 @@
(wid_caps & AC_WCAP_DELAY) >>
AC_WCAP_DELAY_SHIFT);
+ if (wid_type == AC_WID_PIN && codec->dp_mst)
+ print_device_list(buffer, codec, nid);
+
if (wid_caps & AC_WCAP_CONN_LIST)
print_conn_list(buffer, codec, nid, wid_type,
conn, conn_len);
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index d97f0d6..0cbdd87 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -32,7 +32,6 @@
#include "hda_jack.h"
#include "hda_generic.h"
-#define ENABLE_AD_STATIC_QUIRKS
struct ad198x_spec {
struct hda_gen_spec gen;
@@ -43,114 +42,8 @@
hda_nid_t eapd_nid;
unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
- const struct snd_kcontrol_new *mixers[6];
- int num_mixers;
- const struct hda_verb *init_verbs[6]; /* initialization verbs
- * don't forget NULL termination!
- */
- unsigned int num_init_verbs;
-
- /* playback */
- struct hda_multi_out multiout; /* playback set-up
- * max_channels, dacs must be set
- * dig_out_nid and hp_nid are optional
- */
- unsigned int cur_eapd;
- unsigned int need_dac_fix;
-
- /* capture */
- unsigned int num_adc_nids;
- const hda_nid_t *adc_nids;
- hda_nid_t dig_in_nid; /* digital-in NID; optional */
-
- /* capture source */
- const struct hda_input_mux *input_mux;
- const hda_nid_t *capsrc_nids;
- unsigned int cur_mux[3];
-
- /* channel model */
- const struct hda_channel_mode *channel_mode;
- int num_channel_mode;
-
- /* PCM information */
- struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */
-
- unsigned int spdif_route;
-
- unsigned int jack_present: 1;
- unsigned int inv_jack_detect: 1;/* inverted jack-detection */
- unsigned int analog_beep: 1; /* analog beep input present */
- unsigned int avoid_init_slave_vol:1;
-
-#ifdef CONFIG_PM
- struct hda_loopback_check loopback;
-#endif
- /* for virtual master */
- hda_nid_t vmaster_nid;
- const char * const *slave_vols;
- const char * const *slave_sws;
-#endif /* ENABLE_AD_STATIC_QUIRKS */
};
-#ifdef ENABLE_AD_STATIC_QUIRKS
-/*
- * input MUX handling (common part)
- */
-static int ad198x_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
-
- return snd_hda_input_mux_info(spec->input_mux, uinfo);
-}
-
-static int ad198x_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
- unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-
- ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
- return 0;
-}
-
-static int ad198x_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
- unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-
- return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
- spec->capsrc_nids[adc_idx],
- &spec->cur_mux[adc_idx]);
-}
-
-/*
- * initialization (common callbacks)
- */
-static int ad198x_init(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
- int i;
-
- for (i = 0; i < spec->num_init_verbs; i++)
- snd_hda_sequence_write(codec, spec->init_verbs[i]);
- return 0;
-}
-
-static const char * const ad_slave_pfxs[] = {
- "Front", "Surround", "Center", "LFE", "Side",
- "Headphone", "Mono", "Speaker", "IEC958",
- NULL
-};
-
-static const char * const ad1988_6stack_fp_slave_pfxs[] = {
- "Front", "Surround", "Center", "LFE", "Side", "IEC958",
- NULL
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
#ifdef CONFIG_SND_HDA_INPUT_BEEP
/* additional beep mixers; the actual parameters are overwritten at build */
@@ -160,12 +53,6 @@
{ } /* end */
};
-static const struct snd_kcontrol_new ad_beep2_mixer[] = {
- HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0, 0, HDA_OUTPUT),
- HDA_CODEC_MUTE_BEEP("Digital Beep Playback Switch", 0, 0, HDA_OUTPUT),
- { } /* end */
-};
-
#define set_beep_amp(spec, nid, idx, dir) \
((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */
#else
@@ -181,8 +68,7 @@
if (!spec->beep_amp)
return 0;
- knew = spec->analog_beep ? ad_beep2_mixer : ad_beep_mixer;
- for ( ; knew->name; knew++) {
+ for (knew = ad_beep_mixer ; knew->name; knew++) {
int err;
struct snd_kcontrol *kctl;
kctl = snd_ctl_new1(knew, codec);
@@ -199,268 +85,6 @@
#define create_beep_ctls(codec) 0
#endif
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int ad198x_build_controls(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
- struct snd_kcontrol *kctl;
- unsigned int i;
- int err;
-
- for (i = 0; i < spec->num_mixers; i++) {
- err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
- if (err < 0)
- return err;
- }
- if (spec->multiout.dig_out_nid) {
- err = snd_hda_create_spdif_out_ctls(codec,
- spec->multiout.dig_out_nid,
- spec->multiout.dig_out_nid);
- if (err < 0)
- return err;
- err = snd_hda_create_spdif_share_sw(codec,
- &spec->multiout);
- if (err < 0)
- return err;
- spec->multiout.share_spdif = 1;
- }
- if (spec->dig_in_nid) {
- err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
- if (err < 0)
- return err;
- }
-
- /* create beep controls if needed */
- err = create_beep_ctls(codec);
- if (err < 0)
- return err;
-
- /* if we have no master control, let's create it */
- if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
- unsigned int vmaster_tlv[4];
- snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
- HDA_OUTPUT, vmaster_tlv);
- err = __snd_hda_add_vmaster(codec, "Master Playback Volume",
- vmaster_tlv,
- (spec->slave_vols ?
- spec->slave_vols : ad_slave_pfxs),
- "Playback Volume",
- !spec->avoid_init_slave_vol, NULL);
- if (err < 0)
- return err;
- }
- if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
- err = snd_hda_add_vmaster(codec, "Master Playback Switch",
- NULL,
- (spec->slave_sws ?
- spec->slave_sws : ad_slave_pfxs),
- "Playback Switch");
- if (err < 0)
- return err;
- }
-
- /* assign Capture Source enums to NID */
- kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
- if (!kctl)
- kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
- for (i = 0; kctl && i < kctl->count; i++) {
- err = snd_hda_add_nid(codec, kctl, i, spec->capsrc_nids[i]);
- if (err < 0)
- return err;
- }
-
- /* assign IEC958 enums to NID */
- kctl = snd_hda_find_mixer_ctl(codec,
- SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source");
- if (kctl) {
- err = snd_hda_add_nid(codec, kctl, 0,
- spec->multiout.dig_out_nid);
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
-#ifdef CONFIG_PM
-static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
-}
-#endif
-
-/*
- * Analog playback callbacks
- */
-static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
- hinfo);
-}
-
-static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
- format, substream);
-}
-
-static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
-}
-
-/*
- * Digital out
- */
-static int ad198x_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_open(codec, &spec->multiout);
-}
-
-static int ad198x_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_close(codec, &spec->multiout);
-}
-
-static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
- format, substream);
-}
-
-static int ad198x_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
-}
-
-/*
- * Analog capture
- */
-static int ad198x_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
- stream_tag, 0, format);
- return 0;
-}
-
-static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
- return 0;
-}
-
-/*
- */
-static const struct hda_pcm_stream ad198x_pcm_analog_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 6, /* changed later */
- .nid = 0, /* fill later */
- .ops = {
- .open = ad198x_playback_pcm_open,
- .prepare = ad198x_playback_pcm_prepare,
- .cleanup = ad198x_playback_pcm_cleanup,
- },
-};
-
-static const struct hda_pcm_stream ad198x_pcm_analog_capture = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0, /* fill later */
- .ops = {
- .prepare = ad198x_capture_pcm_prepare,
- .cleanup = ad198x_capture_pcm_cleanup
- },
-};
-
-static const struct hda_pcm_stream ad198x_pcm_digital_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0, /* fill later */
- .ops = {
- .open = ad198x_dig_playback_pcm_open,
- .close = ad198x_dig_playback_pcm_close,
- .prepare = ad198x_dig_playback_pcm_prepare,
- .cleanup = ad198x_dig_playback_pcm_cleanup
- },
-};
-
-static const struct hda_pcm_stream ad198x_pcm_digital_capture = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- /* NID is set in alc_build_pcms */
-};
-
-static int ad198x_build_pcms(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
- struct hda_pcm *info = spec->pcm_rec;
-
- codec->num_pcms = 1;
- codec->pcm_info = info;
-
- info->name = "AD198x Analog";
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_analog_playback;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->multiout.max_channels;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_analog_capture;
- info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids;
- info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
-
- if (spec->multiout.dig_out_nid) {
- info++;
- codec->num_pcms++;
- codec->spdif_status_reset = 1;
- info->name = "AD198x Digital";
- info->pcm_type = HDA_PCM_TYPE_SPDIF;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
- if (spec->dig_in_nid) {
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_digital_capture;
- info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
- }
- }
-
- return 0;
-}
-#endif /* ENABLE_AD_STATIC_QUIRKS */
static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front,
hda_nid_t hp)
@@ -507,18 +131,6 @@
ad198x_power_eapd(codec);
}
-static void ad198x_free(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
-
- if (!spec)
- return;
-
- snd_hda_gen_spec_free(&spec->gen);
- kfree(spec);
- snd_hda_detach_beep_device(codec);
-}
-
#ifdef CONFIG_PM
static int ad198x_suspend(struct hda_codec *codec)
{
@@ -527,65 +139,6 @@
}
#endif
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const struct hda_codec_ops ad198x_patch_ops = {
- .build_controls = ad198x_build_controls,
- .build_pcms = ad198x_build_pcms,
- .init = ad198x_init,
- .free = ad198x_free,
-#ifdef CONFIG_PM
- .check_power_status = ad198x_check_power_status,
- .suspend = ad198x_suspend,
-#endif
- .reboot_notify = ad198x_shutup,
-};
-
-
-/*
- * EAPD control
- * the private value = nid
- */
-#define ad198x_eapd_info snd_ctl_boolean_mono_info
-
-static int ad198x_eapd_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
- if (codec->inv_eapd)
- ucontrol->value.integer.value[0] = ! spec->cur_eapd;
- else
- ucontrol->value.integer.value[0] = spec->cur_eapd;
- return 0;
-}
-
-static int ad198x_eapd_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
- hda_nid_t nid = kcontrol->private_value & 0xff;
- unsigned int eapd;
- eapd = !!ucontrol->value.integer.value[0];
- if (codec->inv_eapd)
- eapd = !eapd;
- if (eapd == spec->cur_eapd)
- return 0;
- spec->cur_eapd = eapd;
- snd_hda_codec_write_cache(codec, nid,
- 0, AC_VERB_SET_EAPD_BTLENABLE,
- eapd ? 0x02 : 0x00);
- return 1;
-}
-
-static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo);
-static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol);
-static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol);
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
/*
* Automatic parse of I/O pins from the BIOS configuration
@@ -646,537 +199,6 @@
* AD1986A specific
*/
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#define AD1986A_SPDIF_OUT 0x02
-#define AD1986A_FRONT_DAC 0x03
-#define AD1986A_SURR_DAC 0x04
-#define AD1986A_CLFE_DAC 0x05
-#define AD1986A_ADC 0x06
-
-static const hda_nid_t ad1986a_dac_nids[3] = {
- AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC
-};
-static const hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC };
-static const hda_nid_t ad1986a_capsrc_nids[1] = { 0x12 };
-
-static const struct hda_input_mux ad1986a_capture_source = {
- .num_items = 7,
- .items = {
- { "Mic", 0x0 },
- { "CD", 0x1 },
- { "Aux", 0x3 },
- { "Line", 0x4 },
- { "Mix", 0x5 },
- { "Mono", 0x6 },
- { "Phone", 0x7 },
- },
-};
-
-
-static const struct hda_bind_ctls ad1986a_bind_pcm_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static const struct hda_bind_ctls ad1986a_bind_pcm_sw = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-/*
- * mixers
- */
-static const struct snd_kcontrol_new ad1986a_mixers[] = {
- /*
- * bind volumes/mutes of 3 DACs as a single PCM control for simplicity
- */
- HDA_BIND_VOL("PCM Playback Volume", &ad1986a_bind_pcm_vol),
- HDA_BIND_SW("PCM Playback Switch", &ad1986a_bind_pcm_sw),
- HDA_CODEC_VOLUME("Front Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Surround Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x1d, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x1d, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x1d, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x1d, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x1a, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- HDA_CODEC_MUTE("Stereo Downmix Switch", 0x09, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-/* additional mixers for 3stack mode */
-static const struct snd_kcontrol_new ad1986a_3st_mixers[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = ad198x_ch_mode_info,
- .get = ad198x_ch_mode_get,
- .put = ad198x_ch_mode_put,
- },
- { } /* end */
-};
-
-/* laptop model - 2ch only */
-static const hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC };
-
-/* master controls both pins 0x1a and 0x1b */
-static const struct hda_bind_ctls ad1986a_laptop_master_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
- 0,
- },
-};
-
-static const struct hda_bind_ctls ad1986a_laptop_master_sw = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
- 0,
- },
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
- HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
- HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT),
- /*
- HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */
- HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- { } /* end */
-};
-
-/* laptop-eapd model - 2ch only */
-
-static const struct hda_input_mux ad1986a_laptop_eapd_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x4 },
- { "Mix", 0x5 },
- },
-};
-
-static const struct hda_input_mux ad1986a_automic_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x0 },
- { "Mix", 0x5 },
- },
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_master_mixers[] = {
- HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
- HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "External Amplifier",
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b,
- .info = ad198x_eapd_info,
- .get = ad198x_eapd_get,
- .put = ad198x_eapd_put,
- .private_value = 0x1b, /* port-D */
- },
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = {
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT),
- { } /* end */
-};
-
-/* re-connect the mic boost input according to the jack sensing */
-static void ad1986a_automic(struct hda_codec *codec)
-{
- unsigned int present;
- present = snd_hda_jack_detect(codec, 0x1f);
- /* 0 = 0x1f, 2 = 0x1d, 4 = mixed */
- snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_CONNECT_SEL,
- present ? 0 : 2);
-}
-
-#define AD1986A_MIC_EVENT 0x36
-
-static void ad1986a_automic_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) != AD1986A_MIC_EVENT)
- return;
- ad1986a_automic(codec);
-}
-
-static int ad1986a_automic_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1986a_automic(codec);
- return 0;
-}
-
-/* laptop-automute - 2ch only */
-
-static void ad1986a_update_hp(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
- unsigned int mute;
-
- if (spec->jack_present)
- mute = HDA_AMP_MUTE; /* mute internal speaker */
- else
- /* unmute internal speaker if necessary */
- mute = snd_hda_codec_amp_read(codec, 0x1a, 0, HDA_OUTPUT, 0);
- snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
-}
-
-static void ad1986a_hp_automute(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
-
- spec->jack_present = snd_hda_jack_detect(codec, 0x1a);
- if (spec->inv_jack_detect)
- spec->jack_present = !spec->jack_present;
- ad1986a_update_hp(codec);
-}
-
-#define AD1986A_HP_EVENT 0x37
-
-static void ad1986a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- if ((res >> 26) != AD1986A_HP_EVENT)
- return;
- ad1986a_hp_automute(codec);
-}
-
-static int ad1986a_hp_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1986a_hp_automute(codec);
- return 0;
-}
-
-/* bind hp and internal speaker mute (with plug check) */
-static int ad1986a_hp_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
- if (change)
- ad1986a_update_hp(codec);
- return change;
-}
-
-static const struct snd_kcontrol_new ad1986a_automute_master_mixers[] = {
- HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_AMP_FLAG,
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = ad1986a_hp_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
- },
- { } /* end */
-};
-
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1986a_init_verbs[] = {
- /* Front, Surround, CLFE DAC; mute as default */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* Downmix - off */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* HP, Line-Out, Surround, CLFE selectors */
- {0x0a, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Mono selector */
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Mic selector: Mic 1/2 pin */
- {0x0f, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Line-in selector: Line-in */
- {0x10, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Mic 1/2 swap */
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Record selector: mic */
- {0x12, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Mic, Phone, CD, Aux, Line-In amp; mute as default */
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* PC beep */
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* HP, Line-Out, Surround, CLFE, Mono pins; mute as default */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* HP Pin */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
- /* Front, Surround, CLFE Pins */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* Mono Pin */
- {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* Mic Pin */
- {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* Line, Aux, CD, Beep-In Pin */
- {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- { } /* end */
-};
-
-static const struct hda_verb ad1986a_ch2_init[] = {
- /* Surround out -> Line In */
- { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- /* Line-in selectors */
- { 0x10, AC_VERB_SET_CONNECT_SEL, 0x1 },
- /* CLFE -> Mic in */
- { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- /* Mic selector, mix C/LFE (backmic) and Mic (frontmic) */
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 },
- { } /* end */
-};
-
-static const struct hda_verb ad1986a_ch4_init[] = {
- /* Surround out -> Surround */
- { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
- /* CLFE -> Mic in */
- { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 },
- { } /* end */
-};
-
-static const struct hda_verb ad1986a_ch6_init[] = {
- /* Surround out -> Surround out */
- { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
- /* CLFE -> CLFE */
- { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x0 },
- { } /* end */
-};
-
-static const struct hda_channel_mode ad1986a_modes[3] = {
- { 2, ad1986a_ch2_init },
- { 4, ad1986a_ch4_init },
- { 6, ad1986a_ch6_init },
-};
-
-/* eapd initialization */
-static const struct hda_verb ad1986a_eapd_init_verbs[] = {
- {0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 },
- {}
-};
-
-static const struct hda_verb ad1986a_automic_verbs[] = {
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- /*{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},*/
- {0x0f, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x1f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_MIC_EVENT},
- {}
-};
-
-/* Ultra initialization */
-static const struct hda_verb ad1986a_ultra_init[] = {
- /* eapd initialization */
- { 0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 },
- /* CLFE -> Mic in */
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2 },
- { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- { 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
- { } /* end */
-};
-
-/* pin sensing on HP jack */
-static const struct hda_verb ad1986a_hp_init_verbs[] = {
- {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_HP_EVENT},
- {}
-};
-
-static void ad1986a_samsung_p50_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case AD1986A_HP_EVENT:
- ad1986a_hp_automute(codec);
- break;
- case AD1986A_MIC_EVENT:
- ad1986a_automic(codec);
- break;
- }
-}
-
-static int ad1986a_samsung_p50_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1986a_hp_automute(codec);
- ad1986a_automic(codec);
- return 0;
-}
-
-
-/* models */
-enum {
- AD1986A_AUTO,
- AD1986A_6STACK,
- AD1986A_3STACK,
- AD1986A_LAPTOP,
- AD1986A_LAPTOP_EAPD,
- AD1986A_LAPTOP_AUTOMUTE,
- AD1986A_ULTRA,
- AD1986A_SAMSUNG,
- AD1986A_SAMSUNG_P50,
- AD1986A_MODELS
-};
-
-static const char * const ad1986a_models[AD1986A_MODELS] = {
- [AD1986A_AUTO] = "auto",
- [AD1986A_6STACK] = "6stack",
- [AD1986A_3STACK] = "3stack",
- [AD1986A_LAPTOP] = "laptop",
- [AD1986A_LAPTOP_EAPD] = "laptop-eapd",
- [AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute",
- [AD1986A_ULTRA] = "ultra",
- [AD1986A_SAMSUNG] = "samsung",
- [AD1986A_SAMSUNG_P50] = "samsung-p50",
-};
-
-static const struct snd_pci_quirk ad1986a_cfg_tbl[] = {
- SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x11f7, "ASUS U5A", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x1213, "ASUS A6J", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x1263, "ASUS U5F", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x1297, "ASUS Z62F", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS V1j", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x1302, "ASUS W3j", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x1443, "ASUS VX1", AD1986A_LAPTOP),
- SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8J", AD1986A_3STACK),
- SND_PCI_QUIRK(0x1043, 0x817f, "ASUS P5", AD1986A_3STACK),
- SND_PCI_QUIRK(0x1043, 0x818f, "ASUS P5", AD1986A_LAPTOP),
- SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS P5", AD1986A_3STACK),
- SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS M2N", AD1986A_3STACK),
- SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK),
- SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK),
- SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba Satellite L40-10Q", AD1986A_3STACK),
- SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK),
- SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP),
- SND_PCI_QUIRK(0x144d, 0xc024, "Samsung P50", AD1986A_SAMSUNG_P50),
- SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA),
- SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_SAMSUNG),
- SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK),
- SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP),
- SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK),
- SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE),
- SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP),
- {}
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1986a_loopbacks[] = {
- { 0x13, HDA_OUTPUT, 0 }, /* Mic */
- { 0x14, HDA_OUTPUT, 0 }, /* Phone */
- { 0x15, HDA_OUTPUT, 0 }, /* CD */
- { 0x16, HDA_OUTPUT, 0 }, /* Aux */
- { 0x17, HDA_OUTPUT, 0 }, /* Line */
- { } /* end */
-};
-#endif
-
-static int is_jack_available(struct hda_codec *codec, hda_nid_t nid)
-{
- unsigned int conf = snd_hda_codec_get_pincfg(codec, nid);
- return get_defcfg_connect(conf) != AC_JACK_PORT_NONE;
-}
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
static int alloc_ad_spec(struct hda_codec *codec)
{
struct ad198x_spec *spec;
@@ -1203,6 +225,11 @@
enum {
AD1986A_FIXUP_INV_JACK_DETECT,
+ AD1986A_FIXUP_ULTRA,
+ AD1986A_FIXUP_SAMSUNG,
+ AD1986A_FIXUP_3STACK,
+ AD1986A_FIXUP_LAPTOP,
+ AD1986A_FIXUP_LAPTOP_IMIC,
};
static const struct hda_fixup ad1986a_fixups[] = {
@@ -1210,16 +237,86 @@
.type = HDA_FIXUP_FUNC,
.v.func = ad_fixup_inv_jack_detect,
},
+ [AD1986A_FIXUP_ULTRA] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1b, 0x90170110 }, /* speaker */
+ { 0x1d, 0x90a7013e }, /* int mic */
+ {}
+ },
+ },
+ [AD1986A_FIXUP_SAMSUNG] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1b, 0x90170110 }, /* speaker */
+ { 0x1d, 0x90a7013e }, /* int mic */
+ { 0x20, 0x411111f0 }, /* N/A */
+ { 0x24, 0x411111f0 }, /* N/A */
+ {}
+ },
+ },
+ [AD1986A_FIXUP_3STACK] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1a, 0x02214021 }, /* headphone */
+ { 0x1b, 0x01014011 }, /* front */
+ { 0x1c, 0x01013012 }, /* surround */
+ { 0x1d, 0x01019015 }, /* clfe */
+ { 0x1e, 0x411111f0 }, /* N/A */
+ { 0x1f, 0x02a190f0 }, /* mic */
+ { 0x20, 0x018130f0 }, /* line-in */
+ {}
+ },
+ },
+ [AD1986A_FIXUP_LAPTOP] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1a, 0x02214021 }, /* headphone */
+ { 0x1b, 0x90170110 }, /* speaker */
+ { 0x1c, 0x411111f0 }, /* N/A */
+ { 0x1d, 0x411111f0 }, /* N/A */
+ { 0x1e, 0x411111f0 }, /* N/A */
+ { 0x1f, 0x02a191f0 }, /* mic */
+ { 0x20, 0x411111f0 }, /* N/A */
+ {}
+ },
+ },
+ [AD1986A_FIXUP_LAPTOP_IMIC] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1d, 0x90a7013e }, /* int mic */
+ {}
+ },
+ .chained_before = 1,
+ .chain_id = AD1986A_FIXUP_LAPTOP,
+ },
};
static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC),
+ SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK),
+ SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK),
+ SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_FIXUP_3STACK),
+ SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_FIXUP_LAPTOP),
+ SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG),
+ SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA),
SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT),
+ SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_FIXUP_3STACK),
+ SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_FIXUP_3STACK),
+ {}
+};
+
+static const struct hda_model_fixup ad1986a_fixup_models[] = {
+ { .id = AD1986A_FIXUP_3STACK, .name = "3stack" },
+ { .id = AD1986A_FIXUP_LAPTOP, .name = "laptop" },
+ { .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-imic" },
+ { .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-eapd" }, /* alias */
{}
};
/*
*/
-static int ad1986a_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1986a(struct hda_codec *codec)
{
int err;
struct ad198x_spec *spec;
@@ -1244,7 +341,8 @@
*/
spec->gen.multiout.no_share_stream = 1;
- snd_hda_pick_fixup(codec, NULL, ad1986a_fixup_tbl, ad1986a_fixups);
+ snd_hda_pick_fixup(codec, ad1986a_fixup_models, ad1986a_fixup_tbl,
+ ad1986a_fixups);
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
err = ad198x_parse_auto_config(codec);
@@ -1258,330 +356,11 @@
return 0;
}
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1986a(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int err, board_config;
-
- board_config = snd_hda_check_board_config(codec, AD1986A_MODELS,
- ad1986a_models,
- ad1986a_cfg_tbl);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1986A_AUTO;
- }
-
- if (board_config == AD1986A_AUTO)
- return ad1986a_parse_auto_config(codec);
-
- err = alloc_ad_spec(codec);
- if (err < 0)
- return err;
- spec = codec->spec;
-
- err = snd_hda_attach_beep_device(codec, 0x19);
- if (err < 0) {
- ad198x_free(codec);
- return err;
- }
- set_beep_amp(spec, 0x18, 0, HDA_OUTPUT);
-
- spec->multiout.max_channels = 6;
- spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids);
- spec->multiout.dac_nids = ad1986a_dac_nids;
- spec->multiout.dig_out_nid = AD1986A_SPDIF_OUT;
- spec->num_adc_nids = 1;
- spec->adc_nids = ad1986a_adc_nids;
- spec->capsrc_nids = ad1986a_capsrc_nids;
- spec->input_mux = &ad1986a_capture_source;
- spec->num_mixers = 1;
- spec->mixers[0] = ad1986a_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1986a_init_verbs;
-#ifdef CONFIG_PM
- spec->loopback.amplist = ad1986a_loopbacks;
-#endif
- spec->vmaster_nid = 0x1b;
- codec->inv_eapd = 1; /* AD1986A has the inverted EAPD implementation */
-
- codec->patch_ops = ad198x_patch_ops;
-
- /* override some parameters */
- switch (board_config) {
- case AD1986A_3STACK:
- spec->num_mixers = 2;
- spec->mixers[1] = ad1986a_3st_mixers;
- spec->num_init_verbs = 2;
- spec->init_verbs[1] = ad1986a_ch2_init;
- spec->channel_mode = ad1986a_modes;
- spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes);
- spec->need_dac_fix = 1;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- break;
- case AD1986A_LAPTOP:
- spec->mixers[0] = ad1986a_laptop_mixers;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
- break;
- case AD1986A_LAPTOP_EAPD:
- spec->num_mixers = 3;
- spec->mixers[0] = ad1986a_laptop_master_mixers;
- spec->mixers[1] = ad1986a_laptop_eapd_mixers;
- spec->mixers[2] = ad1986a_laptop_intmic_mixers;
- spec->num_init_verbs = 2;
- spec->init_verbs[1] = ad1986a_eapd_init_verbs;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
- if (!is_jack_available(codec, 0x25))
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1986a_laptop_eapd_capture_source;
- break;
- case AD1986A_SAMSUNG:
- spec->num_mixers = 2;
- spec->mixers[0] = ad1986a_laptop_master_mixers;
- spec->mixers[1] = ad1986a_laptop_eapd_mixers;
- spec->num_init_verbs = 3;
- spec->init_verbs[1] = ad1986a_eapd_init_verbs;
- spec->init_verbs[2] = ad1986a_automic_verbs;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
- if (!is_jack_available(codec, 0x25))
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1986a_automic_capture_source;
- codec->patch_ops.unsol_event = ad1986a_automic_unsol_event;
- codec->patch_ops.init = ad1986a_automic_init;
- break;
- case AD1986A_SAMSUNG_P50:
- spec->num_mixers = 2;
- spec->mixers[0] = ad1986a_automute_master_mixers;
- spec->mixers[1] = ad1986a_laptop_eapd_mixers;
- spec->num_init_verbs = 4;
- spec->init_verbs[1] = ad1986a_eapd_init_verbs;
- spec->init_verbs[2] = ad1986a_automic_verbs;
- spec->init_verbs[3] = ad1986a_hp_init_verbs;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
- if (!is_jack_available(codec, 0x25))
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1986a_automic_capture_source;
- codec->patch_ops.unsol_event = ad1986a_samsung_p50_unsol_event;
- codec->patch_ops.init = ad1986a_samsung_p50_init;
- break;
- case AD1986A_LAPTOP_AUTOMUTE:
- spec->num_mixers = 3;
- spec->mixers[0] = ad1986a_automute_master_mixers;
- spec->mixers[1] = ad1986a_laptop_eapd_mixers;
- spec->mixers[2] = ad1986a_laptop_intmic_mixers;
- spec->num_init_verbs = 3;
- spec->init_verbs[1] = ad1986a_eapd_init_verbs;
- spec->init_verbs[2] = ad1986a_hp_init_verbs;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
- if (!is_jack_available(codec, 0x25))
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1986a_laptop_eapd_capture_source;
- codec->patch_ops.unsol_event = ad1986a_hp_unsol_event;
- codec->patch_ops.init = ad1986a_hp_init;
- /* Lenovo N100 seems to report the reversed bit
- * for HP jack-sensing
- */
- spec->inv_jack_detect = 1;
- break;
- case AD1986A_ULTRA:
- spec->mixers[0] = ad1986a_laptop_eapd_mixers;
- spec->num_init_verbs = 2;
- spec->init_verbs[1] = ad1986a_ultra_init;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
- spec->multiout.dig_out_nid = 0;
- break;
- }
-
- /* AD1986A has a hardware problem that it can't share a stream
- * with multiple output pins. The copy of front to surrounds
- * causes noisy or silent outputs at a certain timing, e.g.
- * changing the volume.
- * So, let's disable the shared stream.
- */
- spec->multiout.no_share_stream = 1;
-
- codec->no_trigger_sense = 1;
- codec->no_sticky_stream = 1;
-
- return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1986a ad1986a_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
/*
* AD1983 specific
*/
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#define AD1983_SPDIF_OUT 0x02
-#define AD1983_DAC 0x03
-#define AD1983_ADC 0x04
-
-static const hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC };
-static const hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC };
-static const hda_nid_t ad1983_capsrc_nids[1] = { 0x15 };
-
-static const struct hda_input_mux ad1983_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x1 },
- { "Mix", 0x2 },
- { "Mix Mono", 0x3 },
- },
-};
-
-/*
- * SPDIF playback route
- */
-static int ad1983_spdif_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- static const char * const texts[] = { "PCM", "ADC" };
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 2;
- if (uinfo->value.enumerated.item > 1)
- uinfo->value.enumerated.item = 1;
- strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
- return 0;
-}
-
-static int ad1983_spdif_route_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
-
- ucontrol->value.enumerated.item[0] = spec->spdif_route;
- return 0;
-}
-
-static int ad1983_spdif_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
-
- if (ucontrol->value.enumerated.item[0] > 1)
- return -EINVAL;
- if (spec->spdif_route != ucontrol->value.enumerated.item[0]) {
- spec->spdif_route = ucontrol->value.enumerated.item[0];
- snd_hda_codec_write_cache(codec, spec->multiout.dig_out_nid, 0,
- AC_VERB_SET_CONNECT_SEL,
- spec->spdif_route);
- return 1;
- }
- return 0;
-}
-
-static const struct snd_kcontrol_new ad1983_mixers[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- .info = ad1983_spdif_route_info,
- .get = ad1983_spdif_route_get,
- .put = ad1983_spdif_route_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb ad1983_init_verbs[] = {
- /* Front, HP, Mono; mute as default */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* Beep, PCM, Mic, Line-In: mute */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* Front, HP selectors; from Mix */
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x06, AC_VERB_SET_CONNECT_SEL, 0x01},
- /* Mono selector; from Mix */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0x03},
- /* Mic selector; Mic */
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Line-in selector: Line-in */
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Mic boost: 0dB */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* Record selector: mic */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* SPDIF route: PCM */
- {0x02, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Front Pin */
- {0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* HP Pin */
- {0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
- /* Mono Pin */
- {0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* Mic Pin */
- {0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* Line Pin */
- {0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- { } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1983_loopbacks[] = {
- { 0x12, HDA_OUTPUT, 0 }, /* Mic */
- { 0x13, HDA_OUTPUT, 0 }, /* Line */
- { } /* end */
-};
-#endif
-
-/* models */
-enum {
- AD1983_AUTO,
- AD1983_BASIC,
- AD1983_MODELS
-};
-
-static const char * const ad1983_models[AD1983_MODELS] = {
- [AD1983_AUTO] = "auto",
- [AD1983_BASIC] = "basic",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
/*
* SPDIF mux control for AD1983 auto-parser
*/
@@ -1656,7 +435,7 @@
return 0;
}
-static int ad1983_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1983(struct hda_codec *codec)
{
struct ad198x_spec *spec;
int err;
@@ -1681,432 +460,11 @@
return err;
}
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1983(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int board_config;
- int err;
-
- board_config = snd_hda_check_board_config(codec, AD1983_MODELS,
- ad1983_models, NULL);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1983_AUTO;
- }
-
- if (board_config == AD1983_AUTO)
- return ad1983_parse_auto_config(codec);
-
- err = alloc_ad_spec(codec);
- if (err < 0)
- return err;
- spec = codec->spec;
-
- err = snd_hda_attach_beep_device(codec, 0x10);
- if (err < 0) {
- ad198x_free(codec);
- return err;
- }
- set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids);
- spec->multiout.dac_nids = ad1983_dac_nids;
- spec->multiout.dig_out_nid = AD1983_SPDIF_OUT;
- spec->num_adc_nids = 1;
- spec->adc_nids = ad1983_adc_nids;
- spec->capsrc_nids = ad1983_capsrc_nids;
- spec->input_mux = &ad1983_capture_source;
- spec->num_mixers = 1;
- spec->mixers[0] = ad1983_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1983_init_verbs;
- spec->spdif_route = 0;
-#ifdef CONFIG_PM
- spec->loopback.amplist = ad1983_loopbacks;
-#endif
- spec->vmaster_nid = 0x05;
-
- codec->patch_ops = ad198x_patch_ops;
-
- codec->no_trigger_sense = 1;
- codec->no_sticky_stream = 1;
-
- return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1983 ad1983_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
/*
* AD1981 HD specific
*/
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#define AD1981_SPDIF_OUT 0x02
-#define AD1981_DAC 0x03
-#define AD1981_ADC 0x04
-
-static const hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC };
-static const hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC };
-static const hda_nid_t ad1981_capsrc_nids[1] = { 0x15 };
-
-/* 0x0c, 0x09, 0x0e, 0x0f, 0x19, 0x05, 0x18, 0x17 */
-static const struct hda_input_mux ad1981_capture_source = {
- .num_items = 7,
- .items = {
- { "Front Mic", 0x0 },
- { "Line", 0x1 },
- { "Mix", 0x2 },
- { "Mix Mono", 0x3 },
- { "CD", 0x4 },
- { "Mic", 0x6 },
- { "Aux", 0x7 },
- },
-};
-
-static const struct snd_kcontrol_new ad1981_mixers[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Aux Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Aux Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- /* identical with AD1983 */
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- .info = ad1983_spdif_route_info,
- .get = ad1983_spdif_route_get,
- .put = ad1983_spdif_route_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb ad1981_init_verbs[] = {
- /* Front, HP, Mono; mute as default */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* Beep, PCM, Front Mic, Line, Rear Mic, Aux, CD-In: mute */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* Front, HP selectors; from Mix */
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x06, AC_VERB_SET_CONNECT_SEL, 0x01},
- /* Mono selector; from Mix */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0x03},
- /* Mic Mixer; select Front Mic */
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* Mic boost: 0dB */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Record selector: Front mic */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* SPDIF route: PCM */
- {0x02, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Front Pin */
- {0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* HP Pin */
- {0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
- /* Mono Pin */
- {0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* Front & Rear Mic Pins */
- {0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* Line Pin */
- {0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- /* Digital Beep */
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Line-Out as Input: disabled */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- { } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1981_loopbacks[] = {
- { 0x12, HDA_OUTPUT, 0 }, /* Front Mic */
- { 0x13, HDA_OUTPUT, 0 }, /* Line */
- { 0x1b, HDA_OUTPUT, 0 }, /* Aux */
- { 0x1c, HDA_OUTPUT, 0 }, /* Mic */
- { 0x1d, HDA_OUTPUT, 0 }, /* CD */
- { } /* end */
-};
-#endif
-
-/*
- * Patch for HP nx6320
- *
- * nx6320 uses EAPD in the reverse way - EAPD-on means the internal
- * speaker output enabled _and_ mute-LED off.
- */
-
-#define AD1981_HP_EVENT 0x37
-#define AD1981_MIC_EVENT 0x38
-
-static const struct hda_verb ad1981_hp_init_verbs[] = {
- {0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, /* default off */
- /* pin sensing on HP and Mic jacks */
- {0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT},
- {0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT},
- {}
-};
-
-/* turn on/off EAPD (+ mute HP) as a master switch */
-static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
-
- if (! ad198x_eapd_put(kcontrol, ucontrol))
- return 0;
- /* change speaker pin appropriately */
- snd_hda_set_pin_ctl(codec, 0x05, spec->cur_eapd ? PIN_OUT : 0);
- /* toggle HP mute appropriately */
- snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- spec->cur_eapd ? 0 : HDA_AMP_MUTE);
- return 1;
-}
-
-/* bind volumes of both NID 0x05 and 0x06 */
-static const struct hda_bind_ctls ad1981_hp_bind_master_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x06, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1981_hp_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x06);
- snd_hda_codec_amp_stereo(codec, 0x05, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-/* toggle input of built-in and mic jack appropriately */
-static void ad1981_hp_automic(struct hda_codec *codec)
-{
- static const struct hda_verb mic_jack_on[] = {
- {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- {}
- };
- static const struct hda_verb mic_jack_off[] = {
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- {}
- };
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x08);
- if (present)
- snd_hda_sequence_write(codec, mic_jack_on);
- else
- snd_hda_sequence_write(codec, mic_jack_off);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1981_hp_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- res >>= 26;
- switch (res) {
- case AD1981_HP_EVENT:
- ad1981_hp_automute(codec);
- break;
- case AD1981_MIC_EVENT:
- ad1981_hp_automic(codec);
- break;
- }
-}
-
-static const struct hda_input_mux ad1981_hp_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Dock Mic", 0x1 },
- { "Mix", 0x2 },
- },
-};
-
-static const struct snd_kcontrol_new ad1981_hp_mixers[] = {
- HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x05,
- .name = "Master Playback Switch",
- .info = ad198x_eapd_info,
- .get = ad198x_eapd_get,
- .put = ad1981_hp_master_sw_put,
- .private_value = 0x05,
- },
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-#if 0
- /* FIXME: analog mic/line loopback doesn't work with my tests...
- * (although recording is OK)
- */
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
- /* FIXME: does this laptop have analog CD connection? */
- HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
-#endif
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x18, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- { } /* end */
-};
-
-/* initialize jack-sensing, too */
-static int ad1981_hp_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1981_hp_automute(codec);
- ad1981_hp_automic(codec);
- return 0;
-}
-
-/* configuration for Toshiba Laptops */
-static const struct hda_verb ad1981_toshiba_init_verbs[] = {
- {0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x01 }, /* default on */
- /* pin sensing on HP and Mic jacks */
- {0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT},
- {0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT},
- {}
-};
-
-static const struct snd_kcontrol_new ad1981_toshiba_mixers[] = {
- HDA_CODEC_VOLUME("Amp Volume", 0x1a, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Amp Switch", 0x1a, 0x0, HDA_OUTPUT),
- { }
-};
-
-/* configuration for Lenovo Thinkpad T60 */
-static const struct snd_kcontrol_new ad1981_thinkpad_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x05, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0x05, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- /* identical with AD1983 */
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- .info = ad1983_spdif_route_info,
- .get = ad1983_spdif_route_get,
- .put = ad1983_spdif_route_put,
- },
- { } /* end */
-};
-
-static const struct hda_input_mux ad1981_thinkpad_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Mix", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-/* models */
-enum {
- AD1981_AUTO,
- AD1981_BASIC,
- AD1981_HP,
- AD1981_THINKPAD,
- AD1981_TOSHIBA,
- AD1981_MODELS
-};
-
-static const char * const ad1981_models[AD1981_MODELS] = {
- [AD1981_AUTO] = "auto",
- [AD1981_HP] = "hp",
- [AD1981_THINKPAD] = "thinkpad",
- [AD1981_BASIC] = "basic",
- [AD1981_TOSHIBA] = "toshiba"
-};
-
-static const struct snd_pci_quirk ad1981_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD),
- SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD),
- /* All HP models */
- SND_PCI_QUIRK_VENDOR(0x103c, "HP nx", AD1981_HP),
- SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA),
- /* Lenovo Thinkpad T60/X60/Z6xx */
- SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1981_THINKPAD),
- /* HP nx6320 (reversed SSID, H/W bug) */
- SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP),
- {}
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
/* follow EAPD via vmaster hook */
static void ad_vmaster_eapd_hook(void *private_data, int enabled)
{
@@ -2172,7 +530,7 @@
{}
};
-static int ad1981_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1981(struct hda_codec *codec)
{
struct ad198x_spec *spec;
int err;
@@ -2205,110 +563,6 @@
return err;
}
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1981(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int err, board_config;
-
- board_config = snd_hda_check_board_config(codec, AD1981_MODELS,
- ad1981_models,
- ad1981_cfg_tbl);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1981_AUTO;
- }
-
- if (board_config == AD1981_AUTO)
- return ad1981_parse_auto_config(codec);
-
- err = alloc_ad_spec(codec);
- if (err < 0)
- return -ENOMEM;
- spec = codec->spec;
-
- err = snd_hda_attach_beep_device(codec, 0x10);
- if (err < 0) {
- ad198x_free(codec);
- return err;
- }
- set_beep_amp(spec, 0x0d, 0, HDA_OUTPUT);
-
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids);
- spec->multiout.dac_nids = ad1981_dac_nids;
- spec->multiout.dig_out_nid = AD1981_SPDIF_OUT;
- spec->num_adc_nids = 1;
- spec->adc_nids = ad1981_adc_nids;
- spec->capsrc_nids = ad1981_capsrc_nids;
- spec->input_mux = &ad1981_capture_source;
- spec->num_mixers = 1;
- spec->mixers[0] = ad1981_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1981_init_verbs;
- spec->spdif_route = 0;
-#ifdef CONFIG_PM
- spec->loopback.amplist = ad1981_loopbacks;
-#endif
- spec->vmaster_nid = 0x05;
-
- codec->patch_ops = ad198x_patch_ops;
-
- /* override some parameters */
- switch (board_config) {
- case AD1981_HP:
- spec->mixers[0] = ad1981_hp_mixers;
- spec->num_init_verbs = 2;
- spec->init_verbs[1] = ad1981_hp_init_verbs;
- if (!is_jack_available(codec, 0x0a))
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1981_hp_capture_source;
-
- codec->patch_ops.init = ad1981_hp_init;
- codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
- /* set the upper-limit for mixer amp to 0dB for avoiding the
- * possible damage by overloading
- */
- snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
- (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
- (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
- (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
- (1 << AC_AMPCAP_MUTE_SHIFT));
- break;
- case AD1981_THINKPAD:
- spec->mixers[0] = ad1981_thinkpad_mixers;
- spec->input_mux = &ad1981_thinkpad_capture_source;
- /* set the upper-limit for mixer amp to 0dB for avoiding the
- * possible damage by overloading
- */
- snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
- (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
- (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
- (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
- (1 << AC_AMPCAP_MUTE_SHIFT));
- break;
- case AD1981_TOSHIBA:
- spec->mixers[0] = ad1981_hp_mixers;
- spec->mixers[1] = ad1981_toshiba_mixers;
- spec->num_init_verbs = 2;
- spec->init_verbs[1] = ad1981_toshiba_init_verbs;
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1981_hp_capture_source;
- codec->patch_ops.init = ad1981_hp_init;
- codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
- break;
- }
-
- codec->no_trigger_sense = 1;
- codec->no_sticky_stream = 1;
-
- return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1981 ad1981_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
/*
* AD1988
@@ -2395,90 +649,7 @@
* E/F quad mic array
*/
-
#ifdef ENABLE_AD_STATIC_QUIRKS
-/* models */
-enum {
- AD1988_AUTO,
- AD1988_6STACK,
- AD1988_6STACK_DIG,
- AD1988_3STACK,
- AD1988_3STACK_DIG,
- AD1988_LAPTOP,
- AD1988_LAPTOP_DIG,
- AD1988_MODEL_LAST,
-};
-
-/* reivision id to check workarounds */
-#define AD1988A_REV2 0x100200
-
-#define is_rev2(codec) \
- ((codec)->vendor_id == 0x11d41988 && \
- (codec)->revision_id == AD1988A_REV2)
-
-/*
- * mixers
- */
-
-static const hda_nid_t ad1988_6stack_dac_nids[4] = {
- 0x04, 0x06, 0x05, 0x0a
-};
-
-static const hda_nid_t ad1988_3stack_dac_nids[3] = {
- 0x04, 0x05, 0x0a
-};
-
-/* for AD1988A revision-2, DAC2-4 are swapped */
-static const hda_nid_t ad1988_6stack_dac_nids_rev2[4] = {
- 0x04, 0x05, 0x0a, 0x06
-};
-
-static const hda_nid_t ad1988_alt_dac_nid[1] = {
- 0x03
-};
-
-static const hda_nid_t ad1988_3stack_dac_nids_rev2[3] = {
- 0x04, 0x0a, 0x06
-};
-
-static const hda_nid_t ad1988_adc_nids[3] = {
- 0x08, 0x09, 0x0f
-};
-
-static const hda_nid_t ad1988_capsrc_nids[3] = {
- 0x0c, 0x0d, 0x0e
-};
-
-#define AD1988_SPDIF_OUT 0x02
-#define AD1988_SPDIF_OUT_HDMI 0x0b
-#define AD1988_SPDIF_IN 0x07
-
-static const hda_nid_t ad1989b_slave_dig_outs[] = {
- AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0
-};
-
-static const struct hda_input_mux ad1988_6stack_capture_source = {
- .num_items = 5,
- .items = {
- { "Front Mic", 0x1 }, /* port-B */
- { "Line", 0x2 }, /* port-C */
- { "Mic", 0x4 }, /* port-E */
- { "CD", 0x5 },
- { "Mix", 0x9 },
- },
-};
-
-static const struct hda_input_mux ad1988_laptop_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic/Line", 0x1 }, /* port-B */
- { "CD", 0x5 },
- { "Mix", 0x9 },
- },
-};
-
-/*
- */
static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
@@ -2509,569 +680,6 @@
spec->multiout.num_dacs = spec->multiout.max_channels / 2;
return err;
}
-
-/* 6-stack mode */
-static const struct snd_kcontrol_new ad1988_6stack_mixers1[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x05, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0a, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Side Playback Volume", 0x06, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT),
- HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
- HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-/* 3-stack mode */
-static const struct snd_kcontrol_new ad1988_3stack_mixers1[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x06, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x06, 2, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x2c, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x26, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x26, 2, 2, HDA_INPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT),
- HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
- HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = ad198x_ch_mode_info,
- .get = ad198x_ch_mode_get,
- .put = ad198x_ch_mode_put,
- },
-
- { } /* end */
-};
-
-/* laptop mode */
-static const struct snd_kcontrol_new ad1988_laptop_mixers[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x29, 0x0, HDA_INPUT),
- HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
- HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
-
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "External Amplifier",
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x12,
- .info = ad198x_eapd_info,
- .get = ad198x_eapd_get,
- .put = ad198x_eapd_put,
- .private_value = 0x12, /* port-D */
- },
-
- { } /* end */
-};
-
-/* capture */
-static const struct snd_kcontrol_new ad1988_capture_mixers[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x0e, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x0e, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 3,
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- { } /* end */
-};
-
-static int ad1988_spdif_playback_source_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- static const char * const texts[] = {
- "PCM", "ADC1", "ADC2", "ADC3"
- };
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 4;
- if (uinfo->value.enumerated.item >= 4)
- uinfo->value.enumerated.item = 3;
- strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
- return 0;
-}
-
-static int ad1988_spdif_playback_source_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- unsigned int sel;
-
- sel = snd_hda_codec_read(codec, 0x1d, 0, AC_VERB_GET_AMP_GAIN_MUTE,
- AC_AMP_GET_INPUT);
- if (!(sel & 0x80))
- ucontrol->value.enumerated.item[0] = 0;
- else {
- sel = snd_hda_codec_read(codec, 0x0b, 0,
- AC_VERB_GET_CONNECT_SEL, 0);
- if (sel < 3)
- sel++;
- else
- sel = 0;
- ucontrol->value.enumerated.item[0] = sel;
- }
- return 0;
-}
-
-static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- unsigned int val, sel;
- int change;
-
- val = ucontrol->value.enumerated.item[0];
- if (val > 3)
- return -EINVAL;
- if (!val) {
- sel = snd_hda_codec_read(codec, 0x1d, 0,
- AC_VERB_GET_AMP_GAIN_MUTE,
- AC_AMP_GET_INPUT);
- change = sel & 0x80;
- if (change) {
- snd_hda_codec_write_cache(codec, 0x1d, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_UNMUTE(0));
- snd_hda_codec_write_cache(codec, 0x1d, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_MUTE(1));
- }
- } else {
- sel = snd_hda_codec_read(codec, 0x1d, 0,
- AC_VERB_GET_AMP_GAIN_MUTE,
- AC_AMP_GET_INPUT | 0x01);
- change = sel & 0x80;
- if (change) {
- snd_hda_codec_write_cache(codec, 0x1d, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_MUTE(0));
- snd_hda_codec_write_cache(codec, 0x1d, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_UNMUTE(1));
- }
- sel = snd_hda_codec_read(codec, 0x0b, 0,
- AC_VERB_GET_CONNECT_SEL, 0) + 1;
- change |= sel != val;
- if (change)
- snd_hda_codec_write_cache(codec, 0x0b, 0,
- AC_VERB_SET_CONNECT_SEL,
- val - 1);
- }
- return change;
-}
-
-static const struct snd_kcontrol_new ad1988_spdif_out_mixers[] = {
- HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "IEC958 Playback Source",
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b,
- .info = ad1988_spdif_playback_source_info,
- .get = ad1988_spdif_playback_source_get,
- .put = ad1988_spdif_playback_source_put,
- },
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_spdif_in_mixers[] = {
- HDA_CODEC_VOLUME("IEC958 Capture Volume", 0x1c, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1989_spdif_out_mixers[] = {
- HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("HDMI Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-/*
- * initialization verbs
- */
-
-/*
- * for 6-stack (+dig)
- */
-static const struct hda_verb ad1988_6stack_init_verbs[] = {
- /* Front, Surround, CLFE, side DAC; unmute as default */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Port-A front headphon path */
- {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Port-D line-out path */
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Port-F surround path */
- {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Port-G CLFE path */
- {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Port-H side path */
- {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Mono out path */
- {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
- /* Port-B front mic-in path */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Port-C line-in path */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Port-E mic-in path */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Analog CD Input */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-
- { }
-};
-
-static const struct hda_verb ad1988_6stack_fp_init_verbs[] = {
- /* Headphone; unmute as default */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Port-A front headphon path */
- {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-
- { }
-};
-
-static const struct hda_verb ad1988_capture_init_verbs[] = {
- /* mute analog mix */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
- /* select ADCs - front-mic */
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-
- { }
-};
-
-static const struct hda_verb ad1988_spdif_init_verbs[] = {
- /* SPDIF out sel */
- {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* SPDIF out pin */
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-
- { }
-};
-
-static const struct hda_verb ad1988_spdif_in_init_verbs[] = {
- /* unmute SPDIF input pin */
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- { }
-};
-
-/* AD1989 has no ADC -> SPDIF route */
-static const struct hda_verb ad1989_spdif_init_verbs[] = {
- /* SPDIF-1 out pin */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
- /* SPDIF-2/HDMI out pin */
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
- { }
-};
-
-/*
- * verbs for 3stack (+dig)
- */
-static const struct hda_verb ad1988_3stack_ch2_init[] = {
- /* set port-C to line-in */
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- /* set port-E to mic-in */
- { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { } /* end */
-};
-
-static const struct hda_verb ad1988_3stack_ch6_init[] = {
- /* set port-C to surround out */
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- /* set port-E to CLFE out */
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { } /* end */
-};
-
-static const struct hda_channel_mode ad1988_3stack_modes[2] = {
- { 2, ad1988_3stack_ch2_init },
- { 6, ad1988_3stack_ch6_init },
-};
-
-static const struct hda_verb ad1988_3stack_init_verbs[] = {
- /* Front, Surround, CLFE, side DAC; unmute as default */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Port-A front headphon path */
- {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Port-D line-out path */
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Mono out path */
- {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
- /* Port-B front mic-in path */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Port-C line-in/surround path - 6ch mode as default */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x31, AC_VERB_SET_CONNECT_SEL, 0x0}, /* output sel: DAC 0x05 */
- {0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Port-E mic-in/CLFE path - 6ch mode as default */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x32, AC_VERB_SET_CONNECT_SEL, 0x1}, /* output sel: DAC 0x0a */
- {0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* mute analog mix */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
- /* select ADCs - front-mic */
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
- { }
-};
-
-/*
- * verbs for laptop mode (+dig)
- */
-static const struct hda_verb ad1988_laptop_hp_on[] = {
- /* unmute port-A and mute port-D */
- { 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-static const struct hda_verb ad1988_laptop_hp_off[] = {
- /* mute port-A and unmute port-D */
- { 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { } /* end */
-};
-
-#define AD1988_HP_EVENT 0x01
-
-static const struct hda_verb ad1988_laptop_init_verbs[] = {
- /* Front, Surround, CLFE, side DAC; unmute as default */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Port-A front headphon path */
- {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* unsolicited event for pin-sense */
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1988_HP_EVENT },
- /* Port-D line-out path + EAPD */
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x00}, /* EAPD-off */
- /* Mono out path */
- {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
- /* Port-B mic-in path */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Port-C docking station - try to output */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* mute analog mix */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
- /* select ADCs - mic */
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
- { }
-};
-
-static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- if ((res >> 26) != AD1988_HP_EVENT)
- return;
- if (snd_hda_jack_detect(codec, 0x11))
- snd_hda_sequence_write(codec, ad1988_laptop_hp_on);
- else
- snd_hda_sequence_write(codec, ad1988_laptop_hp_off);
-}
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1988_loopbacks[] = {
- { 0x20, HDA_INPUT, 0 }, /* Front Mic */
- { 0x20, HDA_INPUT, 1 }, /* Line */
- { 0x20, HDA_INPUT, 4 }, /* Mic */
- { 0x20, HDA_INPUT, 6 }, /* CD */
- { } /* end */
-};
-#endif
#endif /* ENABLE_AD_STATIC_QUIRKS */
static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol,
@@ -3220,7 +828,34 @@
/*
*/
-static int ad1988_parse_auto_config(struct hda_codec *codec)
+enum {
+ AD1988_FIXUP_6STACK_DIG,
+};
+
+static const struct hda_fixup ad1988_fixups[] = {
+ [AD1988_FIXUP_6STACK_DIG] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x11, 0x02214130 }, /* front-hp */
+ { 0x12, 0x01014010 }, /* line-out */
+ { 0x14, 0x02a19122 }, /* front-mic */
+ { 0x15, 0x01813021 }, /* line-in */
+ { 0x16, 0x01011012 }, /* line-out */
+ { 0x17, 0x01a19020 }, /* mic */
+ { 0x1b, 0x0145f1f0 }, /* SPDIF */
+ { 0x24, 0x01016011 }, /* line-out */
+ { 0x25, 0x01012013 }, /* line-out */
+ { }
+ }
+ },
+};
+
+static const struct hda_model_fixup ad1988_fixup_models[] = {
+ { .id = AD1988_FIXUP_6STACK_DIG, .name = "6stack-dig" },
+ {}
+};
+
+static int patch_ad1988(struct hda_codec *codec)
{
struct ad198x_spec *spec;
int err;
@@ -3234,12 +869,19 @@
spec->gen.mixer_merge_nid = 0x21;
spec->gen.beep_nid = 0x10;
set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
+
+ snd_hda_pick_fixup(codec, ad1988_fixup_models, NULL, ad1988_fixups);
+ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
+
err = ad198x_parse_auto_config(codec);
if (err < 0)
goto error;
err = ad1988_add_spdif_mux_ctl(codec);
if (err < 0)
goto error;
+
+ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
+
return 0;
error:
@@ -3247,169 +889,6 @@
return err;
}
-/*
- */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const char * const ad1988_models[AD1988_MODEL_LAST] = {
- [AD1988_6STACK] = "6stack",
- [AD1988_6STACK_DIG] = "6stack-dig",
- [AD1988_3STACK] = "3stack",
- [AD1988_3STACK_DIG] = "3stack-dig",
- [AD1988_LAPTOP] = "laptop",
- [AD1988_LAPTOP_DIG] = "laptop-dig",
- [AD1988_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk ad1988_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG),
- SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG),
- SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG),
- SND_PCI_QUIRK(0x1043, 0x82c0, "Asus M3N-HT Deluxe", AD1988_6STACK_DIG),
- SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG),
- {}
-};
-
-static int patch_ad1988(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int err, board_config;
-
- board_config = snd_hda_check_board_config(codec, AD1988_MODEL_LAST,
- ad1988_models, ad1988_cfg_tbl);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1988_AUTO;
- }
-
- if (board_config == AD1988_AUTO)
- return ad1988_parse_auto_config(codec);
-
- err = alloc_ad_spec(codec);
- if (err < 0)
- return err;
- spec = codec->spec;
-
- if (is_rev2(codec))
- snd_printk(KERN_INFO "patch_analog: AD1988A rev.2 is detected, enable workarounds\n");
-
- err = snd_hda_attach_beep_device(codec, 0x10);
- if (err < 0) {
- ad198x_free(codec);
- return err;
- }
- set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
- if (!spec->multiout.hp_nid)
- spec->multiout.hp_nid = ad1988_alt_dac_nid[0];
- switch (board_config) {
- case AD1988_6STACK:
- case AD1988_6STACK_DIG:
- spec->multiout.max_channels = 8;
- spec->multiout.num_dacs = 4;
- if (is_rev2(codec))
- spec->multiout.dac_nids = ad1988_6stack_dac_nids_rev2;
- else
- spec->multiout.dac_nids = ad1988_6stack_dac_nids;
- spec->input_mux = &ad1988_6stack_capture_source;
- spec->num_mixers = 2;
- if (is_rev2(codec))
- spec->mixers[0] = ad1988_6stack_mixers1_rev2;
- else
- spec->mixers[0] = ad1988_6stack_mixers1;
- spec->mixers[1] = ad1988_6stack_mixers2;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1988_6stack_init_verbs;
- if (board_config == AD1988_6STACK_DIG) {
- spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
- spec->dig_in_nid = AD1988_SPDIF_IN;
- }
- break;
- case AD1988_3STACK:
- case AD1988_3STACK_DIG:
- spec->multiout.max_channels = 6;
- spec->multiout.num_dacs = 3;
- if (is_rev2(codec))
- spec->multiout.dac_nids = ad1988_3stack_dac_nids_rev2;
- else
- spec->multiout.dac_nids = ad1988_3stack_dac_nids;
- spec->input_mux = &ad1988_6stack_capture_source;
- spec->channel_mode = ad1988_3stack_modes;
- spec->num_channel_mode = ARRAY_SIZE(ad1988_3stack_modes);
- spec->num_mixers = 2;
- if (is_rev2(codec))
- spec->mixers[0] = ad1988_3stack_mixers1_rev2;
- else
- spec->mixers[0] = ad1988_3stack_mixers1;
- spec->mixers[1] = ad1988_3stack_mixers2;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1988_3stack_init_verbs;
- if (board_config == AD1988_3STACK_DIG)
- spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
- break;
- case AD1988_LAPTOP:
- case AD1988_LAPTOP_DIG:
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = ad1988_3stack_dac_nids;
- spec->input_mux = &ad1988_laptop_capture_source;
- spec->num_mixers = 1;
- spec->mixers[0] = ad1988_laptop_mixers;
- codec->inv_eapd = 1; /* inverted EAPD */
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1988_laptop_init_verbs;
- if (board_config == AD1988_LAPTOP_DIG)
- spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
- break;
- }
-
- spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids);
- spec->adc_nids = ad1988_adc_nids;
- spec->capsrc_nids = ad1988_capsrc_nids;
- spec->mixers[spec->num_mixers++] = ad1988_capture_mixers;
- spec->init_verbs[spec->num_init_verbs++] = ad1988_capture_init_verbs;
- if (spec->multiout.dig_out_nid) {
- if (codec->vendor_id >= 0x11d4989a) {
- spec->mixers[spec->num_mixers++] =
- ad1989_spdif_out_mixers;
- spec->init_verbs[spec->num_init_verbs++] =
- ad1989_spdif_init_verbs;
- codec->slave_dig_outs = ad1989b_slave_dig_outs;
- } else {
- spec->mixers[spec->num_mixers++] =
- ad1988_spdif_out_mixers;
- spec->init_verbs[spec->num_init_verbs++] =
- ad1988_spdif_init_verbs;
- }
- }
- if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) {
- spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers;
- spec->init_verbs[spec->num_init_verbs++] =
- ad1988_spdif_in_init_verbs;
- }
-
- codec->patch_ops = ad198x_patch_ops;
- switch (board_config) {
- case AD1988_LAPTOP:
- case AD1988_LAPTOP_DIG:
- codec->patch_ops.unsol_event = ad1988_laptop_unsol_event;
- break;
- }
-#ifdef CONFIG_PM
- spec->loopback.amplist = ad1988_loopbacks;
-#endif
- spec->vmaster_nid = 0x04;
-
- codec->no_trigger_sense = 1;
- codec->no_sticky_stream = 1;
-
- return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1988 ad1988_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
/*
* AD1884 / AD1984
@@ -3423,168 +902,20 @@
*
* AD1984 = AD1884 + two digital mic-ins
*
- * FIXME:
- * For simplicity, we share the single DAC for both HP and line-outs
- * right now. The inidividual playbacks could be easily implemented,
- * but no build-up framework is given, so far.
+ * AD1883 / AD1884A / AD1984A / AD1984B
+ *
+ * port-B (0x14) - front mic-in
+ * port-E (0x1c) - rear mic-in
+ * port-F (0x16) - CD / ext out
+ * port-C (0x15) - rear line-in
+ * port-D (0x12) - rear line-out
+ * port-A (0x11) - front hp-out
+ *
+ * AD1984A = AD1884A + digital-mic
+ * AD1883 = equivalent with AD1984A
+ * AD1984B = AD1984A + extra SPDIF-out
*/
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1884_dac_nids[1] = {
- 0x04,
-};
-
-static const hda_nid_t ad1884_adc_nids[2] = {
- 0x08, 0x09,
-};
-
-static const hda_nid_t ad1884_capsrc_nids[2] = {
- 0x0c, 0x0d,
-};
-
-#define AD1884_SPDIF_OUT 0x02
-
-static const struct hda_input_mux ad1884_capture_source = {
- .num_items = 4,
- .items = {
- { "Front Mic", 0x0 },
- { "Mic", 0x1 },
- { "CD", 0x2 },
- { "Mix", 0x3 },
- },
-};
-
-static const struct snd_kcontrol_new ad1884_base_mixers[] = {
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- /* SPDIF controls */
- HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- /* identical with ad1983 */
- .info = ad1983_spdif_route_info,
- .get = ad1983_spdif_route_get,
- .put = ad1983_spdif_route_put,
- },
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984_dmic_mixers[] = {
- HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0,
- HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Digital Mic Capture Switch", 1, 0x06, 0x0,
- HDA_INPUT),
- { } /* end */
-};
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1884_init_verbs[] = {
- /* DACs; mute as default */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Port-A (HP) mixer */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-A pin */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* HP selector - select DAC2 */
- {0x22, AC_VERB_SET_CONNECT_SEL, 0x1},
- /* Port-D (Line-out) mixer */
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-D pin */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Mono-out mixer */
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Mono-out pin */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Mono selector */
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
- /* Port-B (front mic) pin */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Port-C (rear mic) pin */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Analog mixer; mute as default */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
- /* SPDIF output selector */
- {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
- { } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1884_loopbacks[] = {
- { 0x20, HDA_INPUT, 0 }, /* Front Mic */
- { 0x20, HDA_INPUT, 1 }, /* Mic */
- { 0x20, HDA_INPUT, 2 }, /* CD */
- { 0x20, HDA_INPUT, 4 }, /* Docking */
- { } /* end */
-};
-#endif
-
-static const char * const ad1884_slave_vols[] = {
- "PCM", "Mic", "Mono", "Front Mic", "Mic", "CD",
- "Internal Mic", "Dock Mic", /* "Beep", */ "IEC958",
- NULL
-};
-
-enum {
- AD1884_AUTO,
- AD1884_BASIC,
- AD1884_MODELS
-};
-
-static const char * const ad1884_models[AD1884_MODELS] = {
- [AD1884_AUTO] = "auto",
- [AD1884_BASIC] = "basic",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
/* set the upper-limit for mixer amp to 0dB for avoiding the possible
* damage by overloading
*/
@@ -3599,14 +930,34 @@
(1 << AC_AMPCAP_MUTE_SHIFT));
}
+/* toggle GPIO1 according to the mute state */
+static void ad1884_vmaster_hp_gpio_hook(void *private_data, int enabled)
+{
+ struct hda_codec *codec = private_data;
+ struct ad198x_spec *spec = codec->spec;
+
+ if (spec->eapd_nid)
+ ad_vmaster_eapd_hook(private_data, enabled);
+ snd_hda_codec_update_cache(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DATA,
+ enabled ? 0x00 : 0x02);
+}
+
static void ad1884_fixup_hp_eapd(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
struct ad198x_spec *spec = codec->spec;
+ static const struct hda_verb gpio_init_verbs[] = {
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x02},
+ {},
+ };
switch (action) {
case HDA_FIXUP_ACT_PRE_PROBE:
- spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook;
+ spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook;
+ snd_hda_sequence_write_cache(codec, gpio_init_verbs);
break;
case HDA_FIXUP_ACT_PROBE:
if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
@@ -3617,9 +968,18 @@
}
}
+/* set magic COEFs for dmic */
+static const struct hda_verb ad1884_dmic_init_verbs[] = {
+ {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
+ {0x01, AC_VERB_SET_PROC_COEF, 0x08},
+ {}
+};
+
enum {
AD1884_FIXUP_AMP_OVERRIDE,
AD1884_FIXUP_HP_EAPD,
+ AD1884_FIXUP_DMIC_COEF,
+ AD1884_FIXUP_HP_TOUCHSMART,
};
static const struct hda_fixup ad1884_fixups[] = {
@@ -3633,15 +993,27 @@
.chained = true,
.chain_id = AD1884_FIXUP_AMP_OVERRIDE,
},
+ [AD1884_FIXUP_DMIC_COEF] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = ad1884_dmic_init_verbs,
+ },
+ [AD1884_FIXUP_HP_TOUCHSMART] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = ad1884_dmic_init_verbs,
+ .chained = true,
+ .chain_id = AD1884_FIXUP_HP_EAPD,
+ },
};
static const struct snd_pci_quirk ad1884_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x103c, 0x2a82, "HP Touchsmart", AD1884_FIXUP_HP_TOUCHSMART),
SND_PCI_QUIRK_VENDOR(0x103c, "HP", AD1884_FIXUP_HP_EAPD),
+ SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_DMIC_COEF),
{}
};
-static int ad1884_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1884(struct hda_codec *codec)
{
struct ad198x_spec *spec;
int err;
@@ -3674,1170 +1046,6 @@
return err;
}
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1884_basic(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int err;
-
- err = alloc_ad_spec(codec);
- if (err < 0)
- return err;
- spec = codec->spec;
-
- err = snd_hda_attach_beep_device(codec, 0x10);
- if (err < 0) {
- ad198x_free(codec);
- return err;
- }
- set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids);
- spec->multiout.dac_nids = ad1884_dac_nids;
- spec->multiout.dig_out_nid = AD1884_SPDIF_OUT;
- spec->num_adc_nids = ARRAY_SIZE(ad1884_adc_nids);
- spec->adc_nids = ad1884_adc_nids;
- spec->capsrc_nids = ad1884_capsrc_nids;
- spec->input_mux = &ad1884_capture_source;
- spec->num_mixers = 1;
- spec->mixers[0] = ad1884_base_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1884_init_verbs;
- spec->spdif_route = 0;
-#ifdef CONFIG_PM
- spec->loopback.amplist = ad1884_loopbacks;
-#endif
- spec->vmaster_nid = 0x04;
- /* we need to cover all playback volumes */
- spec->slave_vols = ad1884_slave_vols;
- /* slaves may contain input volumes, so we can't raise to 0dB blindly */
- spec->avoid_init_slave_vol = 1;
-
- codec->patch_ops = ad198x_patch_ops;
-
- codec->no_trigger_sense = 1;
- codec->no_sticky_stream = 1;
-
- return 0;
-}
-
-static int patch_ad1884(struct hda_codec *codec)
-{
- int board_config;
-
- board_config = snd_hda_check_board_config(codec, AD1884_MODELS,
- ad1884_models, NULL);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1884_AUTO;
- }
-
- if (board_config == AD1884_AUTO)
- return ad1884_parse_auto_config(codec);
- else
- return patch_ad1884_basic(codec);
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1884 ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-/*
- * Lenovo Thinkpad T61/X61
- */
-static const struct hda_input_mux ad1984_thinkpad_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x1 },
- { "Mix", 0x3 },
- { "Dock Mic", 0x4 },
- },
-};
-
-
-/*
- * Dell Precision T3400
- */
-static const struct hda_input_mux ad1984_dell_desktop_capture_source = {
- .num_items = 3,
- .items = {
- { "Front Mic", 0x0 },
- { "Line-In", 0x1 },
- { "Mix", 0x3 },
- },
-};
-
-
-static const struct snd_kcontrol_new ad1984_thinkpad_mixers[] = {
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
- HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
- HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- /* SPDIF controls */
- HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- /* identical with ad1983 */
- .info = ad1983_spdif_route_info,
- .get = ad1983_spdif_route_get,
- .put = ad1983_spdif_route_put,
- },
- { } /* end */
-};
-
-/* additional verbs */
-static const struct hda_verb ad1984_thinkpad_init_verbs[] = {
- /* Port-E (docking station mic) pin */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* docking mic boost */
- {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Analog PC Beeper - allow firmware/ACPI beeps */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3) | 0x1a},
- /* Analog mixer - docking mic; mute as default */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* enable EAPD bit */
- {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
- { } /* end */
-};
-
-/*
- * Dell Precision T3400
- */
-static const struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = {
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Line-In Boost Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- { } /* end */
-};
-
-/* Digial MIC ADC NID 0x05 + 0x06 */
-static int ad1984_pcm_dmic_prepare(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- struct snd_pcm_substream *substream)
-{
- snd_hda_codec_setup_stream(codec, 0x05 + substream->number,
- stream_tag, 0, format);
- return 0;
-}
-
-static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- snd_hda_codec_cleanup_stream(codec, 0x05 + substream->number);
- return 0;
-}
-
-static const struct hda_pcm_stream ad1984_pcm_dmic_capture = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0x05,
- .ops = {
- .prepare = ad1984_pcm_dmic_prepare,
- .cleanup = ad1984_pcm_dmic_cleanup
- },
-};
-
-static int ad1984_build_pcms(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
- struct hda_pcm *info;
- int err;
-
- err = ad198x_build_pcms(codec);
- if (err < 0)
- return err;
-
- info = spec->pcm_rec + codec->num_pcms;
- codec->num_pcms++;
- info->name = "AD1984 Digital Mic";
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1984_pcm_dmic_capture;
- return 0;
-}
-
-/* models */
-enum {
- AD1984_AUTO,
- AD1984_BASIC,
- AD1984_THINKPAD,
- AD1984_DELL_DESKTOP,
- AD1984_MODELS
-};
-
-static const char * const ad1984_models[AD1984_MODELS] = {
- [AD1984_AUTO] = "auto",
- [AD1984_BASIC] = "basic",
- [AD1984_THINKPAD] = "thinkpad",
- [AD1984_DELL_DESKTOP] = "dell_desktop",
-};
-
-static const struct snd_pci_quirk ad1984_cfg_tbl[] = {
- /* Lenovo Thinkpad T61/X61 */
- SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD),
- SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP),
- SND_PCI_QUIRK(0x1028, 0x0233, "Dell Latitude E6400", AD1984_DELL_DESKTOP),
- {}
-};
-
-static int patch_ad1984(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int board_config, err;
-
- board_config = snd_hda_check_board_config(codec, AD1984_MODELS,
- ad1984_models, ad1984_cfg_tbl);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1984_AUTO;
- }
-
- if (board_config == AD1984_AUTO)
- return ad1884_parse_auto_config(codec);
-
- err = patch_ad1884_basic(codec);
- if (err < 0)
- return err;
- spec = codec->spec;
-
- switch (board_config) {
- case AD1984_BASIC:
- /* additional digital mics */
- spec->mixers[spec->num_mixers++] = ad1984_dmic_mixers;
- codec->patch_ops.build_pcms = ad1984_build_pcms;
- break;
- case AD1984_THINKPAD:
- if (codec->subsystem_id == 0x17aa20fb) {
- /* Thinpad X300 does not have the ability to do SPDIF,
- or attach to docking station to use SPDIF */
- spec->multiout.dig_out_nid = 0;
- } else
- spec->multiout.dig_out_nid = AD1884_SPDIF_OUT;
- spec->input_mux = &ad1984_thinkpad_capture_source;
- spec->mixers[0] = ad1984_thinkpad_mixers;
- spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs;
- spec->analog_beep = 1;
- break;
- case AD1984_DELL_DESKTOP:
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1984_dell_desktop_capture_source;
- spec->mixers[0] = ad1984_dell_desktop_mixers;
- break;
- }
- return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1984 ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
-/*
- * AD1883 / AD1884A / AD1984A / AD1984B
- *
- * port-B (0x14) - front mic-in
- * port-E (0x1c) - rear mic-in
- * port-F (0x16) - CD / ext out
- * port-C (0x15) - rear line-in
- * port-D (0x12) - rear line-out
- * port-A (0x11) - front hp-out
- *
- * AD1984A = AD1884A + digital-mic
- * AD1883 = equivalent with AD1984A
- * AD1984B = AD1984A + extra SPDIF-out
- *
- * FIXME:
- * We share the single DAC for both HP and line-outs (see AD1884/1984).
- */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1884a_dac_nids[1] = {
- 0x03,
-};
-
-#define ad1884a_adc_nids ad1884_adc_nids
-#define ad1884a_capsrc_nids ad1884_capsrc_nids
-
-#define AD1884A_SPDIF_OUT 0x02
-
-static const struct hda_input_mux ad1884a_capture_source = {
- .num_items = 5,
- .items = {
- { "Front Mic", 0x0 },
- { "Mic", 0x4 },
- { "Line", 0x1 },
- { "CD", 0x2 },
- { "Mix", 0x3 },
- },
-};
-
-static const struct snd_kcontrol_new ad1884a_base_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- /* SPDIF controls */
- HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- /* identical with ad1983 */
- .info = ad1983_spdif_route_info,
- .get = ad1983_spdif_route_get,
- .put = ad1983_spdif_route_put,
- },
- { } /* end */
-};
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1884a_init_verbs[] = {
- /* DACs; unmute as default */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
- /* Port-A (HP) mixer - route only from analog mixer */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-A pin */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Port-D (Line-out) mixer - route only from analog mixer */
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-D pin */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Mono-out mixer - route only from analog mixer */
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Mono-out pin */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Port-B (front mic) pin */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Port-C (rear line-in) pin */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Port-E (rear mic) pin */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* no boost */
- /* Port-F (CD) pin */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Analog mixer; mute as default */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, /* aux */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* capture sources */
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* SPDIF output amp */
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
- { } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1884a_loopbacks[] = {
- { 0x20, HDA_INPUT, 0 }, /* Front Mic */
- { 0x20, HDA_INPUT, 1 }, /* Mic */
- { 0x20, HDA_INPUT, 2 }, /* CD */
- { 0x20, HDA_INPUT, 4 }, /* Docking */
- { } /* end */
-};
-#endif
-
-/*
- * Laptop model
- *
- * Port A: Headphone jack
- * Port B: MIC jack
- * Port C: Internal MIC
- * Port D: Dock Line Out (if enabled)
- * Port E: Dock Line In (if enabled)
- * Port F: Internal speakers
- */
-
-static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
- int mute = (!ucontrol->value.integer.value[0] &&
- !ucontrol->value.integer.value[1]);
- /* toggle GPIO1 according to the mute state */
- snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
- mute ? 0x02 : 0x0);
- return ret;
-}
-
-static const struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_AMP_FLAG,
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = ad1884a_mobile_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
- },
- HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- /*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_AMP_FLAG,
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = ad1884a_mobile_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
- },
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1884a_hp_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x11);
- snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
- present ? 0x00 : 0x02);
-}
-
-/* switch to external mic if plugged */
-static void ad1884a_hp_automic(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x14);
- snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL,
- present ? 0 : 1);
-}
-
-#define AD1884A_HP_EVENT 0x37
-#define AD1884A_MIC_EVENT 0x36
-
-/* unsolicited event for HP jack sensing */
-static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- switch (res >> 26) {
- case AD1884A_HP_EVENT:
- ad1884a_hp_automute(codec);
- break;
- case AD1884A_MIC_EVENT:
- ad1884a_hp_automic(codec);
- break;
- }
-}
-
-/* initialize jack-sensing, too */
-static int ad1884a_hp_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1884a_hp_automute(codec);
- ad1884a_hp_automic(codec);
- return 0;
-}
-
-/* mute internal speaker if HP or docking HP is plugged */
-static void ad1884a_laptop_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x11);
- if (!present)
- present = snd_hda_jack_detect(codec, 0x12);
- snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
- present ? 0x00 : 0x02);
-}
-
-/* switch to external mic if plugged */
-static void ad1884a_laptop_automic(struct hda_codec *codec)
-{
- unsigned int idx;
-
- if (snd_hda_jack_detect(codec, 0x14))
- idx = 0;
- else if (snd_hda_jack_detect(codec, 0x1c))
- idx = 4;
- else
- idx = 1;
- snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, idx);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1884a_laptop_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case AD1884A_HP_EVENT:
- ad1884a_laptop_automute(codec);
- break;
- case AD1884A_MIC_EVENT:
- ad1884a_laptop_automic(codec);
- break;
- }
-}
-
-/* initialize jack-sensing, too */
-static int ad1884a_laptop_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1884a_laptop_automute(codec);
- ad1884a_laptop_automic(codec);
- return 0;
-}
-
-/* additional verbs for laptop model */
-static const struct hda_verb ad1884a_laptop_verbs[] = {
- /* Port-A (HP) pin - always unmuted */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Port-F (int speaker) mixer - route only from analog mixer */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-F (int speaker) pin */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* required for compaq 6530s/6531s speaker output */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Port-C pin - internal mic-in */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
- /* Port-D (docking line-out) pin - default unmuted */
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* analog mix */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* unsolicited event for pin-sense */
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
- {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
- {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
- /* allow to touch GPIO1 (for mute control) */
- {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
- { } /* end */
-};
-
-static const struct hda_verb ad1884a_mobile_verbs[] = {
- /* DACs; unmute as default */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
- /* Port-A (HP) mixer - route only from analog mixer */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-A pin */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Port-A (HP) pin - always unmuted */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Port-B (mic jack) pin */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
- /* Port-C (int mic) pin */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
- /* Port-F (int speaker) mixer - route only from analog mixer */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-F pin */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Analog mixer; mute as default */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* capture sources */
- /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* unsolicited event for pin-sense */
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
- /* allow to touch GPIO1 (for mute control) */
- {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
- { } /* end */
-};
-
-/*
- * Thinkpad X300
- * 0x11 - HP
- * 0x12 - speaker
- * 0x14 - mic-in
- * 0x17 - built-in mic
- */
-
-static const struct hda_verb ad1984a_thinkpad_verbs[] = {
- /* HP unmute */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* analog mix */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* turn on EAPD */
- {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
- /* unsolicited event for pin-sense */
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
- /* internal mic - dmic */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* set magic COEFs for dmic */
- {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
- {0x01, AC_VERB_SET_PROC_COEF, 0x08},
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- { } /* end */
-};
-
-static const struct hda_input_mux ad1984a_thinkpad_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x5 },
- { "Mix", 0x3 },
- },
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1984a_thinkpad_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x11);
- snd_hda_codec_amp_stereo(codec, 0x12, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_thinkpad_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) != AD1884A_HP_EVENT)
- return;
- ad1984a_thinkpad_automute(codec);
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_thinkpad_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1984a_thinkpad_automute(codec);
- return 0;
-}
-
-/*
- * Precision R5500
- * 0x12 - HP/line-out
- * 0x13 - speaker (mono)
- * 0x15 - mic-in
- */
-
-static const struct hda_verb ad1984a_precision_verbs[] = {
- /* Unmute main output path */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x1f}, /* 0dB */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) + 0x17}, /* 0dB */
- /* Analog mixer; mute as default */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Select mic as input */
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x27}, /* 0dB */
- /* Configure as mic */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
- /* HP unmute */
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* turn on EAPD */
- {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
- /* unsolicited event for pin-sense */
- {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_precision_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-
-/* mute internal speaker if HP is plugged */
-static void ad1984a_precision_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x12);
- snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_precision_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) != AD1884A_HP_EVENT)
- return;
- ad1984a_precision_automute(codec);
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_precision_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1984a_precision_automute(codec);
- return 0;
-}
-
-
-/*
- * HP Touchsmart
- * port-A (0x11) - front hp-out
- * port-B (0x14) - unused
- * port-C (0x15) - unused
- * port-D (0x12) - rear line out
- * port-E (0x1c) - front mic-in
- * port-F (0x16) - Internal speakers
- * digital-mic (0x17) - Internal mic
- */
-
-static const struct hda_verb ad1984a_touchsmart_verbs[] = {
- /* DACs; unmute as default */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
- /* Port-A (HP) mixer - route only from analog mixer */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-A pin */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Port-A (HP) pin - always unmuted */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Port-E (int speaker) mixer - route only from analog mixer */
- {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03},
- /* Port-E pin */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- /* Port-F (int speaker) mixer - route only from analog mixer */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-F pin */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Analog mixer; mute as default */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* capture sources */
- /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* unsolicited event for pin-sense */
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
- {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
- /* allow to touch GPIO1 (for mute control) */
- {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
- /* internal mic - dmic */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* set magic COEFs for dmic */
- {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
- {0x01, AC_VERB_SET_PROC_COEF, 0x08},
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .subdevice = HDA_SUBDEV_AMP_FLAG,
- .name = "Master Playback Switch",
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = ad1884a_mobile_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
- },
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-/* switch to external mic if plugged */
-static void ad1984a_touchsmart_automic(struct hda_codec *codec)
-{
- if (snd_hda_jack_detect(codec, 0x1c))
- snd_hda_codec_write(codec, 0x0c, 0,
- AC_VERB_SET_CONNECT_SEL, 0x4);
- else
- snd_hda_codec_write(codec, 0x0c, 0,
- AC_VERB_SET_CONNECT_SEL, 0x5);
-}
-
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case AD1884A_HP_EVENT:
- ad1884a_hp_automute(codec);
- break;
- case AD1884A_MIC_EVENT:
- ad1984a_touchsmart_automic(codec);
- break;
- }
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_touchsmart_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1884a_hp_automute(codec);
- ad1984a_touchsmart_automic(codec);
- return 0;
-}
-
-
-/*
- */
-
-enum {
- AD1884A_AUTO,
- AD1884A_DESKTOP,
- AD1884A_LAPTOP,
- AD1884A_MOBILE,
- AD1884A_THINKPAD,
- AD1984A_TOUCHSMART,
- AD1984A_PRECISION,
- AD1884A_MODELS
-};
-
-static const char * const ad1884a_models[AD1884A_MODELS] = {
- [AD1884A_AUTO] = "auto",
- [AD1884A_DESKTOP] = "desktop",
- [AD1884A_LAPTOP] = "laptop",
- [AD1884A_MOBILE] = "mobile",
- [AD1884A_THINKPAD] = "thinkpad",
- [AD1984A_TOUCHSMART] = "touchsmart",
- [AD1984A_PRECISION] = "precision",
-};
-
-static const struct snd_pci_quirk ad1884a_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1028, 0x04ac, "Precision R5500", AD1984A_PRECISION),
- SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
- SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
- SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
- SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE),
- SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP),
- SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP),
- SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
- SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE),
- SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
- SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART),
- {}
-};
-
-static int patch_ad1884a(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int err, board_config;
-
- board_config = snd_hda_check_board_config(codec, AD1884A_MODELS,
- ad1884a_models,
- ad1884a_cfg_tbl);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1884A_AUTO;
- }
-
- if (board_config == AD1884A_AUTO)
- return ad1884_parse_auto_config(codec);
-
- err = alloc_ad_spec(codec);
- if (err < 0)
- return err;
- spec = codec->spec;
-
- err = snd_hda_attach_beep_device(codec, 0x10);
- if (err < 0) {
- ad198x_free(codec);
- return err;
- }
- set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids);
- spec->multiout.dac_nids = ad1884a_dac_nids;
- spec->multiout.dig_out_nid = AD1884A_SPDIF_OUT;
- spec->num_adc_nids = ARRAY_SIZE(ad1884a_adc_nids);
- spec->adc_nids = ad1884a_adc_nids;
- spec->capsrc_nids = ad1884a_capsrc_nids;
- spec->input_mux = &ad1884a_capture_source;
- spec->num_mixers = 1;
- spec->mixers[0] = ad1884a_base_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1884a_init_verbs;
- spec->spdif_route = 0;
-#ifdef CONFIG_PM
- spec->loopback.amplist = ad1884a_loopbacks;
-#endif
- codec->patch_ops = ad198x_patch_ops;
-
- /* override some parameters */
- switch (board_config) {
- case AD1884A_LAPTOP:
- spec->mixers[0] = ad1884a_laptop_mixers;
- spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
- spec->multiout.dig_out_nid = 0;
- codec->patch_ops.unsol_event = ad1884a_laptop_unsol_event;
- codec->patch_ops.init = ad1884a_laptop_init;
- /* set the upper-limit for mixer amp to 0dB for avoiding the
- * possible damage by overloading
- */
- snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
- (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
- (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
- (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
- (1 << AC_AMPCAP_MUTE_SHIFT));
- break;
- case AD1884A_MOBILE:
- spec->mixers[0] = ad1884a_mobile_mixers;
- spec->init_verbs[0] = ad1884a_mobile_verbs;
- spec->multiout.dig_out_nid = 0;
- codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
- codec->patch_ops.init = ad1884a_hp_init;
- /* set the upper-limit for mixer amp to 0dB for avoiding the
- * possible damage by overloading
- */
- snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
- (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
- (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
- (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
- (1 << AC_AMPCAP_MUTE_SHIFT));
- break;
- case AD1884A_THINKPAD:
- spec->mixers[0] = ad1984a_thinkpad_mixers;
- spec->init_verbs[spec->num_init_verbs++] =
- ad1984a_thinkpad_verbs;
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1984a_thinkpad_capture_source;
- codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event;
- codec->patch_ops.init = ad1984a_thinkpad_init;
- break;
- case AD1984A_PRECISION:
- spec->mixers[0] = ad1984a_precision_mixers;
- spec->init_verbs[spec->num_init_verbs++] =
- ad1984a_precision_verbs;
- spec->multiout.dig_out_nid = 0;
- codec->patch_ops.unsol_event = ad1984a_precision_unsol_event;
- codec->patch_ops.init = ad1984a_precision_init;
- break;
- case AD1984A_TOUCHSMART:
- spec->mixers[0] = ad1984a_touchsmart_mixers;
- spec->init_verbs[0] = ad1984a_touchsmart_verbs;
- spec->multiout.dig_out_nid = 0;
- codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event;
- codec->patch_ops.init = ad1984a_touchsmart_init;
- /* set the upper-limit for mixer amp to 0dB for avoiding the
- * possible damage by overloading
- */
- snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
- (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
- (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
- (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
- (1 << AC_AMPCAP_MUTE_SHIFT));
- break;
- }
-
- codec->no_trigger_sense = 1;
- codec->no_sticky_stream = 1;
-
- return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1884a ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
/*
* AD1882 / AD1882A
*
@@ -4850,299 +1058,7 @@
* port-G - rear clfe-out (6stack)
*/
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1882_dac_nids[3] = {
- 0x04, 0x03, 0x05
-};
-
-static const hda_nid_t ad1882_adc_nids[2] = {
- 0x08, 0x09,
-};
-
-static const hda_nid_t ad1882_capsrc_nids[2] = {
- 0x0c, 0x0d,
-};
-
-#define AD1882_SPDIF_OUT 0x02
-
-/* list: 0x11, 0x39, 0x3a, 0x18, 0x3c, 0x3b, 0x12, 0x20 */
-static const struct hda_input_mux ad1882_capture_source = {
- .num_items = 5,
- .items = {
- { "Front Mic", 0x1 },
- { "Mic", 0x4 },
- { "Line", 0x2 },
- { "CD", 0x3 },
- { "Mix", 0x7 },
- },
-};
-
-/* list: 0x11, 0x39, 0x3a, 0x3c, 0x18, 0x1f, 0x12, 0x20 */
-static const struct hda_input_mux ad1882a_capture_source = {
- .num_items = 5,
- .items = {
- { "Front Mic", 0x1 },
- { "Mic", 0x4},
- { "Line", 0x2 },
- { "Digital Mic", 0x06 },
- { "Mix", 0x7 },
- },
-};
-
-static const struct snd_kcontrol_new ad1882_base_mixers[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line-In Boost Volume", 0x3a, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- /* SPDIF controls */
- HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- /* identical with ad1983 */
- .info = ad1983_spdif_route_info,
- .get = ad1983_spdif_route_get,
- .put = ad1983_spdif_route_put,
- },
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882_loopback_mixers[] = {
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882a_loopback_mixers[] = {
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
- HDA_CODEC_VOLUME("Digital Mic Boost Volume", 0x1f, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882_3stack_mixers[] = {
- HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x17, 2, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = ad198x_ch_mode_info,
- .get = ad198x_ch_mode_get,
- .put = ad198x_ch_mode_put,
- },
- { } /* end */
-};
-
-/* simple auto-mute control for AD1882 3-stack board */
-#define AD1882_HP_EVENT 0x01
-
-static void ad1882_3stack_automute(struct hda_codec *codec)
-{
- bool mute = snd_hda_jack_detect(codec, 0x11);
- snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- mute ? 0 : PIN_OUT);
-}
-
-static int ad1882_3stack_automute_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1882_3stack_automute(codec);
- return 0;
-}
-
-static void ad1882_3stack_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- switch (res >> 26) {
- case AD1882_HP_EVENT:
- ad1882_3stack_automute(codec);
- break;
- }
-}
-
-static const struct snd_kcontrol_new ad1882_6stack_mixers[] = {
- HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x24, 2, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct hda_verb ad1882_ch2_init[] = {
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- { } /* end */
-};
-
-static const struct hda_verb ad1882_ch4_init[] = {
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- { } /* end */
-};
-
-static const struct hda_verb ad1882_ch6_init[] = {
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- { } /* end */
-};
-
-static const struct hda_channel_mode ad1882_modes[3] = {
- { 2, ad1882_ch2_init },
- { 4, ad1882_ch4_init },
- { 6, ad1882_ch6_init },
-};
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1882_init_verbs[] = {
- /* DACs; mute as default */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Port-A (HP) mixer */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-A pin */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* HP selector - select DAC2 */
- {0x37, AC_VERB_SET_CONNECT_SEL, 0x1},
- /* Port-D (Line-out) mixer */
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-D pin */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Mono-out mixer */
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Mono-out pin */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Port-B (front mic) pin */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
- /* Port-C (line-in) pin */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
- /* Port-C mixer - mute as input */
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Port-E (mic-in) pin */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
- /* Port-E mixer - mute as input */
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Port-F (surround) */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Port-G (CLFE) */
- {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Analog mixer; mute as default */
- /* list: 0x39, 0x3a, 0x11, 0x12, 0x3c, 0x3b, 0x18, 0x1a */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
- /* SPDIF output selector */
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
- {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
- { } /* end */
-};
-
-static const struct hda_verb ad1882_3stack_automute_verbs[] = {
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1882_HP_EVENT},
- { } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1882_loopbacks[] = {
- { 0x20, HDA_INPUT, 0 }, /* Front Mic */
- { 0x20, HDA_INPUT, 1 }, /* Mic */
- { 0x20, HDA_INPUT, 4 }, /* Line */
- { 0x20, HDA_INPUT, 6 }, /* CD */
- { } /* end */
-};
-#endif
-
-/* models */
-enum {
- AD1882_AUTO,
- AD1882_3STACK,
- AD1882_6STACK,
- AD1882_3STACK_AUTOMUTE,
- AD1882_MODELS
-};
-
-static const char * const ad1882_models[AD1986A_MODELS] = {
- [AD1882_AUTO] = "auto",
- [AD1882_3STACK] = "3stack",
- [AD1882_6STACK] = "6stack",
- [AD1882_3STACK_AUTOMUTE] = "3stack-automute",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-static int ad1882_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1882(struct hda_codec *codec)
{
struct ad198x_spec *spec;
int err;
@@ -5169,110 +1085,20 @@
return err;
}
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1882(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int err, board_config;
-
- board_config = snd_hda_check_board_config(codec, AD1882_MODELS,
- ad1882_models, NULL);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1882_AUTO;
- }
-
- if (board_config == AD1882_AUTO)
- return ad1882_parse_auto_config(codec);
-
- err = alloc_ad_spec(codec);
- if (err < 0)
- return err;
- spec = codec->spec;
-
- err = snd_hda_attach_beep_device(codec, 0x10);
- if (err < 0) {
- ad198x_free(codec);
- return err;
- }
- set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
- spec->multiout.max_channels = 6;
- spec->multiout.num_dacs = 3;
- spec->multiout.dac_nids = ad1882_dac_nids;
- spec->multiout.dig_out_nid = AD1882_SPDIF_OUT;
- spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids);
- spec->adc_nids = ad1882_adc_nids;
- spec->capsrc_nids = ad1882_capsrc_nids;
- if (codec->vendor_id == 0x11d41882)
- spec->input_mux = &ad1882_capture_source;
- else
- spec->input_mux = &ad1882a_capture_source;
- spec->num_mixers = 2;
- spec->mixers[0] = ad1882_base_mixers;
- if (codec->vendor_id == 0x11d41882)
- spec->mixers[1] = ad1882_loopback_mixers;
- else
- spec->mixers[1] = ad1882a_loopback_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1882_init_verbs;
- spec->spdif_route = 0;
-#ifdef CONFIG_PM
- spec->loopback.amplist = ad1882_loopbacks;
-#endif
- spec->vmaster_nid = 0x04;
-
- codec->patch_ops = ad198x_patch_ops;
-
- /* override some parameters */
- switch (board_config) {
- default:
- case AD1882_3STACK:
- case AD1882_3STACK_AUTOMUTE:
- spec->num_mixers = 3;
- spec->mixers[2] = ad1882_3stack_mixers;
- spec->channel_mode = ad1882_modes;
- spec->num_channel_mode = ARRAY_SIZE(ad1882_modes);
- spec->need_dac_fix = 1;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- if (board_config != AD1882_3STACK) {
- spec->init_verbs[spec->num_init_verbs++] =
- ad1882_3stack_automute_verbs;
- codec->patch_ops.unsol_event = ad1882_3stack_unsol_event;
- codec->patch_ops.init = ad1882_3stack_automute_init;
- }
- break;
- case AD1882_6STACK:
- spec->num_mixers = 3;
- spec->mixers[2] = ad1882_6stack_mixers;
- break;
- }
-
- codec->no_trigger_sense = 1;
- codec->no_sticky_stream = 1;
-
- return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1882 ad1882_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
/*
* patch entries
*/
static const struct hda_codec_preset snd_hda_preset_analog[] = {
- { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a },
+ { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884 },
{ .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 },
- { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a },
+ { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884 },
{ .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 },
- { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884a },
- { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884a },
+ { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884 },
+ { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884 },
{ .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 },
{ .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 },
- { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 },
+ { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1884 },
{ .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },
{ .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 },
{ .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 },
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index de00ce1..4edd2d0 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -66,6 +66,8 @@
hda_nid_t eapds[4];
bool dynamic_eapd;
+ unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */
+
#ifdef ENABLE_CXT_STATIC_QUIRKS
const struct snd_kcontrol_new *mixers[5];
int num_mixers;
@@ -3200,6 +3202,9 @@
snd_hda_gen_init(codec);
if (!spec->dynamic_eapd)
cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
+
+ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT);
+
return 0;
}
@@ -3224,6 +3229,8 @@
CXT_PINCFG_LEMOTE_A1205,
CXT_FIXUP_STEREO_DMIC,
CXT_FIXUP_INC_MIC_BOOST,
+ CXT_FIXUP_HEADPHONE_MIC_PIN,
+ CXT_FIXUP_HEADPHONE_MIC,
};
static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
@@ -3246,6 +3253,59 @@
(0 << AC_AMPCAP_MUTE_SHIFT));
}
+static void cxt_update_headset_mode(struct hda_codec *codec)
+{
+ /* The verbs used in this function were tested on a Conexant CX20751/2 codec. */
+ int i;
+ bool mic_mode = false;
+ struct conexant_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+
+ hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]];
+
+ for (i = 0; i < cfg->num_inputs; i++)
+ if (cfg->inputs[i].pin == mux_pin) {
+ mic_mode = !!cfg->inputs[i].is_headphone_mic;
+ break;
+ }
+
+ if (mic_mode) {
+ snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x7c); /* enable merged mode for analog int-mic */
+ spec->gen.hp_jack_present = false;
+ } else {
+ snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x54); /* disable merged mode for analog int-mic */
+ spec->gen.hp_jack_present = snd_hda_jack_detect(codec, spec->gen.autocfg.hp_pins[0]);
+ }
+
+ snd_hda_gen_update_outputs(codec);
+}
+
+static void cxt_update_headset_mode_hook(struct hda_codec *codec,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ cxt_update_headset_mode(codec);
+}
+
+static void cxt_fixup_headphone_mic(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct conexant_spec *spec = codec->spec;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ spec->parse_flags |= HDA_PINCFG_HEADPHONE_MIC;
+ break;
+ case HDA_FIXUP_ACT_PROBE:
+ spec->gen.cap_sync_hook = cxt_update_headset_mode_hook;
+ spec->gen.automute_hook = cxt_update_headset_mode;
+ break;
+ case HDA_FIXUP_ACT_INIT:
+ cxt_update_headset_mode(codec);
+ break;
+ }
+}
+
+
/* ThinkPad X200 & co with cxt5051 */
static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
@@ -3302,6 +3362,19 @@
.type = HDA_FIXUP_FUNC,
.v.func = cxt5066_increase_mic_boost,
},
+ [CXT_FIXUP_HEADPHONE_MIC_PIN] = {
+ .type = HDA_FIXUP_PINS,
+ .chained = true,
+ .chain_id = CXT_FIXUP_HEADPHONE_MIC,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x18, 0x03a1913d }, /* use as headphone mic, without its own jack detect */
+ { }
+ }
+ },
+ [CXT_FIXUP_HEADPHONE_MIC] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = cxt_fixup_headphone_mic,
+ },
};
static const struct snd_pci_quirk cxt5051_fixups[] = {
@@ -3311,6 +3384,7 @@
static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
@@ -3395,7 +3469,8 @@
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
- err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0);
+ err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL,
+ spec->parse_flags);
if (err < 0)
goto error;
@@ -3416,6 +3491,8 @@
codec->bus->allow_bus_reset = 1;
}
+ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
+
return 0;
error:
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 030ca86..895a0d3 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -959,6 +959,7 @@
int pin_nid;
int pin_idx;
struct hda_jack_tbl *jack;
+ int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT;
jack = snd_hda_jack_tbl_get_from_tag(codec, tag);
if (!jack)
@@ -967,8 +968,8 @@
jack->jack_dirty = 1;
_snd_printd(SND_PR_VERBOSE,
- "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
- codec->addr, pin_nid,
+ "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n",
+ codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA),
!!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV));
pin_idx = pin_nid_to_pin_index(spec, pin_nid);
@@ -1989,8 +1990,10 @@
return -EINVAL;
}
codec->patch_ops = generic_hdmi_patch_ops;
- if (codec->vendor_id == 0x80862807)
+ if (codec->vendor_id == 0x80862807) {
codec->patch_ops.set_power_state = haswell_set_power_state;
+ codec->dp_mst = true;
+ }
generic_hdmi_init_per_pins(codec);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f303cd8..4a90917 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -282,6 +282,7 @@
{
alc_auto_setup_eapd(codec, false);
msleep(200);
+ snd_hda_shutup_pins(codec);
}
/* generic EAPD initialization */
@@ -826,7 +827,8 @@
if (spec && spec->shutup)
spec->shutup(codec);
- snd_hda_shutup_pins(codec);
+ else
+ snd_hda_shutup_pins(codec);
}
#define alc_free snd_hda_gen_free
@@ -1853,8 +1855,10 @@
const struct hda_fixup *fix, int action)
{
struct alc_spec *spec = codec->spec;
- if (action == HDA_FIXUP_ACT_PRE_PROBE)
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
spec->gen.no_primary_hp = 1;
+ spec->gen.no_multi_io = 1;
+ }
}
static const struct hda_fixup alc882_fixups[] = {
@@ -2533,6 +2537,7 @@
ALC269_TYPE_ALC269VD,
ALC269_TYPE_ALC280,
ALC269_TYPE_ALC282,
+ ALC269_TYPE_ALC283,
ALC269_TYPE_ALC284,
ALC269_TYPE_ALC286,
};
@@ -2558,6 +2563,7 @@
case ALC269_TYPE_ALC269VB:
case ALC269_TYPE_ALC269VD:
case ALC269_TYPE_ALC282:
+ case ALC269_TYPE_ALC283:
case ALC269_TYPE_ALC286:
ssids = alc269_ssids;
break;
@@ -2583,15 +2589,81 @@
{
struct alc_spec *spec = codec->spec;
- if (spec->codec_variant != ALC269_TYPE_ALC269VB)
- return;
-
if (spec->codec_variant == ALC269_TYPE_ALC269VB)
alc269vb_toggle_power_output(codec, 0);
if (spec->codec_variant == ALC269_TYPE_ALC269VB &&
(alc_get_coef0(codec) & 0x00ff) == 0x018) {
msleep(150);
}
+ snd_hda_shutup_pins(codec);
+}
+
+static void alc283_init(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+ bool hp_pin_sense;
+ int val;
+
+ if (!hp_pin)
+ return;
+ hp_pin_sense = snd_hda_jack_detect(codec, hp_pin);
+
+ /* Index 0x43 Direct Drive HP AMP LPM Control 1 */
+ /* Headphone capless set to high power mode */
+ alc_write_coef_idx(codec, 0x43, 0x9004);
+
+ snd_hda_codec_write(codec, hp_pin, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+ if (hp_pin_sense)
+ msleep(85);
+
+ snd_hda_codec_write(codec, hp_pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+
+ if (hp_pin_sense)
+ msleep(85);
+ /* Index 0x46 Combo jack auto switch control 2 */
+ /* 3k pull low control for Headset jack. */
+ val = alc_read_coef_idx(codec, 0x46);
+ alc_write_coef_idx(codec, 0x46, val & ~(3 << 12));
+ /* Headphone capless set to normal mode */
+ alc_write_coef_idx(codec, 0x43, 0x9614);
+}
+
+static void alc283_shutup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+ bool hp_pin_sense;
+ int val;
+
+ if (!hp_pin) {
+ alc269_shutup(codec);
+ return;
+ }
+
+ hp_pin_sense = snd_hda_jack_detect(codec, hp_pin);
+
+ alc_write_coef_idx(codec, 0x43, 0x9004);
+
+ snd_hda_codec_write(codec, hp_pin, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+ if (hp_pin_sense)
+ msleep(85);
+
+ snd_hda_codec_write(codec, hp_pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0);
+
+ val = alc_read_coef_idx(codec, 0x46);
+ alc_write_coef_idx(codec, 0x46, val | (3 << 12));
+
+ if (hp_pin_sense)
+ msleep(85);
+ snd_hda_shutup_pins(codec);
+ alc_write_coef_idx(codec, 0x43, 0x9614);
}
static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg,
@@ -2722,6 +2794,7 @@
hda_call_check_power_status(codec, 0x01);
if (spec->has_alc5505_dsp)
alc5505_dsp_resume(codec);
+
return 0;
}
#endif /* CONFIG_PM */
@@ -3261,6 +3334,28 @@
alc_fixup_headset_mode(codec, fix, action);
}
+/* Returns the nid of the external mic input pin, or 0 if it cannot be found. */
+static int find_ext_mic_pin(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+ hda_nid_t nid;
+ unsigned int defcfg;
+ int i;
+
+ for (i = 0; i < cfg->num_inputs; i++) {
+ if (cfg->inputs[i].type != AUTO_PIN_MIC)
+ continue;
+ nid = cfg->inputs[i].pin;
+ defcfg = snd_hda_codec_get_pincfg(codec, nid);
+ if (snd_hda_get_input_pin_attr(defcfg) == INPUT_PIN_ATTR_INT)
+ continue;
+ return nid;
+ }
+
+ return 0;
+}
+
static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
const struct hda_fixup *fix,
int action)
@@ -3268,11 +3363,12 @@
struct alc_spec *spec = codec->spec;
if (action == HDA_FIXUP_ACT_PROBE) {
- if (snd_BUG_ON(!spec->gen.am_entry[1].pin ||
- !spec->gen.autocfg.hp_pins[0]))
+ int mic_pin = find_ext_mic_pin(codec);
+ int hp_pin = spec->gen.autocfg.hp_pins[0];
+
+ if (snd_BUG_ON(!mic_pin || !hp_pin))
return;
- snd_hda_jack_set_gating_jack(codec, spec->gen.am_entry[1].pin,
- spec->gen.autocfg.hp_pins[0]);
+ snd_hda_jack_set_gating_jack(codec, mic_pin, hp_pin);
}
}
@@ -3308,6 +3404,45 @@
}
}
+static void alc283_hp_automute_hook(struct hda_codec *codec,
+ struct hda_jack_tbl *jack)
+{
+ struct alc_spec *spec = codec->spec;
+ int vref;
+
+ msleep(200);
+ snd_hda_gen_hp_automute(codec, jack);
+
+ vref = spec->gen.hp_jack_present ? PIN_VREF80 : 0;
+
+ msleep(600);
+ snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ vref);
+}
+
+static void alc283_chromebook_caps(struct hda_codec *codec)
+{
+ snd_hda_override_wcaps(codec, 0x03, 0);
+}
+
+static void alc283_fixup_chromebook(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ int val;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ alc283_chromebook_caps(codec);
+ spec->gen.hp_automute_hook = alc283_hp_automute_hook;
+ /* MIC2-VREF control */
+ /* Set to manual mode */
+ val = alc_read_coef_idx(codec, 0x06);
+ alc_write_coef_idx(codec, 0x06, val & ~0x000c);
+ break;
+ }
+}
+
enum {
ALC269_FIXUP_SONY_VAIO,
ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -3344,6 +3479,7 @@
ALC269_FIXUP_ACER_AC700,
ALC269_FIXUP_LIMIT_INT_MIC_BOOST,
ALC269VB_FIXUP_ORDISSIMO_EVE2,
+ ALC283_FIXUP_CHROME_BOOK,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -3595,11 +3731,20 @@
{ }
},
},
+ [ALC283_FIXUP_CHROME_BOOK] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc283_fixup_chromebook,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC),
+ SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),
+ SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
+ SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
+ SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
+ SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05c4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
@@ -3637,6 +3782,7 @@
SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK),
SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED),
SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -3655,11 +3801,6 @@
SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
- SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
- SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),
- SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
- SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
- SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
@@ -3670,8 +3811,16 @@
SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
- SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
+ SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
+ SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
@@ -3840,11 +3989,15 @@
case 0x10ec0290:
spec->codec_variant = ALC269_TYPE_ALC280;
break;
- case 0x10ec0233:
case 0x10ec0282:
- case 0x10ec0283:
spec->codec_variant = ALC269_TYPE_ALC282;
break;
+ case 0x10ec0233:
+ case 0x10ec0283:
+ spec->codec_variant = ALC269_TYPE_ALC283;
+ spec->shutup = alc283_shutup;
+ spec->init_hook = alc283_init;
+ break;
case 0x10ec0284:
case 0x10ec0292:
spec->codec_variant = ALC269_TYPE_ALC284;
@@ -3872,7 +4025,8 @@
codec->patch_ops.suspend = alc269_suspend;
codec->patch_ops.resume = alc269_resume;
#endif
- spec->shutup = alc269_shutup;
+ if (!spec->shutup)
+ spec->shutup = alc269_shutup;
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6d1924c..fba0cef 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -158,6 +158,7 @@
STAC_D965_VERBS,
STAC_DELL_3ST,
STAC_DELL_BIOS,
+ STAC_DELL_BIOS_AMIC,
STAC_DELL_BIOS_SPDIF,
STAC_927X_DELL_DMIC,
STAC_927X_VOLKNOB,
@@ -3231,8 +3232,6 @@
[STAC_DELL_BIOS] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
- /* configure the analog microphone on some laptops */
- { 0x0c, 0x90a79130 },
/* correct the front output jack as a hp out */
{ 0x0f, 0x0221101f },
/* correct the front input jack as a mic */
@@ -3242,6 +3241,16 @@
.chained = true,
.chain_id = STAC_927X_DELL_DMIC,
},
+ [STAC_DELL_BIOS_AMIC] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ /* configure the analog microphone on some laptops */
+ { 0x0c, 0x90a79130 },
+ {}
+ },
+ .chained = true,
+ .chain_id = STAC_DELL_BIOS,
+ },
[STAC_DELL_BIOS_SPDIF] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
@@ -3270,6 +3279,7 @@
{ .id = STAC_D965_5ST_NO_FP, .name = "5stack-no-fp" },
{ .id = STAC_DELL_3ST, .name = "dell-3stack" },
{ .id = STAC_DELL_BIOS, .name = "dell-bios" },
+ { .id = STAC_DELL_BIOS_AMIC, .name = "dell-bios-amic" },
{ .id = STAC_927X_VOLKNOB, .name = "volknob" },
{}
};
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index e2481ba..0bc20ef 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -207,9 +207,9 @@
return;
if (spec->hp_work_active) {
snd_hda_codec_write(codec, 0x1, 0, 0xf81, 1);
+ codec->jackpoll_interval = 0;
cancel_delayed_work_sync(&codec->jackpoll_work);
spec->hp_work_active = false;
- codec->jackpoll_interval = 0;
}
}
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 2a8ad9d..bb9ebc5 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -28,6 +28,7 @@
#include <linux/interrupt.h>
#include <linux/pci.h>
#include <linux/module.h>
+#include <linux/vmalloc.h>
#include <sound/core.h>
#include <sound/info.h>
@@ -198,6 +199,31 @@
#define RME96_AD1852_VOL_BITS 14
#define RME96_AD1855_VOL_BITS 10
+/* Defines for snd_rme96_trigger */
+#define RME96_TB_START_PLAYBACK 1
+#define RME96_TB_START_CAPTURE 2
+#define RME96_TB_STOP_PLAYBACK 4
+#define RME96_TB_STOP_CAPTURE 8
+#define RME96_TB_RESET_PLAYPOS 16
+#define RME96_TB_RESET_CAPTUREPOS 32
+#define RME96_TB_CLEAR_PLAYBACK_IRQ 64
+#define RME96_TB_CLEAR_CAPTURE_IRQ 128
+#define RME96_RESUME_PLAYBACK (RME96_TB_START_PLAYBACK)
+#define RME96_RESUME_CAPTURE (RME96_TB_START_CAPTURE)
+#define RME96_RESUME_BOTH (RME96_RESUME_PLAYBACK \
+ | RME96_RESUME_CAPTURE)
+#define RME96_START_PLAYBACK (RME96_TB_START_PLAYBACK \
+ | RME96_TB_RESET_PLAYPOS)
+#define RME96_START_CAPTURE (RME96_TB_START_CAPTURE \
+ | RME96_TB_RESET_CAPTUREPOS)
+#define RME96_START_BOTH (RME96_START_PLAYBACK \
+ | RME96_START_CAPTURE)
+#define RME96_STOP_PLAYBACK (RME96_TB_STOP_PLAYBACK \
+ | RME96_TB_CLEAR_PLAYBACK_IRQ)
+#define RME96_STOP_CAPTURE (RME96_TB_STOP_CAPTURE \
+ | RME96_TB_CLEAR_CAPTURE_IRQ)
+#define RME96_STOP_BOTH (RME96_STOP_PLAYBACK \
+ | RME96_STOP_CAPTURE)
struct rme96 {
spinlock_t lock;
@@ -214,6 +240,13 @@
u8 rev; /* card revision number */
+#ifdef CONFIG_PM
+ u32 playback_pointer;
+ u32 capture_pointer;
+ void *playback_suspend_buffer;
+ void *capture_suspend_buffer;
+#endif
+
struct snd_pcm_substream *playback_substream;
struct snd_pcm_substream *capture_substream;
@@ -344,6 +377,8 @@
{
.info = (SNDRV_PCM_INFO_MMAP_IOMEM |
SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
@@ -373,6 +408,8 @@
{
.info = (SNDRV_PCM_INFO_MMAP_IOMEM |
SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
@@ -402,6 +439,8 @@
{
.info = (SNDRV_PCM_INFO_MMAP_IOMEM |
SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
@@ -427,6 +466,8 @@
{
.info = (SNDRV_PCM_INFO_MMAP_IOMEM |
SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
@@ -1045,54 +1086,35 @@
}
static void
-snd_rme96_playback_start(struct rme96 *rme96,
- int from_pause)
+snd_rme96_trigger(struct rme96 *rme96,
+ int op)
{
- if (!from_pause) {
+ if (op & RME96_TB_RESET_PLAYPOS)
writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
- }
-
- rme96->wcreg |= RME96_WCR_START;
- writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
-
-static void
-snd_rme96_capture_start(struct rme96 *rme96,
- int from_pause)
-{
- if (!from_pause) {
+ if (op & RME96_TB_RESET_CAPTUREPOS)
writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
+ if (op & RME96_TB_CLEAR_PLAYBACK_IRQ) {
+ rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ if (rme96->rcreg & RME96_RCR_IRQ)
+ writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ);
}
-
- rme96->wcreg |= RME96_WCR_START_2;
+ if (op & RME96_TB_CLEAR_CAPTURE_IRQ) {
+ rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ if (rme96->rcreg & RME96_RCR_IRQ_2)
+ writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ);
+ }
+ if (op & RME96_TB_START_PLAYBACK)
+ rme96->wcreg |= RME96_WCR_START;
+ if (op & RME96_TB_STOP_PLAYBACK)
+ rme96->wcreg &= ~RME96_WCR_START;
+ if (op & RME96_TB_START_CAPTURE)
+ rme96->wcreg |= RME96_WCR_START_2;
+ if (op & RME96_TB_STOP_CAPTURE)
+ rme96->wcreg &= ~RME96_WCR_START_2;
writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
}
-static void
-snd_rme96_playback_stop(struct rme96 *rme96)
-{
- /*
- * Check if there is an unconfirmed IRQ, if so confirm it, or else
- * the hardware will not stop generating interrupts
- */
- rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
- if (rme96->rcreg & RME96_RCR_IRQ) {
- writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ);
- }
- rme96->wcreg &= ~RME96_WCR_START;
- writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
-static void
-snd_rme96_capture_stop(struct rme96 *rme96)
-{
- rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
- if (rme96->rcreg & RME96_RCR_IRQ_2) {
- writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ);
- }
- rme96->wcreg &= ~RME96_WCR_START_2;
- writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
static irqreturn_t
snd_rme96_interrupt(int irq,
@@ -1155,6 +1177,7 @@
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_set_sync(substream);
spin_lock_irq(&rme96->lock);
if (rme96->playback_substream != NULL) {
spin_unlock_irq(&rme96->lock);
@@ -1191,6 +1214,7 @@
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_set_sync(substream);
runtime->hw = snd_rme96_capture_spdif_info;
if (snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG &&
(rate = snd_rme96_capture_getrate(rme96, &isadat)) > 0)
@@ -1222,6 +1246,7 @@
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_set_sync(substream);
spin_lock_irq(&rme96->lock);
if (rme96->playback_substream != NULL) {
spin_unlock_irq(&rme96->lock);
@@ -1253,6 +1278,7 @@
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_set_sync(substream);
runtime->hw = snd_rme96_capture_adat_info;
if (snd_rme96_getinputtype(rme96) == RME96_INPUT_ANALOG) {
/* makes no sense to use analog input. Note that analog
@@ -1288,7 +1314,7 @@
spin_lock_irq(&rme96->lock);
if (RME96_ISPLAYING(rme96)) {
- snd_rme96_playback_stop(rme96);
+ snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK);
}
rme96->playback_substream = NULL;
rme96->playback_periodsize = 0;
@@ -1309,7 +1335,7 @@
spin_lock_irq(&rme96->lock);
if (RME96_ISRECORDING(rme96)) {
- snd_rme96_capture_stop(rme96);
+ snd_rme96_trigger(rme96, RME96_STOP_CAPTURE);
}
rme96->capture_substream = NULL;
rme96->capture_periodsize = 0;
@@ -1324,7 +1350,7 @@
spin_lock_irq(&rme96->lock);
if (RME96_ISPLAYING(rme96)) {
- snd_rme96_playback_stop(rme96);
+ snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK);
}
writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
spin_unlock_irq(&rme96->lock);
@@ -1338,7 +1364,7 @@
spin_lock_irq(&rme96->lock);
if (RME96_ISRECORDING(rme96)) {
- snd_rme96_capture_stop(rme96);
+ snd_rme96_trigger(rme96, RME96_STOP_CAPTURE);
}
writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
spin_unlock_irq(&rme96->lock);
@@ -1350,41 +1376,55 @@
int cmd)
{
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_substream *s;
+ bool sync;
+
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (snd_pcm_substream_chip(s) == rme96)
+ snd_pcm_trigger_done(s, substream);
+ }
+
+ sync = (rme96->playback_substream && rme96->capture_substream) &&
+ (rme96->playback_substream->group ==
+ rme96->capture_substream->group);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
if (!RME96_ISPLAYING(rme96)) {
- if (substream != rme96->playback_substream) {
+ if (substream != rme96->playback_substream)
return -EBUSY;
- }
- snd_rme96_playback_start(rme96, 0);
+ snd_rme96_trigger(rme96, sync ? RME96_START_BOTH
+ : RME96_START_PLAYBACK);
}
break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_STOP:
if (RME96_ISPLAYING(rme96)) {
- if (substream != rme96->playback_substream) {
+ if (substream != rme96->playback_substream)
return -EBUSY;
- }
- snd_rme96_playback_stop(rme96);
+ snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+ : RME96_STOP_PLAYBACK);
}
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- if (RME96_ISPLAYING(rme96)) {
- snd_rme96_playback_stop(rme96);
- }
+ if (RME96_ISPLAYING(rme96))
+ snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+ : RME96_STOP_PLAYBACK);
break;
+ case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- if (!RME96_ISPLAYING(rme96)) {
- snd_rme96_playback_start(rme96, 1);
- }
+ if (!RME96_ISPLAYING(rme96))
+ snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH
+ : RME96_RESUME_PLAYBACK);
break;
-
+
default:
return -EINVAL;
}
+
return 0;
}
@@ -1393,38 +1433,51 @@
int cmd)
{
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_substream *s;
+ bool sync;
+
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (snd_pcm_substream_chip(s) == rme96)
+ snd_pcm_trigger_done(s, substream);
+ }
+
+ sync = (rme96->playback_substream && rme96->capture_substream) &&
+ (rme96->playback_substream->group ==
+ rme96->capture_substream->group);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
if (!RME96_ISRECORDING(rme96)) {
- if (substream != rme96->capture_substream) {
+ if (substream != rme96->capture_substream)
return -EBUSY;
- }
- snd_rme96_capture_start(rme96, 0);
+ snd_rme96_trigger(rme96, sync ? RME96_START_BOTH
+ : RME96_START_CAPTURE);
}
break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_STOP:
if (RME96_ISRECORDING(rme96)) {
- if (substream != rme96->capture_substream) {
+ if (substream != rme96->capture_substream)
return -EBUSY;
- }
- snd_rme96_capture_stop(rme96);
+ snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+ : RME96_STOP_CAPTURE);
}
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- if (RME96_ISRECORDING(rme96)) {
- snd_rme96_capture_stop(rme96);
- }
+ if (RME96_ISRECORDING(rme96))
+ snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+ : RME96_STOP_CAPTURE);
break;
+ case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- if (!RME96_ISRECORDING(rme96)) {
- snd_rme96_capture_start(rme96, 1);
- }
+ if (!RME96_ISRECORDING(rme96))
+ snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH
+ : RME96_RESUME_CAPTURE);
break;
-
+
default:
return -EINVAL;
}
@@ -1505,8 +1558,7 @@
return;
}
if (rme96->irq >= 0) {
- snd_rme96_playback_stop(rme96);
- snd_rme96_capture_stop(rme96);
+ snd_rme96_trigger(rme96, RME96_STOP_BOTH);
rme96->areg &= ~RME96_AR_DAC_EN;
writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
free_irq(rme96->irq, (void *)rme96);
@@ -1520,6 +1572,10 @@
pci_release_regions(rme96->pci);
rme96->port = 0;
}
+#ifdef CONFIG_PM
+ vfree(rme96->playback_suspend_buffer);
+ vfree(rme96->capture_suspend_buffer);
+#endif
pci_disable_device(rme96->pci);
}
@@ -1606,8 +1662,7 @@
rme96->capture_periodsize = 0;
/* make sure playback/capture is stopped, if by some reason active */
- snd_rme96_playback_stop(rme96);
- snd_rme96_capture_stop(rme96);
+ snd_rme96_trigger(rme96, RME96_STOP_BOTH);
/* set default values in registers */
rme96->wcreg =
@@ -2319,6 +2374,87 @@
* Card initialisation
*/
+#ifdef CONFIG_PM
+
+static int
+snd_rme96_suspend(struct pci_dev *pci,
+ pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct rme96 *rme96 = card->private_data;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend(rme96->playback_substream);
+ snd_pcm_suspend(rme96->capture_substream);
+
+ /* save capture & playback pointers */
+ rme96->playback_pointer = readl(rme96->iobase + RME96_IO_GET_PLAY_POS)
+ & RME96_RCR_AUDIO_ADDR_MASK;
+ rme96->capture_pointer = readl(rme96->iobase + RME96_IO_GET_REC_POS)
+ & RME96_RCR_AUDIO_ADDR_MASK;
+
+ /* save playback and capture buffers */
+ memcpy_fromio(rme96->playback_suspend_buffer,
+ rme96->iobase + RME96_IO_PLAY_BUFFER, RME96_BUFFER_SIZE);
+ memcpy_fromio(rme96->capture_suspend_buffer,
+ rme96->iobase + RME96_IO_REC_BUFFER, RME96_BUFFER_SIZE);
+
+ /* disable the DAC */
+ rme96->areg &= ~RME96_AR_DAC_EN;
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+
+ return 0;
+}
+
+static int
+snd_rme96_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct rme96 *rme96 = card->private_data;
+
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "rme96: pci_enable_device failed, disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+
+ /* reset playback and record buffer pointers */
+ writel(0, rme96->iobase + RME96_IO_SET_PLAY_POS
+ + rme96->playback_pointer);
+ writel(0, rme96->iobase + RME96_IO_SET_REC_POS
+ + rme96->capture_pointer);
+
+ /* restore playback and capture buffers */
+ memcpy_toio(rme96->iobase + RME96_IO_PLAY_BUFFER,
+ rme96->playback_suspend_buffer, RME96_BUFFER_SIZE);
+ memcpy_toio(rme96->iobase + RME96_IO_REC_BUFFER,
+ rme96->capture_suspend_buffer, RME96_BUFFER_SIZE);
+
+ /* reset the ADC */
+ writel(rme96->areg | RME96_AR_PD2,
+ rme96->iobase + RME96_IO_ADDITIONAL_REG);
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+ /* reset and enable DAC, restore analog volume */
+ snd_rme96_reset_dac(rme96);
+ rme96->areg |= RME96_AR_DAC_EN;
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+ if (RME96_HAS_ANALOG_OUT(rme96)) {
+ usleep_range(3000, 10000);
+ snd_rme96_apply_dac_volume(rme96);
+ }
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+
+ return 0;
+}
+
+#endif
+
static void snd_rme96_card_free(struct snd_card *card)
{
snd_rme96_free(card->private_data);
@@ -2355,6 +2491,23 @@
return err;
}
+#ifdef CONFIG_PM
+ rme96->playback_suspend_buffer = vmalloc(RME96_BUFFER_SIZE);
+ if (!rme96->playback_suspend_buffer) {
+ snd_printk(KERN_ERR
+ "Failed to allocate playback suspend buffer!\n");
+ snd_card_free(card);
+ return -ENOMEM;
+ }
+ rme96->capture_suspend_buffer = vmalloc(RME96_BUFFER_SIZE);
+ if (!rme96->capture_suspend_buffer) {
+ snd_printk(KERN_ERR
+ "Failed to allocate capture suspend buffer!\n");
+ snd_card_free(card);
+ return -ENOMEM;
+ }
+#endif
+
strcpy(card->driver, "Digi96");
switch (rme96->pci->device) {
case PCI_DEVICE_ID_RME_DIGI96:
@@ -2397,6 +2550,10 @@
.id_table = snd_rme96_ids,
.probe = snd_rme96_probe,
.remove = snd_rme96_remove,
+#ifdef CONFIG_PM
+ .suspend = snd_rme96_suspend,
+ .resume = snd_rme96_resume,
+#endif
};
module_pci_driver(rme96_driver);
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index bd50193..3cde55b 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -38,6 +38,97 @@
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
+
+/* ************* Register Documentation *******************************************************
+ *
+ * Work in progress! Documentation is based on the code in this file.
+ *
+ * --------- HDSPM_controlRegister ---------
+ * :7654.3210:7654.3210:7654.3210:7654.3210: bit number per byte
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number
+ * :1098.7654:3210.9876:5432.1098:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ * : . : . : . : x . : HDSPM_AudioInterruptEnable \_ setting both bits
+ * : . : . : . : . x: HDSPM_Start / enables audio IO
+ * : . : . : . : x. : HDSPM_ClockModeMaster - 1: Master, 0: Slave
+ * : . : . : . : .210 : HDSPM_LatencyMask - 3 Bit value for latency
+ * : . : . : . : . : 0:64, 1:128, 2:256, 3:512,
+ * : . : . : . : . : 4:1024, 5:2048, 6:4096, 7:8192
+ * :x . : . : . x:xx . : HDSPM_FrequencyMask
+ * : . : . : . :10 . : HDSPM_Frequency1|HDSPM_Frequency0: 1=32K,2=44.1K,3=48K,0=??
+ * : . : . : . x: . : <MADI> HDSPM_DoubleSpeed
+ * :x . : . : . : . : <MADI> HDSPM_QuadSpeed
+ * : . 3 : . 10: 2 . : . : HDSPM_SyncRefMask :
+ * : . : . x: . : . : HDSPM_SyncRef0
+ * : . : . x : . : . : HDSPM_SyncRef1
+ * : . : . : x . : . : <AES32> HDSPM_SyncRef2
+ * : . x : . : . : . : <AES32> HDSPM_SyncRef3
+ * : . : . 10: . : . : <MADI> sync ref: 0:WC, 1:Madi, 2:TCO, 3:SyncIn
+ * : . 3 : . 10: 2 . : . : <AES32> 0:WC, 1:AES1 ... 8:AES8, 9: TCO, 10:SyncIn?
+ * : . x : . : . : . : <MADIe> HDSPe_FLOAT_FORMAT
+ * : . : . : x . : . : <MADI> HDSPM_InputSelect0 : 0=optical,1=coax
+ * : . : . :x . : . : <MADI> HDSPM_InputSelect1
+ * : . : .x : . : . : <MADI> HDSPM_clr_tms
+ * : . : . : . x : . : <MADI> HDSPM_TX_64ch
+ * : . : . : . x : . : <AES32> HDSPM_Emphasis
+ * : . : . : .x : . : <MADI> HDSPM_AutoInp
+ * : . : . x : . : . : <MADI> HDSPM_SMUX
+ * : . : .x : . : . : <MADI> HDSPM_clr_tms
+ * : . : x. : . : . : <MADI> HDSPM_taxi_reset
+ * : . x: . : . : . : <MADI> HDSPM_LineOut
+ * : . x: . : . : . : <AES32> ??????????????????
+ * : . : x. : . : . : <AES32> HDSPM_WCK48
+ * : . : . : .x : . : <AES32> HDSPM_Dolby
+ * : . : x . : . : . : HDSPM_Midi0InterruptEnable
+ * : . :x . : . : . : HDSPM_Midi1InterruptEnable
+ * : . : x . : . : . : HDSPM_Midi2InterruptEnable
+ * : . x : . : . : . : <MADI> HDSPM_Midi3InterruptEnable
+ * : . x : . : . : . : <AES32> HDSPM_DS_DoubleWire
+ * : .x : . : . : . : <AES32> HDSPM_QS_DoubleWire
+ * : x. : . : . : . : <AES32> HDSPM_QS_QuadWire
+ * : . : . : . x : . : <AES32> HDSPM_Professional
+ * : x . : . : . : . : HDSPM_wclk_sel
+ * : . : . : . : . :
+ * :7654.3210:7654.3210:7654.3210:7654.3210: bit number per byte
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number
+ * :1098.7654:3210.9876:5432.1098:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421:hex digit
+ *
+ *
+ *
+ * AIO / RayDAT only
+ *
+ * ------------ HDSPM_WR_SETTINGS ----------
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number per byte
+ * :1098.7654:3210.9876:5432.1098:7654.3210:
+ * :||||.||||:||||.||||:||||.||||:||||.||||: bit number
+ * :7654.3210:7654.3210:7654.3210:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ * : . : . : . : . x: HDSPM_c0Master 1: Master, 0: Slave
+ * : . : . : . : . x : HDSPM_c0_SyncRef0
+ * : . : . : . : . x : HDSPM_c0_SyncRef1
+ * : . : . : . : .x : HDSPM_c0_SyncRef2
+ * : . : . : . : x. : HDSPM_c0_SyncRef3
+ * : . : . : . : 3.210 : HDSPM_c0_SyncRefMask:
+ * : . : . : . : . : RayDat: 0:WC, 1:AES, 2:SPDIF, 3..6: ADAT1..4,
+ * : . : . : . : . : 9:TCO, 10:SyncIn
+ * : . : . : . : . : AIO: 0:WC, 1:AES, 2: SPDIF, 3: ATAT,
+ * : . : . : . : . : 9:TCO, 10:SyncIn
+ * : . : . : . : . :
+ * : . : . : . : . :
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number per byte
+ * :1098.7654:3210.9876:5432.1098:7654.3210:
+ * :||||.||||:||||.||||:||||.||||:||||.||||: bit number
+ * :7654.3210:7654.3210:7654.3210:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ *
+ */
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/interrupt.h>
@@ -95,7 +186,7 @@
#define HDSPM_controlRegister 64
#define HDSPM_interruptConfirmation 96
#define HDSPM_control2Reg 256 /* not in specs ???????? */
-#define HDSPM_freqReg 256 /* for AES32 */
+#define HDSPM_freqReg 256 /* for setting arbitrary clock values (DDS feature) */
#define HDSPM_midiDataOut0 352 /* just believe in old code */
#define HDSPM_midiDataOut1 356
#define HDSPM_eeprom_wr 384 /* for AES32 */
@@ -258,6 +349,25 @@
#define HDSPM_wclk_sel (1<<30)
+/* additional control register bits for AIO*/
+#define HDSPM_c0_Wck48 0x20 /* also RayDAT */
+#define HDSPM_c0_Input0 0x1000
+#define HDSPM_c0_Input1 0x2000
+#define HDSPM_c0_Spdif_Opt 0x4000
+#define HDSPM_c0_Pro 0x8000
+#define HDSPM_c0_clr_tms 0x10000
+#define HDSPM_c0_AEB1 0x20000
+#define HDSPM_c0_AEB2 0x40000
+#define HDSPM_c0_LineOut 0x80000
+#define HDSPM_c0_AD_GAIN0 0x100000
+#define HDSPM_c0_AD_GAIN1 0x200000
+#define HDSPM_c0_DA_GAIN0 0x400000
+#define HDSPM_c0_DA_GAIN1 0x800000
+#define HDSPM_c0_PH_GAIN0 0x1000000
+#define HDSPM_c0_PH_GAIN1 0x2000000
+#define HDSPM_c0_Sym6db 0x4000000
+
+
/* --- bit helper defines */
#define HDSPM_LatencyMask (HDSPM_Latency0|HDSPM_Latency1|HDSPM_Latency2)
#define HDSPM_FrequencyMask (HDSPM_Frequency0|HDSPM_Frequency1|\
@@ -341,11 +451,11 @@
#define HDSPM_madiLock (1<<3) /* MADI Locked =1, no=0 */
#define HDSPM_madiSync (1<<18) /* MADI is in sync */
-#define HDSPM_tcoLock 0x00000020 /* Optional TCO locked status FOR HDSPe MADI! */
-#define HDSPM_tcoSync 0x10000000 /* Optional TCO sync status */
+#define HDSPM_tcoLockMadi 0x00000020 /* Optional TCO locked status for HDSPe MADI*/
+#define HDSPM_tcoSync 0x10000000 /* Optional TCO sync status for HDSPe MADI and AES32!*/
-#define HDSPM_syncInLock 0x00010000 /* Sync In lock status FOR HDSPe MADI! */
-#define HDSPM_syncInSync 0x00020000 /* Sync In sync status FOR HDSPe MADI! */
+#define HDSPM_syncInLock 0x00010000 /* Sync In lock status for HDSPe MADI! */
+#define HDSPM_syncInSync 0x00020000 /* Sync In sync status for HDSPe MADI! */
#define HDSPM_BufferPositionMask 0x000FFC0 /* Bit 6..15 : h/w buffer pointer */
/* since 64byte accurate, last 6 bits are not used */
@@ -363,7 +473,7 @@
* Interrupt
*/
#define HDSPM_tco_detect 0x08000000
-#define HDSPM_tco_lock 0x20000000
+#define HDSPM_tcoLockAes 0x20000000 /* Optional TCO locked status for HDSPe AES */
#define HDSPM_s2_tco_detect 0x00000040
#define HDSPM_s2_AEBO_D 0x00000080
@@ -461,7 +571,9 @@
#define HDSPM_AES32_AUTOSYNC_FROM_AES6 6
#define HDSPM_AES32_AUTOSYNC_FROM_AES7 7
#define HDSPM_AES32_AUTOSYNC_FROM_AES8 8
-#define HDSPM_AES32_AUTOSYNC_FROM_NONE 9
+#define HDSPM_AES32_AUTOSYNC_FROM_TCO 9
+#define HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN 10
+#define HDSPM_AES32_AUTOSYNC_FROM_NONE 11
/* status2 */
/* HDSPM_LockAES_bit is given by HDSPM_LockAES >> (AES# - 1) */
@@ -537,36 +649,39 @@
/* names for speed modes */
static char *hdspm_speed_names[] = { "single", "double", "quad" };
-static char *texts_autosync_aes_tco[] = { "Word Clock",
+static const char *const texts_autosync_aes_tco[] = { "Word Clock",
"AES1", "AES2", "AES3", "AES4",
"AES5", "AES6", "AES7", "AES8",
- "TCO" };
-static char *texts_autosync_aes[] = { "Word Clock",
+ "TCO", "Sync In"
+};
+static const char *const texts_autosync_aes[] = { "Word Clock",
"AES1", "AES2", "AES3", "AES4",
- "AES5", "AES6", "AES7", "AES8" };
-static char *texts_autosync_madi_tco[] = { "Word Clock",
+ "AES5", "AES6", "AES7", "AES8",
+ "Sync In"
+};
+static const char *const texts_autosync_madi_tco[] = { "Word Clock",
"MADI", "TCO", "Sync In" };
-static char *texts_autosync_madi[] = { "Word Clock",
+static const char *const texts_autosync_madi[] = { "Word Clock",
"MADI", "Sync In" };
-static char *texts_autosync_raydat_tco[] = {
+static const char *const texts_autosync_raydat_tco[] = {
"Word Clock",
"ADAT 1", "ADAT 2", "ADAT 3", "ADAT 4",
"AES", "SPDIF", "TCO", "Sync In"
};
-static char *texts_autosync_raydat[] = {
+static const char *const texts_autosync_raydat[] = {
"Word Clock",
"ADAT 1", "ADAT 2", "ADAT 3", "ADAT 4",
"AES", "SPDIF", "Sync In"
};
-static char *texts_autosync_aio_tco[] = {
+static const char *const texts_autosync_aio_tco[] = {
"Word Clock",
"ADAT", "AES", "SPDIF", "TCO", "Sync In"
};
-static char *texts_autosync_aio[] = { "Word Clock",
+static const char *const texts_autosync_aio[] = { "Word Clock",
"ADAT", "AES", "SPDIF", "Sync In" };
-static char *texts_freq[] = {
+static const char *const texts_freq[] = {
"No Lock",
"32 kHz",
"44.1 kHz",
@@ -629,7 +744,8 @@
"AES.L", "AES.R",
"SPDIF.L", "SPDIF.R",
"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6",
- "ADAT.7", "ADAT.8"
+ "ADAT.7", "ADAT.8",
+ "AEB.1", "AEB.2", "AEB.3", "AEB.4"
};
static char *texts_ports_aio_out_ss[] = {
@@ -638,14 +754,16 @@
"SPDIF.L", "SPDIF.R",
"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6",
"ADAT.7", "ADAT.8",
- "Phone.L", "Phone.R"
+ "Phone.L", "Phone.R",
+ "AEB.1", "AEB.2", "AEB.3", "AEB.4"
};
static char *texts_ports_aio_in_ds[] = {
"Analogue.L", "Analogue.R",
"AES.L", "AES.R",
"SPDIF.L", "SPDIF.R",
- "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4"
+ "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
+ "AEB.1", "AEB.2", "AEB.3", "AEB.4"
};
static char *texts_ports_aio_out_ds[] = {
@@ -653,14 +771,16 @@
"AES.L", "AES.R",
"SPDIF.L", "SPDIF.R",
"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
- "Phone.L", "Phone.R"
+ "Phone.L", "Phone.R",
+ "AEB.1", "AEB.2", "AEB.3", "AEB.4"
};
static char *texts_ports_aio_in_qs[] = {
"Analogue.L", "Analogue.R",
"AES.L", "AES.R",
"SPDIF.L", "SPDIF.R",
- "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4"
+ "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
+ "AEB.1", "AEB.2", "AEB.3", "AEB.4"
};
static char *texts_ports_aio_out_qs[] = {
@@ -668,7 +788,8 @@
"AES.L", "AES.R",
"SPDIF.L", "SPDIF.R",
"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
- "Phone.L", "Phone.R"
+ "Phone.L", "Phone.R",
+ "AEB.1", "AEB.2", "AEB.3", "AEB.4"
};
static char *texts_ports_aes32[] = {
@@ -745,8 +866,8 @@
8, 9, /* aes in, */
10, 11, /* spdif in */
12, 13, 14, 15, 16, 17, 18, 19, /* ADAT in */
- -1, -1,
- -1, -1, -1, -1, -1, -1, -1, -1,
+ 2, 3, 4, 5, /* AEB */
+ -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
@@ -760,7 +881,8 @@
10, 11, /* spdif out */
12, 13, 14, 15, 16, 17, 18, 19, /* ADAT out */
6, 7, /* phone out */
- -1, -1, -1, -1, -1, -1, -1, -1,
+ 2, 3, 4, 5, /* AEB */
+ -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
@@ -773,7 +895,8 @@
8, 9, /* aes in */
10, 11, /* spdif in */
12, 14, 16, 18, /* adat in */
- -1, -1, -1, -1, -1, -1,
+ 2, 3, 4, 5, /* AEB */
+ -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
@@ -788,7 +911,7 @@
10, 11, /* spdif out */
12, 14, 16, 18, /* adat out */
6, 7, /* phone out */
- -1, -1, -1, -1,
+ 2, 3, 4, 5, /* AEB */
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
@@ -802,7 +925,8 @@
8, 9, /* aes in */
10, 11, /* spdif in */
12, 16, /* adat in */
- -1, -1, -1, -1, -1, -1, -1, -1,
+ 2, 3, 4, 5, /* AEB */
+ -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
@@ -817,7 +941,8 @@
10, 11, /* spdif out */
12, 16, /* adat out */
6, 7, /* phone out */
- -1, -1, -1, -1, -1, -1,
+ 2, 3, 4, 5, /* AEB */
+ -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
@@ -856,11 +981,11 @@
};
struct hdspm_tco {
- int input;
- int framerate;
- int wordclock;
- int samplerate;
- int pull;
+ int input; /* 0: LTC, 1:Video, 2: WC*/
+ int framerate; /* 0=24, 1=25, 2=29.97, 3=29.97d, 4=30, 5=30d */
+ int wordclock; /* 0=1:1, 1=44.1->48, 2=48->44.1 */
+ int samplerate; /* 0=44.1, 1=48, 2= freq from app */
+ int pull; /* 0=0, 1=+0.1%, 2=-0.1%, 3=+4%, 4=-4%*/
int term; /* 0 = off, 1 = on */
};
@@ -879,7 +1004,7 @@
u32 control_register; /* cached value */
u32 control2_register; /* cached value */
- u32 settings_register;
+ u32 settings_register; /* cached value for AIO / RayDat (sync reference, master/slave) */
struct hdspm_midi midi[4];
struct tasklet_struct midi_tasklet;
@@ -941,7 +1066,7 @@
struct hdspm_tco *tco; /* NULL if no TCO detected */
- char **texts_autosync;
+ const char *const *texts_autosync;
int texts_autosync_items;
cycles_t last_interrupt;
@@ -976,12 +1101,24 @@
static inline int hdspm_get_pll_freq(struct hdspm *hdspm);
static int hdspm_update_simple_mixer_controls(struct hdspm *hdspm);
static int hdspm_autosync_ref(struct hdspm *hdspm);
+static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out);
static int snd_hdspm_set_defaults(struct hdspm *hdspm);
static int hdspm_system_clock_mode(struct hdspm *hdspm);
static void hdspm_set_sgbuf(struct hdspm *hdspm,
struct snd_pcm_substream *substream,
unsigned int reg, int channels);
+static int hdspm_aes_sync_check(struct hdspm *hdspm, int idx);
+static int hdspm_wc_sync_check(struct hdspm *hdspm);
+static int hdspm_tco_sync_check(struct hdspm *hdspm);
+static int hdspm_sync_in_sync_check(struct hdspm *hdspm);
+
+static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index);
+static int hdspm_get_tco_sample_rate(struct hdspm *hdspm);
+static int hdspm_get_wc_sample_rate(struct hdspm *hdspm);
+
+
+
static inline int HDSPM_bit2freq(int n)
{
static const int bit2freq_tab[] = {
@@ -992,6 +1129,12 @@
return bit2freq_tab[n];
}
+static bool hdspm_is_raydat_or_aio(struct hdspm *hdspm)
+{
+ return ((AIO == hdspm->io_type) || (RayDAT == hdspm->io_type));
+}
+
+
/* Write/read to/from HDSPM with Adresses in Bytes
not words but only 32Bit writes are allowed */
@@ -1107,14 +1250,11 @@
else if (hdspm->control_register &
HDSPM_DoubleSpeed)
return rate * 2;
- };
+ }
return rate;
}
-static int hdspm_tco_sync_check(struct hdspm *hdspm);
-static int hdspm_sync_in_sync_check(struct hdspm *hdspm);
-
-/* check for external sample rate */
+/* check for external sample rate, returns the sample rate in Hz*/
static int hdspm_external_sample_rate(struct hdspm *hdspm)
{
unsigned int status, status2, timecode;
@@ -1127,17 +1267,36 @@
timecode = hdspm_read(hdspm, HDSPM_timecodeRegister);
syncref = hdspm_autosync_ref(hdspm);
+ switch (syncref) {
+ case HDSPM_AES32_AUTOSYNC_FROM_WORD:
+ /* Check WC sync and get sample rate */
+ if (hdspm_wc_sync_check(hdspm))
+ return HDSPM_bit2freq(hdspm_get_wc_sample_rate(hdspm));
+ break;
- if (syncref == HDSPM_AES32_AUTOSYNC_FROM_WORD &&
- status & HDSPM_AES32_wcLock)
- return HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF);
+ case HDSPM_AES32_AUTOSYNC_FROM_AES1:
+ case HDSPM_AES32_AUTOSYNC_FROM_AES2:
+ case HDSPM_AES32_AUTOSYNC_FROM_AES3:
+ case HDSPM_AES32_AUTOSYNC_FROM_AES4:
+ case HDSPM_AES32_AUTOSYNC_FROM_AES5:
+ case HDSPM_AES32_AUTOSYNC_FROM_AES6:
+ case HDSPM_AES32_AUTOSYNC_FROM_AES7:
+ case HDSPM_AES32_AUTOSYNC_FROM_AES8:
+ /* Check AES sync and get sample rate */
+ if (hdspm_aes_sync_check(hdspm, syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1))
+ return HDSPM_bit2freq(hdspm_get_aes_sample_rate(hdspm,
+ syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1));
+ break;
- if (syncref >= HDSPM_AES32_AUTOSYNC_FROM_AES1 &&
- syncref <= HDSPM_AES32_AUTOSYNC_FROM_AES8 &&
- status2 & (HDSPM_LockAES >>
- (syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1)))
- return HDSPM_bit2freq((timecode >> (4*(syncref-HDSPM_AES32_AUTOSYNC_FROM_AES1))) & 0xF);
- return 0;
+
+ case HDSPM_AES32_AUTOSYNC_FROM_TCO:
+ /* Check TCO sync and get sample rate */
+ if (hdspm_tco_sync_check(hdspm))
+ return HDSPM_bit2freq(hdspm_get_tco_sample_rate(hdspm));
+ break;
+ default:
+ return 0;
+ } /* end switch(syncref) */
break;
case MADIface:
@@ -2129,6 +2288,9 @@
status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);
return (status >> 16) & 0xF;
break;
+ case AES32:
+ status = hdspm_read(hdspm, HDSPM_statusRegister);
+ return (status >> HDSPM_AES32_wcFreq_bit) & 0xF;
default:
break;
}
@@ -2152,6 +2314,9 @@
status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);
return (status >> 20) & 0xF;
break;
+ case AES32:
+ status = hdspm_read(hdspm, HDSPM_statusRegister);
+ return (status >> 1) & 0xF;
default:
break;
}
@@ -2183,6 +2348,23 @@
return 0;
}
+/**
+ * Returns the AES sample rate class for the given card.
+ **/
+static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index)
+{
+ int timecode;
+
+ switch (hdspm->io_type) {
+ case AES32:
+ timecode = hdspm_read(hdspm, HDSPM_timecodeRegister);
+ return (timecode >> (4*index)) & 0xF;
+ break;
+ default:
+ break;
+ }
+ return 0;
+}
/**
* Returns the sample rate class for input source <idx> for
@@ -2196,16 +2378,24 @@
}
#define ENUMERATED_CTL_INFO(info, texts) \
-{ \
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; \
- uinfo->count = 1; \
- uinfo->value.enumerated.items = ARRAY_SIZE(texts); \
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) \
- uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; \
- strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); \
-}
+ snd_ctl_enum_info(info, 1, ARRAY_SIZE(texts), texts)
+/* Helper function to query the external sample rate and return the
+ * corresponding enum to be returned to userspace.
+ */
+static int hdspm_external_rate_to_enum(struct hdspm *hdspm)
+{
+ int rate = hdspm_external_sample_rate(hdspm);
+ int i, selected_rate = 0;
+ for (i = 1; i < 10; i++)
+ if (HDSPM_bit2freq(i) == rate) {
+ selected_rate = i;
+ break;
+ }
+ return selected_rate;
+}
+
#define HDSPM_AUTOSYNC_SAMPLE_RATE(xname, xindex) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
@@ -2270,7 +2460,7 @@
default:
ucontrol->value.enumerated.item[0] =
hdspm_get_s1_sample_rate(hdspm,
- ucontrol->id.index-1);
+ kcontrol->private_value-1);
}
break;
@@ -2289,28 +2479,24 @@
ucontrol->value.enumerated.item[0] =
hdspm_get_sync_in_sample_rate(hdspm);
break;
+ case 11: /* External Rate */
+ ucontrol->value.enumerated.item[0] =
+ hdspm_external_rate_to_enum(hdspm);
+ break;
default: /* AES1 to AES8 */
ucontrol->value.enumerated.item[0] =
- hdspm_get_s1_sample_rate(hdspm,
- kcontrol->private_value-1);
+ hdspm_get_aes_sample_rate(hdspm,
+ kcontrol->private_value -
+ HDSPM_AES32_AUTOSYNC_FROM_AES1);
break;
}
break;
case MADI:
case MADIface:
- {
- int rate = hdspm_external_sample_rate(hdspm);
- int i, selected_rate = 0;
- for (i = 1; i < 10; i++)
- if (HDSPM_bit2freq(i) == rate) {
- selected_rate = i;
- break;
- }
- ucontrol->value.enumerated.item[0] = selected_rate;
- }
+ ucontrol->value.enumerated.item[0] =
+ hdspm_external_rate_to_enum(hdspm);
break;
-
default:
break;
}
@@ -2359,33 +2545,17 @@
**/
static void hdspm_set_system_clock_mode(struct hdspm *hdspm, int mode)
{
- switch (hdspm->io_type) {
- case AIO:
- case RayDAT:
- if (0 == mode)
- hdspm->settings_register |= HDSPM_c0Master;
- else
- hdspm->settings_register &= ~HDSPM_c0Master;
-
- hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
- break;
-
- default:
- if (0 == mode)
- hdspm->control_register |= HDSPM_ClockModeMaster;
- else
- hdspm->control_register &= ~HDSPM_ClockModeMaster;
-
- hdspm_write(hdspm, HDSPM_controlRegister,
- hdspm->control_register);
- }
+ hdspm_set_toggle_setting(hdspm,
+ (hdspm_is_raydat_or_aio(hdspm)) ?
+ HDSPM_c0Master : HDSPM_ClockModeMaster,
+ (0 == mode));
}
static int snd_hdspm_info_system_clock_mode(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "Master", "AutoSync" };
+ static const char *const texts[] = { "Master", "AutoSync" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -2809,16 +2979,7 @@
{
struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = hdspm->texts_autosync_items;
-
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
-
- strcpy(uinfo->value.enumerated.name,
- hdspm->texts_autosync[uinfo->value.enumerated.item]);
+ snd_ctl_enum_info(uinfo, 1, hdspm->texts_autosync_items, hdspm->texts_autosync);
return 0;
}
@@ -2873,19 +3034,20 @@
static int hdspm_autosync_ref(struct hdspm *hdspm)
{
+ /* This looks at the autosync selected sync reference */
if (AES32 == hdspm->io_type) {
- unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister);
- unsigned int syncref =
- (status >> HDSPM_AES32_syncref_bit) & 0xF;
- if (syncref == 0)
- return HDSPM_AES32_AUTOSYNC_FROM_WORD;
- if (syncref <= 8)
- return syncref;
- return HDSPM_AES32_AUTOSYNC_FROM_NONE;
- } else if (MADI == hdspm->io_type) {
- /* This looks at the autosync selected sync reference */
- unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+ unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister);
+ unsigned int syncref = (status >> HDSPM_AES32_syncref_bit) & 0xF;
+ if ((syncref >= HDSPM_AES32_AUTOSYNC_FROM_WORD) &&
+ (syncref <= HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN)) {
+ return syncref;
+ }
+ return HDSPM_AES32_AUTOSYNC_FROM_NONE;
+
+ } else if (MADI == hdspm->io_type) {
+
+ unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
switch (status2 & HDSPM_SelSyncRefMask) {
case HDSPM_SelSyncRef_WORD:
return HDSPM_AUTOSYNC_FROM_WORD;
@@ -2898,7 +3060,7 @@
case HDSPM_SelSyncRef_NVALID:
return HDSPM_AUTOSYNC_FROM_NONE;
default:
- return 0;
+ return HDSPM_AUTOSYNC_FROM_NONE;
}
}
@@ -2912,31 +3074,15 @@
struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
if (AES32 == hdspm->io_type) {
- static char *texts[] = { "WordClock", "AES1", "AES2", "AES3",
- "AES4", "AES5", "AES6", "AES7", "AES8", "None"};
+ static const char *const texts[] = { "WordClock", "AES1", "AES2", "AES3",
+ "AES4", "AES5", "AES6", "AES7", "AES8", "TCO", "Sync In", "None"};
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 10;
- if (uinfo->value.enumerated.item >=
- uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
+ ENUMERATED_CTL_INFO(uinfo, texts);
} else if (MADI == hdspm->io_type) {
- static char *texts[] = {"Word Clock", "MADI", "TCO",
+ static const char *const texts[] = {"Word Clock", "MADI", "TCO",
"Sync In", "None" };
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 5;
- if (uinfo->value.enumerated.item >=
- uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
+ ENUMERATED_CTL_INFO(uinfo, texts);
}
return 0;
}
@@ -2964,7 +3110,7 @@
static int snd_hdspm_info_tco_video_input_format(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = {"No video", "NTSC", "PAL"};
+ static const char *const texts[] = {"No video", "NTSC", "PAL"};
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3010,7 +3156,7 @@
static int snd_hdspm_info_tco_ltc_frames(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps",
+ static const char *const texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps",
"30 fps"};
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
@@ -3027,19 +3173,19 @@
HDSPM_TCO1_LTC_Format_MSB)) {
case 0:
/* 24 fps */
- ret = 1;
+ ret = fps_24;
break;
case HDSPM_TCO1_LTC_Format_LSB:
/* 25 fps */
- ret = 2;
+ ret = fps_25;
break;
case HDSPM_TCO1_LTC_Format_MSB:
- /* 25 fps */
- ret = 3;
+ /* 29.97 fps */
+ ret = fps_2997;
break;
default:
/* 30 fps */
- ret = 4;
+ ret = fps_30;
break;
}
}
@@ -3067,16 +3213,35 @@
static int hdspm_toggle_setting(struct hdspm *hdspm, u32 regmask)
{
- return (hdspm->control_register & regmask) ? 1 : 0;
+ u32 reg;
+
+ if (hdspm_is_raydat_or_aio(hdspm))
+ reg = hdspm->settings_register;
+ else
+ reg = hdspm->control_register;
+
+ return (reg & regmask) ? 1 : 0;
}
static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out)
{
+ u32 *reg;
+ u32 target_reg;
+
+ if (hdspm_is_raydat_or_aio(hdspm)) {
+ reg = &(hdspm->settings_register);
+ target_reg = HDSPM_WR_SETTINGS;
+ } else {
+ reg = &(hdspm->control_register);
+ target_reg = HDSPM_controlRegister;
+ }
+
if (out)
- hdspm->control_register |= regmask;
+ *reg |= regmask;
else
- hdspm->control_register &= ~regmask;
- hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+ *reg &= ~regmask;
+
+ hdspm_write(hdspm, target_reg, *reg);
return 0;
}
@@ -3141,7 +3306,7 @@
static int snd_hdspm_info_input_select(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "optical", "coaxial" };
+ static const char *const texts[] = { "optical", "coaxial" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3203,7 +3368,7 @@
static int snd_hdspm_info_ds_wire(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "Single", "Double" };
+ static const char *const texts[] = { "Single", "Double" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3276,7 +3441,7 @@
static int snd_hdspm_info_qs_wire(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "Single", "Double", "Quad" };
+ static const char *const texts[] = { "Single", "Double", "Quad" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3313,6 +3478,84 @@
return change;
}
+#define HDSPM_CONTROL_TRISTATE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .private_value = xindex, \
+ .info = snd_hdspm_info_tristate, \
+ .get = snd_hdspm_get_tristate, \
+ .put = snd_hdspm_put_tristate \
+}
+
+static int hdspm_tristate(struct hdspm *hdspm, u32 regmask)
+{
+ u32 reg = hdspm->settings_register & (regmask * 3);
+ return reg / regmask;
+}
+
+static int hdspm_set_tristate(struct hdspm *hdspm, int mode, u32 regmask)
+{
+ hdspm->settings_register &= ~(regmask * 3);
+ hdspm->settings_register |= (regmask * mode);
+ hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
+
+ return 0;
+}
+
+static int snd_hdspm_info_tristate(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ u32 regmask = kcontrol->private_value;
+
+ static const char *const texts_spdif[] = { "Optical", "Coaxial", "Internal" };
+ static const char *const texts_levels[] = { "Hi Gain", "+4 dBu", "-10 dBV" };
+
+ switch (regmask) {
+ case HDSPM_c0_Input0:
+ ENUMERATED_CTL_INFO(uinfo, texts_spdif);
+ break;
+ default:
+ ENUMERATED_CTL_INFO(uinfo, texts_levels);
+ break;
+ }
+ return 0;
+}
+
+static int snd_hdspm_get_tristate(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ u32 regmask = kcontrol->private_value;
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.enumerated.item[0] = hdspm_tristate(hdspm, regmask);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_tristate(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ u32 regmask = kcontrol->private_value;
+ int change;
+ int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0];
+ if (val < 0)
+ val = 0;
+ if (val > 2)
+ val = 2;
+
+ spin_lock_irq(&hdspm->lock);
+ change = val != hdspm_tristate(hdspm, regmask);
+ hdspm_set_tristate(hdspm, val, regmask);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
#define HDSPM_MADI_SPEEDMODE(xname, xindex) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
@@ -3352,7 +3595,7 @@
static int snd_hdspm_info_madi_speedmode(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "Single", "Double", "Quad" };
+ static const char *const texts[] = { "Single", "Double", "Quad" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3587,7 +3830,7 @@
static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "No Lock", "Lock", "Sync", "N/A" };
+ static const char *const texts[] = { "No Lock", "Lock", "Sync", "N/A" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3595,7 +3838,7 @@
static int snd_hdspm_tco_info_lock_check(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "No Lock", "Lock" };
+ static const char *const texts[] = { "No Lock", "Lock" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3745,9 +3988,18 @@
if (hdspm->tco) {
switch (hdspm->io_type) {
case MADI:
+ status = hdspm_read(hdspm, HDSPM_statusRegister);
+ if (status & HDSPM_tcoLockMadi) {
+ if (status & HDSPM_tcoSync)
+ return 2;
+ else
+ return 1;
+ }
+ return 0;
+ break;
case AES32:
status = hdspm_read(hdspm, HDSPM_statusRegister);
- if (status & HDSPM_tcoLock) {
+ if (status & HDSPM_tcoLockAes) {
if (status & HDSPM_tcoSync)
return 2;
else
@@ -3807,7 +4059,8 @@
case 5: /* SYNC IN */
val = hdspm_sync_in_sync_check(hdspm); break;
default:
- val = hdspm_s1_sync_check(hdspm, ucontrol->id.index-1);
+ val = hdspm_s1_sync_check(hdspm,
+ kcontrol->private_value-1);
}
break;
@@ -3975,7 +4228,8 @@
static int snd_hdspm_info_tco_sample_rate(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "44.1 kHz", "48 kHz" };
+ /* TODO freq from app could be supported here, see tco->samplerate */
+ static const char *const texts[] = { "44.1 kHz", "48 kHz" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -4021,7 +4275,8 @@
static int snd_hdspm_info_tco_pull(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "0", "+ 0.1 %", "- 0.1 %", "+ 4 %", "- 4 %" };
+ static const char *const texts[] = { "0", "+ 0.1 %", "- 0.1 %",
+ "+ 4 %", "- 4 %" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -4066,7 +4321,7 @@
static int snd_hdspm_info_tco_wck_conversion(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" };
+ static const char *const texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -4112,7 +4367,7 @@
static int snd_hdspm_info_tco_frame_rate(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "24 fps", "25 fps", "29.97fps",
+ static const char *const texts[] = { "24 fps", "25 fps", "29.97fps",
"29.97 dfps", "30 fps", "30 dfps" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
@@ -4159,7 +4414,7 @@
static int snd_hdspm_info_tco_sync_source(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "LTC", "Video", "WCK" };
+ static const char *const texts[] = { "LTC", "Video", "WCK" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -4284,7 +4539,6 @@
HDSPM_INTERNAL_CLOCK("Internal Clock", 0),
HDSPM_SYSTEM_CLOCK_MODE("System Clock Mode", 0),
HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0),
- HDSPM_AUTOSYNC_REF("AutoSync Reference", 0),
HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0),
HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0),
HDSPM_SYNC_CHECK("WC SyncCheck", 0),
@@ -4298,7 +4552,16 @@
HDSPM_AUTOSYNC_SAMPLE_RATE("SPDIF Frequency", 2),
HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT Frequency", 3),
HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 4),
- HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5)
+ HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5),
+ HDSPM_CONTROL_TRISTATE("S/PDIF Input", HDSPM_c0_Input0),
+ HDSPM_TOGGLE_SETTING("S/PDIF Out Optical", HDSPM_c0_Spdif_Opt),
+ HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro),
+ HDSPM_TOGGLE_SETTING("ADAT internal (AEB/TEB)", HDSPM_c0_AEB1),
+ HDSPM_TOGGLE_SETTING("XLR Breakout Cable", HDSPM_c0_Sym6db),
+ HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48),
+ HDSPM_CONTROL_TRISTATE("Input Level", HDSPM_c0_AD_GAIN0),
+ HDSPM_CONTROL_TRISTATE("Output Level", HDSPM_c0_DA_GAIN0),
+ HDSPM_CONTROL_TRISTATE("Phones Level", HDSPM_c0_PH_GAIN0)
/*
HDSPM_INPUT_SELECT("Input Select", 0),
@@ -4335,7 +4598,9 @@
HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT3 Frequency", 5),
HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT4 Frequency", 6),
HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 7),
- HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 8)
+ HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 8),
+ HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro),
+ HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48)
};
static struct snd_kcontrol_new snd_hdspm_controls_aes32[] = {
@@ -4345,7 +4610,7 @@
HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0),
HDSPM_AUTOSYNC_REF("AutoSync Reference", 0),
HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0),
- HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0),
+ HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 11),
HDSPM_SYNC_CHECK("WC Sync Check", 0),
HDSPM_SYNC_CHECK("AES1 Sync Check", 1),
HDSPM_SYNC_CHECK("AES2 Sync Check", 2),
@@ -4501,77 +4766,22 @@
------------------------------------------------------------*/
static void
-snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
- struct snd_info_buffer *buffer)
+snd_hdspm_proc_read_tco(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
{
struct hdspm *hdspm = entry->private_data;
- unsigned int status, status2, control, freq;
-
- char *pref_sync_ref;
- char *autosync_ref;
- char *system_clock_mode;
- char *insel;
- int x, x2;
-
- /* TCO stuff */
+ unsigned int status, control;
int a, ltc, frames, seconds, minutes, hours;
unsigned int period;
u64 freq_const = 0;
u32 rate;
+ snd_iprintf(buffer, "--- TCO ---\n");
+
status = hdspm_read(hdspm, HDSPM_statusRegister);
- status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
control = hdspm->control_register;
- freq = hdspm_read(hdspm, HDSPM_timecodeRegister);
- snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n",
- hdspm->card_name, hdspm->card->number + 1,
- hdspm->firmware_rev,
- (status2 & HDSPM_version0) |
- (status2 & HDSPM_version1) | (status2 &
- HDSPM_version2));
- snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n",
- (hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF,
- hdspm->serial);
-
- snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
- hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase);
-
- snd_iprintf(buffer, "--- System ---\n");
-
- snd_iprintf(buffer,
- "IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n",
- status & HDSPM_audioIRQPending,
- (status & HDSPM_midi0IRQPending) ? 1 : 0,
- (status & HDSPM_midi1IRQPending) ? 1 : 0,
- hdspm->irq_count);
- snd_iprintf(buffer,
- "HW pointer: id = %d, rawptr = %d (%d->%d) "
- "estimated= %ld (bytes)\n",
- ((status & HDSPM_BufferID) ? 1 : 0),
- (status & HDSPM_BufferPositionMask),
- (status & HDSPM_BufferPositionMask) %
- (2 * (int)hdspm->period_bytes),
- ((status & HDSPM_BufferPositionMask) - 64) %
- (2 * (int)hdspm->period_bytes),
- (long) hdspm_hw_pointer(hdspm) * 4);
-
- snd_iprintf(buffer,
- "MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n",
- hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF,
- hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF,
- hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
- hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
- snd_iprintf(buffer,
- "MIDIoverMADI FIFO: In=0x%x, Out=0x%x \n",
- hdspm_read(hdspm, HDSPM_midiStatusIn2) & 0xFF,
- hdspm_read(hdspm, HDSPM_midiStatusOut2) & 0xFF);
- snd_iprintf(buffer,
- "Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, "
- "status2=0x%x\n",
- hdspm->control_register, hdspm->control2_register,
- status, status2);
if (status & HDSPM_tco_detect) {
snd_iprintf(buffer, "TCO module detected.\n");
a = hdspm_read(hdspm, HDSPM_RD_TCO+4);
@@ -4665,6 +4875,75 @@
} else {
snd_iprintf(buffer, "No TCO module detected.\n");
}
+}
+
+static void
+snd_hdspm_proc_read_madi(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct hdspm *hdspm = entry->private_data;
+ unsigned int status, status2, control, freq;
+
+ char *pref_sync_ref;
+ char *autosync_ref;
+ char *system_clock_mode;
+ char *insel;
+ int x, x2;
+
+ status = hdspm_read(hdspm, HDSPM_statusRegister);
+ status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+ control = hdspm->control_register;
+ freq = hdspm_read(hdspm, HDSPM_timecodeRegister);
+
+ snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n",
+ hdspm->card_name, hdspm->card->number + 1,
+ hdspm->firmware_rev,
+ (status2 & HDSPM_version0) |
+ (status2 & HDSPM_version1) | (status2 &
+ HDSPM_version2));
+
+ snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n",
+ (hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF,
+ hdspm->serial);
+
+ snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
+ hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase);
+
+ snd_iprintf(buffer, "--- System ---\n");
+
+ snd_iprintf(buffer,
+ "IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n",
+ status & HDSPM_audioIRQPending,
+ (status & HDSPM_midi0IRQPending) ? 1 : 0,
+ (status & HDSPM_midi1IRQPending) ? 1 : 0,
+ hdspm->irq_count);
+ snd_iprintf(buffer,
+ "HW pointer: id = %d, rawptr = %d (%d->%d) "
+ "estimated= %ld (bytes)\n",
+ ((status & HDSPM_BufferID) ? 1 : 0),
+ (status & HDSPM_BufferPositionMask),
+ (status & HDSPM_BufferPositionMask) %
+ (2 * (int)hdspm->period_bytes),
+ ((status & HDSPM_BufferPositionMask) - 64) %
+ (2 * (int)hdspm->period_bytes),
+ (long) hdspm_hw_pointer(hdspm) * 4);
+
+ snd_iprintf(buffer,
+ "MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n",
+ hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
+ snd_iprintf(buffer,
+ "MIDIoverMADI FIFO: In=0x%x, Out=0x%x \n",
+ hdspm_read(hdspm, HDSPM_midiStatusIn2) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusOut2) & 0xFF);
+ snd_iprintf(buffer,
+ "Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, "
+ "status2=0x%x\n",
+ hdspm->control_register, hdspm->control2_register,
+ status, status2);
+
snd_iprintf(buffer, "--- Settings ---\n");
@@ -4768,6 +5047,9 @@
(status & HDSPM_RX_64ch) ? "64 channels" :
"56 channels");
+ /* call readout function for TCO specific status */
+ snd_hdspm_proc_read_tco(entry, buffer);
+
snd_iprintf(buffer, "\n");
}
@@ -4909,11 +5191,18 @@
autosync_ref = "AES7"; break;
case HDSPM_AES32_AUTOSYNC_FROM_AES8:
autosync_ref = "AES8"; break;
+ case HDSPM_AES32_AUTOSYNC_FROM_TCO:
+ autosync_ref = "TCO"; break;
+ case HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN:
+ autosync_ref = "Sync In"; break;
default:
autosync_ref = "---"; break;
}
snd_iprintf(buffer, "AutoSync ref = %s\n", autosync_ref);
+ /* call readout function for TCO specific status */
+ snd_hdspm_proc_read_tco(entry, buffer);
+
snd_iprintf(buffer, "\n");
}
@@ -5097,7 +5386,7 @@
case AES32:
hdspm->control_register =
- HDSPM_ClockModeMaster | /* Master Cloack Mode on */
+ HDSPM_ClockModeMaster | /* Master Clock Mode on */
hdspm_encode_latency(7) | /* latency max=8192samples */
HDSPM_SyncRef0 | /* AES1 is syncclock */
HDSPM_LineOut | /* Analog output in */
@@ -5123,9 +5412,8 @@
all_in_all_mixer(hdspm, 0 * UNITY_GAIN);
- if (hdspm->io_type == AIO || hdspm->io_type == RayDAT) {
+ if (hdspm_is_raydat_or_aio(hdspm))
hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
- }
/* set a default rate so that the channel map is set up. */
hdspm_set_rate(hdspm, 48000, 1);
@@ -5371,6 +5659,16 @@
*/
+ /* For AES cards, the float format bit is the same as the
+ * preferred sync reference. Since we don't want to break
+ * sync settings, we have to skip the remaining part of this
+ * function.
+ */
+ if (hdspm->io_type == AES32) {
+ return 0;
+ }
+
+
/* Switch to native float format if requested */
if (SNDRV_PCM_FORMAT_FLOAT_LE == params_format(params)) {
if (!(hdspm->control_register & HDSPe_FLOAT_FORMAT))
@@ -6013,7 +6311,7 @@
ltc.format = fps_2997;
break;
default:
- ltc.format = 30;
+ ltc.format = fps_30;
break;
}
if (i & HDSPM_TCO1_set_drop_frame_flag) {
@@ -6479,10 +6777,6 @@
break;
case AIO:
- if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) {
- snd_printk(KERN_INFO "HDSPM: AEB input board found, but not supported\n");
- }
-
hdspm->ss_in_channels = AIO_IN_SS_CHANNELS;
hdspm->ds_in_channels = AIO_IN_DS_CHANNELS;
hdspm->qs_in_channels = AIO_IN_QS_CHANNELS;
@@ -6490,6 +6784,20 @@
hdspm->ds_out_channels = AIO_OUT_DS_CHANNELS;
hdspm->qs_out_channels = AIO_OUT_QS_CHANNELS;
+ if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) {
+ snd_printk(KERN_INFO "HDSPM: AEB input board found\n");
+ hdspm->ss_in_channels += 4;
+ hdspm->ds_in_channels += 4;
+ hdspm->qs_in_channels += 4;
+ }
+
+ if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBO_D)) {
+ snd_printk(KERN_INFO "HDSPM: AEB output board found\n");
+ hdspm->ss_out_channels += 4;
+ hdspm->ds_out_channels += 4;
+ hdspm->qs_out_channels += 4;
+ }
+
hdspm->channel_map_out_ss = channel_map_aio_out_ss;
hdspm->channel_map_out_ds = channel_map_aio_out_ds;
hdspm->channel_map_out_qs = channel_map_aio_out_qs;
@@ -6558,6 +6866,7 @@
break;
case MADI:
+ case AES32:
if (hdspm_read(hdspm, HDSPM_statusRegister) & HDSPM_tco_detect) {
hdspm->midiPorts++;
hdspm->tco = kzalloc(sizeof(struct hdspm_tco),
@@ -6565,7 +6874,7 @@
if (NULL != hdspm->tco) {
hdspm_tco_write(hdspm);
}
- snd_printk(KERN_INFO "HDSPM: MADI TCO module found\n");
+ snd_printk(KERN_INFO "HDSPM: MADI/AES TCO module found\n");
} else {
hdspm->tco = NULL;
}
@@ -6580,10 +6889,12 @@
case AES32:
if (hdspm->tco) {
hdspm->texts_autosync = texts_autosync_aes_tco;
- hdspm->texts_autosync_items = 10;
+ hdspm->texts_autosync_items =
+ ARRAY_SIZE(texts_autosync_aes_tco);
} else {
hdspm->texts_autosync = texts_autosync_aes;
- hdspm->texts_autosync_items = 9;
+ hdspm->texts_autosync_items =
+ ARRAY_SIZE(texts_autosync_aes);
}
break;
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 45eeaa9..5138b84 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -26,12 +26,9 @@
config SND_SOC_AC97_BUS
bool
-config SND_SOC_DMAENGINE_PCM
- bool
-
config SND_SOC_GENERIC_DMAENGINE_PCM
bool
- select SND_SOC_DMAENGINE_PCM
+ select SND_DMAENGINE_PCM
# All the supported SoCs
source "sound/soc/atmel/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index bc02614..61a64d2 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,10 +1,6 @@
snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o
-ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),)
-snd-soc-core-objs += soc-dmaengine-pcm.o
-endif
-
ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),)
snd-soc-core-objs += soc-generic-dmaengine-pcm.o
endif
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index 3fdd87f..e48d38a 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -13,6 +13,7 @@
config SND_ATMEL_SOC_DMA
tristate
depends on SND_ATMEL_SOC
+ select SND_SOC_GENERIC_DMAENGINE_PCM
config SND_ATMEL_SOC_SSC
tristate
@@ -32,6 +33,26 @@
Say Y if you want to add support for SoC audio on WM8731-based
AT91sam9g20 evaluation board.
+config SND_ATMEL_SOC_WM8904
+ tristate "Atmel ASoC driver for boards using WM8904 codec"
+ depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC
+ select SND_ATMEL_SOC_SSC
+ select SND_ATMEL_SOC_DMA
+ select SND_SOC_WM8904
+ help
+ Say Y if you want to add support for Atmel ASoC driver for boards using
+ WM8904 codec.
+
+config SND_AT91_SOC_SAM9X5_WM8731
+ tristate "SoC Audio support for WM8731-based at91sam9x5 board"
+ depends on ATMEL_SSC && SND_ATMEL_SOC && SOC_AT91SAM9X5
+ select SND_ATMEL_SOC_SSC
+ select SND_ATMEL_SOC_DMA
+ select SND_SOC_WM8731
+ help
+ Say Y if you want to add support for audio SoC on an
+ at91sam9x5 based board that is using WM8731 codec.
+
config SND_AT91_SOC_AFEB9260
tristate "SoC Audio support for AFEB9260 board"
depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index 41967cc..5baabc8 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -11,6 +11,10 @@
# AT91 Machine Support
snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
+snd-atmel-soc-wm8904-objs := atmel_wm8904.o
+snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
+obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
+obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o
obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c
index d128265..06082e5 100644
--- a/sound/soc/atmel/atmel-pcm-dma.c
+++ b/sound/soc/atmel/atmel-pcm-dma.c
@@ -91,138 +91,52 @@
}
}
-/*--------------------------------------------------------------------------*\
- * DMAENGINE operations
-\*--------------------------------------------------------------------------*/
-static bool filter(struct dma_chan *chan, void *slave)
-{
- struct at_dma_slave *sl = slave;
-
- if (sl->dma_dev == chan->device->dev) {
- chan->private = sl;
- return true;
- } else {
- return false;
- }
-}
-
static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params, struct atmel_pcm_dma_params *prtd)
+ struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config)
{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct atmel_pcm_dma_params *prtd;
struct ssc_device *ssc;
- struct dma_chan *dma_chan;
- struct dma_slave_config slave_config;
int ret;
+ prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
ssc = prtd->ssc;
- ret = snd_hwparams_to_dma_slave_config(substream, params,
- &slave_config);
+ ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config);
if (ret) {
pr_err("atmel-pcm: hwparams to dma slave configure failed\n");
return ret;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- slave_config.dst_addr = (dma_addr_t)ssc->phybase + SSC_THR;
- slave_config.dst_maxburst = 1;
+ slave_config->dst_addr = ssc->phybase + SSC_THR;
+ slave_config->dst_maxburst = 1;
} else {
- slave_config.src_addr = (dma_addr_t)ssc->phybase + SSC_RHR;
- slave_config.src_maxburst = 1;
- }
-
- dma_chan = snd_dmaengine_pcm_get_chan(substream);
- if (dmaengine_slave_config(dma_chan, &slave_config)) {
- pr_err("atmel-pcm: failed to configure dma channel\n");
- ret = -EBUSY;
- return ret;
- }
-
- return 0;
-}
-
-static int atmel_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct atmel_pcm_dma_params *prtd;
- struct ssc_device *ssc;
- struct at_dma_slave *sdata = NULL;
- int ret;
-
- snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-
- prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
- ssc = prtd->ssc;
- if (ssc->pdev)
- sdata = ssc->pdev->dev.platform_data;
-
- ret = snd_dmaengine_pcm_open_request_chan(substream, filter, sdata);
- if (ret) {
- pr_err("atmel-pcm: dmaengine pcm open failed\n");
- return -EINVAL;
- }
-
- ret = atmel_pcm_configure_dma(substream, params, prtd);
- if (ret) {
- pr_err("atmel-pcm: failed to configure dmai\n");
- goto err;
+ slave_config->src_addr = ssc->phybase + SSC_RHR;
+ slave_config->src_maxburst = 1;
}
prtd->dma_intr_handler = atmel_pcm_dma_irq;
return 0;
-err:
- snd_dmaengine_pcm_close_release_chan(substream);
- return ret;
}
-static int atmel_pcm_dma_prepare(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct atmel_pcm_dma_params *prtd;
-
- prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
- ssc_writex(prtd->ssc->regs, SSC_IER, prtd->mask->ssc_error);
- ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_enable);
-
- return 0;
-}
-
-static int atmel_pcm_open(struct snd_pcm_substream *substream)
-{
- snd_soc_set_runtime_hwparams(substream, &atmel_pcm_dma_hardware);
-
- return 0;
-}
-
-static struct snd_pcm_ops atmel_pcm_ops = {
- .open = atmel_pcm_open,
- .close = snd_dmaengine_pcm_close_release_chan,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = atmel_pcm_hw_params,
- .prepare = atmel_pcm_dma_prepare,
- .trigger = snd_dmaengine_pcm_trigger,
- .pointer = snd_dmaengine_pcm_pointer_no_residue,
- .mmap = atmel_pcm_mmap,
-};
-
-static struct snd_soc_platform_driver atmel_soc_platform = {
- .ops = &atmel_pcm_ops,
- .pcm_new = atmel_pcm_new,
- .pcm_free = atmel_pcm_free,
+static const struct snd_dmaengine_pcm_config atmel_dmaengine_pcm_config = {
+ .prepare_slave_config = atmel_pcm_configure_dma,
+ .pcm_hardware = &atmel_pcm_dma_hardware,
+ .prealloc_buffer_size = ATMEL_SSC_DMABUF_SIZE,
};
int atmel_pcm_dma_platform_register(struct device *dev)
{
- return snd_soc_register_platform(dev, &atmel_soc_platform);
+ return snd_dmaengine_pcm_register(dev, &atmel_dmaengine_pcm_config,
+ SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
}
EXPORT_SYMBOL(atmel_pcm_dma_platform_register);
void atmel_pcm_dma_platform_unregister(struct device *dev)
{
- snd_soc_unregister_platform(dev);
+ snd_dmaengine_pcm_unregister(dev);
}
EXPORT_SYMBOL(atmel_pcm_dma_platform_unregister);
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index f3fdfa0..0ecf356 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -73,6 +73,7 @@
.ssc_disable = SSC_BIT(CR_TXDIS),
.ssc_endx = SSC_BIT(SR_ENDTX),
.ssc_endbuf = SSC_BIT(SR_TXBUFE),
+ .ssc_error = SSC_BIT(SR_OVRUN),
.pdc_enable = ATMEL_PDC_TXTEN,
.pdc_disable = ATMEL_PDC_TXTDIS,
};
@@ -82,6 +83,7 @@
.ssc_disable = SSC_BIT(CR_RXDIS),
.ssc_endx = SSC_BIT(SR_ENDRX),
.ssc_endbuf = SSC_BIT(SR_RXBUFF),
+ .ssc_error = SSC_BIT(SR_OVRUN),
.pdc_enable = ATMEL_PDC_RXTEN,
.pdc_disable = ATMEL_PDC_RXTDIS,
};
@@ -196,15 +198,27 @@
struct snd_soc_dai *dai)
{
struct atmel_ssc_info *ssc_p = &ssc_info[dai->id];
- int dir_mask;
+ struct atmel_pcm_dma_params *dma_params;
+ int dir, dir_mask;
pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n",
ssc_readl(ssc_p->ssc->regs, SR));
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ dir = 0;
dir_mask = SSC_DIR_MASK_PLAYBACK;
- else
+ } else {
+ dir = 1;
dir_mask = SSC_DIR_MASK_CAPTURE;
+ }
+
+ dma_params = &ssc_dma_params[dai->id][dir];
+ dma_params->ssc = ssc_p->ssc;
+ dma_params->substream = substream;
+
+ ssc_p->dma_params[dir] = dma_params;
+
+ snd_soc_dai_set_dma_data(dai, substream, dma_params);
spin_lock_irq(&ssc_p->lock);
if (ssc_p->dir_mask & dir_mask) {
@@ -325,7 +339,6 @@
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
int id = dai->id;
struct atmel_ssc_info *ssc_p = &ssc_info[id];
struct atmel_pcm_dma_params *dma_params;
@@ -344,19 +357,7 @@
else
dir = 1;
- dma_params = &ssc_dma_params[id][dir];
- dma_params->ssc = ssc_p->ssc;
- dma_params->substream = substream;
-
- ssc_p->dma_params[dir] = dma_params;
-
- /*
- * The snd_soc_pcm_stream->dma_data field is only used to communicate
- * the appropriate DMA parameters to the pcm driver hw_params()
- * function. It should not be used for other purposes
- * as it is common to all substreams.
- */
- snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_params);
+ dma_params = ssc_p->dma_params[dir];
channels = params_channels(params);
@@ -648,6 +649,7 @@
dma_params = ssc_p->dma_params[dir];
ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+ ssc_writel(ssc_p->ssc->regs, IER, dma_params->mask->ssc_error);
pr_debug("%s enabled SSC_SR=0x%08x\n",
dir ? "receive" : "transmit",
diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c
new file mode 100644
index 0000000..7222380
--- /dev/null
+++ b/sound/soc/atmel/atmel_wm8904.c
@@ -0,0 +1,254 @@
+/*
+ * atmel_wm8904 - Atmel ASoC driver for boards with WM8904 codec.
+ *
+ * Copyright (C) 2012 Atmel
+ *
+ * Author: Bo Shen <voice.shen@atmel.com>
+ *
+ * GPLv2 or later
+ */
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/pinctrl/consumer.h>
+
+#include <sound/soc.h>
+
+#include "../codecs/wm8904.h"
+#include "atmel_ssc_dai.h"
+
+#define MCLK_RATE 32768
+
+static struct clk *mclk;
+
+static const struct snd_soc_dapm_widget atmel_asoc_wm8904_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Mic", NULL),
+ SND_SOC_DAPM_LINE("Line In Jack", NULL),
+};
+
+static int atmel_asoc_wm8904_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_pll(codec_dai, WM8904_FLL_MCLK, WM8904_FLL_MCLK,
+ 32768, params_rate(params) * 256);
+ if (ret < 0) {
+ pr_err("%s - failed to set wm8904 codec PLL.", __func__);
+ return ret;
+ }
+
+ /*
+ * As here wm8904 use FLL output as its system clock
+ * so calling set_sysclk won't care freq parameter
+ * then we pass 0
+ */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8904_CLK_FLL,
+ 0, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ pr_err("%s -failed to set wm8904 SYSCLK\n", __func__);
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops atmel_asoc_wm8904_ops = {
+ .hw_params = atmel_asoc_wm8904_hw_params,
+};
+
+static int atmel_set_bias_level(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
+{
+ if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ clk_prepare_enable(mclk);
+ break;
+ case SND_SOC_BIAS_OFF:
+ clk_disable_unprepare(mclk);
+ break;
+ default:
+ break;
+ }
+ }
+
+ return 0;
+};
+
+static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = {
+ .name = "WM8904",
+ .stream_name = "WM8904 PCM",
+ .codec_dai_name = "wm8904-hifi",
+ .dai_fmt = SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM,
+ .ops = &atmel_asoc_wm8904_ops,
+};
+
+static struct snd_soc_card atmel_asoc_wm8904_card = {
+ .name = "atmel_asoc_wm8904",
+ .owner = THIS_MODULE,
+ .set_bias_level = atmel_set_bias_level,
+ .dai_link = &atmel_asoc_wm8904_dailink,
+ .num_links = 1,
+ .dapm_widgets = atmel_asoc_wm8904_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(atmel_asoc_wm8904_dapm_widgets),
+ .fully_routed = true,
+};
+
+static int atmel_asoc_wm8904_dt_init(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *codec_np, *cpu_np;
+ struct snd_soc_card *card = &atmel_asoc_wm8904_card;
+ struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink;
+ int ret;
+
+ if (!np) {
+ dev_err(&pdev->dev, "only device tree supported\n");
+ return -EINVAL;
+ }
+
+ ret = snd_soc_of_parse_card_name(card, "atmel,model");
+ if (ret) {
+ dev_err(&pdev->dev, "failed to parse card name\n");
+ return ret;
+ }
+
+ ret = snd_soc_of_parse_audio_routing(card, "atmel,audio-routing");
+ if (ret) {
+ dev_err(&pdev->dev, "failed to parse audio routing\n");
+ return ret;
+ }
+
+ cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
+ if (!cpu_np) {
+ dev_err(&pdev->dev, "failed to get dai and pcm info\n");
+ ret = -EINVAL;
+ return ret;
+ }
+ dailink->cpu_of_node = cpu_np;
+ dailink->platform_of_node = cpu_np;
+ of_node_put(cpu_np);
+
+ codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
+ if (!codec_np) {
+ dev_err(&pdev->dev, "failed to get codec info\n");
+ ret = -EINVAL;
+ return ret;
+ }
+ dailink->codec_of_node = codec_np;
+ of_node_put(codec_np);
+
+ return 0;
+}
+
+static int atmel_asoc_wm8904_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &atmel_asoc_wm8904_card;
+ struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink;
+ struct clk *clk_src;
+ struct pinctrl *pinctrl;
+ int id, ret;
+
+ pinctrl = devm_pinctrl_get_select_default(&pdev->dev);
+ if (IS_ERR(pinctrl)) {
+ dev_err(&pdev->dev, "failed to request pinctrl\n");
+ return PTR_ERR(pinctrl);
+ }
+
+ card->dev = &pdev->dev;
+ ret = atmel_asoc_wm8904_dt_init(pdev);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to init dt info\n");
+ return ret;
+ }
+
+ id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc");
+ ret = atmel_ssc_set_audio(id);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "failed to set SSC %d for audio\n", id);
+ return ret;
+ }
+
+ mclk = clk_get(NULL, "pck0");
+ if (IS_ERR(mclk)) {
+ dev_err(&pdev->dev, "failed to get pck0\n");
+ ret = PTR_ERR(mclk);
+ goto err_set_audio;
+ }
+
+ clk_src = clk_get(NULL, "clk32k");
+ if (IS_ERR(clk_src)) {
+ dev_err(&pdev->dev, "failed to get clk32k\n");
+ ret = PTR_ERR(clk_src);
+ goto err_set_audio;
+ }
+
+ ret = clk_set_parent(mclk, clk_src);
+ clk_put(clk_src);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "failed to set MCLK parent\n");
+ goto err_set_audio;
+ }
+
+ dev_info(&pdev->dev, "setting pck0 to %dHz\n", MCLK_RATE);
+ clk_set_rate(mclk, MCLK_RATE);
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed\n");
+ goto err_set_audio;
+ }
+
+ return 0;
+
+err_set_audio:
+ atmel_ssc_put_audio(id);
+ return ret;
+}
+
+static int atmel_asoc_wm8904_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink;
+ int id;
+
+ id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc");
+
+ snd_soc_unregister_card(card);
+ atmel_ssc_put_audio(id);
+
+ return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = {
+ { .compatible = "atmel,asoc-wm8904", },
+ { }
+};
+#endif
+
+static struct platform_driver atmel_asoc_wm8904_driver = {
+ .driver = {
+ .name = "atmel-wm8904-audio",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(atmel_asoc_wm8904_dt_ids),
+ },
+ .probe = atmel_asoc_wm8904_probe,
+ .remove = atmel_asoc_wm8904_remove,
+};
+
+module_platform_driver(atmel_asoc_wm8904_driver);
+
+/* Module information */
+MODULE_AUTHOR("Bo Shen <voice.shen@atmel.com>");
+MODULE_DESCRIPTION("ALSA SoC machine driver for Atmel EK with WM8904 codec");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c
new file mode 100644
index 0000000..992ae38
--- /dev/null
+++ b/sound/soc/atmel/sam9x5_wm8731.c
@@ -0,0 +1,208 @@
+/*
+ * sam9x5_wm8731 -- SoC audio for AT91SAM9X5-based boards
+ * that are using WM8731 as codec.
+ *
+ * Copyright (C) 2011 Atmel,
+ * Nicolas Ferre <nicolas.ferre@atmel.com>
+ *
+ * Copyright (C) 2013 Paratronic,
+ * Richard Genoud <richard.genoud@gmail.com>
+ *
+ * Based on sam9g20_wm8731.c by:
+ * Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+#include <linux/of.h>
+#include <linux/export.h>
+#include <linux/module.h>
+#include <linux/mod_devicetable.h>
+#include <linux/platform_device.h>
+#include <linux/device.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
+
+#include "../codecs/wm8731.h"
+#include "atmel_ssc_dai.h"
+
+
+#define MCLK_RATE 12288000
+
+#define DRV_NAME "sam9x5-snd-wm8731"
+
+struct sam9x5_drvdata {
+ int ssc_id;
+};
+
+/*
+ * Logic for a wm8731 as connected on a at91sam9x5ek based board.
+ */
+static int sam9x5_wm8731_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct device *dev = rtd->dev;
+ int ret;
+
+ dev_dbg(dev, "ASoC: %s called\n", __func__);
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
+ MCLK_RATE, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(dev, "ASoC: Failed to set WM8731 SYSCLK: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+/*
+ * Audio paths on at91sam9x5ek board:
+ *
+ * |A| ------------> | | ---R----> Headphone Jack
+ * |T| <----\ | WM | ---L--/
+ * |9| ---> CLK <--> | 8731 | <--R----- Line In Jack
+ * |1| <------------ | | <--L--/
+ */
+static const struct snd_soc_dapm_widget sam9x5_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In Jack", NULL),
+};
+
+static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *codec_np, *cpu_np;
+ struct snd_soc_card *card;
+ struct snd_soc_dai_link *dai;
+ struct sam9x5_drvdata *priv;
+ int ret;
+
+ if (!np) {
+ dev_err(&pdev->dev, "No device node supplied\n");
+ return -EINVAL;
+ }
+
+ card = devm_kzalloc(&pdev->dev, sizeof(*card), GFP_KERNEL);
+ priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+ dai = devm_kzalloc(&pdev->dev, sizeof(*dai), GFP_KERNEL);
+ if (!dai || !card || !priv) {
+ ret = -ENOMEM;
+ goto out;
+ }
+
+ card->dev = &pdev->dev;
+ card->owner = THIS_MODULE;
+ card->dai_link = dai;
+ card->num_links = 1;
+ card->dapm_widgets = sam9x5_dapm_widgets;
+ card->num_dapm_widgets = ARRAY_SIZE(sam9x5_dapm_widgets);
+ dai->name = "WM8731";
+ dai->stream_name = "WM8731 PCM";
+ dai->codec_dai_name = "wm8731-hifi";
+ dai->init = sam9x5_wm8731_init;
+ dai->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM;
+
+ ret = snd_soc_of_parse_card_name(card, "atmel,model");
+ if (ret) {
+ dev_err(&pdev->dev, "atmel,model node missing\n");
+ goto out;
+ }
+
+ ret = snd_soc_of_parse_audio_routing(card, "atmel,audio-routing");
+ if (ret) {
+ dev_err(&pdev->dev, "atmel,audio-routing node missing\n");
+ goto out;
+ }
+
+ codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
+ if (!codec_np) {
+ dev_err(&pdev->dev, "atmel,audio-codec node missing\n");
+ ret = -EINVAL;
+ goto out;
+ }
+
+ dai->codec_of_node = codec_np;
+
+ cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
+ if (!cpu_np) {
+ dev_err(&pdev->dev, "atmel,ssc-controller node missing\n");
+ ret = -EINVAL;
+ goto out;
+ }
+ dai->cpu_of_node = cpu_np;
+ dai->platform_of_node = cpu_np;
+
+ priv->ssc_id = of_alias_get_id(cpu_np, "ssc");
+
+ ret = atmel_ssc_set_audio(priv->ssc_id);
+ if (ret != 0) {
+ dev_err(&pdev->dev,
+ "ASoC: Failed to set SSC %d for audio: %d\n",
+ ret, priv->ssc_id);
+ goto out;
+ }
+
+ of_node_put(codec_np);
+ of_node_put(cpu_np);
+
+ platform_set_drvdata(pdev, card);
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "ASoC: Platform device allocation failed\n");
+ goto out_put_audio;
+ }
+
+ dev_dbg(&pdev->dev, "ASoC: %s ok\n", __func__);
+
+ return ret;
+
+out_put_audio:
+ atmel_ssc_put_audio(priv->ssc_id);
+out:
+ return ret;
+}
+
+static int sam9x5_wm8731_driver_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct sam9x5_drvdata *priv = card->drvdata;
+
+ snd_soc_unregister_card(card);
+ atmel_ssc_put_audio(priv->ssc_id);
+
+ return 0;
+}
+
+static const struct of_device_id sam9x5_wm8731_of_match[] = {
+ { .compatible = "atmel,sam9x5-wm8731-audio", },
+ {},
+};
+MODULE_DEVICE_TABLE(of, sam9x5_wm8731_of_match);
+
+static struct platform_driver sam9x5_wm8731_driver = {
+ .driver = {
+ .name = DRV_NAME,
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(sam9x5_wm8731_of_match),
+ },
+ .probe = sam9x5_wm8731_driver_probe,
+ .remove = sam9x5_wm8731_driver_remove,
+};
+module_platform_driver(sam9x5_wm8731_driver);
+
+/* Module information */
+MODULE_AUTHOR("Nicolas Ferre <nicolas.ferre@atmel.com>");
+MODULE_AUTHOR("Richard Genoud <richard.genoud@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC machine driver for AT91SAM9x5 - WM8731");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index a497a0c..decba87 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -73,12 +73,14 @@
static struct snd_soc_card db1300_ac97_machine = {
.name = "DB1300_AC97",
+ .owner = THIS_MODULE,
.dai_link = &db1300_ac97_dai,
.num_links = 1,
};
static struct snd_soc_card db1550_ac97_machine = {
.name = "DB1550_AC97",
+ .owner = THIS_MODULE,
.dai_link = &db1200_ac97_dai,
.num_links = 1,
};
@@ -145,6 +147,7 @@
static struct snd_soc_card db1300_i2s_machine = {
.name = "DB1300_I2S",
+ .owner = THIS_MODULE,
.dai_link = &db1300_i2s_dai,
.num_links = 1,
};
@@ -161,6 +164,7 @@
static struct snd_soc_card db1550_i2s_machine = {
.name = "DB1550_I2S",
+ .owner = THIS_MODULE,
.dai_link = &db1550_i2s_dai,
.num_links = 1,
};
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index a822ab8..986dcec 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -379,9 +379,6 @@
mutex_init(&wd->lock);
iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!iores)
- return -ENODEV;
-
wd->mmio = devm_ioremap_resource(&pdev->dev, iores);
if (IS_ERR(wd->mmio))
return PTR_ERR(wd->mmio);
diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h
index 0c3e22d..a680fdc 100644
--- a/sound/soc/blackfin/bf5xx-ac97.h
+++ b/sound/soc/blackfin/bf5xx-ac97.h
@@ -9,7 +9,6 @@
#ifndef _BF5XX_AC97_H
#define _BF5XX_AC97_H
-extern struct snd_ac97 *ac97;
/* Frame format in memory, only support stereo currently */
struct ac97_frame {
u16 ac97_tag; /* slot 0 */
diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c
index 04491f0..efa75b5 100644
--- a/sound/soc/cirrus/ep93xx-ac97.c
+++ b/sound/soc/cirrus/ep93xx-ac97.c
@@ -363,9 +363,6 @@
return -ENOMEM;
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!res)
- return -ENODEV;
-
info->regs = devm_ioremap_resource(&pdev->dev, res);
if (IS_ERR(info->regs))
return PTR_ERR(info->regs);
diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c
index 17ad70b..a57643d 100644
--- a/sound/soc/cirrus/ep93xx-i2s.c
+++ b/sound/soc/cirrus/ep93xx-i2s.c
@@ -376,9 +376,6 @@
return -ENOMEM;
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!res)
- return -ENODEV;
-
info->regs = devm_ioremap_resource(&pdev->dev, res);
if (IS_ERR(info->regs))
return PTR_ERR(info->regs);
@@ -411,7 +408,6 @@
return 0;
fail_put_lrclk:
- dev_set_drvdata(&pdev->dev, NULL);
clk_put(info->lrclk);
fail_put_sclk:
clk_put(info->sclk);
@@ -426,7 +422,6 @@
struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev);
snd_soc_unregister_component(&pdev->dev);
- dev_set_drvdata(&pdev->dev, NULL);
clk_put(info->lrclk);
clk_put(info->sclk);
clk_put(info->mclk);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 01d112b..15106c0 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -21,6 +21,7 @@
select SND_SOC_AD73311
select SND_SOC_ADAU1373 if I2C
select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI
+ select SND_SOC_ADAU1701 if I2C
select SND_SOC_ADS117X
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
@@ -55,6 +56,8 @@
select SND_SOC_MC13783 if MFD_MC13XXX
select SND_SOC_ML26124 if I2C
select SND_SOC_HDMI_CODEC
+ select SND_SOC_PCM1681 if I2C
+ select SND_SOC_PCM1792A if SPI_MASTER
select SND_SOC_PCM3008
select SND_SOC_RT5631 if I2C
select SND_SOC_RT5640 if I2C
@@ -123,6 +126,7 @@
select SND_SOC_WM8994 if MFD_WM8994
select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8996 if I2C
+ select SND_SOC_WM8997 if MFD_WM8997
select SND_SOC_WM9081 if I2C
select SND_SOC_WM9090 if I2C
select SND_SOC_WM9705 if SND_SOC_AC97_BUS
@@ -146,8 +150,10 @@
tristate
default y if SND_SOC_WM5102=y
default y if SND_SOC_WM5110=y
+ default y if SND_SOC_WM8997=y
default m if SND_SOC_WM5102=m
default m if SND_SOC_WM5110=m
+ default m if SND_SOC_WM8997=m
config SND_SOC_WM_HUBS
tristate
@@ -199,6 +205,9 @@
config SND_SOC_AK4535
tristate
+config SND_SOC_AK4554
+ tristate
+
config SND_SOC_AK4641
tristate
@@ -293,6 +302,12 @@
config SND_SOC_HDMI_CODEC
tristate
+config SND_SOC_PCM1681
+ tristate
+
+config SND_SOC_PCM1792A
+ tristate
+
config SND_SOC_PCM3008
tristate
@@ -501,6 +516,9 @@
config SND_SOC_WM8996
tristate
+config SND_SOC_WM8997
+ tristate
+
config SND_SOC_WM9081
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 70fd806..bc12676 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -11,6 +11,7 @@
snd-soc-ads117x-objs := ads117x.o
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
+snd-soc-ak4554-objs := ak4554.o
snd-soc-ak4641-objs := ak4641.o
snd-soc-ak4642-objs := ak4642.o
snd-soc-ak4671-objs := ak4671.o
@@ -42,6 +43,8 @@
snd-soc-mc13783-objs := mc13783.o
snd-soc-ml26124-objs := ml26124.o
snd-soc-hdmi-codec-objs := hdmi.o
+snd-soc-pcm1681-objs := pcm1681.o
+snd-soc-pcm1792a-codec-objs := pcm1792a.o
snd-soc-pcm3008-objs := pcm3008.o
snd-soc-rt5631-objs := rt5631.o
snd-soc-rt5640-objs := rt5640.o
@@ -114,6 +117,7 @@
snd-soc-wm8993-objs := wm8993.o
snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o
snd-soc-wm8995-objs := wm8995.o
+snd-soc-wm8997-objs := wm8997.o
snd-soc-wm9081-objs := wm9081.o
snd-soc-wm9090-objs := wm9090.o
snd-soc-wm9705-objs := wm9705.o
@@ -138,6 +142,7 @@
obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
+obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o
obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o
obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o
@@ -171,6 +176,8 @@
obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o
obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o
obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o
+obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o
+obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o
obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o
@@ -239,6 +246,7 @@
obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o
obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o
obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o
+obj-$(CONFIG_SND_SOC_WM8997) += snd-soc-wm8997.o
obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o
obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o
obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index ec73518..8d9ba4b 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -23,6 +23,16 @@
#include <sound/initval.h>
#include <sound/soc.h>
+static const struct snd_soc_dapm_widget ac97_widgets[] = {
+ SND_SOC_DAPM_INPUT("RX"),
+ SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route ac97_routes[] = {
+ { "AC97 Capture", NULL, "RX" },
+ { "TX", NULL, "AC97 Playback" },
+};
+
static int ac97_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
@@ -117,6 +127,11 @@
.probe = ac97_soc_probe,
.suspend = ac97_soc_suspend,
.resume = ac97_soc_resume,
+
+ .dapm_widgets = ac97_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ac97_widgets),
+ .dapm_routes = ac97_routes,
+ .num_dapm_routes = ARRAY_SIZE(ac97_routes),
};
static int ac97_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 89fcf7d..7257a88 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -96,6 +96,44 @@
SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
};
+static const struct snd_soc_dapm_widget ad1980_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2"),
+SND_SOC_DAPM_INPUT("CD_L"),
+SND_SOC_DAPM_INPUT("CD_R"),
+SND_SOC_DAPM_INPUT("AUX_L"),
+SND_SOC_DAPM_INPUT("AUX_R"),
+SND_SOC_DAPM_INPUT("LINE_IN_L"),
+SND_SOC_DAPM_INPUT("LINE_IN_R"),
+
+SND_SOC_DAPM_OUTPUT("LFE_OUT"),
+SND_SOC_DAPM_OUTPUT("CENTER_OUT"),
+SND_SOC_DAPM_OUTPUT("LINE_OUT_L"),
+SND_SOC_DAPM_OUTPUT("LINE_OUT_R"),
+SND_SOC_DAPM_OUTPUT("MONO_OUT"),
+SND_SOC_DAPM_OUTPUT("HP_OUT_L"),
+SND_SOC_DAPM_OUTPUT("HP_OUT_R"),
+};
+
+static const struct snd_soc_dapm_route ad1980_dapm_routes[] = {
+ { "Capture", NULL, "MIC1" },
+ { "Capture", NULL, "MIC2" },
+ { "Capture", NULL, "CD_L" },
+ { "Capture", NULL, "CD_R" },
+ { "Capture", NULL, "AUX_L" },
+ { "Capture", NULL, "AUX_R" },
+ { "Capture", NULL, "LINE_IN_L" },
+ { "Capture", NULL, "LINE_IN_R" },
+
+ { "LFE_OUT", NULL, "Playback" },
+ { "CENTER_OUT", NULL, "Playback" },
+ { "LINE_OUT_L", NULL, "Playback" },
+ { "LINE_OUT_R", NULL, "Playback" },
+ { "MONO_OUT", NULL, "Playback" },
+ { "HP_OUT_L", NULL, "Playback" },
+ { "HP_OUT_R", NULL, "Playback" },
+};
+
static unsigned int ac97_read(struct snd_soc_codec *codec,
unsigned int reg)
{
@@ -253,6 +291,11 @@
.reg_cache_step = 2,
.write = ac97_write,
.read = ac97_read,
+
+ .dapm_widgets = ad1980_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ad1980_dapm_widgets),
+ .dapm_routes = ad1980_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(ad1980_dapm_routes),
};
static int ad1980_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
index b1f2baf..5fac8ad 100644
--- a/sound/soc/codecs/ad73311.c
+++ b/sound/soc/codecs/ad73311.c
@@ -23,6 +23,21 @@
#include "ad73311.h"
+static const struct snd_soc_dapm_widget ad73311_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("VINP"),
+SND_SOC_DAPM_INPUT("VINN"),
+SND_SOC_DAPM_OUTPUT("VOUTN"),
+SND_SOC_DAPM_OUTPUT("VOUTP"),
+};
+
+static const struct snd_soc_dapm_route ad73311_dapm_routes[] = {
+ { "Capture", NULL, "VINP" },
+ { "Capture", NULL, "VINN" },
+
+ { "VOUTN", NULL, "Playback" },
+ { "VOUTP", NULL, "Playback" },
+};
+
static struct snd_soc_dai_driver ad73311_dai = {
.name = "ad73311-hifi",
.playback = {
@@ -39,7 +54,12 @@
.formats = SNDRV_PCM_FMTBIT_S16_LE, },
};
-static struct snd_soc_codec_driver soc_codec_dev_ad73311;
+static struct snd_soc_codec_driver soc_codec_dev_ad73311 = {
+ .dapm_widgets = ad73311_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ad73311_dapm_widgets),
+ .dapm_routes = ad73311_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(ad73311_dapm_routes),
+};
static int ad73311_probe(struct platform_device *pdev)
{
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index d1124a5..ebff1128b 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -91,7 +91,7 @@
#define ADAU1701_OSCIPOW_OPD 0x04
#define ADAU1701_DACSET_DACINIT 1
-#define ADAU1707_CLKDIV_UNSET (-1UL)
+#define ADAU1707_CLKDIV_UNSET (-1U)
#define ADAU1701_FIRMWARE "adau1701.bin"
@@ -247,21 +247,21 @@
gpio_is_valid(adau1701->gpio_pll_mode[1])) {
switch (clkdiv) {
case 64:
- gpio_set_value(adau1701->gpio_pll_mode[0], 0);
- gpio_set_value(adau1701->gpio_pll_mode[1], 0);
+ gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0);
+ gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0);
break;
case 256:
- gpio_set_value(adau1701->gpio_pll_mode[0], 0);
- gpio_set_value(adau1701->gpio_pll_mode[1], 1);
+ gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0);
+ gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1);
break;
case 384:
- gpio_set_value(adau1701->gpio_pll_mode[0], 1);
- gpio_set_value(adau1701->gpio_pll_mode[1], 0);
+ gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1);
+ gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0);
break;
case 0: /* fallback */
case 512:
- gpio_set_value(adau1701->gpio_pll_mode[0], 1);
- gpio_set_value(adau1701->gpio_pll_mode[1], 1);
+ gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1);
+ gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1);
break;
}
}
@@ -269,10 +269,10 @@
adau1701->pll_clkdiv = clkdiv;
if (gpio_is_valid(adau1701->gpio_nreset)) {
- gpio_set_value(adau1701->gpio_nreset, 0);
+ gpio_set_value_cansleep(adau1701->gpio_nreset, 0);
/* minimum reset time is 20ns */
udelay(1);
- gpio_set_value(adau1701->gpio_nreset, 1);
+ gpio_set_value_cansleep(adau1701->gpio_nreset, 1);
/* power-up time may be as long as 85ms */
mdelay(85);
}
@@ -734,7 +734,10 @@
}
static const struct i2c_device_id adau1701_i2c_id[] = {
+ { "adau1401", 0 },
+ { "adau1401a", 0 },
{ "adau1701", 0 },
+ { "adau1702", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id);
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 3c839cc..15b012d0 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -868,6 +868,12 @@
}
#if defined(CONFIG_SPI_MASTER)
+static const struct spi_device_id adav80x_spi_id[] = {
+ { "adav801", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(spi, adav80x_spi_id);
+
static int adav80x_spi_probe(struct spi_device *spi)
{
return adav80x_bus_probe(&spi->dev, SND_SOC_SPI);
@@ -885,15 +891,16 @@
},
.probe = adav80x_spi_probe,
.remove = adav80x_spi_remove,
+ .id_table = adav80x_spi_id,
};
#endif
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-static const struct i2c_device_id adav80x_id[] = {
+static const struct i2c_device_id adav80x_i2c_id[] = {
{ "adav803", 0 },
{ }
};
-MODULE_DEVICE_TABLE(i2c, adav80x_id);
+MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id);
static int adav80x_i2c_probe(struct i2c_client *client,
const struct i2c_device_id *id)
@@ -913,7 +920,7 @@
},
.probe = adav80x_i2c_probe,
.remove = adav80x_i2c_remove,
- .id_table = adav80x_id,
+ .id_table = adav80x_i2c_id,
};
#endif
diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c
index 506d474..8f388ed 100644
--- a/sound/soc/codecs/ads117x.c
+++ b/sound/soc/codecs/ads117x.c
@@ -23,6 +23,28 @@
#define ADS117X_RATES (SNDRV_PCM_RATE_8000_48000)
#define ADS117X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+static const struct snd_soc_dapm_widget ads117x_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("Input1"),
+SND_SOC_DAPM_INPUT("Input2"),
+SND_SOC_DAPM_INPUT("Input3"),
+SND_SOC_DAPM_INPUT("Input4"),
+SND_SOC_DAPM_INPUT("Input5"),
+SND_SOC_DAPM_INPUT("Input6"),
+SND_SOC_DAPM_INPUT("Input7"),
+SND_SOC_DAPM_INPUT("Input8"),
+};
+
+static const struct snd_soc_dapm_route ads117x_dapm_routes[] = {
+ { "Capture", NULL, "Input1" },
+ { "Capture", NULL, "Input2" },
+ { "Capture", NULL, "Input3" },
+ { "Capture", NULL, "Input4" },
+ { "Capture", NULL, "Input5" },
+ { "Capture", NULL, "Input6" },
+ { "Capture", NULL, "Input7" },
+ { "Capture", NULL, "Input8" },
+};
+
static struct snd_soc_dai_driver ads117x_dai = {
/* ADC */
.name = "ads117x-hifi",
@@ -34,7 +56,12 @@
.formats = ADS117X_FORMATS,},
};
-static struct snd_soc_codec_driver soc_codec_dev_ads117x;
+static struct snd_soc_codec_driver soc_codec_dev_ads117x = {
+ .dapm_widgets = ads117x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ads117x_dapm_widgets),
+ .dapm_routes = ads117x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(ads117x_dapm_routes),
+};
static int ads117x_probe(struct platform_device *pdev)
{
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index c7cfdf9..71059c0 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -51,6 +51,17 @@
struct regmap *regmap;
};
+static const struct snd_soc_dapm_widget ak4104_dapm_widgets[] = {
+SND_SOC_DAPM_PGA("TXE", AK4104_REG_TX, AK4104_TX_TXE, 0, NULL, 0),
+
+SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route ak4104_dapm_routes[] = {
+ { "TXE", NULL, "Playback" },
+ { "TX", NULL, "TXE" },
+};
+
static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int format)
{
@@ -138,29 +149,11 @@
if (ret < 0)
return ret;
- /* enable transmitter */
- ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
- AK4104_TX_TXE, AK4104_TX_TXE);
- if (ret < 0)
- return ret;
-
return 0;
}
-static int ak4104_hw_free(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_codec *codec = dai->codec;
- struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
-
- /* disable transmitter */
- return regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
- AK4104_TX_TXE, 0);
-}
-
static const struct snd_soc_dai_ops ak4101_dai_ops = {
.hw_params = ak4104_hw_params,
- .hw_free = ak4104_hw_free,
.set_fmt = ak4104_set_dai_fmt,
};
@@ -214,6 +207,11 @@
static struct snd_soc_codec_driver soc_codec_device_ak4104 = {
.probe = ak4104_probe,
.remove = ak4104_remove,
+
+ .dapm_widgets = ak4104_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ak4104_dapm_widgets),
+ .dapm_routes = ak4104_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(ak4104_dapm_routes),
};
static const struct regmap_config ak4104_regmap = {
diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c
new file mode 100644
index 0000000..79e9555
--- /dev/null
+++ b/sound/soc/codecs/ak4554.c
@@ -0,0 +1,106 @@
+/*
+ * ak4554.c
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <sound/soc.h>
+
+/*
+ * ak4554 is very simple DA/AD converter which has no setting register.
+ *
+ * CAUTION
+ *
+ * ak4554 playback format is SND_SOC_DAIFMT_RIGHT_J,
+ * and, capture format is SND_SOC_DAIFMT_LEFT_J
+ * on same bit clock, LR clock.
+ * But, this driver doesn't have snd_soc_dai_ops :: set_fmt
+ *
+ * CPU/Codec DAI image
+ *
+ * CPU-DAI1 (plaback only fmt = RIGHT_J) --+-- ak4554
+ * |
+ * CPU-DAI2 (capture only fmt = LEFT_J) ---+
+ */
+
+static const struct snd_soc_dapm_widget ak4554_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("AINL"),
+SND_SOC_DAPM_INPUT("AINR"),
+
+SND_SOC_DAPM_OUTPUT("AOUTL"),
+SND_SOC_DAPM_OUTPUT("AOUTR"),
+};
+
+static const struct snd_soc_dapm_route ak4554_dapm_routes[] = {
+ { "Capture", NULL, "AINL" },
+ { "Capture", NULL, "AINR" },
+
+ { "AOUTL", NULL, "Playback" },
+ { "AOUTR", NULL, "Playback" },
+};
+
+static struct snd_soc_dai_driver ak4554_dai = {
+ .name = "ak4554-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .symmetric_rates = 1,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_ak4554 = {
+ .dapm_widgets = ak4554_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ak4554_dapm_widgets),
+ .dapm_routes = ak4554_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(ak4554_dapm_routes),
+};
+
+static int ak4554_soc_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev,
+ &soc_codec_dev_ak4554,
+ &ak4554_dai, 1);
+}
+
+static int ak4554_soc_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+static struct of_device_id ak4554_of_match[] = {
+ { .compatible = "asahi-kasei,ak4554" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, ak4554_of_match);
+
+static struct platform_driver ak4554_driver = {
+ .driver = {
+ .name = "ak4554-adc-dac",
+ .owner = THIS_MODULE,
+ .of_match_table = ak4554_of_match,
+ },
+ .probe = ak4554_soc_probe,
+ .remove = ak4554_soc_remove,
+};
+module_platform_driver(ak4554_driver);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("SoC AK4554 driver");
+MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c
index 1f30398..72e953b 100644
--- a/sound/soc/codecs/ak5386.c
+++ b/sound/soc/codecs/ak5386.c
@@ -22,7 +22,22 @@
int reset_gpio;
};
-static struct snd_soc_codec_driver soc_codec_ak5386;
+static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("AINL"),
+SND_SOC_DAPM_INPUT("AINR"),
+};
+
+static const struct snd_soc_dapm_route ak5386_dapm_routes[] = {
+ { "Capture", NULL, "AINL" },
+ { "Capture", NULL, "AINR" },
+};
+
+static struct snd_soc_codec_driver soc_codec_ak5386 = {
+ .dapm_widgets = ak5386_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ak5386_dapm_widgets),
+ .dapm_routes = ak5386_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(ak5386_dapm_routes),
+};
static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int format)
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index de62581..657808b 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -19,6 +19,7 @@
#include <sound/tlv.h>
#include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/gpio.h>
#include <linux/mfd/arizona/registers.h>
#include "arizona.h"
@@ -199,9 +200,16 @@
if (ret != 0)
return ret;
- ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkr, 1);
- if (ret != 0)
- return ret;
+ switch (arizona->type) {
+ case WM8997:
+ break;
+ default:
+ ret = snd_soc_dapm_new_controls(&codec->dapm,
+ &arizona_spkr, 1);
+ if (ret != 0)
+ return ret;
+ break;
+ }
ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN_WARN,
"Thermal warning", arizona_thermal_warn,
@@ -223,6 +231,41 @@
}
EXPORT_SYMBOL_GPL(arizona_init_spk);
+int arizona_init_gpio(struct snd_soc_codec *codec)
+{
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+ int i;
+
+ switch (arizona->type) {
+ case WM5110:
+ snd_soc_dapm_disable_pin(&codec->dapm, "DRC2 Signal Activity");
+ break;
+ default:
+ break;
+ }
+
+ snd_soc_dapm_disable_pin(&codec->dapm, "DRC1 Signal Activity");
+
+ for (i = 0; i < ARRAY_SIZE(arizona->pdata.gpio_defaults); i++) {
+ switch (arizona->pdata.gpio_defaults[i] & ARIZONA_GPN_FN_MASK) {
+ case ARIZONA_GP_FN_DRC1_SIGNAL_DETECT:
+ snd_soc_dapm_enable_pin(&codec->dapm,
+ "DRC1 Signal Activity");
+ break;
+ case ARIZONA_GP_FN_DRC2_SIGNAL_DETECT:
+ snd_soc_dapm_enable_pin(&codec->dapm,
+ "DRC2 Signal Activity");
+ break;
+ default:
+ break;
+ }
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_gpio);
+
const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
"None",
"Tone Generator 1",
@@ -517,6 +560,26 @@
4, arizona_ng_hold_text);
EXPORT_SYMBOL_GPL(arizona_ng_hold);
+static const char * const arizona_in_dmic_osr_text[] = {
+ "1.536MHz", "3.072MHz", "6.144MHz",
+};
+
+const struct soc_enum arizona_in_dmic_osr[] = {
+ SOC_ENUM_SINGLE(ARIZONA_IN1L_CONTROL, ARIZONA_IN1_OSR_SHIFT,
+ ARRAY_SIZE(arizona_in_dmic_osr_text),
+ arizona_in_dmic_osr_text),
+ SOC_ENUM_SINGLE(ARIZONA_IN2L_CONTROL, ARIZONA_IN2_OSR_SHIFT,
+ ARRAY_SIZE(arizona_in_dmic_osr_text),
+ arizona_in_dmic_osr_text),
+ SOC_ENUM_SINGLE(ARIZONA_IN3L_CONTROL, ARIZONA_IN3_OSR_SHIFT,
+ ARRAY_SIZE(arizona_in_dmic_osr_text),
+ arizona_in_dmic_osr_text),
+ SOC_ENUM_SINGLE(ARIZONA_IN4L_CONTROL, ARIZONA_IN4_OSR_SHIFT,
+ ARRAY_SIZE(arizona_in_dmic_osr_text),
+ arizona_in_dmic_osr_text),
+};
+EXPORT_SYMBOL_GPL(arizona_in_dmic_osr);
+
static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena)
{
struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index b60b08c..9e81b63 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -150,7 +150,8 @@
ARIZONA_MUX(name_str " Aux 5", &name##_aux5_mux), \
ARIZONA_MUX(name_str " Aux 6", &name##_aux6_mux)
-#define ARIZONA_MUX_ROUTES(name) \
+#define ARIZONA_MUX_ROUTES(widget, name) \
+ { widget, NULL, name " Input" }, \
ARIZONA_MIXER_INPUT_ROUTES(name " Input")
#define ARIZONA_MIXER_ROUTES(widget, name) \
@@ -198,6 +199,7 @@
extern const struct soc_enum arizona_lhpf4_mode;
extern const struct soc_enum arizona_ng_hold;
+extern const struct soc_enum arizona_in_dmic_osr[];
extern int arizona_in_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol,
@@ -242,6 +244,7 @@
unsigned int Fref, unsigned int Fout);
extern int arizona_init_spk(struct snd_soc_codec *codec);
+extern int arizona_init_gpio(struct snd_soc_codec *codec);
extern int arizona_init_dai(struct arizona_priv *priv, int dai);
diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c
index a081d9f..c4cf069 100644
--- a/sound/soc/codecs/bt-sco.c
+++ b/sound/soc/codecs/bt-sco.c
@@ -15,15 +15,27 @@
#include <sound/soc.h>
+static const struct snd_soc_dapm_widget bt_sco_widgets[] = {
+ SND_SOC_DAPM_INPUT("RX"),
+ SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route bt_sco_routes[] = {
+ { "Capture", NULL, "RX" },
+ { "TX", NULL, "Playback" },
+};
+
static struct snd_soc_dai_driver bt_sco_dai = {
.name = "bt-sco-pcm",
.playback = {
+ .stream_name = "Playback",
.channels_min = 1,
.channels_max = 1,
.rates = SNDRV_PCM_RATE_8000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
+ .stream_name = "Capture",
.channels_min = 1,
.channels_max = 1,
.rates = SNDRV_PCM_RATE_8000,
@@ -31,7 +43,12 @@
},
};
-static struct snd_soc_codec_driver soc_codec_dev_bt_sco;
+static struct snd_soc_codec_driver soc_codec_dev_bt_sco = {
+ .dapm_widgets = bt_sco_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(bt_sco_widgets),
+ .dapm_routes = bt_sco_routes,
+ .num_dapm_routes = ARRAY_SIZE(bt_sco_routes),
+};
static int bt_sco_probe(struct platform_device *pdev)
{
@@ -50,6 +67,9 @@
{
.name = "dfbmcs320",
},
+ {
+ .name = "bt-sco",
+ },
{},
};
MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids);
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 8e47798..83c835d 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -139,6 +139,22 @@
struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
};
+static const struct snd_soc_dapm_widget cs4270_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("AINL"),
+SND_SOC_DAPM_INPUT("AINR"),
+
+SND_SOC_DAPM_OUTPUT("AOUTL"),
+SND_SOC_DAPM_OUTPUT("AOUTR"),
+};
+
+static const struct snd_soc_dapm_route cs4270_dapm_routes[] = {
+ { "Capture", NULL, "AINA" },
+ { "Capture", NULL, "AINB" },
+
+ { "AOUTA", NULL, "Playback" },
+ { "AOUTB", NULL, "Playback" },
+};
+
/**
* struct cs4270_mode_ratios - clock ratio tables
* @ratio: the ratio of MCLK to the sample rate
@@ -612,6 +628,10 @@
.controls = cs4270_snd_controls,
.num_controls = ARRAY_SIZE(cs4270_snd_controls),
+ .dapm_widgets = cs4270_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cs4270_dapm_widgets),
+ .dapm_routes = cs4270_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(cs4270_dapm_routes),
};
/*
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 03036b3..a20f1bb 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -173,6 +173,26 @@
bool enable_soft_reset;
};
+static const struct snd_soc_dapm_widget cs4271_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("AINA"),
+SND_SOC_DAPM_INPUT("AINB"),
+
+SND_SOC_DAPM_OUTPUT("AOUTA+"),
+SND_SOC_DAPM_OUTPUT("AOUTA-"),
+SND_SOC_DAPM_OUTPUT("AOUTB+"),
+SND_SOC_DAPM_OUTPUT("AOUTB-"),
+};
+
+static const struct snd_soc_dapm_route cs4271_dapm_routes[] = {
+ { "Capture", NULL, "AINA" },
+ { "Capture", NULL, "AINB" },
+
+ { "AOUTA+", NULL, "Playback" },
+ { "AOUTA-", NULL, "Playback" },
+ { "AOUTB+", NULL, "Playback" },
+ { "AOUTB-", NULL, "Playback" },
+};
+
/*
* @freq is the desired MCLK rate
* MCLK rate should (c) be the sample rate, multiplied by one of the
@@ -576,8 +596,7 @@
CS4271_MODE2_MUTECAEQUB,
CS4271_MODE2_MUTECAEQUB);
- return snd_soc_add_codec_controls(codec, cs4271_snd_controls,
- ARRAY_SIZE(cs4271_snd_controls));
+ return 0;
}
static int cs4271_remove(struct snd_soc_codec *codec)
@@ -596,6 +615,13 @@
.remove = cs4271_remove,
.suspend = cs4271_soc_suspend,
.resume = cs4271_soc_resume,
+
+ .controls = cs4271_snd_controls,
+ .num_controls = ARRAY_SIZE(cs4271_snd_controls),
+ .dapm_widgets = cs4271_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cs4271_dapm_widgets),
+ .dapm_routes = cs4271_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(cs4271_dapm_routes),
};
#if defined(CONFIG_SPI_MASTER)
diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c
index 2bcae2b..68342b1 100644
--- a/sound/soc/codecs/hdmi.c
+++ b/sound/soc/codecs/hdmi.c
@@ -23,11 +23,20 @@
#define DRV_NAME "hdmi-audio-codec"
-static struct snd_soc_codec_driver hdmi_codec;
+static const struct snd_soc_dapm_widget hdmi_widgets[] = {
+ SND_SOC_DAPM_INPUT("RX"),
+ SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route hdmi_routes[] = {
+ { "Capture", NULL, "RX" },
+ { "TX", NULL, "Playback" },
+};
static struct snd_soc_dai_driver hdmi_codec_dai = {
.name = "hdmi-hifi",
.playback = {
+ .stream_name = "Playback",
.channels_min = 2,
.channels_max = 8,
.rates = SNDRV_PCM_RATE_32000 |
@@ -37,6 +46,25 @@
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE,
},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ },
+
+};
+
+static struct snd_soc_codec_driver hdmi_codec = {
+ .dapm_widgets = hdmi_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
+ .dapm_routes = hdmi_routes,
+ .num_dapm_routes = ARRAY_SIZE(hdmi_routes),
};
static int hdmi_codec_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index 9f9f595..0e5743e 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -16,6 +16,7 @@
#include <linux/init.h>
#include <linux/module.h>
#include <linux/i2c.h>
+#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/core.h>
@@ -23,12 +24,15 @@
#include <sound/tlv.h>
struct lm4857 {
- struct i2c_client *i2c;
+ struct regmap *regmap;
uint8_t mode;
};
-static const uint8_t lm4857_default_regs[] = {
- 0x00, 0x00, 0x00, 0x00,
+static const struct reg_default lm4857_default_regs[] = {
+ { 0x0, 0x00 },
+ { 0x1, 0x00 },
+ { 0x2, 0x00 },
+ { 0x3, 0x00 },
};
/* The register offsets in the cache array */
@@ -42,39 +46,6 @@
#define LM4857_WAKEUP 5
#define LM4857_EPGAIN 4
-static int lm4857_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- uint8_t data;
- int ret;
-
- ret = snd_soc_cache_write(codec, reg, value);
- if (ret < 0)
- return ret;
-
- data = (reg << 6) | value;
- ret = i2c_master_send(codec->control_data, &data, 1);
- if (ret != 1) {
- dev_err(codec->dev, "Failed to write register: %d\n", ret);
- return ret;
- }
-
- return 0;
-}
-
-static unsigned int lm4857_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- unsigned int val;
- int ret;
-
- ret = snd_soc_cache_read(codec, reg, &val);
- if (ret)
- return -1;
-
- return val;
-}
-
static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -96,7 +67,7 @@
lm4857->mode = value;
if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
- snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, value + 6);
+ regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, value + 6);
return 1;
}
@@ -108,10 +79,11 @@
switch (level) {
case SND_SOC_BIAS_ON:
- snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, lm4857->mode + 6);
+ regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F,
+ lm4857->mode + 6);
break;
case SND_SOC_BIAS_STANDBY:
- snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, 0);
+ regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, 0);
break;
default:
break;
@@ -171,49 +143,32 @@
{"EP", NULL, "IN"},
};
-static int lm4857_probe(struct snd_soc_codec *codec)
-{
- struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
-
- codec->control_data = lm4857->i2c;
-
- ret = snd_soc_add_codec_controls(codec, lm4857_controls,
- ARRAY_SIZE(lm4857_controls));
- if (ret)
- return ret;
-
- ret = snd_soc_dapm_new_controls(dapm, lm4857_dapm_widgets,
- ARRAY_SIZE(lm4857_dapm_widgets));
- if (ret)
- return ret;
-
- ret = snd_soc_dapm_add_routes(dapm, lm4857_routes,
- ARRAY_SIZE(lm4857_routes));
- if (ret)
- return ret;
-
- snd_soc_dapm_new_widgets(dapm);
-
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_lm4857 = {
- .write = lm4857_write,
- .read = lm4857_read,
- .probe = lm4857_probe,
- .reg_cache_size = ARRAY_SIZE(lm4857_default_regs),
- .reg_word_size = sizeof(uint8_t),
- .reg_cache_default = lm4857_default_regs,
.set_bias_level = lm4857_set_bias_level,
+
+ .controls = lm4857_controls,
+ .num_controls = ARRAY_SIZE(lm4857_controls),
+ .dapm_widgets = lm4857_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(lm4857_dapm_widgets),
+ .dapm_routes = lm4857_routes,
+ .num_dapm_routes = ARRAY_SIZE(lm4857_routes),
+};
+
+static const struct regmap_config lm4857_regmap_config = {
+ .val_bits = 6,
+ .reg_bits = 2,
+
+ .max_register = LM4857_CTRL,
+
+ .cache_type = REGCACHE_FLAT,
+ .reg_defaults = lm4857_default_regs,
+ .num_reg_defaults = ARRAY_SIZE(lm4857_default_regs),
};
static int lm4857_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct lm4857 *lm4857;
- int ret;
lm4857 = devm_kzalloc(&i2c->dev, sizeof(*lm4857), GFP_KERNEL);
if (!lm4857)
@@ -221,11 +176,11 @@
i2c_set_clientdata(i2c, lm4857);
- lm4857->i2c = i2c;
+ lm4857->regmap = devm_regmap_init_i2c(i2c, &lm4857_regmap_config);
+ if (IS_ERR(lm4857->regmap))
+ return PTR_ERR(lm4857->regmap);
- ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0);
-
- return ret;
+ return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0);
}
static int lm4857_i2c_remove(struct i2c_client *i2c)
diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c
index a6ac231..31f9156 100644
--- a/sound/soc/codecs/max9768.c
+++ b/sound/soc/codecs/max9768.c
@@ -118,6 +118,18 @@
SOC_SINGLE_BOOL_EXT("Playback Switch", 0, max9768_get_gpio, max9768_set_gpio),
};
+static const struct snd_soc_dapm_widget max9768_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("IN"),
+
+SND_SOC_DAPM_OUTPUT("OUT+"),
+SND_SOC_DAPM_OUTPUT("OUT-"),
+};
+
+static const struct snd_soc_dapm_route max9768_dapm_routes[] = {
+ { "OUT+", NULL, "IN" },
+ { "OUT-", NULL, "IN" },
+};
+
static int max9768_probe(struct snd_soc_codec *codec)
{
struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec);
@@ -148,6 +160,10 @@
.probe = max9768_probe,
.controls = max9768_volume,
.num_controls = ARRAY_SIZE(max9768_volume),
+ .dapm_widgets = max9768_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(max9768_dapm_widgets),
+ .dapm_routes = max9768_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(max9768_dapm_routes),
};
static const struct regmap_config max9768_i2c_regmap_config = {
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index ad5313f..0569a4c 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2084,8 +2084,9 @@
pm_wakeup_event(codec->dev, 100);
- schedule_delayed_work(&max98090->jack_work,
- msecs_to_jiffies(100));
+ queue_delayed_work(system_power_efficient_wq,
+ &max98090->jack_work,
+ msecs_to_jiffies(100));
}
if (active & M98090_DRCACT_MASK)
@@ -2132,8 +2133,9 @@
snd_soc_jack_report(max98090->jack, 0,
SND_JACK_HEADSET | SND_JACK_BTN_0);
- schedule_delayed_work(&max98090->jack_work,
- msecs_to_jiffies(100));
+ queue_delayed_work(system_power_efficient_wq,
+ &max98090->jack_work,
+ msecs_to_jiffies(100));
return 0;
}
diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c
index 6b6c74c..29549cd 100644
--- a/sound/soc/codecs/max9877.c
+++ b/sound/soc/codecs/max9877.c
@@ -14,170 +14,21 @@
#include <linux/module.h>
#include <linux/init.h>
#include <linux/i2c.h>
+#include <linux/regmap.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#include "max9877.h"
-static struct i2c_client *i2c;
+static struct regmap *regmap;
-static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 };
-
-static void max9877_write_regs(void)
-{
- unsigned int i;
- u8 data[6];
-
- data[0] = MAX9877_INPUT_MODE;
- for (i = 0; i < ARRAY_SIZE(max9877_regs); i++)
- data[i + 1] = max9877_regs[i];
-
- if (i2c_master_send(i2c, data, 6) != 6)
- dev_err(&i2c->dev, "i2c write failed\n");
-}
-
-static int max9877_get_reg(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct soc_mixer_control *mc =
- (struct soc_mixer_control *)kcontrol->private_value;
- unsigned int reg = mc->reg;
- unsigned int shift = mc->shift;
- unsigned int mask = mc->max;
- unsigned int invert = mc->invert;
-
- ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask;
-
- if (invert)
- ucontrol->value.integer.value[0] =
- mask - ucontrol->value.integer.value[0];
-
- return 0;
-}
-
-static int max9877_set_reg(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct soc_mixer_control *mc =
- (struct soc_mixer_control *)kcontrol->private_value;
- unsigned int reg = mc->reg;
- unsigned int shift = mc->shift;
- unsigned int mask = mc->max;
- unsigned int invert = mc->invert;
- unsigned int val = (ucontrol->value.integer.value[0] & mask);
-
- if (invert)
- val = mask - val;
-
- if (((max9877_regs[reg] >> shift) & mask) == val)
- return 0;
-
- max9877_regs[reg] &= ~(mask << shift);
- max9877_regs[reg] |= val << shift;
- max9877_write_regs();
-
- return 1;
-}
-
-static int max9877_get_2reg(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct soc_mixer_control *mc =
- (struct soc_mixer_control *)kcontrol->private_value;
- unsigned int reg = mc->reg;
- unsigned int reg2 = mc->rreg;
- unsigned int shift = mc->shift;
- unsigned int mask = mc->max;
-
- ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask;
- ucontrol->value.integer.value[1] = (max9877_regs[reg2] >> shift) & mask;
-
- return 0;
-}
-
-static int max9877_set_2reg(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct soc_mixer_control *mc =
- (struct soc_mixer_control *)kcontrol->private_value;
- unsigned int reg = mc->reg;
- unsigned int reg2 = mc->rreg;
- unsigned int shift = mc->shift;
- unsigned int mask = mc->max;
- unsigned int val = (ucontrol->value.integer.value[0] & mask);
- unsigned int val2 = (ucontrol->value.integer.value[1] & mask);
- unsigned int change = 0;
-
- if (((max9877_regs[reg] >> shift) & mask) != val)
- change = 1;
-
- if (((max9877_regs[reg2] >> shift) & mask) != val2)
- change = 1;
-
- if (change) {
- max9877_regs[reg] &= ~(mask << shift);
- max9877_regs[reg] |= val << shift;
- max9877_regs[reg2] &= ~(mask << shift);
- max9877_regs[reg2] |= val2 << shift;
- max9877_write_regs();
- }
-
- return change;
-}
-
-static int max9877_get_out_mode(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- u8 value = max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK;
-
- if (value)
- value -= 1;
-
- ucontrol->value.integer.value[0] = value;
- return 0;
-}
-
-static int max9877_set_out_mode(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- u8 value = ucontrol->value.integer.value[0];
-
- value += 1;
-
- if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value)
- return 0;
-
- max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OUTMODE_MASK;
- max9877_regs[MAX9877_OUTPUT_MODE] |= value;
- max9877_write_regs();
- return 1;
-}
-
-static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- u8 value = (max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK);
-
- value = value >> MAX9877_OSC_OFFSET;
-
- ucontrol->value.integer.value[0] = value;
- return 0;
-}
-
-static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- u8 value = ucontrol->value.integer.value[0];
-
- value = value << MAX9877_OSC_OFFSET;
- if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value)
- return 0;
-
- max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OSC_MASK;
- max9877_regs[MAX9877_OUTPUT_MODE] |= value;
- max9877_write_regs();
- return 1;
-}
+static struct reg_default max9877_regs[] = {
+ { 0, 0x40 },
+ { 1, 0x00 },
+ { 2, 0x00 },
+ { 3, 0x00 },
+ { 4, 0x49 },
+};
static const unsigned int max9877_pgain_tlv[] = {
TLV_DB_RANGE_HEAD(2),
@@ -212,65 +63,104 @@
};
static const struct soc_enum max9877_enum[] = {
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_out_mode), max9877_out_mode),
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode),
+ SOC_ENUM_SINGLE(MAX9877_OUTPUT_MODE, 0, ARRAY_SIZE(max9877_out_mode),
+ max9877_out_mode),
+ SOC_ENUM_SINGLE(MAX9877_OUTPUT_MODE, MAX9877_OSC_OFFSET,
+ ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode),
};
static const struct snd_kcontrol_new max9877_controls[] = {
- SOC_SINGLE_EXT_TLV("MAX9877 PGAINA Playback Volume",
- MAX9877_INPUT_MODE, 0, 2, 0,
- max9877_get_reg, max9877_set_reg, max9877_pgain_tlv),
- SOC_SINGLE_EXT_TLV("MAX9877 PGAINB Playback Volume",
- MAX9877_INPUT_MODE, 2, 2, 0,
- max9877_get_reg, max9877_set_reg, max9877_pgain_tlv),
- SOC_SINGLE_EXT_TLV("MAX9877 Amp Speaker Playback Volume",
- MAX9877_SPK_VOLUME, 0, 31, 0,
- max9877_get_reg, max9877_set_reg, max9877_output_tlv),
- SOC_DOUBLE_R_EXT_TLV("MAX9877 Amp HP Playback Volume",
- MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0,
- max9877_get_2reg, max9877_set_2reg, max9877_output_tlv),
- SOC_SINGLE_EXT("MAX9877 INB Stereo Switch",
- MAX9877_INPUT_MODE, 4, 1, 1,
- max9877_get_reg, max9877_set_reg),
- SOC_SINGLE_EXT("MAX9877 INA Stereo Switch",
- MAX9877_INPUT_MODE, 5, 1, 1,
- max9877_get_reg, max9877_set_reg),
- SOC_SINGLE_EXT("MAX9877 Zero-crossing detection Switch",
- MAX9877_INPUT_MODE, 6, 1, 0,
- max9877_get_reg, max9877_set_reg),
- SOC_SINGLE_EXT("MAX9877 Bypass Mode Switch",
- MAX9877_OUTPUT_MODE, 6, 1, 0,
- max9877_get_reg, max9877_set_reg),
- SOC_SINGLE_EXT("MAX9877 Shutdown Mode Switch",
- MAX9877_OUTPUT_MODE, 7, 1, 1,
- max9877_get_reg, max9877_set_reg),
- SOC_ENUM_EXT("MAX9877 Output Mode", max9877_enum[0],
- max9877_get_out_mode, max9877_set_out_mode),
- SOC_ENUM_EXT("MAX9877 Oscillator Mode", max9877_enum[1],
- max9877_get_osc_mode, max9877_set_osc_mode),
+ SOC_SINGLE_TLV("MAX9877 PGAINA Playback Volume",
+ MAX9877_INPUT_MODE, 0, 2, 0, max9877_pgain_tlv),
+ SOC_SINGLE_TLV("MAX9877 PGAINB Playback Volume",
+ MAX9877_INPUT_MODE, 2, 2, 0, max9877_pgain_tlv),
+ SOC_SINGLE_TLV("MAX9877 Amp Speaker Playback Volume",
+ MAX9877_SPK_VOLUME, 0, 31, 0, max9877_output_tlv),
+ SOC_DOUBLE_R_TLV("MAX9877 Amp HP Playback Volume",
+ MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0,
+ max9877_output_tlv),
+ SOC_SINGLE("MAX9877 INB Stereo Switch",
+ MAX9877_INPUT_MODE, 4, 1, 1),
+ SOC_SINGLE("MAX9877 INA Stereo Switch",
+ MAX9877_INPUT_MODE, 5, 1, 1),
+ SOC_SINGLE("MAX9877 Zero-crossing detection Switch",
+ MAX9877_INPUT_MODE, 6, 1, 0),
+ SOC_SINGLE("MAX9877 Bypass Mode Switch",
+ MAX9877_OUTPUT_MODE, 6, 1, 0),
+ SOC_ENUM("MAX9877 Output Mode", max9877_enum[0]),
+ SOC_ENUM("MAX9877 Oscillator Mode", max9877_enum[1]),
};
-/* This function is called from ASoC machine driver */
-int max9877_add_controls(struct snd_soc_codec *codec)
-{
- return snd_soc_add_codec_controls(codec, max9877_controls,
- ARRAY_SIZE(max9877_controls));
-}
-EXPORT_SYMBOL_GPL(max9877_add_controls);
+static const struct snd_soc_dapm_widget max9877_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("INA1"),
+SND_SOC_DAPM_INPUT("INA2"),
+SND_SOC_DAPM_INPUT("INB1"),
+SND_SOC_DAPM_INPUT("INB2"),
+SND_SOC_DAPM_INPUT("RXIN+"),
+SND_SOC_DAPM_INPUT("RXIN-"),
+
+SND_SOC_DAPM_PGA("SHDN", MAX9877_OUTPUT_MODE, 7, 1, NULL, 0),
+
+SND_SOC_DAPM_OUTPUT("OUT+"),
+SND_SOC_DAPM_OUTPUT("OUT-"),
+SND_SOC_DAPM_OUTPUT("HPL"),
+SND_SOC_DAPM_OUTPUT("HPR"),
+};
+
+static const struct snd_soc_dapm_route max9877_dapm_routes[] = {
+ { "SHDN", NULL, "INA1" },
+ { "SHDN", NULL, "INA2" },
+ { "SHDN", NULL, "INB1" },
+ { "SHDN", NULL, "INB2" },
+
+ { "OUT+", NULL, "RXIN+" },
+ { "OUT+", NULL, "SHDN" },
+
+ { "OUT-", NULL, "SHDN" },
+ { "OUT-", NULL, "RXIN-" },
+
+ { "HPL", NULL, "SHDN" },
+ { "HPR", NULL, "SHDN" },
+};
+
+static const struct snd_soc_codec_driver max9877_codec = {
+ .controls = max9877_controls,
+ .num_controls = ARRAY_SIZE(max9877_controls),
+
+ .dapm_widgets = max9877_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(max9877_dapm_widgets),
+ .dapm_routes = max9877_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(max9877_dapm_routes),
+};
+
+static const struct regmap_config max9877_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .reg_defaults = max9877_regs,
+ .num_reg_defaults = ARRAY_SIZE(max9877_regs),
+ .cache_type = REGCACHE_RBTREE,
+};
static int max9877_i2c_probe(struct i2c_client *client,
const struct i2c_device_id *id)
{
- i2c = client;
+ int i;
- max9877_write_regs();
+ regmap = devm_regmap_init_i2c(client, &max9877_regmap);
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
- return 0;
+ /* Ensure the device is in reset state */
+ for (i = 0; i < ARRAY_SIZE(max9877_regs); i++)
+ regmap_write(regmap, max9877_regs[i].reg, max9877_regs[i].def);
+
+ return snd_soc_register_codec(&client->dev, &max9877_codec, NULL, 0);
}
static int max9877_i2c_remove(struct i2c_client *client)
{
- i2c = NULL;
+ snd_soc_unregister_codec(&client->dev);
return 0;
}
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 5402dfb..4d3c8fd 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -94,7 +94,6 @@
#define AUDIO_DAC_CFS_DLY_B (1 << 10)
struct mc13783_priv {
- struct snd_soc_codec codec;
struct mc13xxx *mc13xxx;
enum mc13783_ssi_port adc_ssi_port;
diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
new file mode 100644
index 0000000..651ce09
--- /dev/null
+++ b/sound/soc/codecs/pcm1681.c
@@ -0,0 +1,339 @@
+/*
+ * PCM1681 ASoC codec driver
+ *
+ * Copyright (c) StreamUnlimited GmbH 2013
+ * Marek Belisko <marek.belisko@streamunlimited.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/of_device.h>
+#include <linux/of_gpio.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#define PCM1681_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+#define PCM1681_PCM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \
+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000)
+
+#define PCM1681_SOFT_MUTE_ALL 0xff
+#define PCM1681_DEEMPH_RATE_MASK 0x18
+#define PCM1681_DEEMPH_MASK 0x01
+
+#define PCM1681_ATT_CONTROL(X) (X <= 6 ? X : X + 9) /* Attenuation level */
+#define PCM1681_SOFT_MUTE 0x07 /* Soft mute control register */
+#define PCM1681_DAC_CONTROL 0x08 /* DAC operation control */
+#define PCM1681_FMT_CONTROL 0x09 /* Audio interface data format */
+#define PCM1681_DEEMPH_CONTROL 0x0a /* De-emphasis control */
+#define PCM1681_ZERO_DETECT_STATUS 0x0e /* Zero detect status reg */
+
+static const struct reg_default pcm1681_reg_defaults[] = {
+ { 0x01, 0xff },
+ { 0x02, 0xff },
+ { 0x03, 0xff },
+ { 0x04, 0xff },
+ { 0x05, 0xff },
+ { 0x06, 0xff },
+ { 0x07, 0x00 },
+ { 0x08, 0x00 },
+ { 0x09, 0x06 },
+ { 0x0A, 0x00 },
+ { 0x0B, 0xff },
+ { 0x0C, 0x0f },
+ { 0x0D, 0x00 },
+ { 0x10, 0xff },
+ { 0x11, 0xff },
+ { 0x12, 0x00 },
+ { 0x13, 0x00 },
+};
+
+static bool pcm1681_accessible_reg(struct device *dev, unsigned int reg)
+{
+ return !((reg == 0x00) || (reg == 0x0f));
+}
+
+static bool pcm1681_writeable_reg(struct device *dev, unsigned register reg)
+{
+ return pcm1681_accessible_reg(dev, reg) &&
+ (reg != PCM1681_ZERO_DETECT_STATUS);
+}
+
+struct pcm1681_private {
+ struct regmap *regmap;
+ unsigned int format;
+ /* Current deemphasis status */
+ unsigned int deemph;
+ /* Current rate for deemphasis control */
+ unsigned int rate;
+};
+
+static const int pcm1681_deemph[] = { 44100, 48000, 32000 };
+
+static int pcm1681_set_deemph(struct snd_soc_codec *codec)
+{
+ struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
+ int i = 0, val = -1, enable = 0;
+
+ if (priv->deemph)
+ for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++)
+ if (pcm1681_deemph[i] == priv->rate)
+ val = i;
+
+ if (val != -1) {
+ regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL,
+ PCM1681_DEEMPH_RATE_MASK, val);
+ enable = 1;
+ } else
+ enable = 0;
+
+ /* enable/disable deemphasis functionality */
+ return regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL,
+ PCM1681_DEEMPH_MASK, enable);
+}
+
+static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.enumerated.item[0] = priv->deemph;
+
+ return 0;
+}
+
+static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ priv->deemph = ucontrol->value.enumerated.item[0];
+
+ return pcm1681_set_deemph(codec);
+}
+
+static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int format)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ /* The PCM1681 can only be slave to all clocks */
+ if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+ dev_err(codec->dev, "Invalid clocking mode\n");
+ return -EINVAL;
+ }
+
+ priv->format = format;
+
+ return 0;
+}
+
+static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
+ int val;
+
+ if (mute)
+ val = PCM1681_SOFT_MUTE_ALL;
+ else
+ val = 0;
+
+ return regmap_write(priv->regmap, PCM1681_SOFT_MUTE, val);
+}
+
+static int pcm1681_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
+ int val = 0, ret;
+ int pcm_format = params_format(params);
+
+ priv->rate = params_rate(params);
+
+ switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_RIGHT_J:
+ if (pcm_format == SNDRV_PCM_FORMAT_S24_LE)
+ val = 0x00;
+ else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE)
+ val = 0x03;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ val = 0x04;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ val = 0x05;
+ break;
+ default:
+ dev_err(codec->dev, "Invalid DAI format\n");
+ return -EINVAL;
+ }
+
+ ret = regmap_update_bits(priv->regmap, PCM1681_FMT_CONTROL, 0x0f, val);
+ if (ret < 0)
+ return ret;
+
+ return pcm1681_set_deemph(codec);
+}
+
+static const struct snd_soc_dai_ops pcm1681_dai_ops = {
+ .set_fmt = pcm1681_set_dai_fmt,
+ .hw_params = pcm1681_hw_params,
+ .digital_mute = pcm1681_digital_mute,
+};
+
+static const struct snd_soc_dapm_widget pcm1681_dapm_widgets[] = {
+SND_SOC_DAPM_OUTPUT("VOUT1"),
+SND_SOC_DAPM_OUTPUT("VOUT2"),
+SND_SOC_DAPM_OUTPUT("VOUT3"),
+SND_SOC_DAPM_OUTPUT("VOUT4"),
+SND_SOC_DAPM_OUTPUT("VOUT5"),
+SND_SOC_DAPM_OUTPUT("VOUT6"),
+SND_SOC_DAPM_OUTPUT("VOUT7"),
+SND_SOC_DAPM_OUTPUT("VOUT8"),
+};
+
+static const struct snd_soc_dapm_route pcm1681_dapm_routes[] = {
+ { "VOUT1", NULL, "Playback" },
+ { "VOUT2", NULL, "Playback" },
+ { "VOUT3", NULL, "Playback" },
+ { "VOUT4", NULL, "Playback" },
+ { "VOUT5", NULL, "Playback" },
+ { "VOUT6", NULL, "Playback" },
+ { "VOUT7", NULL, "Playback" },
+ { "VOUT8", NULL, "Playback" },
+};
+
+static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1);
+
+static const struct snd_kcontrol_new pcm1681_controls[] = {
+ SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume",
+ PCM1681_ATT_CONTROL(1), PCM1681_ATT_CONTROL(2), 0,
+ 0x7f, 0, pcm1681_dac_tlv),
+ SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume",
+ PCM1681_ATT_CONTROL(3), PCM1681_ATT_CONTROL(4), 0,
+ 0x7f, 0, pcm1681_dac_tlv),
+ SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume",
+ PCM1681_ATT_CONTROL(5), PCM1681_ATT_CONTROL(6), 0,
+ 0x7f, 0, pcm1681_dac_tlv),
+ SOC_DOUBLE_R_TLV("Channel 7/8 Playback Volume",
+ PCM1681_ATT_CONTROL(7), PCM1681_ATT_CONTROL(8), 0,
+ 0x7f, 0, pcm1681_dac_tlv),
+ SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0,
+ pcm1681_get_deemph, pcm1681_put_deemph),
+};
+
+static struct snd_soc_dai_driver pcm1681_dai = {
+ .name = "pcm1681-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = PCM1681_PCM_RATES,
+ .formats = PCM1681_PCM_FORMATS,
+ },
+ .ops = &pcm1681_dai_ops,
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id pcm1681_dt_ids[] = {
+ { .compatible = "ti,pcm1681", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, pcm1681_dt_ids);
+#endif
+
+static const struct regmap_config pcm1681_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .max_register = ARRAY_SIZE(pcm1681_reg_defaults) + 1,
+ .reg_defaults = pcm1681_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults),
+ .writeable_reg = pcm1681_writeable_reg,
+ .readable_reg = pcm1681_accessible_reg,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = {
+ .controls = pcm1681_controls,
+ .num_controls = ARRAY_SIZE(pcm1681_controls),
+ .dapm_widgets = pcm1681_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(pcm1681_dapm_widgets),
+ .dapm_routes = pcm1681_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(pcm1681_dapm_routes),
+};
+
+static const struct i2c_device_id pcm1681_i2c_id[] = {
+ {"pcm1681", 0},
+ {}
+};
+MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id);
+
+static int pcm1681_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ int ret;
+ struct pcm1681_private *priv;
+
+ priv = devm_kzalloc(&client->dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ priv->regmap = devm_regmap_init_i2c(client, &pcm1681_regmap);
+ if (IS_ERR(priv->regmap)) {
+ ret = PTR_ERR(priv->regmap);
+ dev_err(&client->dev, "Failed to create regmap: %d\n", ret);
+ return ret;
+ }
+
+ i2c_set_clientdata(client, priv);
+
+ return snd_soc_register_codec(&client->dev, &soc_codec_dev_pcm1681,
+ &pcm1681_dai, 1);
+}
+
+static int pcm1681_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static struct i2c_driver pcm1681_i2c_driver = {
+ .driver = {
+ .name = "pcm1681",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(pcm1681_dt_ids),
+ },
+ .id_table = pcm1681_i2c_id,
+ .probe = pcm1681_i2c_probe,
+ .remove = pcm1681_i2c_remove,
+};
+
+module_i2c_driver(pcm1681_i2c_driver);
+
+MODULE_DESCRIPTION("Texas Instruments PCM1681 ALSA SoC Codec Driver");
+MODULE_AUTHOR("Marek Belisko <marek.belisko@streamunlimited.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c
new file mode 100644
index 0000000..2a8eccf
--- /dev/null
+++ b/sound/soc/codecs/pcm1792a.c
@@ -0,0 +1,257 @@
+/*
+ * PCM1792A ASoC codec driver
+ *
+ * Copyright (c) Amarula Solutions B.V. 2013
+ *
+ * Michael Trimarchi <michael@amarulasolutions.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <linux/spi/spi.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <linux/of_device.h>
+
+#include "pcm1792a.h"
+
+#define PCM1792A_DAC_VOL_LEFT 0x10
+#define PCM1792A_DAC_VOL_RIGHT 0x11
+#define PCM1792A_FMT_CONTROL 0x12
+#define PCM1792A_SOFT_MUTE PCM1792A_FMT_CONTROL
+
+#define PCM1792A_FMT_MASK 0x70
+#define PCM1792A_FMT_SHIFT 4
+#define PCM1792A_MUTE_MASK 0x01
+#define PCM1792A_MUTE_SHIFT 0
+#define PCM1792A_ATLD_ENABLE (1 << 7)
+
+static const struct reg_default pcm1792a_reg_defaults[] = {
+ { 0x10, 0xff },
+ { 0x11, 0xff },
+ { 0x12, 0x50 },
+ { 0x13, 0x00 },
+ { 0x14, 0x00 },
+ { 0x15, 0x01 },
+ { 0x16, 0x00 },
+ { 0x17, 0x00 },
+};
+
+static bool pcm1792a_accessible_reg(struct device *dev, unsigned int reg)
+{
+ return reg >= 0x10 && reg <= 0x17;
+}
+
+static bool pcm1792a_writeable_reg(struct device *dev, unsigned register reg)
+{
+ bool accessible;
+
+ accessible = pcm1792a_accessible_reg(dev, reg);
+
+ return accessible && reg != 0x16 && reg != 0x17;
+}
+
+struct pcm1792a_private {
+ struct regmap *regmap;
+ unsigned int format;
+ unsigned int rate;
+};
+
+static int pcm1792a_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int format)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ priv->format = format;
+
+ return 0;
+}
+
+static int pcm1792a_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ ret = regmap_update_bits(priv->regmap, PCM1792A_SOFT_MUTE,
+ PCM1792A_MUTE_MASK, !!mute);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int pcm1792a_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec);
+ int val = 0, ret;
+ int pcm_format = params_format(params);
+
+ priv->rate = params_rate(params);
+
+ switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_RIGHT_J:
+ if (pcm_format == SNDRV_PCM_FORMAT_S24_LE ||
+ pcm_format == SNDRV_PCM_FORMAT_S32_LE)
+ val = 0x02;
+ else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE)
+ val = 0x00;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ if (pcm_format == SNDRV_PCM_FORMAT_S24_LE ||
+ pcm_format == SNDRV_PCM_FORMAT_S32_LE)
+ val = 0x05;
+ else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE)
+ val = 0x04;
+ break;
+ default:
+ dev_err(codec->dev, "Invalid DAI format\n");
+ return -EINVAL;
+ }
+
+ val = val << PCM1792A_FMT_SHIFT | PCM1792A_ATLD_ENABLE;
+
+ ret = regmap_update_bits(priv->regmap, PCM1792A_FMT_CONTROL,
+ PCM1792A_FMT_MASK | PCM1792A_ATLD_ENABLE, val);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops pcm1792a_dai_ops = {
+ .set_fmt = pcm1792a_set_dai_fmt,
+ .hw_params = pcm1792a_hw_params,
+ .digital_mute = pcm1792a_digital_mute,
+};
+
+static const DECLARE_TLV_DB_SCALE(pcm1792a_dac_tlv, -12000, 50, 1);
+
+static const struct snd_kcontrol_new pcm1792a_controls[] = {
+ SOC_DOUBLE_R_RANGE_TLV("DAC Playback Volume", PCM1792A_DAC_VOL_LEFT,
+ PCM1792A_DAC_VOL_RIGHT, 0, 0xf, 0xff, 0,
+ pcm1792a_dac_tlv),
+};
+
+static const struct snd_soc_dapm_widget pcm1792a_dapm_widgets[] = {
+SND_SOC_DAPM_OUTPUT("IOUTL+"),
+SND_SOC_DAPM_OUTPUT("IOUTL-"),
+SND_SOC_DAPM_OUTPUT("IOUTR+"),
+SND_SOC_DAPM_OUTPUT("IOUTR-"),
+};
+
+static const struct snd_soc_dapm_route pcm1792a_dapm_routes[] = {
+ { "IOUTL+", NULL, "Playback" },
+ { "IOUTL-", NULL, "Playback" },
+ { "IOUTR+", NULL, "Playback" },
+ { "IOUTR-", NULL, "Playback" },
+};
+
+static struct snd_soc_dai_driver pcm1792a_dai = {
+ .name = "pcm1792a-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PCM1792A_RATES,
+ .formats = PCM1792A_FORMATS, },
+ .ops = &pcm1792a_dai_ops,
+};
+
+static const struct of_device_id pcm1792a_of_match[] = {
+ { .compatible = "ti,pcm1792a", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, pcm1792a_of_match);
+
+static const struct regmap_config pcm1792a_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .max_register = 24,
+ .reg_defaults = pcm1792a_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(pcm1792a_reg_defaults),
+ .writeable_reg = pcm1792a_writeable_reg,
+ .readable_reg = pcm1792a_accessible_reg,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_pcm1792a = {
+ .controls = pcm1792a_controls,
+ .num_controls = ARRAY_SIZE(pcm1792a_controls),
+ .dapm_widgets = pcm1792a_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(pcm1792a_dapm_widgets),
+ .dapm_routes = pcm1792a_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(pcm1792a_dapm_routes),
+};
+
+static int pcm1792a_spi_probe(struct spi_device *spi)
+{
+ struct pcm1792a_private *pcm1792a;
+ int ret;
+
+ pcm1792a = devm_kzalloc(&spi->dev, sizeof(struct pcm1792a_private),
+ GFP_KERNEL);
+ if (!pcm1792a)
+ return -ENOMEM;
+
+ spi_set_drvdata(spi, pcm1792a);
+
+ pcm1792a->regmap = devm_regmap_init_spi(spi, &pcm1792a_regmap);
+ if (IS_ERR(pcm1792a->regmap)) {
+ ret = PTR_ERR(pcm1792a->regmap);
+ dev_err(&spi->dev, "Failed to register regmap: %d\n", ret);
+ return ret;
+ }
+
+ return snd_soc_register_codec(&spi->dev,
+ &soc_codec_dev_pcm1792a, &pcm1792a_dai, 1);
+}
+
+static int pcm1792a_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ return 0;
+}
+
+static const struct spi_device_id pcm1792a_spi_ids[] = {
+ { "pcm1792a", 0 },
+ { },
+};
+MODULE_DEVICE_TABLE(spi, pcm1792a_spi_ids);
+
+static struct spi_driver pcm1792a_codec_driver = {
+ .driver = {
+ .name = "pcm1792a",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(pcm1792a_of_match),
+ },
+ .id_table = pcm1792a_spi_ids,
+ .probe = pcm1792a_spi_probe,
+ .remove = pcm1792a_spi_remove,
+};
+
+module_spi_driver(pcm1792a_codec_driver);
+
+MODULE_DESCRIPTION("ASoC PCM1792A driver");
+MODULE_AUTHOR("Michael Trimarchi <michael@amarulasolutions.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/pcm1792a.h b/sound/soc/codecs/pcm1792a.h
new file mode 100644
index 0000000..7a83d1f
--- /dev/null
+++ b/sound/soc/codecs/pcm1792a.h
@@ -0,0 +1,26 @@
+/*
+ * definitions for PCM1792A
+ *
+ * Copyright 2013 Amarula Solutions
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#ifndef __PCM1792A_H__
+#define __PCM1792A_H__
+
+#define PCM1792A_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_8000_48000 | \
+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000)
+
+#define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S16_LE)
+
+#endif
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
index f2a6282..b6618c4 100644
--- a/sound/soc/codecs/pcm3008.c
+++ b/sound/soc/codecs/pcm3008.c
@@ -28,7 +28,54 @@
#include "pcm3008.h"
-#define PCM3008_VERSION "0.2"
+static int pcm3008_dac_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct pcm3008_setup_data *setup = codec->dev->platform_data;
+
+ gpio_set_value_cansleep(setup->pdda_pin,
+ SND_SOC_DAPM_EVENT_ON(event));
+
+ return 0;
+}
+
+static int pcm3008_adc_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct pcm3008_setup_data *setup = codec->dev->platform_data;
+
+ gpio_set_value_cansleep(setup->pdad_pin,
+ SND_SOC_DAPM_EVENT_ON(event));
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget pcm3008_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("VINL"),
+SND_SOC_DAPM_INPUT("VINR"),
+
+SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, pcm3008_dac_ev,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_ADC_E("ADC", NULL, SND_SOC_NOPM, 0, 0, pcm3008_adc_ev,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+SND_SOC_DAPM_OUTPUT("VOUTL"),
+SND_SOC_DAPM_OUTPUT("VOUTR"),
+};
+
+static const struct snd_soc_dapm_route pcm3008_dapm_routes[] = {
+ { "PCM3008 Capture", NULL, "ADC" },
+ { "ADC", NULL, "VINL" },
+ { "ADC", NULL, "VINR" },
+
+ { "DAC", NULL, "PCM3008 Playback" },
+ { "VOUTL", NULL, "DAC" },
+ { "VOUTR", NULL, "DAC" },
+};
#define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000)
@@ -51,20 +98,20 @@
},
};
-static void pcm3008_gpio_free(struct pcm3008_setup_data *setup)
-{
- gpio_free(setup->dem0_pin);
- gpio_free(setup->dem1_pin);
- gpio_free(setup->pdad_pin);
- gpio_free(setup->pdda_pin);
-}
+static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = {
+ .dapm_widgets = pcm3008_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(pcm3008_dapm_widgets),
+ .dapm_routes = pcm3008_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(pcm3008_dapm_routes),
+};
-static int pcm3008_soc_probe(struct snd_soc_codec *codec)
+static int pcm3008_codec_probe(struct platform_device *pdev)
{
- struct pcm3008_setup_data *setup = codec->dev->platform_data;
- int ret = 0;
+ struct pcm3008_setup_data *setup = pdev->dev.platform_data;
+ int ret;
- printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION);
+ if (!setup)
+ return -EINVAL;
/* DEM1 DEM0 DE-EMPHASIS_MODE
* Low Low De-emphasis 44.1 kHz ON
@@ -74,83 +121,29 @@
*/
/* Configure DEM0 GPIO (turning OFF DAC De-emphasis). */
- ret = gpio_request(setup->dem0_pin, "codec_dem0");
- if (ret == 0)
- ret = gpio_direction_output(setup->dem0_pin, 1);
+ ret = devm_gpio_request_one(&pdev->dev, setup->dem0_pin,
+ GPIOF_OUT_INIT_HIGH, "codec_dem0");
if (ret != 0)
- goto gpio_err;
+ return ret;
/* Configure DEM1 GPIO (turning OFF DAC De-emphasis). */
- ret = gpio_request(setup->dem1_pin, "codec_dem1");
- if (ret == 0)
- ret = gpio_direction_output(setup->dem1_pin, 0);
+ ret = devm_gpio_request_one(&pdev->dev, setup->dem1_pin,
+ GPIOF_OUT_INIT_LOW, "codec_dem1");
if (ret != 0)
- goto gpio_err;
+ return ret;
/* Configure PDAD GPIO. */
- ret = gpio_request(setup->pdad_pin, "codec_pdad");
- if (ret == 0)
- ret = gpio_direction_output(setup->pdad_pin, 1);
+ ret = devm_gpio_request_one(&pdev->dev, setup->pdad_pin,
+ GPIOF_OUT_INIT_LOW, "codec_pdad");
if (ret != 0)
- goto gpio_err;
+ return ret;
/* Configure PDDA GPIO. */
- ret = gpio_request(setup->pdda_pin, "codec_pdda");
- if (ret == 0)
- ret = gpio_direction_output(setup->pdda_pin, 1);
+ ret = devm_gpio_request_one(&pdev->dev, setup->pdda_pin,
+ GPIOF_OUT_INIT_LOW, "codec_pdda");
if (ret != 0)
- goto gpio_err;
+ return ret;
- return ret;
-
-gpio_err:
- pcm3008_gpio_free(setup);
-
- return ret;
-}
-
-static int pcm3008_soc_remove(struct snd_soc_codec *codec)
-{
- struct pcm3008_setup_data *setup = codec->dev->platform_data;
-
- pcm3008_gpio_free(setup);
- return 0;
-}
-
-#ifdef CONFIG_PM
-static int pcm3008_soc_suspend(struct snd_soc_codec *codec)
-{
- struct pcm3008_setup_data *setup = codec->dev->platform_data;
-
- gpio_set_value(setup->pdad_pin, 0);
- gpio_set_value(setup->pdda_pin, 0);
-
- return 0;
-}
-
-static int pcm3008_soc_resume(struct snd_soc_codec *codec)
-{
- struct pcm3008_setup_data *setup = codec->dev->platform_data;
-
- gpio_set_value(setup->pdad_pin, 1);
- gpio_set_value(setup->pdda_pin, 1);
-
- return 0;
-}
-#else
-#define pcm3008_soc_suspend NULL
-#define pcm3008_soc_resume NULL
-#endif
-
-static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = {
- .probe = pcm3008_soc_probe,
- .remove = pcm3008_soc_remove,
- .suspend = pcm3008_soc_suspend,
- .resume = pcm3008_soc_resume,
-};
-
-static int pcm3008_codec_probe(struct platform_device *pdev)
-{
return snd_soc_register_codec(&pdev->dev,
&soc_codec_dev_pcm3008, &pcm3008_dai, 1);
}
@@ -158,6 +151,7 @@
static int pcm3008_codec_remove(struct platform_device *pdev)
{
snd_soc_unregister_codec(&pdev->dev);
+
return 0;
}
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index ce585e3..4db7314 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -737,29 +737,6 @@
RT5640_M_BST1_MM_SFT, 1, 1),
};
-/* INL/R source */
-static const char * const rt5640_inl_src[] = {
- "IN2P", "MONOP"
-};
-
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_inl_enum, RT5640_INL_INR_VOL,
- RT5640_INL_SEL_SFT, rt5640_inl_src);
-
-static const struct snd_kcontrol_new rt5640_inl_mux =
- SOC_DAPM_ENUM("INL source", rt5640_inl_enum);
-
-static const char * const rt5640_inr_src[] = {
- "IN2N", "MONON"
-};
-
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_inr_enum, RT5640_INL_INR_VOL,
- RT5640_INR_SEL_SFT, rt5640_inr_src);
-
-static const struct snd_kcontrol_new rt5640_inr_mux =
- SOC_DAPM_ENUM("INR source", rt5640_inr_enum);
-
/* Stereo ADC source */
static const char * const rt5640_stereo_adc1_src[] = {
"DIG MIX", "ADC"
@@ -1005,9 +982,6 @@
RT5640_PWR_IN_L_BIT, 0, NULL, 0),
SND_SOC_DAPM_PGA("INR VOL", RT5640_PWR_VOL,
RT5640_PWR_IN_R_BIT, 0, NULL, 0),
- /* IN Mux */
- SND_SOC_DAPM_MUX("INL Mux", SND_SOC_NOPM, 0, 0, &rt5640_inl_mux),
- SND_SOC_DAPM_MUX("INR Mux", SND_SOC_NOPM, 0, 0, &rt5640_inr_mux),
/* REC Mixer */
SND_SOC_DAPM_MIXER("RECMIXL", RT5640_PWR_MIXER, RT5640_PWR_RM_L_BIT, 0,
rt5640_rec_l_mix, ARRAY_SIZE(rt5640_rec_l_mix)),
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 760e8bf..1f4093f 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -654,16 +654,19 @@
snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP,
SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP);
+
+ /* if using pll, clk_ctrl must be set after pll power up */
+ snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl);
} else {
+ /* otherwise, clk_ctrl must be set before pll power down */
+ snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl);
+
/* power down pll */
snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP,
0);
}
- /* if using pll, clk_ctrl must be set after pll power up */
- snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl);
-
return 0;
}
@@ -1480,6 +1483,7 @@
static const struct regmap_config sgtl5000_regmap = {
.reg_bits = 16,
.val_bits = 16,
+ .reg_stride = 2,
.max_register = SGTL5000_MAX_REG_OFFSET,
.volatile_reg = sgtl5000_volatile,
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
index 73e205c..38f3b10 100644
--- a/sound/soc/codecs/si476x.c
+++ b/sound/soc/codecs/si476x.c
@@ -102,6 +102,16 @@
return err;
}
+static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = {
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+};
+
+static const struct snd_soc_dapm_route si476x_dapm_routes[] = {
+ { "Capture", NULL, "LOUT" },
+ { "Capture", NULL, "ROUT" },
+};
+
static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
@@ -260,6 +270,10 @@
.probe = si476x_codec_probe,
.read = si476x_codec_read,
.write = si476x_codec_write,
+ .dapm_widgets = si476x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets),
+ .dapm_routes = si476x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(si476x_dapm_routes),
};
static int si476x_platform_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c
index e9d7881..e3501f4 100644
--- a/sound/soc/codecs/spdif_receiver.c
+++ b/sound/soc/codecs/spdif_receiver.c
@@ -23,11 +23,26 @@
#include <sound/initval.h>
#include <linux/of.h>
+static const struct snd_soc_dapm_widget dir_widgets[] = {
+ SND_SOC_DAPM_INPUT("spdif-in"),
+};
+
+static const struct snd_soc_dapm_route dir_routes[] = {
+ { "Capture", NULL, "spdif-in" },
+};
+
#define STUB_RATES SNDRV_PCM_RATE_8000_192000
#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
-static struct snd_soc_codec_driver soc_codec_spdif_dir;
+static struct snd_soc_codec_driver soc_codec_spdif_dir = {
+ .dapm_widgets = dir_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(dir_widgets),
+ .dapm_routes = dir_routes,
+ .num_dapm_routes = ARRAY_SIZE(dir_routes),
+};
static struct snd_soc_dai_driver dir_stub_dai = {
.name = "dir-hifi",
diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c
index 1828049..a078aa3 100644
--- a/sound/soc/codecs/spdif_transmitter.c
+++ b/sound/soc/codecs/spdif_transmitter.c
@@ -25,10 +25,24 @@
#define DRV_NAME "spdif-dit"
#define STUB_RATES SNDRV_PCM_RATE_8000_96000
-#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+static const struct snd_soc_dapm_widget dit_widgets[] = {
+ SND_SOC_DAPM_OUTPUT("spdif-out"),
+};
-static struct snd_soc_codec_driver soc_codec_spdif_dit;
+static const struct snd_soc_dapm_route dit_routes[] = {
+ { "spdif-out", NULL, "Playback" },
+};
+
+static struct snd_soc_codec_driver soc_codec_spdif_dit = {
+ .dapm_widgets = dit_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(dit_widgets),
+ .dapm_routes = dit_routes,
+ .num_dapm_routes = ARRAY_SIZE(dit_routes),
+};
static struct snd_soc_dai_driver dit_stub_dai = {
.name = "dit-hifi",
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index cfb55fe..06edb39 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -363,16 +363,18 @@
}
if (!sta32x->shutdown)
- schedule_delayed_work(&sta32x->watchdog_work,
- round_jiffies_relative(HZ));
+ queue_delayed_work(system_power_efficient_wq,
+ &sta32x->watchdog_work,
+ round_jiffies_relative(HZ));
}
static void sta32x_watchdog_start(struct sta32x_priv *sta32x)
{
if (sta32x->pdata->needs_esd_watchdog) {
sta32x->shutdown = 0;
- schedule_delayed_work(&sta32x->watchdog_work,
- round_jiffies_relative(HZ));
+ queue_delayed_work(system_power_efficient_wq,
+ &sta32x->watchdog_work,
+ round_jiffies_relative(HZ));
}
}
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index b1f6982..7b8f3d9 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -29,7 +29,7 @@
/* AIC26 driver private data */
struct aic26 {
struct spi_device *spi;
- struct snd_soc_codec codec;
+ struct snd_soc_codec *codec;
int master;
int datfm;
int mclk;
@@ -119,6 +119,22 @@
return 0;
}
+static const struct snd_soc_dapm_widget tlv320aic26_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("MICIN"),
+SND_SOC_DAPM_INPUT("AUX"),
+
+SND_SOC_DAPM_OUTPUT("HPL"),
+SND_SOC_DAPM_OUTPUT("HPR"),
+};
+
+static const struct snd_soc_dapm_route tlv320aic26_dapm_routes[] = {
+ { "Capture", NULL, "MICIN" },
+ { "Capture", NULL, "AUX" },
+
+ { "HPL", NULL, "Playback" },
+ { "HPR", NULL, "Playback" },
+};
+
/* ---------------------------------------------------------------------
* Digital Audio Interface Operations
*/
@@ -174,9 +190,9 @@
dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval);
qval = 0;
reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
- aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
+ snd_soc_write(codec, AIC26_REG_PLL_PROG1, reg);
reg = dval << 2;
- aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg);
+ snd_soc_write(codec, AIC26_REG_PLL_PROG2, reg);
/* Audio Control 3 (master mode, fsref rate) */
reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3);
@@ -185,13 +201,13 @@
reg |= 0x0800;
if (fsref == 48000)
reg |= 0x2000;
- aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
+ snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
/* Audio Control 1 (FSref divisor) */
reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1);
reg &= ~0x0fff;
reg |= wlen | aic26->datfm | (divisor << 3) | divisor;
- aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg);
+ snd_soc_write(codec, AIC26_REG_AUDIO_CTRL1, reg);
return 0;
}
@@ -212,7 +228,7 @@
reg |= 0x8080;
else
reg &= ~0x8080;
- aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg);
+ snd_soc_write(codec, AIC26_REG_DAC_GAIN, reg);
return 0;
}
@@ -330,7 +346,7 @@
struct aic26 *aic26 = dev_get_drvdata(dev);
int val, amp, freq, len;
- val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
+ val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2);
amp = (val >> 12) & 0x7;
freq = (125 << ((val >> 8) & 0x7)) >> 1;
len = 2 * (1 + ((val >> 4) & 0xf));
@@ -346,9 +362,9 @@
struct aic26 *aic26 = dev_get_drvdata(dev);
int val;
- val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
+ val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2);
val |= 0x8000;
- aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val);
+ snd_soc_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val);
return count;
}
@@ -360,25 +376,26 @@
*/
static int aic26_probe(struct snd_soc_codec *codec)
{
+ struct aic26 *aic26 = dev_get_drvdata(codec->dev);
int ret, err, i, reg;
- dev_info(codec->dev, "Probing AIC26 SoC CODEC driver\n");
+ aic26->codec = codec;
/* Reset the codec to power on defaults */
- aic26_reg_write(codec, AIC26_REG_RESET, 0xBB00);
+ snd_soc_write(codec, AIC26_REG_RESET, 0xBB00);
/* Power up CODEC */
- aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0);
+ snd_soc_write(codec, AIC26_REG_POWER_CTRL, 0);
/* Audio Control 3 (master mode, fsref rate) */
- reg = aic26_reg_read(codec, AIC26_REG_AUDIO_CTRL3);
+ reg = snd_soc_read(codec, AIC26_REG_AUDIO_CTRL3);
reg &= ~0xf800;
reg |= 0x0800; /* set master mode */
- aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
+ snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
/* Fill register cache */
for (i = 0; i < codec->driver->reg_cache_size; i++)
- aic26_reg_read(codec, i);
+ snd_soc_read(codec, i);
/* Register the sysfs files for debugging */
/* Create SysFS files */
@@ -401,6 +418,10 @@
.write = aic26_reg_write,
.reg_cache_size = AIC26_NUM_REGS,
.reg_word_size = sizeof(u16),
+ .dapm_widgets = tlv320aic26_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(tlv320aic26_dapm_widgets),
+ .dapm_routes = tlv320aic26_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(tlv320aic26_dapm_routes),
};
/* ---------------------------------------------------------------------
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index fec0db0..6e3f269 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1474,6 +1474,7 @@
{ "tlv320aic3x", AIC3X_MODEL_3X },
{ "tlv320aic33", AIC3X_MODEL_33 },
{ "tlv320aic3007", AIC3X_MODEL_3007 },
+ { "tlv320aic3106", AIC3X_MODEL_3X },
{ }
};
MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
@@ -1564,6 +1565,9 @@
#if defined(CONFIG_OF)
static const struct of_device_id tlv320aic3x_of_match[] = {
{ .compatible = "ti,tlv320aic3x", },
+ { .compatible = "ti,tlv320aic33" },
+ { .compatible = "ti,tlv320aic3007" },
+ { .compatible = "ti,tlv320aic3106" },
{},
};
MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match);
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 8e6e5b0..1e3884d 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -137,8 +137,6 @@
/* codec private data */
struct twl4030_priv {
- struct snd_soc_codec codec;
-
unsigned int codec_powered;
/* reference counts of AIF/APLL users */
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index d6c5bf1..3c79dbb 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -429,7 +429,8 @@
struct snd_soc_codec *codec = data;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
- schedule_delayed_work(&priv->hs_jack.work, msecs_to_jiffies(200));
+ queue_delayed_work(system_power_efficient_wq,
+ &priv->hs_jack.work, msecs_to_jiffies(200));
return IRQ_HANDLED;
}
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 6d0aa44..c94d4c1 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -325,7 +325,6 @@
static int uda134x_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
- u8 reg;
struct uda134x_platform_data *pd = codec->control_data;
int i;
u8 *cache = codec->reg_cache;
@@ -334,23 +333,6 @@
switch (level) {
case SND_SOC_BIAS_ON:
- /* ADC, DAC on */
- switch (pd->model) {
- case UDA134X_UDA1340:
- case UDA134X_UDA1344:
- case UDA134X_UDA1345:
- reg = uda134x_read_reg_cache(codec, UDA134X_DATA011);
- uda134x_write(codec, UDA134X_DATA011, reg | 0x03);
- break;
- case UDA134X_UDA1341:
- reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
- uda134x_write(codec, UDA134X_STATUS1, reg | 0x03);
- break;
- default:
- printk(KERN_ERR "UDA134X SoC codec: "
- "unsupported model %d\n", pd->model);
- return -EINVAL;
- }
break;
case SND_SOC_BIAS_PREPARE:
/* power on */
@@ -362,23 +344,6 @@
}
break;
case SND_SOC_BIAS_STANDBY:
- /* ADC, DAC power off */
- switch (pd->model) {
- case UDA134X_UDA1340:
- case UDA134X_UDA1344:
- case UDA134X_UDA1345:
- reg = uda134x_read_reg_cache(codec, UDA134X_DATA011);
- uda134x_write(codec, UDA134X_DATA011, reg & ~(0x03));
- break;
- case UDA134X_UDA1341:
- reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
- uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03));
- break;
- default:
- printk(KERN_ERR "UDA134X SoC codec: "
- "unsupported model %d\n", pd->model);
- return -EINVAL;
- }
break;
case SND_SOC_BIAS_OFF:
/* power off */
@@ -450,6 +415,37 @@
SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
};
+/* UDA1341 has the DAC/ADC power down in STATUS1 */
+static const struct snd_soc_dapm_widget uda1341_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_STATUS1, 0, 0),
+ SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_STATUS1, 1, 0),
+};
+
+/* UDA1340/4/5 has the DAC/ADC pwoer down in DATA0 11 */
+static const struct snd_soc_dapm_widget uda1340_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_DATA011, 0, 0),
+ SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_DATA011, 1, 0),
+};
+
+/* Common DAPM widgets */
+static const struct snd_soc_dapm_widget uda134x_dapm_widgets[] = {
+ SND_SOC_DAPM_INPUT("VINL1"),
+ SND_SOC_DAPM_INPUT("VINR1"),
+ SND_SOC_DAPM_INPUT("VINL2"),
+ SND_SOC_DAPM_INPUT("VINR2"),
+ SND_SOC_DAPM_OUTPUT("VOUTL"),
+ SND_SOC_DAPM_OUTPUT("VOUTR"),
+};
+
+static const struct snd_soc_dapm_route uda134x_dapm_routes[] = {
+ { "ADC", NULL, "VINL1" },
+ { "ADC", NULL, "VINR1" },
+ { "ADC", NULL, "VINL2" },
+ { "ADC", NULL, "VINR2" },
+ { "VOUTL", NULL, "DAC" },
+ { "VOUTR", NULL, "DAC" },
+};
+
static const struct snd_soc_dai_ops uda134x_dai_ops = {
.startup = uda134x_startup,
.shutdown = uda134x_shutdown,
@@ -485,6 +481,8 @@
{
struct uda134x_priv *uda134x;
struct uda134x_platform_data *pd = codec->card->dev->platform_data;
+ const struct snd_soc_dapm_widget *widgets;
+ unsigned num_widgets;
int ret;
@@ -526,6 +524,22 @@
else
uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ if (pd->model == UDA134X_UDA1341) {
+ widgets = uda1341_dapm_widgets;
+ num_widgets = ARRAY_SIZE(uda1341_dapm_widgets);
+ } else {
+ widgets = uda1340_dapm_widgets;
+ num_widgets = ARRAY_SIZE(uda1340_dapm_widgets);
+ }
+
+ ret = snd_soc_dapm_new_controls(&codec->dapm, widgets, num_widgets);
+ if (ret) {
+ printk(KERN_ERR "%s failed to register dapm controls: %d",
+ __func__, ret);
+ kfree(uda134x);
+ return ret;
+ }
+
switch (pd->model) {
case UDA134X_UDA1340:
case UDA134X_UDA1344:
@@ -599,6 +613,10 @@
.read = uda134x_read_reg_cache,
.write = uda134x_write,
.set_bias_level = uda134x_set_bias_level,
+ .dapm_widgets = uda134x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(uda134x_dapm_widgets),
+ .dapm_routes = uda134x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(uda134x_dapm_routes),
};
static int uda134x_codec_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index 54cd3da..b7ab2ef 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -290,6 +290,18 @@
snd_wl1273_fm_volume_get, snd_wl1273_fm_volume_put),
};
+static const struct snd_soc_dapm_widget wl1273_dapm_widgets[] = {
+ SND_SOC_DAPM_INPUT("RX"),
+
+ SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route wl1273_dapm_routes[] = {
+ { "Capture", NULL, "RX" },
+
+ { "TX", NULL, "Playback" },
+};
+
static int wl1273_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
@@ -483,6 +495,11 @@
static struct snd_soc_codec_driver soc_codec_dev_wl1273 = {
.probe = wl1273_probe,
.remove = wl1273_remove,
+
+ .dapm_widgets = wl1273_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wl1273_dapm_widgets),
+ .dapm_routes = wl1273_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(wl1273_dapm_routes),
};
static int wl1273_platform_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index 10adc41..d5ebcb0 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -420,7 +420,7 @@
xfer->codec = codec;
list_add_tail(&xfer->list, &xfer_list);
- out = kzalloc(len, GFP_KERNEL);
+ out = kzalloc(len, GFP_KERNEL | GFP_DMA);
if (!out) {
dev_err(codec->dev,
"Failed to allocate RX buffer\n");
@@ -429,7 +429,7 @@
}
xfer->t.rx_buf = out;
- img = kzalloc(len, GFP_KERNEL);
+ img = kzalloc(len, GFP_KERNEL | GFP_DMA);
if (!img) {
dev_err(codec->dev,
"Failed to allocate image buffer\n");
@@ -523,14 +523,14 @@
dev_dbg(codec->dev, "Downloading %zu byte stage 2 loader\n", fw->size);
/* Copy to local buffer first as vmalloc causes problems for dma */
- img = kzalloc(fw->size, GFP_KERNEL);
+ img = kzalloc(fw->size, GFP_KERNEL | GFP_DMA);
if (!img) {
dev_err(codec->dev, "Failed to allocate image buffer\n");
ret = -ENOMEM;
goto abort2;
}
- out = kzalloc(fw->size, GFP_KERNEL);
+ out = kzalloc(fw->size, GFP_KERNEL | GFP_DMA);
if (!out) {
dev_err(codec->dev, "Failed to allocate output buffer\n");
ret = -ENOMEM;
@@ -670,14 +670,14 @@
ret = -ENOMEM;
len = pll_rec.length + 8;
- out = kzalloc(len, GFP_KERNEL);
+ out = kzalloc(len, GFP_KERNEL | GFP_DMA);
if (!out) {
dev_err(codec->dev,
"Failed to allocate RX buffer\n");
goto abort;
}
- img_swap = kzalloc(len, GFP_KERNEL);
+ img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA);
if (!img_swap) {
dev_err(codec->dev,
"Failed to allocate image buffer\n");
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 282fd23..8bbddc1 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -998,6 +998,8 @@
SND_SOC_DAPM_INPUT("IN3L"),
SND_SOC_DAPM_INPUT("IN3R"),
+SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"),
+
SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
@@ -1421,9 +1423,6 @@
{ "Tone Generator 1", NULL, "TONE" },
{ "Tone Generator 2", NULL, "TONE" },
- { "Mic Mute Mixer", NULL, "Noise Mixer" },
- { "Mic Mute Mixer", NULL, "Mic Mixer" },
-
{ "AIF1 Capture", NULL, "AIF1TX1" },
{ "AIF1 Capture", NULL, "AIF1TX2" },
{ "AIF1 Capture", NULL, "AIF1TX3" },
@@ -1499,23 +1498,6 @@
{ "IN3L PGA", NULL, "IN3L" },
{ "IN3R PGA", NULL, "IN3R" },
- { "ASRC1L", NULL, "ASRC1L Input" },
- { "ASRC1R", NULL, "ASRC1R Input" },
- { "ASRC2L", NULL, "ASRC2L Input" },
- { "ASRC2R", NULL, "ASRC2R Input" },
-
- { "ISRC1DEC1", NULL, "ISRC1DEC1 Input" },
- { "ISRC1DEC2", NULL, "ISRC1DEC2 Input" },
-
- { "ISRC1INT1", NULL, "ISRC1INT1 Input" },
- { "ISRC1INT2", NULL, "ISRC1INT2 Input" },
-
- { "ISRC2DEC1", NULL, "ISRC2DEC1 Input" },
- { "ISRC2DEC2", NULL, "ISRC2DEC2 Input" },
-
- { "ISRC2INT1", NULL, "ISRC2INT1 Input" },
- { "ISRC2INT2", NULL, "ISRC2INT2 Input" },
-
ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
@@ -1567,22 +1549,25 @@
ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
- ARIZONA_MUX_ROUTES("ASRC1L"),
- ARIZONA_MUX_ROUTES("ASRC1R"),
- ARIZONA_MUX_ROUTES("ASRC2L"),
- ARIZONA_MUX_ROUTES("ASRC2R"),
+ ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"),
+ ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"),
- ARIZONA_MUX_ROUTES("ISRC1INT1"),
- ARIZONA_MUX_ROUTES("ISRC1INT2"),
+ ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"),
+ ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"),
+ ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"),
+ ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"),
- ARIZONA_MUX_ROUTES("ISRC1DEC1"),
- ARIZONA_MUX_ROUTES("ISRC1DEC2"),
+ ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"),
+ ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"),
- ARIZONA_MUX_ROUTES("ISRC2INT1"),
- ARIZONA_MUX_ROUTES("ISRC2INT2"),
+ ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"),
+ ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"),
- ARIZONA_MUX_ROUTES("ISRC2DEC1"),
- ARIZONA_MUX_ROUTES("ISRC2DEC2"),
+ ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"),
+ ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"),
+
+ ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"),
+ ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"),
ARIZONA_DSP_ROUTES("DSP1"),
@@ -1614,6 +1599,9 @@
{ "SPKDAT1R", NULL, "OUT5R" },
{ "MICSUPP", NULL, "SYSCLK" },
+
+ { "DRC1 Signal Activity", NULL, "DRC1L" },
+ { "DRC1 Signal Activity", NULL, "DRC1R" },
};
static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
@@ -1781,6 +1769,7 @@
return ret;
arizona_init_spk(codec);
+ arizona_init_gpio(codec);
snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 2e7cb4b..bbd6438 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -58,14 +58,10 @@
SOC_SINGLE(name " NG SPKDAT2R Switch", base, 11, 1, 0)
static const struct snd_kcontrol_new wm5110_snd_controls[] = {
-SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL,
- ARIZONA_IN1_OSR_SHIFT, 1, 0),
-SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL,
- ARIZONA_IN2_OSR_SHIFT, 1, 0),
-SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL,
- ARIZONA_IN3_OSR_SHIFT, 1, 0),
-SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL,
- ARIZONA_IN4_OSR_SHIFT, 1, 0),
+SOC_ENUM("IN1 OSR", arizona_in_dmic_osr[0]),
+SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]),
+SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]),
+SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]),
SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL,
ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
@@ -432,6 +428,9 @@
SND_SOC_DAPM_INPUT("IN4L"),
SND_SOC_DAPM_INPUT("IN4R"),
+SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"),
+SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"),
+
SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
@@ -842,9 +841,6 @@
{ "Tone Generator 1", NULL, "TONE" },
{ "Tone Generator 2", NULL, "TONE" },
- { "Mic Mute Mixer", NULL, "Noise Mixer" },
- { "Mic Mute Mixer", NULL, "Mic Mixer" },
-
{ "AIF1 Capture", NULL, "AIF1TX1" },
{ "AIF1 Capture", NULL, "AIF1TX2" },
{ "AIF1 Capture", NULL, "AIF1TX3" },
@@ -979,10 +975,13 @@
ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
- ARIZONA_MUX_ROUTES("ASRC1L"),
- ARIZONA_MUX_ROUTES("ASRC1R"),
- ARIZONA_MUX_ROUTES("ASRC2L"),
- ARIZONA_MUX_ROUTES("ASRC2R"),
+ ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"),
+ ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"),
+
+ ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"),
+ ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"),
+ ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"),
+ ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"),
{ "HPOUT1L", NULL, "OUT1L" },
{ "HPOUT1R", NULL, "OUT1R" },
@@ -1006,6 +1005,11 @@
{ "SPKDAT2R", NULL, "OUT6R" },
{ "MICSUPP", NULL, "SYSCLK" },
+
+ { "DRC1 Signal Activity", NULL, "DRC1L" },
+ { "DRC1 Signal Activity", NULL, "DRC1R" },
+ { "DRC2 Signal Activity", NULL, "DRC2L" },
+ { "DRC2 Signal Activity", NULL, "DRC2R" },
};
static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
@@ -1170,6 +1174,7 @@
return ret;
arizona_init_spk(codec);
+ arizona_init_gpio(codec);
snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 0e8b3aa..af1318d 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1301,7 +1301,8 @@
if (device_may_wakeup(wm8350->dev))
pm_wakeup_event(wm8350->dev, 250);
- schedule_delayed_work(&priv->hpl.work, msecs_to_jiffies(200));
+ queue_delayed_work(system_power_efficient_wq,
+ &priv->hpl.work, msecs_to_jiffies(200));
return IRQ_HANDLED;
}
@@ -1318,7 +1319,8 @@
if (device_may_wakeup(wm8350->dev))
pm_wakeup_event(wm8350->dev, 250);
- schedule_delayed_work(&priv->hpr.work, msecs_to_jiffies(200));
+ queue_delayed_work(system_power_efficient_wq,
+ &priv->hpr.work, msecs_to_jiffies(200));
return IRQ_HANDLED;
}
diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c
index 462f5e4..7b1a6d5 100644
--- a/sound/soc/codecs/wm8727.c
+++ b/sound/soc/codecs/wm8727.c
@@ -23,6 +23,16 @@
#include <sound/initval.h>
#include <sound/soc.h>
+static const struct snd_soc_dapm_widget wm8727_dapm_widgets[] = {
+SND_SOC_DAPM_OUTPUT("VOUTL"),
+SND_SOC_DAPM_OUTPUT("VOUTR"),
+};
+
+static const struct snd_soc_dapm_route wm8727_dapm_routes[] = {
+ { "VOUTL", NULL, "Playback" },
+ { "VOUTR", NULL, "Playback" },
+};
+
/*
* Note this is a simple chip with no configuration interface, sample rate is
* determined automatically by examining the Master clock and Bit clock ratios
@@ -43,7 +53,12 @@
},
};
-static struct snd_soc_codec_driver soc_codec_dev_wm8727;
+static struct snd_soc_codec_driver soc_codec_dev_wm8727 = {
+ .dapm_widgets = wm8727_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm8727_dapm_widgets),
+ .dapm_routes = wm8727_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(wm8727_dapm_routes),
+};
static int wm8727_probe(struct platform_device *pdev)
{
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 5276062..456bb8c 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -45,6 +45,7 @@
struct wm8731_priv {
struct regmap *regmap;
struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES];
+ const struct snd_pcm_hw_constraint_list *constraints;
unsigned int sysclk;
int sysclk_type;
int playback_fs;
@@ -290,6 +291,36 @@
{12000000, 88200, 136, 0xf, 0x1, 0x1},
};
+/* rates constraints */
+static const unsigned int wm8731_rates_12000000[] = {
+ 8000, 32000, 44100, 48000, 96000, 88200,
+};
+
+static const unsigned int wm8731_rates_12288000_18432000[] = {
+ 8000, 32000, 48000, 96000,
+};
+
+static const unsigned int wm8731_rates_11289600_16934400[] = {
+ 8000, 44100, 88200,
+};
+
+static const struct snd_pcm_hw_constraint_list wm8731_constraints_12000000 = {
+ .list = wm8731_rates_12000000,
+ .count = ARRAY_SIZE(wm8731_rates_12000000),
+};
+
+static const
+struct snd_pcm_hw_constraint_list wm8731_constraints_12288000_18432000 = {
+ .list = wm8731_rates_12288000_18432000,
+ .count = ARRAY_SIZE(wm8731_rates_12288000_18432000),
+};
+
+static const
+struct snd_pcm_hw_constraint_list wm8731_constraints_11289600_16934400 = {
+ .list = wm8731_rates_11289600_16934400,
+ .count = ARRAY_SIZE(wm8731_rates_11289600_16934400),
+};
+
static inline int get_coeff(int mclk, int rate)
{
int i;
@@ -362,17 +393,26 @@
}
switch (freq) {
- case 11289600:
+ case 0:
+ wm8731->constraints = NULL;
+ break;
case 12000000:
+ wm8731->constraints = &wm8731_constraints_12000000;
+ break;
case 12288000:
- case 16934400:
case 18432000:
- wm8731->sysclk = freq;
+ wm8731->constraints = &wm8731_constraints_12288000_18432000;
+ break;
+ case 16934400:
+ case 11289600:
+ wm8731->constraints = &wm8731_constraints_11289600_16934400;
break;
default:
return -EINVAL;
}
+ wm8731->sysclk = freq;
+
snd_soc_dapm_sync(&codec->dapm);
return 0;
@@ -475,12 +515,26 @@
return 0;
}
+static int wm8731_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(dai->codec);
+
+ if (wm8731->constraints)
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ wm8731->constraints);
+
+ return 0;
+}
+
#define WM8731_RATES SNDRV_PCM_RATE_8000_96000
#define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
static const struct snd_soc_dai_ops wm8731_dai_ops = {
+ .startup = wm8731_startup,
.hw_params = wm8731_hw_params,
.digital_mute = wm8731_mute,
.set_sysclk = wm8731_set_dai_sysclk,
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 0a4ab4c..d96ebf5 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1456,8 +1456,9 @@
if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
codec->dapm.bias_level = SND_SOC_BIAS_ON;
- schedule_delayed_work(&codec->dapm.delayed_work,
- msecs_to_jiffies(caps_charge));
+ queue_delayed_work(system_power_efficient_wq,
+ &codec->dapm.delayed_work,
+ msecs_to_jiffies(caps_charge));
}
return 0;
diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c
index f1fdbf6..8092495 100644
--- a/sound/soc/codecs/wm8782.c
+++ b/sound/soc/codecs/wm8782.c
@@ -26,6 +26,16 @@
#include <sound/initval.h>
#include <sound/soc.h>
+static const struct snd_soc_dapm_widget wm8782_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("AINL"),
+SND_SOC_DAPM_INPUT("AINR"),
+};
+
+static const struct snd_soc_dapm_route wm8782_dapm_routes[] = {
+ { "Capture", NULL, "AINL" },
+ { "Capture", NULL, "AINR" },
+};
+
static struct snd_soc_dai_driver wm8782_dai = {
.name = "wm8782",
.capture = {
@@ -40,7 +50,12 @@
},
};
-static struct snd_soc_codec_driver soc_codec_dev_wm8782;
+static struct snd_soc_codec_driver soc_codec_dev_wm8782 = {
+ .dapm_widgets = wm8782_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm8782_dapm_widgets),
+ .dapm_routes = wm8782_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(wm8782_dapm_routes),
+};
static int wm8782_probe(struct platform_device *pdev)
{
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 4c9fb14..4dfa8dc 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1012,7 +1012,7 @@
SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text);
static const struct snd_kcontrol_new liner_mux =
- SOC_DAPM_ENUM("LINEL Mux", liner_enum);
+ SOC_DAPM_ENUM("LINER Mux", liner_enum);
static const char *sidetone_text[] = {
"None", "Left", "Right"
@@ -1202,7 +1202,6 @@
break;
}
- snd_soc_dapm_new_widgets(dapm);
return 0;
}
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 0a4ffdd..f156010 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -263,8 +263,8 @@
SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0),
SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0),
-SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH,
- 0, 127, 0),
+SOC_DOUBLE_R_TLV("ADC PCM Capture Volume", WM8960_LADC, WM8960_RADC,
+ 0, 255, 0, adc_tlv),
SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume",
WM8960_BYPASS1, 4, 7, 1, bypass_tlv),
@@ -857,9 +857,9 @@
if (pll_div.k) {
reg |= 0x20;
- snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f);
- snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff);
- snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0x1ff);
+ snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 16) & 0xff);
+ snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 8) & 0xff);
+ snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0xff);
}
snd_soc_write(codec, WM8960_PLL1, reg);
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index e2de9ec..11d80f3 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2621,8 +2621,6 @@
wm8962->sysclk_rate = freq;
- wm8962_configure_bclk(codec);
-
return 0;
}
@@ -3046,8 +3044,9 @@
pm_wakeup_event(dev, 300);
- schedule_delayed_work(&wm8962->mic_work,
- msecs_to_jiffies(250));
+ queue_delayed_work(system_power_efficient_wq,
+ &wm8962->mic_work,
+ msecs_to_jiffies(250));
}
return IRQ_HANDLED;
@@ -3175,7 +3174,7 @@
long int time;
int ret;
- ret = strict_strtol(buf, 10, &time);
+ ret = kstrtol(buf, 10, &time);
if (ret != 0)
return ret;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index eee2a01..86426a1 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -819,8 +819,9 @@
* don't want false reports.
*/
if (wm8994->jackdet && !wm8994->clk_has_run) {
- schedule_delayed_work(&wm8994->jackdet_bootstrap,
- msecs_to_jiffies(1000));
+ queue_delayed_work(system_power_efficient_wq,
+ &wm8994->jackdet_bootstrap,
+ msecs_to_jiffies(1000));
wm8994->clk_has_run = true;
}
break;
@@ -1432,7 +1433,7 @@
#define WM8994_CLASS_W_SWITCH(xname, reg, shift, max, invert) \
SOC_SINGLE_EXT(xname, reg, shift, max, invert, \
- snd_soc_get_volsw, wm8994_put_class_w)
+ snd_soc_dapm_get_volsw, wm8994_put_class_w)
static int wm8994_put_class_w(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -3485,7 +3486,8 @@
pm_wakeup_event(codec->dev, 300);
- schedule_delayed_work(&priv->mic_work, msecs_to_jiffies(250));
+ queue_delayed_work(system_power_efficient_wq,
+ &priv->mic_work, msecs_to_jiffies(250));
return IRQ_HANDLED;
}
@@ -3573,8 +3575,9 @@
/* If nothing present then clear our statuses */
dev_dbg(codec->dev, "Detected open circuit\n");
- schedule_delayed_work(&wm8994->open_circuit_work,
- msecs_to_jiffies(2500));
+ queue_delayed_work(system_power_efficient_wq,
+ &wm8994->open_circuit_work,
+ msecs_to_jiffies(2500));
return;
}
@@ -3688,8 +3691,9 @@
WM1811_JACKDET_DB, 0);
delay = control->pdata.micdet_delay;
- schedule_delayed_work(&wm8994->mic_work,
- msecs_to_jiffies(delay));
+ queue_delayed_work(system_power_efficient_wq,
+ &wm8994->mic_work,
+ msecs_to_jiffies(delay));
} else {
dev_dbg(codec->dev, "Jack not detected\n");
@@ -3934,8 +3938,9 @@
id_delay = wm8994->wm8994->pdata.mic_id_delay;
if (wm8994->mic_detecting)
- schedule_delayed_work(&wm8994->mic_complete_work,
- msecs_to_jiffies(id_delay));
+ queue_delayed_work(system_power_efficient_wq,
+ &wm8994->mic_complete_work,
+ msecs_to_jiffies(id_delay));
else
wm8958_button_det(codec, reg);
@@ -4008,9 +4013,6 @@
wm8994->micdet_irq = control->pdata.micdet_irq;
- pm_runtime_enable(codec->dev);
- pm_runtime_idle(codec->dev);
-
/* By default use idle_bias_off, will override for WM8994 */
codec->dapm.idle_bias_off = 1;
@@ -4383,8 +4385,6 @@
wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF);
- pm_runtime_disable(codec->dev);
-
for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++)
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FLL1_LOCK + i,
&wm8994->fll_locked[i]);
@@ -4443,6 +4443,9 @@
wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent);
+ pm_runtime_enable(&pdev->dev);
+ pm_runtime_idle(&pdev->dev);
+
return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8994,
wm8994_dai, ARRAY_SIZE(wm8994_dai));
}
@@ -4450,6 +4453,8 @@
static int wm8994_remove(struct platform_device *pdev)
{
snd_soc_unregister_codec(&pdev->dev);
+ pm_runtime_disable(&pdev->dev);
+
return 0;
}
diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c
new file mode 100644
index 0000000..6ec3de3
--- /dev/null
+++ b/sound/soc/codecs/wm8997.c
@@ -0,0 +1,1175 @@
+/*
+ * wm8997.c -- WM8997 ALSA SoC Audio driver
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/registers.h>
+
+#include "arizona.h"
+#include "wm8997.h"
+
+struct wm8997_priv {
+ struct arizona_priv core;
+ struct arizona_fll fll[2];
+};
+
+static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
+static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
+static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0);
+static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0);
+
+static const struct reg_default wm8997_sysclk_reva_patch[] = {
+ { 0x301D, 0x7B15 },
+ { 0x301B, 0x0050 },
+ { 0x305D, 0x7B17 },
+ { 0x305B, 0x0050 },
+ { 0x3001, 0x08FE },
+ { 0x3003, 0x00F4 },
+ { 0x3041, 0x08FF },
+ { 0x3043, 0x0005 },
+ { 0x3020, 0x0225 },
+ { 0x3021, 0x0A00 },
+ { 0x3022, 0xE24D },
+ { 0x3023, 0x0800 },
+ { 0x3024, 0xE24D },
+ { 0x3025, 0xF000 },
+ { 0x3060, 0x0226 },
+ { 0x3061, 0x0A00 },
+ { 0x3062, 0xE252 },
+ { 0x3063, 0x0800 },
+ { 0x3064, 0xE252 },
+ { 0x3065, 0xF000 },
+ { 0x3116, 0x022B },
+ { 0x3117, 0xFA00 },
+ { 0x3110, 0x246C },
+ { 0x3111, 0x0A03 },
+ { 0x3112, 0x246E },
+ { 0x3113, 0x0A03 },
+ { 0x3114, 0x2470 },
+ { 0x3115, 0x0A03 },
+ { 0x3126, 0x246C },
+ { 0x3127, 0x0A02 },
+ { 0x3128, 0x246E },
+ { 0x3129, 0x0A02 },
+ { 0x312A, 0x2470 },
+ { 0x312B, 0xFA02 },
+ { 0x3125, 0x0800 },
+};
+
+static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+ struct regmap *regmap = codec->control_data;
+ const struct reg_default *patch = NULL;
+ int i, patch_size;
+
+ switch (arizona->rev) {
+ case 0:
+ patch = wm8997_sysclk_reva_patch;
+ patch_size = ARRAY_SIZE(wm8997_sysclk_reva_patch);
+ break;
+ default:
+ break;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ if (patch)
+ for (i = 0; i < patch_size; i++)
+ regmap_write(regmap, patch[i].reg,
+ patch[i].def);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static const char *wm8997_osr_text[] = {
+ "Low power", "Normal", "High performance",
+};
+
+static const unsigned int wm8997_osr_val[] = {
+ 0x0, 0x3, 0x5,
+};
+
+static const struct soc_enum wm8997_hpout_osr[] = {
+ SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L,
+ ARIZONA_OUT1_OSR_SHIFT, 0x7, 3,
+ wm8997_osr_text, wm8997_osr_val),
+ SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L,
+ ARIZONA_OUT3_OSR_SHIFT, 0x7, 3,
+ wm8997_osr_text, wm8997_osr_val),
+};
+
+#define WM8997_NG_SRC(name, base) \
+ SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \
+ SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \
+ SOC_SINGLE(name " NG EPOUT Switch", base, 4, 1, 0), \
+ SOC_SINGLE(name " NG SPKOUT Switch", base, 6, 1, 0), \
+ SOC_SINGLE(name " NG SPKDAT1L Switch", base, 8, 1, 0), \
+ SOC_SINGLE(name " NG SPKDAT1R Switch", base, 9, 1, 0)
+
+static const struct snd_kcontrol_new wm8997_snd_controls[] = {
+SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL,
+ ARIZONA_IN1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL,
+ ARIZONA_IN2_OSR_SHIFT, 1, 0),
+
+SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL,
+ ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL,
+ ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL,
+ ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL,
+ ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+
+SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+ ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R,
+ ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+ ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R,
+ ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv),
+
+SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp),
+SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp),
+
+ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES_MASK("EQ1 Coefficeints", ARIZONA_EQ1_1, 21,
+ ARIZONA_EQ1_ENA_MASK),
+SND_SOC_BYTES_MASK("EQ2 Coefficeints", ARIZONA_EQ2_1, 21,
+ ARIZONA_EQ2_ENA_MASK),
+SND_SOC_BYTES_MASK("EQ3 Coefficeints", ARIZONA_EQ3_1, 21,
+ ARIZONA_EQ3_ENA_MASK),
+SND_SOC_BYTES_MASK("EQ4 Coefficeints", ARIZONA_EQ4_1, 21,
+ ARIZONA_EQ4_ENA_MASK),
+
+SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5,
+ ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA),
+
+ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE),
+
+SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode),
+SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode),
+SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode),
+SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode),
+
+SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1),
+SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1),
+SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1),
+SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1),
+
+SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]),
+SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]),
+
+ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR,
+ ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv),
+
+ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUT", ARIZONA_OUT4LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
+ ARIZONA_OUT4_OSR_SHIFT, 1, 0),
+SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
+ ARIZONA_OUT5_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+ ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
+SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+ ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
+SOC_SINGLE("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+ ARIZONA_OUT4L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+ ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+ ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+ ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+ ARIZONA_OUT4L_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+ ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+
+SOC_VALUE_ENUM("HPOUT1 OSR", wm8997_hpout_osr[0]),
+SOC_VALUE_ENUM("EPOUT OSR", wm8997_hpout_osr[1]),
+
+SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp),
+SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp),
+
+SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
+ ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
+
+SOC_SINGLE("Noise Gate Switch", ARIZONA_NOISE_GATE_CONTROL,
+ ARIZONA_NGATE_ENA_SHIFT, 1, 0),
+SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL,
+ ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv),
+SOC_ENUM("Noise Gate Hold", arizona_ng_hold),
+
+WM8997_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L),
+WM8997_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R),
+WM8997_NG_SRC("EPOUT", ARIZONA_NOISE_GATE_SELECT_3L),
+WM8997_NG_SRC("SPKOUT", ARIZONA_NOISE_GATE_SELECT_4L),
+WM8997_NG_SRC("SPKDAT1L", ARIZONA_NOISE_GATE_SELECT_5L),
+WM8997_NG_SRC("SPKDAT1R", ARIZONA_NOISE_GATE_SELECT_5R),
+
+ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE),
+};
+
+ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUT, ARIZONA_OUT4LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SLIMTX7, ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SLIMTX8, ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE);
+
+static const char *wm8997_aec_loopback_texts[] = {
+ "HPOUT1L", "HPOUT1R", "EPOUT", "SPKOUT", "SPKDAT1L", "SPKDAT1R",
+};
+
+static const unsigned int wm8997_aec_loopback_values[] = {
+ 0, 1, 4, 6, 8, 9,
+};
+
+static const struct soc_enum wm8997_aec_loopback =
+ SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1,
+ ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf,
+ ARRAY_SIZE(wm8997_aec_loopback_texts),
+ wm8997_aec_loopback_texts,
+ wm8997_aec_loopback_values);
+
+static const struct snd_kcontrol_new wm8997_aec_loopback_mux =
+ SOC_DAPM_VALUE_ENUM("AEC Loopback", wm8997_aec_loopback);
+
+static const struct snd_soc_dapm_widget wm8997_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT,
+ 0, wm8997_sysclk_ev, SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1,
+ ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK,
+ ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK,
+ ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDD", 0, 0),
+
+SND_SOC_DAPM_SIGGEN("TONE"),
+SND_SOC_DAPM_SIGGEN("NOISE"),
+SND_SOC_DAPM_SIGGEN("HAPTICS"),
+
+SND_SOC_DAPM_INPUT("IN1L"),
+SND_SOC_DAPM_INPUT("IN1R"),
+SND_SOC_DAPM_INPUT("IN2L"),
+SND_SOC_DAPM_INPUT("IN2R"),
+
+SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+
+SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1,
+ ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2,
+ ARIZONA_MICB2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3,
+ ARIZONA_MICB3_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR,
+ ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1,
+ ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1,
+ ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1,
+ ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT,
+ 0, NULL, 0),
+SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT,
+ 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3,
+ ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3,
+ ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3,
+ ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3,
+ ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3,
+ ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3,
+ ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3,
+ ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3,
+ ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0,
+ ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0,
+ ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0,
+ ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0,
+ ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0,
+ ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+ ARIZONA_SLIMTX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0,
+ ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+ ARIZONA_SLIMTX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0,
+ ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+ ARIZONA_SLIMTX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0,
+ ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+ ARIZONA_SLIMTX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0,
+ ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+ ARIZONA_SLIMTX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0,
+ ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+ ARIZONA_SLIMTX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX7", NULL, 0,
+ ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+ ARIZONA_SLIMTX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX8", NULL, 0,
+ ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+ ARIZONA_SLIMTX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0,
+ ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+ ARIZONA_SLIMRX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0,
+ ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+ ARIZONA_SLIMRX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0,
+ ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+ ARIZONA_SLIMRX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0,
+ ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+ ARIZONA_SLIMRX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX5", NULL, 0,
+ ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+ ARIZONA_SLIMRX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX6", NULL, 0,
+ ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+ ARIZONA_SLIMRX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX7", NULL, 0,
+ ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+ ARIZONA_SLIMRX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0,
+ ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+ ARIZONA_SLIMRX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
+ ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+ &wm8997_aec_loopback_mux),
+
+SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
+ ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM,
+ ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"),
+ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"),
+ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"),
+ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"),
+
+ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"),
+ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"),
+
+ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"),
+ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"),
+ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"),
+ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"),
+
+ARIZONA_MIXER_WIDGETS(Mic, "Mic"),
+ARIZONA_MIXER_WIDGETS(Noise, "Noise"),
+
+ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"),
+ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"),
+
+ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"),
+ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"),
+ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"),
+ARIZONA_MIXER_WIDGETS(SPKOUT, "SPKOUT"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"),
+
+ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"),
+ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"),
+ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"),
+ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"),
+ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"),
+ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"),
+ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"),
+ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"),
+
+ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"),
+ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"),
+
+ARIZONA_MIXER_WIDGETS(SLIMTX1, "SLIMTX1"),
+ARIZONA_MIXER_WIDGETS(SLIMTX2, "SLIMTX2"),
+ARIZONA_MIXER_WIDGETS(SLIMTX3, "SLIMTX3"),
+ARIZONA_MIXER_WIDGETS(SLIMTX4, "SLIMTX4"),
+ARIZONA_MIXER_WIDGETS(SLIMTX5, "SLIMTX5"),
+ARIZONA_MIXER_WIDGETS(SLIMTX6, "SLIMTX6"),
+ARIZONA_MIXER_WIDGETS(SLIMTX7, "SLIMTX7"),
+ARIZONA_MIXER_WIDGETS(SLIMTX8, "SLIMTX8"),
+
+ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"),
+ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"),
+
+ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"),
+ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"),
+
+ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"),
+ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"),
+
+ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"),
+ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"),
+
+SND_SOC_DAPM_OUTPUT("HPOUT1L"),
+SND_SOC_DAPM_OUTPUT("HPOUT1R"),
+SND_SOC_DAPM_OUTPUT("EPOUTN"),
+SND_SOC_DAPM_OUTPUT("EPOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1L"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
+
+SND_SOC_DAPM_OUTPUT("MICSUPP"),
+};
+
+#define ARIZONA_MIXER_INPUT_ROUTES(name) \
+ { name, "Noise Generator", "Noise Generator" }, \
+ { name, "Tone Generator 1", "Tone Generator 1" }, \
+ { name, "Tone Generator 2", "Tone Generator 2" }, \
+ { name, "Haptics", "HAPTICS" }, \
+ { name, "AEC", "AEC Loopback" }, \
+ { name, "IN1L", "IN1L PGA" }, \
+ { name, "IN1R", "IN1R PGA" }, \
+ { name, "IN2L", "IN2L PGA" }, \
+ { name, "IN2R", "IN2R PGA" }, \
+ { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \
+ { name, "AIF1RX1", "AIF1RX1" }, \
+ { name, "AIF1RX2", "AIF1RX2" }, \
+ { name, "AIF1RX3", "AIF1RX3" }, \
+ { name, "AIF1RX4", "AIF1RX4" }, \
+ { name, "AIF1RX5", "AIF1RX5" }, \
+ { name, "AIF1RX6", "AIF1RX6" }, \
+ { name, "AIF1RX7", "AIF1RX7" }, \
+ { name, "AIF1RX8", "AIF1RX8" }, \
+ { name, "AIF2RX1", "AIF2RX1" }, \
+ { name, "AIF2RX2", "AIF2RX2" }, \
+ { name, "SLIMRX1", "SLIMRX1" }, \
+ { name, "SLIMRX2", "SLIMRX2" }, \
+ { name, "SLIMRX3", "SLIMRX3" }, \
+ { name, "SLIMRX4", "SLIMRX4" }, \
+ { name, "SLIMRX5", "SLIMRX5" }, \
+ { name, "SLIMRX6", "SLIMRX6" }, \
+ { name, "SLIMRX7", "SLIMRX7" }, \
+ { name, "SLIMRX8", "SLIMRX8" }, \
+ { name, "EQ1", "EQ1" }, \
+ { name, "EQ2", "EQ2" }, \
+ { name, "EQ3", "EQ3" }, \
+ { name, "EQ4", "EQ4" }, \
+ { name, "DRC1L", "DRC1L" }, \
+ { name, "DRC1R", "DRC1R" }, \
+ { name, "LHPF1", "LHPF1" }, \
+ { name, "LHPF2", "LHPF2" }, \
+ { name, "LHPF3", "LHPF3" }, \
+ { name, "LHPF4", "LHPF4" }, \
+ { name, "ISRC1DEC1", "ISRC1DEC1" }, \
+ { name, "ISRC1DEC2", "ISRC1DEC2" }, \
+ { name, "ISRC1INT1", "ISRC1INT1" }, \
+ { name, "ISRC1INT2", "ISRC1INT2" }, \
+ { name, "ISRC2DEC1", "ISRC2DEC1" }, \
+ { name, "ISRC2DEC2", "ISRC2DEC2" }, \
+ { name, "ISRC2INT1", "ISRC2INT1" }, \
+ { name, "ISRC2INT2", "ISRC2INT2" }
+
+static const struct snd_soc_dapm_route wm8997_dapm_routes[] = {
+ { "AIF2 Capture", NULL, "DBVDD2" },
+ { "AIF2 Playback", NULL, "DBVDD2" },
+
+ { "OUT1L", NULL, "CPVDD" },
+ { "OUT1R", NULL, "CPVDD" },
+ { "OUT3L", NULL, "CPVDD" },
+
+ { "OUT4L", NULL, "SPKVDD" },
+
+ { "OUT1L", NULL, "SYSCLK" },
+ { "OUT1R", NULL, "SYSCLK" },
+ { "OUT3L", NULL, "SYSCLK" },
+ { "OUT4L", NULL, "SYSCLK" },
+
+ { "IN1L", NULL, "SYSCLK" },
+ { "IN1R", NULL, "SYSCLK" },
+ { "IN2L", NULL, "SYSCLK" },
+ { "IN2R", NULL, "SYSCLK" },
+
+ { "MICBIAS1", NULL, "MICVDD" },
+ { "MICBIAS2", NULL, "MICVDD" },
+ { "MICBIAS3", NULL, "MICVDD" },
+
+ { "Noise Generator", NULL, "SYSCLK" },
+ { "Tone Generator 1", NULL, "SYSCLK" },
+ { "Tone Generator 2", NULL, "SYSCLK" },
+
+ { "Noise Generator", NULL, "NOISE" },
+ { "Tone Generator 1", NULL, "TONE" },
+ { "Tone Generator 2", NULL, "TONE" },
+
+ { "AIF1 Capture", NULL, "AIF1TX1" },
+ { "AIF1 Capture", NULL, "AIF1TX2" },
+ { "AIF1 Capture", NULL, "AIF1TX3" },
+ { "AIF1 Capture", NULL, "AIF1TX4" },
+ { "AIF1 Capture", NULL, "AIF1TX5" },
+ { "AIF1 Capture", NULL, "AIF1TX6" },
+ { "AIF1 Capture", NULL, "AIF1TX7" },
+ { "AIF1 Capture", NULL, "AIF1TX8" },
+
+ { "AIF1RX1", NULL, "AIF1 Playback" },
+ { "AIF1RX2", NULL, "AIF1 Playback" },
+ { "AIF1RX3", NULL, "AIF1 Playback" },
+ { "AIF1RX4", NULL, "AIF1 Playback" },
+ { "AIF1RX5", NULL, "AIF1 Playback" },
+ { "AIF1RX6", NULL, "AIF1 Playback" },
+ { "AIF1RX7", NULL, "AIF1 Playback" },
+ { "AIF1RX8", NULL, "AIF1 Playback" },
+
+ { "AIF2 Capture", NULL, "AIF2TX1" },
+ { "AIF2 Capture", NULL, "AIF2TX2" },
+
+ { "AIF2RX1", NULL, "AIF2 Playback" },
+ { "AIF2RX2", NULL, "AIF2 Playback" },
+
+ { "Slim1 Capture", NULL, "SLIMTX1" },
+ { "Slim1 Capture", NULL, "SLIMTX2" },
+ { "Slim1 Capture", NULL, "SLIMTX3" },
+ { "Slim1 Capture", NULL, "SLIMTX4" },
+
+ { "SLIMRX1", NULL, "Slim1 Playback" },
+ { "SLIMRX2", NULL, "Slim1 Playback" },
+ { "SLIMRX3", NULL, "Slim1 Playback" },
+ { "SLIMRX4", NULL, "Slim1 Playback" },
+
+ { "Slim2 Capture", NULL, "SLIMTX5" },
+ { "Slim2 Capture", NULL, "SLIMTX6" },
+
+ { "SLIMRX5", NULL, "Slim2 Playback" },
+ { "SLIMRX6", NULL, "Slim2 Playback" },
+
+ { "Slim3 Capture", NULL, "SLIMTX7" },
+ { "Slim3 Capture", NULL, "SLIMTX8" },
+
+ { "SLIMRX7", NULL, "Slim3 Playback" },
+ { "SLIMRX8", NULL, "Slim3 Playback" },
+
+ { "AIF1 Playback", NULL, "SYSCLK" },
+ { "AIF2 Playback", NULL, "SYSCLK" },
+ { "Slim1 Playback", NULL, "SYSCLK" },
+ { "Slim2 Playback", NULL, "SYSCLK" },
+ { "Slim3 Playback", NULL, "SYSCLK" },
+
+ { "AIF1 Capture", NULL, "SYSCLK" },
+ { "AIF2 Capture", NULL, "SYSCLK" },
+ { "Slim1 Capture", NULL, "SYSCLK" },
+ { "Slim2 Capture", NULL, "SYSCLK" },
+ { "Slim3 Capture", NULL, "SYSCLK" },
+
+ { "IN1L PGA", NULL, "IN1L" },
+ { "IN1R PGA", NULL, "IN1R" },
+
+ { "IN2L PGA", NULL, "IN2L" },
+ { "IN2R PGA", NULL, "IN2R" },
+
+ ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
+ ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
+ ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"),
+
+ ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUT"),
+ ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"),
+ ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"),
+
+ ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"),
+ ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"),
+
+ ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"),
+ ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"),
+ ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"),
+ ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"),
+ ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"),
+ ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"),
+ ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"),
+ ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"),
+
+ ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"),
+ ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"),
+
+ ARIZONA_MIXER_ROUTES("SLIMTX1", "SLIMTX1"),
+ ARIZONA_MIXER_ROUTES("SLIMTX2", "SLIMTX2"),
+ ARIZONA_MIXER_ROUTES("SLIMTX3", "SLIMTX3"),
+ ARIZONA_MIXER_ROUTES("SLIMTX4", "SLIMTX4"),
+ ARIZONA_MIXER_ROUTES("SLIMTX5", "SLIMTX5"),
+ ARIZONA_MIXER_ROUTES("SLIMTX6", "SLIMTX6"),
+ ARIZONA_MIXER_ROUTES("SLIMTX7", "SLIMTX7"),
+ ARIZONA_MIXER_ROUTES("SLIMTX8", "SLIMTX8"),
+
+ ARIZONA_MIXER_ROUTES("EQ1", "EQ1"),
+ ARIZONA_MIXER_ROUTES("EQ2", "EQ2"),
+ ARIZONA_MIXER_ROUTES("EQ3", "EQ3"),
+ ARIZONA_MIXER_ROUTES("EQ4", "EQ4"),
+
+ ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"),
+ ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"),
+
+ ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"),
+ ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"),
+ ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
+ ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
+
+ ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"),
+ ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"),
+
+ ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"),
+ ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC2INT2"),
+
+ ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"),
+ ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"),
+
+ ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"),
+ ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"),
+
+ ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"),
+ ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"),
+
+ { "AEC Loopback", "HPOUT1L", "OUT1L" },
+ { "AEC Loopback", "HPOUT1R", "OUT1R" },
+ { "HPOUT1L", NULL, "OUT1L" },
+ { "HPOUT1R", NULL, "OUT1R" },
+
+ { "AEC Loopback", "EPOUT", "OUT3L" },
+ { "EPOUTN", NULL, "OUT3L" },
+ { "EPOUTP", NULL, "OUT3L" },
+
+ { "AEC Loopback", "SPKOUT", "OUT4L" },
+ { "SPKOUTN", NULL, "OUT4L" },
+ { "SPKOUTP", NULL, "OUT4L" },
+
+ { "AEC Loopback", "SPKDAT1L", "OUT5L" },
+ { "AEC Loopback", "SPKDAT1R", "OUT5R" },
+ { "SPKDAT1L", NULL, "OUT5L" },
+ { "SPKDAT1R", NULL, "OUT5R" },
+
+ { "MICSUPP", NULL, "SYSCLK" },
+};
+
+static int wm8997_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
+ unsigned int Fref, unsigned int Fout)
+{
+ struct wm8997_priv *wm8997 = snd_soc_codec_get_drvdata(codec);
+
+ switch (fll_id) {
+ case WM8997_FLL1:
+ return arizona_set_fll(&wm8997->fll[0], source, Fref, Fout);
+ case WM8997_FLL2:
+ return arizona_set_fll(&wm8997->fll[1], source, Fref, Fout);
+ case WM8997_FLL1_REFCLK:
+ return arizona_set_fll_refclk(&wm8997->fll[0], source, Fref,
+ Fout);
+ case WM8997_FLL2_REFCLK:
+ return arizona_set_fll_refclk(&wm8997->fll[1], source, Fref,
+ Fout);
+ default:
+ return -EINVAL;
+ }
+}
+
+#define WM8997_RATES SNDRV_PCM_RATE_8000_192000
+
+#define WM8997_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver wm8997_dai[] = {
+ {
+ .name = "wm8997-aif1",
+ .id = 1,
+ .base = ARIZONA_AIF1_BCLK_CTRL,
+ .playback = {
+ .stream_name = "AIF1 Playback",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = WM8997_RATES,
+ .formats = WM8997_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF1 Capture",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = WM8997_RATES,
+ .formats = WM8997_FORMATS,
+ },
+ .ops = &arizona_dai_ops,
+ .symmetric_rates = 1,
+ },
+ {
+ .name = "wm8997-aif2",
+ .id = 2,
+ .base = ARIZONA_AIF2_BCLK_CTRL,
+ .playback = {
+ .stream_name = "AIF2 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8997_RATES,
+ .formats = WM8997_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF2 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8997_RATES,
+ .formats = WM8997_FORMATS,
+ },
+ .ops = &arizona_dai_ops,
+ .symmetric_rates = 1,
+ },
+ {
+ .name = "wm8997-slim1",
+ .id = 3,
+ .playback = {
+ .stream_name = "Slim1 Playback",
+ .channels_min = 1,
+ .channels_max = 4,
+ .rates = WM8997_RATES,
+ .formats = WM8997_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Slim1 Capture",
+ .channels_min = 1,
+ .channels_max = 4,
+ .rates = WM8997_RATES,
+ .formats = WM8997_FORMATS,
+ },
+ .ops = &arizona_simple_dai_ops,
+ },
+ {
+ .name = "wm8997-slim2",
+ .id = 4,
+ .playback = {
+ .stream_name = "Slim2 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8997_RATES,
+ .formats = WM8997_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Slim2 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8997_RATES,
+ .formats = WM8997_FORMATS,
+ },
+ .ops = &arizona_simple_dai_ops,
+ },
+ {
+ .name = "wm8997-slim3",
+ .id = 5,
+ .playback = {
+ .stream_name = "Slim3 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8997_RATES,
+ .formats = WM8997_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Slim3 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8997_RATES,
+ .formats = WM8997_FORMATS,
+ },
+ .ops = &arizona_simple_dai_ops,
+ },
+};
+
+static int wm8997_codec_probe(struct snd_soc_codec *codec)
+{
+ struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ codec->control_data = priv->core.arizona->regmap;
+
+ ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP);
+ if (ret != 0)
+ return ret;
+
+ arizona_init_spk(codec);
+
+ snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
+
+ priv->core.arizona->dapm = &codec->dapm;
+
+ return 0;
+}
+
+static int wm8997_codec_remove(struct snd_soc_codec *codec)
+{
+ struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ priv->core.arizona->dapm = NULL;
+
+ return 0;
+}
+
+#define WM8997_DIG_VU 0x0200
+
+static unsigned int wm8997_digital_vu[] = {
+ ARIZONA_DAC_DIGITAL_VOLUME_1L,
+ ARIZONA_DAC_DIGITAL_VOLUME_1R,
+ ARIZONA_DAC_DIGITAL_VOLUME_3L,
+ ARIZONA_DAC_DIGITAL_VOLUME_4L,
+ ARIZONA_DAC_DIGITAL_VOLUME_5L,
+ ARIZONA_DAC_DIGITAL_VOLUME_5R,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_wm8997 = {
+ .probe = wm8997_codec_probe,
+ .remove = wm8997_codec_remove,
+
+ .idle_bias_off = true,
+
+ .set_sysclk = arizona_set_sysclk,
+ .set_pll = wm8997_set_fll,
+
+ .controls = wm8997_snd_controls,
+ .num_controls = ARRAY_SIZE(wm8997_snd_controls),
+ .dapm_widgets = wm8997_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm8997_dapm_widgets),
+ .dapm_routes = wm8997_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(wm8997_dapm_routes),
+};
+
+static int wm8997_probe(struct platform_device *pdev)
+{
+ struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
+ struct wm8997_priv *wm8997;
+ int i;
+
+ wm8997 = devm_kzalloc(&pdev->dev, sizeof(struct wm8997_priv),
+ GFP_KERNEL);
+ if (wm8997 == NULL)
+ return -ENOMEM;
+ platform_set_drvdata(pdev, wm8997);
+
+ wm8997->core.arizona = arizona;
+ wm8997->core.num_inputs = 4;
+
+ for (i = 0; i < ARRAY_SIZE(wm8997->fll); i++)
+ wm8997->fll[i].vco_mult = 1;
+
+ arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1,
+ ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK,
+ &wm8997->fll[0]);
+ arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1,
+ ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK,
+ &wm8997->fll[1]);
+
+ /* SR2 fixed at 8kHz, SR3 fixed at 16kHz */
+ regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2,
+ ARIZONA_SAMPLE_RATE_2_MASK, 0x11);
+ regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3,
+ ARIZONA_SAMPLE_RATE_3_MASK, 0x12);
+
+ for (i = 0; i < ARRAY_SIZE(wm8997_dai); i++)
+ arizona_init_dai(&wm8997->core, i);
+
+ /* Latch volume update bits */
+ for (i = 0; i < ARRAY_SIZE(wm8997_digital_vu); i++)
+ regmap_update_bits(arizona->regmap, wm8997_digital_vu[i],
+ WM8997_DIG_VU, WM8997_DIG_VU);
+
+ pm_runtime_enable(&pdev->dev);
+ pm_runtime_idle(&pdev->dev);
+
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8997,
+ wm8997_dai, ARRAY_SIZE(wm8997_dai));
+}
+
+static int wm8997_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ pm_runtime_disable(&pdev->dev);
+
+ return 0;
+}
+
+static struct platform_driver wm8997_codec_driver = {
+ .driver = {
+ .name = "wm8997-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8997_probe,
+ .remove = wm8997_remove,
+};
+
+module_platform_driver(wm8997_codec_driver);
+
+MODULE_DESCRIPTION("ASoC WM8997 driver");
+MODULE_AUTHOR("Charles Keepax <ckeepax@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm8997-codec");
diff --git a/sound/soc/codecs/wm8997.h b/sound/soc/codecs/wm8997.h
new file mode 100644
index 0000000..5e91c6a
--- /dev/null
+++ b/sound/soc/codecs/wm8997.h
@@ -0,0 +1,23 @@
+/*
+ * wm8997.h -- WM8997 ALSA SoC Audio driver
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8997_H
+#define _WM8997_H
+
+#include "arizona.h"
+
+#define WM8997_FLL1 1
+#define WM8997_FLL2 2
+#define WM8997_FLL1_REFCLK 3
+#define WM8997_FLL2_REFCLK 4
+
+#endif
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 05252ac..b38f350 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -225,15 +225,8 @@
struct snd_ctl_elem_info *uinfo);
};
-struct wm_coeff {
- struct device *dev;
- struct list_head ctl_list;
- struct regmap *regmap;
-};
-
struct wm_coeff_ctl {
const char *name;
- struct snd_card *card;
struct wm_adsp_alg_region region;
struct wm_coeff_ctl_ops ops;
struct wm_adsp *adsp;
@@ -378,7 +371,6 @@
static int wm_coeff_write_control(struct snd_kcontrol *kcontrol,
const void *buf, size_t len)
{
- struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol);
struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value;
struct wm_adsp_alg_region *region = &ctl->region;
const struct wm_adsp_region *mem;
@@ -401,7 +393,7 @@
if (!scratch)
return -ENOMEM;
- ret = regmap_raw_write(wm_coeff->regmap, reg, scratch,
+ ret = regmap_raw_write(adsp->regmap, reg, scratch,
ctl->len);
if (ret) {
adsp_err(adsp, "Failed to write %zu bytes to %x\n",
@@ -434,7 +426,6 @@
static int wm_coeff_read_control(struct snd_kcontrol *kcontrol,
void *buf, size_t len)
{
- struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol);
struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value;
struct wm_adsp_alg_region *region = &ctl->region;
const struct wm_adsp_region *mem;
@@ -457,7 +448,7 @@
if (!scratch)
return -ENOMEM;
- ret = regmap_raw_read(wm_coeff->regmap, reg, scratch, ctl->len);
+ ret = regmap_raw_read(adsp->regmap, reg, scratch, ctl->len);
if (ret) {
adsp_err(adsp, "Failed to read %zu bytes from %x\n",
ctl->len, reg);
@@ -481,37 +472,18 @@
return 0;
}
-static int wm_coeff_add_kcontrol(struct wm_coeff *wm_coeff,
- struct wm_coeff_ctl *ctl,
- const struct snd_kcontrol_new *kctl)
-{
- int ret;
- struct snd_kcontrol *kcontrol;
-
- kcontrol = snd_ctl_new1(kctl, wm_coeff);
- ret = snd_ctl_add(ctl->card, kcontrol);
- if (ret < 0) {
- dev_err(wm_coeff->dev, "Failed to add %s: %d\n",
- kctl->name, ret);
- return ret;
- }
- ctl->kcontrol = kcontrol;
- return 0;
-}
-
struct wmfw_ctl_work {
- struct wm_coeff *wm_coeff;
+ struct wm_adsp *adsp;
struct wm_coeff_ctl *ctl;
struct work_struct work;
};
-static int wmfw_add_ctl(struct wm_coeff *wm_coeff,
- struct wm_coeff_ctl *ctl)
+static int wmfw_add_ctl(struct wm_adsp *adsp, struct wm_coeff_ctl *ctl)
{
struct snd_kcontrol_new *kcontrol;
int ret;
- if (!wm_coeff || !ctl || !ctl->name || !ctl->card)
+ if (!ctl || !ctl->name)
return -EINVAL;
kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL);
@@ -525,14 +497,17 @@
kcontrol->put = wm_coeff_put;
kcontrol->private_value = (unsigned long)ctl;
- ret = wm_coeff_add_kcontrol(wm_coeff,
- ctl, kcontrol);
+ ret = snd_soc_add_card_controls(adsp->card,
+ kcontrol, 1);
if (ret < 0)
goto err_kcontrol;
kfree(kcontrol);
- list_add(&ctl->list, &wm_coeff->ctl_list);
+ ctl->kcontrol = snd_soc_card_get_kcontrol(adsp->card,
+ ctl->name);
+
+ list_add(&ctl->list, &adsp->ctl_list);
return 0;
err_kcontrol:
@@ -753,13 +728,12 @@
return ret;
}
-static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff)
+static int wm_coeff_init_control_caches(struct wm_adsp *adsp)
{
struct wm_coeff_ctl *ctl;
int ret;
- list_for_each_entry(ctl, &wm_coeff->ctl_list,
- list) {
+ list_for_each_entry(ctl, &adsp->ctl_list, list) {
if (!ctl->enabled || ctl->set)
continue;
ret = wm_coeff_read_control(ctl->kcontrol,
@@ -772,13 +746,12 @@
return 0;
}
-static int wm_coeff_sync_controls(struct wm_coeff *wm_coeff)
+static int wm_coeff_sync_controls(struct wm_adsp *adsp)
{
struct wm_coeff_ctl *ctl;
int ret;
- list_for_each_entry(ctl, &wm_coeff->ctl_list,
- list) {
+ list_for_each_entry(ctl, &adsp->ctl_list, list) {
if (!ctl->enabled)
continue;
if (ctl->set) {
@@ -799,15 +772,14 @@
struct wmfw_ctl_work,
work);
- wmfw_add_ctl(ctl_work->wm_coeff, ctl_work->ctl);
+ wmfw_add_ctl(ctl_work->adsp, ctl_work->ctl);
kfree(ctl_work);
}
-static int wm_adsp_create_control(struct snd_soc_codec *codec,
+static int wm_adsp_create_control(struct wm_adsp *dsp,
const struct wm_adsp_alg_region *region)
{
- struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec);
struct wm_coeff_ctl *ctl;
struct wmfw_ctl_work *ctl_work;
char *name;
@@ -842,7 +814,7 @@
snprintf(name, PAGE_SIZE, "DSP%d %s %x",
dsp->num, region_name, region->alg);
- list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list,
+ list_for_each_entry(ctl, &dsp->ctl_list,
list) {
if (!strcmp(ctl->name, name)) {
if (!ctl->enabled)
@@ -866,7 +838,6 @@
ctl->set = 0;
ctl->ops.xget = wm_coeff_get;
ctl->ops.xput = wm_coeff_put;
- ctl->card = codec->card->snd_card;
ctl->adsp = dsp;
ctl->len = region->len;
@@ -882,7 +853,7 @@
goto err_ctl_cache;
}
- ctl_work->wm_coeff = dsp->wm_coeff;
+ ctl_work->adsp = dsp;
ctl_work->ctl = ctl;
INIT_WORK(&ctl_work->work, wm_adsp_ctl_work);
schedule_work(&ctl_work->work);
@@ -903,7 +874,7 @@
return ret;
}
-static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec)
+static int wm_adsp_setup_algs(struct wm_adsp *dsp)
{
struct regmap *regmap = dsp->regmap;
struct wmfw_adsp1_id_hdr adsp1_id;
@@ -1091,7 +1062,7 @@
if (i + 1 < algs) {
region->len = be32_to_cpu(adsp1_alg[i + 1].dm);
region->len -= be32_to_cpu(adsp1_alg[i].dm);
- wm_adsp_create_control(codec, region);
+ wm_adsp_create_control(dsp, region);
} else {
adsp_warn(dsp, "Missing length info for region DM with ID %x\n",
be32_to_cpu(adsp1_alg[i].alg.id));
@@ -1108,7 +1079,7 @@
if (i + 1 < algs) {
region->len = be32_to_cpu(adsp1_alg[i + 1].zm);
region->len -= be32_to_cpu(adsp1_alg[i].zm);
- wm_adsp_create_control(codec, region);
+ wm_adsp_create_control(dsp, region);
} else {
adsp_warn(dsp, "Missing length info for region ZM with ID %x\n",
be32_to_cpu(adsp1_alg[i].alg.id));
@@ -1137,7 +1108,7 @@
if (i + 1 < algs) {
region->len = be32_to_cpu(adsp2_alg[i + 1].xm);
region->len -= be32_to_cpu(adsp2_alg[i].xm);
- wm_adsp_create_control(codec, region);
+ wm_adsp_create_control(dsp, region);
} else {
adsp_warn(dsp, "Missing length info for region XM with ID %x\n",
be32_to_cpu(adsp2_alg[i].alg.id));
@@ -1154,7 +1125,7 @@
if (i + 1 < algs) {
region->len = be32_to_cpu(adsp2_alg[i + 1].ym);
region->len -= be32_to_cpu(adsp2_alg[i].ym);
- wm_adsp_create_control(codec, region);
+ wm_adsp_create_control(dsp, region);
} else {
adsp_warn(dsp, "Missing length info for region YM with ID %x\n",
be32_to_cpu(adsp2_alg[i].alg.id));
@@ -1171,7 +1142,7 @@
if (i + 1 < algs) {
region->len = be32_to_cpu(adsp2_alg[i + 1].zm);
region->len -= be32_to_cpu(adsp2_alg[i].zm);
- wm_adsp_create_control(codec, region);
+ wm_adsp_create_control(dsp, region);
} else {
adsp_warn(dsp, "Missing length info for region ZM with ID %x\n",
be32_to_cpu(adsp2_alg[i].alg.id));
@@ -1391,6 +1362,8 @@
int ret;
int val;
+ dsp->card = codec->card;
+
switch (event) {
case SND_SOC_DAPM_POST_PMU:
regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30,
@@ -1425,7 +1398,7 @@
if (ret != 0)
goto err;
- ret = wm_adsp_setup_algs(dsp, codec);
+ ret = wm_adsp_setup_algs(dsp);
if (ret != 0)
goto err;
@@ -1434,12 +1407,12 @@
goto err;
/* Initialize caches for enabled and unset controls */
- ret = wm_coeff_init_control_caches(dsp->wm_coeff);
+ ret = wm_coeff_init_control_caches(dsp);
if (ret != 0)
goto err;
/* Sync set controls */
- ret = wm_coeff_sync_controls(dsp->wm_coeff);
+ ret = wm_coeff_sync_controls(dsp);
if (ret != 0)
goto err;
@@ -1460,10 +1433,8 @@
regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30,
ADSP1_SYS_ENA, 0);
- list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list,
- list) {
+ list_for_each_entry(ctl, &dsp->ctl_list, list)
ctl->enabled = 0;
- }
break;
default:
@@ -1520,6 +1491,8 @@
unsigned int val;
int ret;
+ dsp->card = codec->card;
+
switch (event) {
case SND_SOC_DAPM_POST_PMU:
/*
@@ -1582,7 +1555,7 @@
if (ret != 0)
goto err;
- ret = wm_adsp_setup_algs(dsp, codec);
+ ret = wm_adsp_setup_algs(dsp);
if (ret != 0)
goto err;
@@ -1591,12 +1564,12 @@
goto err;
/* Initialize caches for enabled and unset controls */
- ret = wm_coeff_init_control_caches(dsp->wm_coeff);
+ ret = wm_coeff_init_control_caches(dsp);
if (ret != 0)
goto err;
/* Sync set controls */
- ret = wm_coeff_sync_controls(dsp->wm_coeff);
+ ret = wm_coeff_sync_controls(dsp);
if (ret != 0)
goto err;
@@ -1637,10 +1610,8 @@
ret);
}
- list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list,
- list) {
+ list_for_each_entry(ctl, &dsp->ctl_list, list)
ctl->enabled = 0;
- }
while (!list_empty(&dsp->alg_regions)) {
alg_region = list_first_entry(&dsp->alg_regions,
@@ -1679,49 +1650,38 @@
}
INIT_LIST_HEAD(&adsp->alg_regions);
-
- adsp->wm_coeff = kzalloc(sizeof(*adsp->wm_coeff),
- GFP_KERNEL);
- if (!adsp->wm_coeff)
- return -ENOMEM;
- adsp->wm_coeff->regmap = adsp->regmap;
- adsp->wm_coeff->dev = adsp->dev;
- INIT_LIST_HEAD(&adsp->wm_coeff->ctl_list);
+ INIT_LIST_HEAD(&adsp->ctl_list);
if (dvfs) {
adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD");
if (IS_ERR(adsp->dvfs)) {
ret = PTR_ERR(adsp->dvfs);
dev_err(adsp->dev, "Failed to get DCVDD: %d\n", ret);
- goto out_coeff;
+ return ret;
}
ret = regulator_enable(adsp->dvfs);
if (ret != 0) {
dev_err(adsp->dev, "Failed to enable DCVDD: %d\n",
ret);
- goto out_coeff;
+ return ret;
}
ret = regulator_set_voltage(adsp->dvfs, 1200000, 1800000);
if (ret != 0) {
dev_err(adsp->dev, "Failed to initialise DVFS: %d\n",
ret);
- goto out_coeff;
+ return ret;
}
ret = regulator_disable(adsp->dvfs);
if (ret != 0) {
dev_err(adsp->dev, "Failed to disable DCVDD: %d\n",
ret);
- goto out_coeff;
+ return ret;
}
}
return 0;
-
-out_coeff:
- kfree(adsp->wm_coeff);
- return ret;
}
EXPORT_SYMBOL_GPL(wm_adsp2_init);
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index 9f922c8..d018dea 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -39,6 +39,7 @@
int type;
struct device *dev;
struct regmap *regmap;
+ struct snd_soc_card *card;
int base;
int sysclk_reg;
@@ -57,7 +58,7 @@
struct regulator *dvfs;
- struct wm_coeff *wm_coeff;
+ struct list_head ctl_list;
};
#define WM_ADSP1(wname, num) \
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index 70eb37a..25c31f1 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -421,13 +421,11 @@
dw_i2s_dai, 1);
if (ret != 0) {
dev_err(&pdev->dev, "not able to register dai\n");
- goto err_set_drvdata;
+ goto err_clk_disable;
}
return 0;
-err_set_drvdata:
- dev_set_drvdata(&pdev->dev, NULL);
err_clk_disable:
clk_disable(dev->clk);
err_clk_put:
@@ -440,7 +438,6 @@
struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
snd_soc_unregister_component(&pdev->dev);
- dev_set_drvdata(&pdev->dev, NULL);
clk_put(dev->clk);
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index aa43854..704e246 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,6 +1,9 @@
config SND_SOC_FSL_SSI
tristate
+config SND_SOC_FSL_SPDIF
+ tristate
+
config SND_SOC_FSL_UTILS
tristate
@@ -98,7 +101,7 @@
menuconfig SND_IMX_SOC
tristate "SoC Audio for Freescale i.MX CPUs"
- depends on ARCH_MXC
+ depends on ARCH_MXC || COMPILE_TEST
help
Say Y or M if you want to add support for codecs attached to
the i.MX CPUs.
@@ -109,11 +112,11 @@
tristate
config SND_SOC_IMX_PCM_FIQ
- bool
+ tristate
select FIQ
config SND_SOC_IMX_PCM_DMA
- bool
+ tristate
select SND_SOC_GENERIC_DMAENGINE_PCM
config SND_SOC_IMX_AUDMUX
@@ -175,7 +178,6 @@
select SND_SOC_IMX_PCM_DMA
select SND_SOC_IMX_AUDMUX
select SND_SOC_FSL_SSI
- select SND_SOC_FSL_UTILS
help
Say Y if you want to add support for SoC audio on an i.MX board with
a wm8962 codec.
@@ -187,14 +189,23 @@
select SND_SOC_IMX_PCM_DMA
select SND_SOC_IMX_AUDMUX
select SND_SOC_FSL_SSI
- select SND_SOC_FSL_UTILS
help
Say Y if you want to add support for SoC audio on an i.MX board with
a sgtl5000 codec.
+config SND_SOC_IMX_SPDIF
+ tristate "SoC Audio support for i.MX boards with S/PDIF"
+ select SND_SOC_IMX_PCM_DMA
+ select SND_SOC_FSL_SPDIF
+ select SND_SOC_SPDIF
+ help
+ SoC Audio support for i.MX boards with S/PDIF
+ Say Y if you want to add support for SoC audio on an i.MX board with
+ a S/DPDIF.
+
config SND_SOC_IMX_MC13783
tristate "SoC Audio support for I.MX boards with mc13783"
- depends on MFD_MC13783
+ depends on MFD_MC13783 && ARM
select SND_SOC_IMX_SSI
select SND_SOC_IMX_AUDMUX
select SND_SOC_MC13783
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index d4b4aa8b..e2aaff7 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -12,9 +12,11 @@
# Freescale PowerPC SSI/DMA Platform Support
snd-soc-fsl-ssi-objs := fsl_ssi.o
+snd-soc-fsl-spdif-objs := fsl_spdif.o
snd-soc-fsl-utils-objs := fsl_utils.o
snd-soc-fsl-dma-objs := fsl_dma.o
obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
+obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o
obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o
obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o
@@ -43,6 +45,7 @@
snd-soc-wm1133-ev1-objs := wm1133-ev1.o
snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
snd-soc-imx-wm8962-objs := imx-wm8962.o
+snd-soc-imx-spdif-objs :=imx-spdif.o
snd-soc-imx-mc13783-objs := imx-mc13783.o
obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
@@ -51,4 +54,5 @@
obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
+obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
new file mode 100644
index 0000000..e93dc0d
--- /dev/null
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -0,0 +1,1225 @@
+/*
+ * Freescale S/PDIF ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Copyright (C) 2013 Freescale Semiconductor, Inc.
+ *
+ * Based on stmp3xxx_spdif_dai.c
+ * Vladimir Barinov <vbarinov@embeddedalley.com>
+ * Copyright 2008 SigmaTel, Inc
+ * Copyright 2008 Embedded Alley Solutions, Inc
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/module.h>
+#include <linux/clk.h>
+#include <linux/clk-private.h>
+#include <linux/bitrev.h>
+#include <linux/regmap.h>
+#include <linux/of_address.h>
+#include <linux/of_device.h>
+#include <linux/of_irq.h>
+
+#include <sound/asoundef.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "fsl_spdif.h"
+#include "imx-pcm.h"
+
+#define FSL_SPDIF_TXFIFO_WML 0x8
+#define FSL_SPDIF_RXFIFO_WML 0x8
+
+#define INTR_FOR_PLAYBACK (INT_TXFIFO_RESYNC)
+#define INTR_FOR_CAPTURE (INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL | INT_URX_OV|\
+ INT_QRX_FUL | INT_QRX_OV | INT_UQ_SYNC | INT_UQ_ERR |\
+ INT_RXFIFO_RESYNC | INT_LOSS_LOCK | INT_DPLL_LOCKED)
+
+/* Index list for the values that has if (DPLL Locked) condition */
+static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb };
+#define SRPC_NODPLL_START1 0x5
+#define SRPC_NODPLL_START2 0xc
+
+#define DEFAULT_RXCLK_SRC 1
+
+/*
+ * SPDIF control structure
+ * Defines channel status, subcode and Q sub
+ */
+struct spdif_mixer_control {
+ /* spinlock to access control data */
+ spinlock_t ctl_lock;
+
+ /* IEC958 channel tx status bit */
+ unsigned char ch_status[4];
+
+ /* User bits */
+ unsigned char subcode[2 * SPDIF_UBITS_SIZE];
+
+ /* Q subcode part of user bits */
+ unsigned char qsub[2 * SPDIF_QSUB_SIZE];
+
+ /* Buffer offset for U/Q */
+ u32 upos;
+ u32 qpos;
+
+ /* Ready buffer index of the two buffers */
+ u32 ready_buf;
+};
+
+struct fsl_spdif_priv {
+ struct spdif_mixer_control fsl_spdif_control;
+ struct snd_soc_dai_driver cpu_dai_drv;
+ struct platform_device *pdev;
+ struct regmap *regmap;
+ bool dpll_locked;
+ u8 txclk_div[SPDIF_TXRATE_MAX];
+ u8 txclk_src[SPDIF_TXRATE_MAX];
+ u8 rxclk_src;
+ struct clk *txclk[SPDIF_TXRATE_MAX];
+ struct clk *rxclk;
+ struct snd_dmaengine_dai_dma_data dma_params_tx;
+ struct snd_dmaengine_dai_dma_data dma_params_rx;
+
+ /* The name space will be allocated dynamically */
+ char name[0];
+};
+
+
+/* DPLL locked and lock loss interrupt handler */
+static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv)
+{
+ struct regmap *regmap = spdif_priv->regmap;
+ struct platform_device *pdev = spdif_priv->pdev;
+ u32 locked;
+
+ regmap_read(regmap, REG_SPDIF_SRPC, &locked);
+ locked &= SRPC_DPLL_LOCKED;
+
+ dev_dbg(&pdev->dev, "isr: Rx dpll %s \n",
+ locked ? "locked" : "loss lock");
+
+ spdif_priv->dpll_locked = locked ? true : false;
+}
+
+/* Receiver found illegal symbol interrupt handler */
+static void spdif_irq_sym_error(struct fsl_spdif_priv *spdif_priv)
+{
+ struct regmap *regmap = spdif_priv->regmap;
+ struct platform_device *pdev = spdif_priv->pdev;
+
+ dev_dbg(&pdev->dev, "isr: receiver found illegal symbol\n");
+
+ if (!spdif_priv->dpll_locked) {
+ /* DPLL unlocked seems no audio stream */
+ regmap_update_bits(regmap, REG_SPDIF_SIE, INT_SYM_ERR, 0);
+ }
+}
+
+/* U/Q Channel receive register full */
+static void spdif_irq_uqrx_full(struct fsl_spdif_priv *spdif_priv, char name)
+{
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ struct regmap *regmap = spdif_priv->regmap;
+ struct platform_device *pdev = spdif_priv->pdev;
+ u32 *pos, size, val, reg;
+
+ switch (name) {
+ case 'U':
+ pos = &ctrl->upos;
+ size = SPDIF_UBITS_SIZE;
+ reg = REG_SPDIF_SRU;
+ break;
+ case 'Q':
+ pos = &ctrl->qpos;
+ size = SPDIF_QSUB_SIZE;
+ reg = REG_SPDIF_SRQ;
+ break;
+ default:
+ dev_err(&pdev->dev, "unsupported channel name\n");
+ return;
+ }
+
+ dev_dbg(&pdev->dev, "isr: %c Channel receive register full\n", name);
+
+ if (*pos >= size * 2) {
+ *pos = 0;
+ } else if (unlikely((*pos % size) + 3 > size)) {
+ dev_err(&pdev->dev, "User bit receivce buffer overflow\n");
+ return;
+ }
+
+ regmap_read(regmap, reg, &val);
+ ctrl->subcode[*pos++] = val >> 16;
+ ctrl->subcode[*pos++] = val >> 8;
+ ctrl->subcode[*pos++] = val;
+}
+
+/* U/Q Channel sync found */
+static void spdif_irq_uq_sync(struct fsl_spdif_priv *spdif_priv)
+{
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ struct platform_device *pdev = spdif_priv->pdev;
+
+ dev_dbg(&pdev->dev, "isr: U/Q Channel sync found\n");
+
+ /* U/Q buffer reset */
+ if (ctrl->qpos == 0)
+ return;
+
+ /* Set ready to this buffer */
+ ctrl->ready_buf = (ctrl->qpos - 1) / SPDIF_QSUB_SIZE + 1;
+}
+
+/* U/Q Channel framing error */
+static void spdif_irq_uq_err(struct fsl_spdif_priv *spdif_priv)
+{
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ struct regmap *regmap = spdif_priv->regmap;
+ struct platform_device *pdev = spdif_priv->pdev;
+ u32 val;
+
+ dev_dbg(&pdev->dev, "isr: U/Q Channel framing error\n");
+
+ /* Read U/Q data to clear the irq and do buffer reset */
+ regmap_read(regmap, REG_SPDIF_SRU, &val);
+ regmap_read(regmap, REG_SPDIF_SRQ, &val);
+
+ /* Drop this U/Q buffer */
+ ctrl->ready_buf = 0;
+ ctrl->upos = 0;
+ ctrl->qpos = 0;
+}
+
+/* Get spdif interrupt status and clear the interrupt */
+static u32 spdif_intr_status_clear(struct fsl_spdif_priv *spdif_priv)
+{
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 val, val2;
+
+ regmap_read(regmap, REG_SPDIF_SIS, &val);
+ regmap_read(regmap, REG_SPDIF_SIE, &val2);
+
+ regmap_write(regmap, REG_SPDIF_SIC, val & val2);
+
+ return val;
+}
+
+static irqreturn_t spdif_isr(int irq, void *devid)
+{
+ struct fsl_spdif_priv *spdif_priv = (struct fsl_spdif_priv *)devid;
+ struct platform_device *pdev = spdif_priv->pdev;
+ u32 sis;
+
+ sis = spdif_intr_status_clear(spdif_priv);
+
+ if (sis & INT_DPLL_LOCKED)
+ spdif_irq_dpll_lock(spdif_priv);
+
+ if (sis & INT_TXFIFO_UNOV)
+ dev_dbg(&pdev->dev, "isr: Tx FIFO under/overrun\n");
+
+ if (sis & INT_TXFIFO_RESYNC)
+ dev_dbg(&pdev->dev, "isr: Tx FIFO resync\n");
+
+ if (sis & INT_CNEW)
+ dev_dbg(&pdev->dev, "isr: cstatus new\n");
+
+ if (sis & INT_VAL_NOGOOD)
+ dev_dbg(&pdev->dev, "isr: validity flag no good\n");
+
+ if (sis & INT_SYM_ERR)
+ spdif_irq_sym_error(spdif_priv);
+
+ if (sis & INT_BIT_ERR)
+ dev_dbg(&pdev->dev, "isr: receiver found parity bit error\n");
+
+ if (sis & INT_URX_FUL)
+ spdif_irq_uqrx_full(spdif_priv, 'U');
+
+ if (sis & INT_URX_OV)
+ dev_dbg(&pdev->dev, "isr: U Channel receive register overrun\n");
+
+ if (sis & INT_QRX_FUL)
+ spdif_irq_uqrx_full(spdif_priv, 'Q');
+
+ if (sis & INT_QRX_OV)
+ dev_dbg(&pdev->dev, "isr: Q Channel receive register overrun\n");
+
+ if (sis & INT_UQ_SYNC)
+ spdif_irq_uq_sync(spdif_priv);
+
+ if (sis & INT_UQ_ERR)
+ spdif_irq_uq_err(spdif_priv);
+
+ if (sis & INT_RXFIFO_UNOV)
+ dev_dbg(&pdev->dev, "isr: Rx FIFO under/overrun\n");
+
+ if (sis & INT_RXFIFO_RESYNC)
+ dev_dbg(&pdev->dev, "isr: Rx FIFO resync\n");
+
+ if (sis & INT_LOSS_LOCK)
+ spdif_irq_dpll_lock(spdif_priv);
+
+ /* FIXME: Write Tx FIFO to clear TxEm */
+ if (sis & INT_TX_EM)
+ dev_dbg(&pdev->dev, "isr: Tx FIFO empty\n");
+
+ /* FIXME: Read Rx FIFO to clear RxFIFOFul */
+ if (sis & INT_RXFIFO_FUL)
+ dev_dbg(&pdev->dev, "isr: Rx FIFO full\n");
+
+ return IRQ_HANDLED;
+}
+
+static int spdif_softreset(struct fsl_spdif_priv *spdif_priv)
+{
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 val, cycle = 1000;
+
+ regmap_write(regmap, REG_SPDIF_SCR, SCR_SOFT_RESET);
+
+ /*
+ * RESET bit would be cleared after finishing its reset procedure,
+ * which typically lasts 8 cycles. 1000 cycles will keep it safe.
+ */
+ do {
+ regmap_read(regmap, REG_SPDIF_SCR, &val);
+ } while ((val & SCR_SOFT_RESET) && cycle--);
+
+ if (cycle)
+ return 0;
+ else
+ return -EBUSY;
+}
+
+static void spdif_set_cstatus(struct spdif_mixer_control *ctrl,
+ u8 mask, u8 cstatus)
+{
+ ctrl->ch_status[3] &= ~mask;
+ ctrl->ch_status[3] |= cstatus & mask;
+}
+
+static void spdif_write_channel_status(struct fsl_spdif_priv *spdif_priv)
+{
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ struct regmap *regmap = spdif_priv->regmap;
+ struct platform_device *pdev = spdif_priv->pdev;
+ u32 ch_status;
+
+ ch_status = (bitrev8(ctrl->ch_status[0]) << 16) |
+ (bitrev8(ctrl->ch_status[1]) << 8) |
+ bitrev8(ctrl->ch_status[2]);
+ regmap_write(regmap, REG_SPDIF_STCSCH, ch_status);
+
+ dev_dbg(&pdev->dev, "STCSCH: 0x%06x\n", ch_status);
+
+ ch_status = bitrev8(ctrl->ch_status[3]) << 16;
+ regmap_write(regmap, REG_SPDIF_STCSCL, ch_status);
+
+ dev_dbg(&pdev->dev, "STCSCL: 0x%06x\n", ch_status);
+}
+
+/* Set SPDIF PhaseConfig register for rx clock */
+static int spdif_set_rx_clksrc(struct fsl_spdif_priv *spdif_priv,
+ enum spdif_gainsel gainsel, int dpll_locked)
+{
+ struct regmap *regmap = spdif_priv->regmap;
+ u8 clksrc = spdif_priv->rxclk_src;
+
+ if (clksrc >= SRPC_CLKSRC_MAX || gainsel >= GAINSEL_MULTI_MAX)
+ return -EINVAL;
+
+ regmap_update_bits(regmap, REG_SPDIF_SRPC,
+ SRPC_CLKSRC_SEL_MASK | SRPC_GAINSEL_MASK,
+ SRPC_CLKSRC_SEL_SET(clksrc) | SRPC_GAINSEL_SET(gainsel));
+
+ return 0;
+}
+
+static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
+ int sample_rate)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ struct regmap *regmap = spdif_priv->regmap;
+ struct platform_device *pdev = spdif_priv->pdev;
+ unsigned long csfs = 0;
+ u32 stc, mask, rate;
+ u8 clk, div;
+ int ret;
+
+ switch (sample_rate) {
+ case 32000:
+ rate = SPDIF_TXRATE_32000;
+ csfs = IEC958_AES3_CON_FS_32000;
+ break;
+ case 44100:
+ rate = SPDIF_TXRATE_44100;
+ csfs = IEC958_AES3_CON_FS_44100;
+ break;
+ case 48000:
+ rate = SPDIF_TXRATE_48000;
+ csfs = IEC958_AES3_CON_FS_48000;
+ break;
+ default:
+ dev_err(&pdev->dev, "unsupported sample rate %d\n", sample_rate);
+ return -EINVAL;
+ }
+
+ clk = spdif_priv->txclk_src[rate];
+ if (clk >= STC_TXCLK_SRC_MAX) {
+ dev_err(&pdev->dev, "tx clock source is out of range\n");
+ return -EINVAL;
+ }
+
+ div = spdif_priv->txclk_div[rate];
+ if (div == 0) {
+ dev_err(&pdev->dev, "the divisor can't be zero\n");
+ return -EINVAL;
+ }
+
+ /*
+ * The S/PDIF block needs a clock of 64 * fs * div. The S/PDIF block
+ * will divide by (div). So request 64 * fs * (div+1) which will
+ * get rounded.
+ */
+ ret = clk_set_rate(spdif_priv->txclk[rate], 64 * sample_rate * (div + 1));
+ if (ret) {
+ dev_err(&pdev->dev, "failed to set tx clock rate\n");
+ return ret;
+ }
+
+ dev_dbg(&pdev->dev, "expected clock rate = %d\n",
+ (64 * sample_rate * div));
+ dev_dbg(&pdev->dev, "actual clock rate = %ld\n",
+ clk_get_rate(spdif_priv->txclk[rate]));
+
+ /* set fs field in consumer channel status */
+ spdif_set_cstatus(ctrl, IEC958_AES3_CON_FS, csfs);
+
+ /* select clock source and divisor */
+ stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) | STC_TXCLK_DIV(div);
+ mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK | STC_TXCLK_DIV_MASK;
+ regmap_update_bits(regmap, REG_SPDIF_STC, mask, stc);
+
+ dev_dbg(&pdev->dev, "set sample rate to %d\n", sample_rate);
+
+ return 0;
+}
+
+int fsl_spdif_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct platform_device *pdev = spdif_priv->pdev;
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 scr, mask, i;
+ int ret;
+
+ /* Reset module and interrupts only for first initialization */
+ if (!cpu_dai->active) {
+ ret = spdif_softreset(spdif_priv);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to soft reset\n");
+ return ret;
+ }
+
+ /* Disable all the interrupts */
+ regmap_update_bits(regmap, REG_SPDIF_SIE, 0xffffff, 0);
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ scr = SCR_TXFIFO_AUTOSYNC | SCR_TXFIFO_CTRL_NORMAL |
+ SCR_TXSEL_NORMAL | SCR_USRC_SEL_CHIP |
+ SCR_TXFIFO_FSEL_IF8;
+ mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK |
+ SCR_TXSEL_MASK | SCR_USRC_SEL_MASK |
+ SCR_TXFIFO_FSEL_MASK;
+ for (i = 0; i < SPDIF_TXRATE_MAX; i++)
+ clk_prepare_enable(spdif_priv->txclk[i]);
+ } else {
+ scr = SCR_RXFIFO_FSEL_IF8 | SCR_RXFIFO_AUTOSYNC;
+ mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK|
+ SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK;
+ clk_prepare_enable(spdif_priv->rxclk);
+ }
+ regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr);
+
+ /* Power up SPDIF module */
+ regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_LOW_POWER, 0);
+
+ return 0;
+}
+
+static void fsl_spdif_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 scr, mask, i;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ scr = 0;
+ mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK |
+ SCR_TXSEL_MASK | SCR_USRC_SEL_MASK |
+ SCR_TXFIFO_FSEL_MASK;
+ for (i = 0; i < SPDIF_TXRATE_MAX; i++)
+ clk_disable_unprepare(spdif_priv->txclk[i]);
+ } else {
+ scr = SCR_RXFIFO_OFF | SCR_RXFIFO_CTL_ZERO;
+ mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK|
+ SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK;
+ clk_disable_unprepare(spdif_priv->rxclk);
+ }
+ regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr);
+
+ /* Power down SPDIF module only if tx&rx are both inactive */
+ if (!cpu_dai->active) {
+ spdif_intr_status_clear(spdif_priv);
+ regmap_update_bits(regmap, REG_SPDIF_SCR,
+ SCR_LOW_POWER, SCR_LOW_POWER);
+ }
+}
+
+static int fsl_spdif_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ struct platform_device *pdev = spdif_priv->pdev;
+ u32 sample_rate = params_rate(params);
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = spdif_set_sample_rate(substream, sample_rate);
+ if (ret) {
+ dev_err(&pdev->dev, "%s: set sample rate failed: %d\n",
+ __func__, sample_rate);
+ return ret;
+ }
+ spdif_set_cstatus(ctrl, IEC958_AES3_CON_CLOCK,
+ IEC958_AES3_CON_CLOCK_1000PPM);
+ spdif_write_channel_status(spdif_priv);
+ } else {
+ /* Setup rx clock source */
+ ret = spdif_set_rx_clksrc(spdif_priv, SPDIF_DEFAULT_GAINSEL, 1);
+ }
+
+ return ret;
+}
+
+static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct regmap *regmap = spdif_priv->regmap;
+ int is_playack = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+ u32 intr = is_playack ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE;
+ u32 dmaen = is_playack ? SCR_DMA_TX_EN : SCR_DMA_RX_EN;;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ regmap_update_bits(regmap, REG_SPDIF_SIE, intr, intr);
+ regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, dmaen);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, 0);
+ regmap_update_bits(regmap, REG_SPDIF_SIE, intr, 0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+struct snd_soc_dai_ops fsl_spdif_dai_ops = {
+ .startup = fsl_spdif_startup,
+ .hw_params = fsl_spdif_hw_params,
+ .trigger = fsl_spdif_trigger,
+ .shutdown = fsl_spdif_shutdown,
+};
+
+
+/*
+ * FSL SPDIF IEC958 controller(mixer) functions
+ *
+ * Channel status get/put control
+ * User bit value get/put control
+ * Valid bit value get control
+ * DPLL lock status get control
+ * User bit sync mode selection control
+ */
+
+static int fsl_spdif_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+
+ return 0;
+}
+
+static int fsl_spdif_pb_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *uvalue)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+
+ uvalue->value.iec958.status[0] = ctrl->ch_status[0];
+ uvalue->value.iec958.status[1] = ctrl->ch_status[1];
+ uvalue->value.iec958.status[2] = ctrl->ch_status[2];
+ uvalue->value.iec958.status[3] = ctrl->ch_status[3];
+
+ return 0;
+}
+
+static int fsl_spdif_pb_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *uvalue)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+
+ ctrl->ch_status[0] = uvalue->value.iec958.status[0];
+ ctrl->ch_status[1] = uvalue->value.iec958.status[1];
+ ctrl->ch_status[2] = uvalue->value.iec958.status[2];
+ ctrl->ch_status[3] = uvalue->value.iec958.status[3];
+
+ spdif_write_channel_status(spdif_priv);
+
+ return 0;
+}
+
+/* Get channel status from SPDIF_RX_CCHAN register */
+static int fsl_spdif_capture_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 cstatus, val;
+
+ regmap_read(regmap, REG_SPDIF_SIS, &val);
+ if (!(val & INT_CNEW)) {
+ return -EAGAIN;
+ }
+
+ regmap_read(regmap, REG_SPDIF_SRCSH, &cstatus);
+ ucontrol->value.iec958.status[0] = (cstatus >> 16) & 0xFF;
+ ucontrol->value.iec958.status[1] = (cstatus >> 8) & 0xFF;
+ ucontrol->value.iec958.status[2] = cstatus & 0xFF;
+
+ regmap_read(regmap, REG_SPDIF_SRCSL, &cstatus);
+ ucontrol->value.iec958.status[3] = (cstatus >> 16) & 0xFF;
+ ucontrol->value.iec958.status[4] = (cstatus >> 8) & 0xFF;
+ ucontrol->value.iec958.status[5] = cstatus & 0xFF;
+
+ /* Clear intr */
+ regmap_write(regmap, REG_SPDIF_SIC, INT_CNEW);
+
+ return 0;
+}
+
+/*
+ * Get User bits (subcode) from chip value which readed out
+ * in UChannel register.
+ */
+static int fsl_spdif_subcode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ unsigned long flags;
+ int ret = 0;
+
+ spin_lock_irqsave(&ctrl->ctl_lock, flags);
+ if (ctrl->ready_buf) {
+ int idx = (ctrl->ready_buf - 1) * SPDIF_UBITS_SIZE;
+ memcpy(&ucontrol->value.iec958.subcode[0],
+ &ctrl->subcode[idx], SPDIF_UBITS_SIZE);
+ } else {
+ ret = -EAGAIN;
+ }
+ spin_unlock_irqrestore(&ctrl->ctl_lock, flags);
+
+ return ret;
+}
+
+/* Q-subcode infomation. The byte size is SPDIF_UBITS_SIZE/8 */
+static int fsl_spdif_qinfo(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ uinfo->count = SPDIF_QSUB_SIZE;
+
+ return 0;
+}
+
+/* Get Q subcode from chip value which readed out in QChannel register */
+static int fsl_spdif_qget(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ unsigned long flags;
+ int ret = 0;
+
+ spin_lock_irqsave(&ctrl->ctl_lock, flags);
+ if (ctrl->ready_buf) {
+ int idx = (ctrl->ready_buf - 1) * SPDIF_QSUB_SIZE;
+ memcpy(&ucontrol->value.bytes.data[0],
+ &ctrl->qsub[idx], SPDIF_QSUB_SIZE);
+ } else {
+ ret = -EAGAIN;
+ }
+ spin_unlock_irqrestore(&ctrl->ctl_lock, flags);
+
+ return ret;
+}
+
+/* Valid bit infomation */
+static int fsl_spdif_vbit_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+
+ return 0;
+}
+
+/* Get valid good bit from interrupt status register */
+static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 val;
+
+ val = regmap_read(regmap, REG_SPDIF_SIS, &val);
+ ucontrol->value.integer.value[0] = (val & INT_VAL_NOGOOD) != 0;
+ regmap_write(regmap, REG_SPDIF_SIC, INT_VAL_NOGOOD);
+
+ return 0;
+}
+
+/* DPLL lock infomation */
+static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 16000;
+ uinfo->value.integer.max = 96000;
+
+ return 0;
+}
+
+static u32 gainsel_multi[GAINSEL_MULTI_MAX] = {
+ 24, 16, 12, 8, 6, 4, 3,
+};
+
+/* Get RX data clock rate given the SPDIF bus_clk */
+static int spdif_get_rxclk_rate(struct fsl_spdif_priv *spdif_priv,
+ enum spdif_gainsel gainsel)
+{
+ struct regmap *regmap = spdif_priv->regmap;
+ struct platform_device *pdev = spdif_priv->pdev;
+ u64 tmpval64, busclk_freq = 0;
+ u32 freqmeas, phaseconf;
+ u8 clksrc;
+
+ regmap_read(regmap, REG_SPDIF_SRFM, &freqmeas);
+ regmap_read(regmap, REG_SPDIF_SRPC, &phaseconf);
+
+ clksrc = (phaseconf >> SRPC_CLKSRC_SEL_OFFSET) & 0xf;
+ if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED)) {
+ /* Get bus clock from system */
+ busclk_freq = clk_get_rate(spdif_priv->rxclk);
+ }
+
+ /* FreqMeas_CLK = (BUS_CLK * FreqMeas) / 2 ^ 10 / GAINSEL / 128 */
+ tmpval64 = (u64) busclk_freq * freqmeas;
+ do_div(tmpval64, gainsel_multi[gainsel] * 1024);
+ do_div(tmpval64, 128 * 1024);
+
+ dev_dbg(&pdev->dev, "FreqMeas: %d\n", freqmeas);
+ dev_dbg(&pdev->dev, "BusclkFreq: %lld\n", busclk_freq);
+ dev_dbg(&pdev->dev, "RxRate: %lld\n", tmpval64);
+
+ return (int)tmpval64;
+}
+
+/*
+ * Get DPLL lock or not info from stable interrupt status register.
+ * User application must use this control to get locked,
+ * then can do next PCM operation
+ */
+static int fsl_spdif_rxrate_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ int rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL);
+
+ if (spdif_priv->dpll_locked)
+ ucontrol->value.integer.value[0] = rate;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ return 0;
+}
+
+/* User bit sync mode info */
+static int fsl_spdif_usync_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+
+ return 0;
+}
+
+/*
+ * User bit sync mode:
+ * 1 CD User channel subcode
+ * 0 Non-CD data
+ */
+static int fsl_spdif_usync_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 val;
+
+ regmap_read(regmap, REG_SPDIF_SRCD, &val);
+ ucontrol->value.integer.value[0] = (val & SRCD_CD_USER) != 0;
+
+ return 0;
+}
+
+/*
+ * User bit sync mode:
+ * 1 CD User channel subcode
+ * 0 Non-CD data
+ */
+static int fsl_spdif_usync_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 val = ucontrol->value.integer.value[0] << SRCD_CD_USER_OFFSET;
+
+ regmap_update_bits(regmap, REG_SPDIF_SRCD, SRCD_CD_USER, val);
+
+ return 0;
+}
+
+/* FSL SPDIF IEC958 controller defines */
+static struct snd_kcontrol_new fsl_spdif_ctrls[] = {
+ /* Status cchanel controller */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_WRITE |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = fsl_spdif_info,
+ .get = fsl_spdif_pb_get,
+ .put = fsl_spdif_pb_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = fsl_spdif_info,
+ .get = fsl_spdif_capture_get,
+ },
+ /* User bits controller */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Subcode Capture Default",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = fsl_spdif_info,
+ .get = fsl_spdif_subcode_get,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Q-subcode Capture Default",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = fsl_spdif_qinfo,
+ .get = fsl_spdif_qget,
+ },
+ /* Valid bit error controller */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 V-Bit Errors",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = fsl_spdif_vbit_info,
+ .get = fsl_spdif_vbit_get,
+ },
+ /* DPLL lock info get controller */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "RX Sample Rate",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = fsl_spdif_rxrate_info,
+ .get = fsl_spdif_rxrate_get,
+ },
+ /* User bit sync mode set/get controller */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 USyncMode CDText",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_WRITE |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = fsl_spdif_usync_info,
+ .get = fsl_spdif_usync_get,
+ .put = fsl_spdif_usync_put,
+ },
+};
+
+static int fsl_spdif_dai_probe(struct snd_soc_dai *dai)
+{
+ struct fsl_spdif_priv *spdif_private = snd_soc_dai_get_drvdata(dai);
+
+ dai->playback_dma_data = &spdif_private->dma_params_tx;
+ dai->capture_dma_data = &spdif_private->dma_params_rx;
+
+ snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls));
+
+ return 0;
+}
+
+struct snd_soc_dai_driver fsl_spdif_dai = {
+ .probe = &fsl_spdif_dai_probe,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = FSL_SPDIF_RATES_PLAYBACK,
+ .formats = FSL_SPDIF_FORMATS_PLAYBACK,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = FSL_SPDIF_RATES_CAPTURE,
+ .formats = FSL_SPDIF_FORMATS_CAPTURE,
+ },
+ .ops = &fsl_spdif_dai_ops,
+};
+
+static const struct snd_soc_component_driver fsl_spdif_component = {
+ .name = "fsl-spdif",
+};
+
+/* FSL SPDIF REGMAP */
+
+static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case REG_SPDIF_SCR:
+ case REG_SPDIF_SRCD:
+ case REG_SPDIF_SRPC:
+ case REG_SPDIF_SIE:
+ case REG_SPDIF_SIS:
+ case REG_SPDIF_SRL:
+ case REG_SPDIF_SRR:
+ case REG_SPDIF_SRCSH:
+ case REG_SPDIF_SRCSL:
+ case REG_SPDIF_SRU:
+ case REG_SPDIF_SRQ:
+ case REG_SPDIF_STCSCH:
+ case REG_SPDIF_STCSCL:
+ case REG_SPDIF_SRFM:
+ case REG_SPDIF_STC:
+ return true;
+ default:
+ return false;
+ };
+}
+
+static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case REG_SPDIF_SCR:
+ case REG_SPDIF_SRCD:
+ case REG_SPDIF_SRPC:
+ case REG_SPDIF_SIE:
+ case REG_SPDIF_SIC:
+ case REG_SPDIF_STL:
+ case REG_SPDIF_STR:
+ case REG_SPDIF_STCSCH:
+ case REG_SPDIF_STCSCL:
+ case REG_SPDIF_STC:
+ return true;
+ default:
+ return false;
+ };
+}
+
+static const struct regmap_config fsl_spdif_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+
+ .max_register = REG_SPDIF_STC,
+ .readable_reg = fsl_spdif_readable_reg,
+ .writeable_reg = fsl_spdif_writeable_reg,
+};
+
+static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
+ struct clk *clk, u64 savesub,
+ enum spdif_txrate index)
+{
+ const u32 rate[] = { 32000, 44100, 48000 };
+ u64 rate_ideal, rate_actual, sub;
+ u32 div, arate;
+
+ for (div = 1; div <= 128; div++) {
+ rate_ideal = rate[index] * (div + 1) * 64;
+ rate_actual = clk_round_rate(clk, rate_ideal);
+
+ arate = rate_actual / 64;
+ arate /= div;
+
+ if (arate == rate[index]) {
+ /* We are lucky */
+ savesub = 0;
+ spdif_priv->txclk_div[index] = div;
+ break;
+ } else if (arate / rate[index] == 1) {
+ /* A little bigger than expect */
+ sub = (arate - rate[index]) * 100000;
+ do_div(sub, rate[index]);
+ if (sub < savesub) {
+ savesub = sub;
+ spdif_priv->txclk_div[index] = div;
+ }
+ } else if (rate[index] / arate == 1) {
+ /* A little smaller than expect */
+ sub = (rate[index] - arate) * 100000;
+ do_div(sub, rate[index]);
+ if (sub < savesub) {
+ savesub = sub;
+ spdif_priv->txclk_div[index] = div;
+ }
+ }
+ }
+
+ return savesub;
+}
+
+static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
+ enum spdif_txrate index)
+{
+ const u32 rate[] = { 32000, 44100, 48000 };
+ struct platform_device *pdev = spdif_priv->pdev;
+ struct device *dev = &pdev->dev;
+ u64 savesub = 100000, ret;
+ struct clk *clk;
+ char tmp[16];
+ int i;
+
+ for (i = 0; i < STC_TXCLK_SRC_MAX; i++) {
+ sprintf(tmp, "rxtx%d", i);
+ clk = devm_clk_get(&pdev->dev, tmp);
+ if (IS_ERR(clk)) {
+ dev_err(dev, "no rxtx%d clock in devicetree\n", i);
+ return PTR_ERR(clk);
+ }
+ if (!clk_get_rate(clk))
+ continue;
+
+ ret = fsl_spdif_txclk_caldiv(spdif_priv, clk, savesub, index);
+ if (savesub == ret)
+ continue;
+
+ savesub = ret;
+ spdif_priv->txclk[index] = clk;
+ spdif_priv->txclk_src[index] = i;
+
+ /* To quick catch a divisor, we allow a 0.1% deviation */
+ if (savesub < 100)
+ break;
+ }
+
+ dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate",
+ spdif_priv->txclk_src[index], rate[index]);
+ dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate",
+ spdif_priv->txclk_div[index], rate[index]);
+
+ return 0;
+}
+
+static int fsl_spdif_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct fsl_spdif_priv *spdif_priv;
+ struct spdif_mixer_control *ctrl;
+ struct resource *res;
+ void __iomem *regs;
+ int irq, ret, i;
+
+ if (!np)
+ return -ENODEV;
+
+ spdif_priv = devm_kzalloc(&pdev->dev,
+ sizeof(struct fsl_spdif_priv) + strlen(np->name) + 1,
+ GFP_KERNEL);
+ if (!spdif_priv)
+ return -ENOMEM;
+
+ strcpy(spdif_priv->name, np->name);
+
+ spdif_priv->pdev = pdev;
+
+ /* Initialize this copy of the CPU DAI driver structure */
+ memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
+ spdif_priv->cpu_dai_drv.name = spdif_priv->name;
+
+ /* Get the addresses and IRQ */
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (IS_ERR(res)) {
+ dev_err(&pdev->dev, "could not determine device resources\n");
+ return PTR_ERR(res);
+ }
+
+ regs = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ spdif_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
+ "core", regs, &fsl_spdif_regmap_config);
+ if (IS_ERR(spdif_priv->regmap)) {
+ dev_err(&pdev->dev, "regmap init failed\n");
+ return PTR_ERR(spdif_priv->regmap);
+ }
+
+ irq = platform_get_irq(pdev, 0);
+ if (irq < 0) {
+ dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+ return irq;
+ }
+
+ ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0,
+ spdif_priv->name, spdif_priv);
+ if (ret) {
+ dev_err(&pdev->dev, "could not claim irq %u\n", irq);
+ return ret;
+ }
+
+ /* Select clock source for rx/tx clock */
+ spdif_priv->rxclk = devm_clk_get(&pdev->dev, "rxtx1");
+ if (IS_ERR(spdif_priv->rxclk)) {
+ dev_err(&pdev->dev, "no rxtx1 clock in devicetree\n");
+ return PTR_ERR(spdif_priv->rxclk);
+ }
+ spdif_priv->rxclk_src = DEFAULT_RXCLK_SRC;
+
+ for (i = 0; i < SPDIF_TXRATE_MAX; i++) {
+ ret = fsl_spdif_probe_txclk(spdif_priv, i);
+ if (ret)
+ return ret;
+ }
+
+ /* Initial spinlock for control data */
+ ctrl = &spdif_priv->fsl_spdif_control;
+ spin_lock_init(&ctrl->ctl_lock);
+
+ /* Init tx channel status default value */
+ ctrl->ch_status[0] =
+ IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_5015;
+ ctrl->ch_status[1] = IEC958_AES1_CON_DIGDIGCONV_ID;
+ ctrl->ch_status[2] = 0x00;
+ ctrl->ch_status[3] =
+ IEC958_AES3_CON_FS_44100 | IEC958_AES3_CON_CLOCK_1000PPM;
+
+ spdif_priv->dpll_locked = false;
+
+ spdif_priv->dma_params_tx.maxburst = FSL_SPDIF_TXFIFO_WML;
+ spdif_priv->dma_params_rx.maxburst = FSL_SPDIF_RXFIFO_WML;
+ spdif_priv->dma_params_tx.addr = res->start + REG_SPDIF_STL;
+ spdif_priv->dma_params_rx.addr = res->start + REG_SPDIF_SRL;
+
+ /* Register with ASoC */
+ dev_set_drvdata(&pdev->dev, spdif_priv);
+
+ ret = snd_soc_register_component(&pdev->dev, &fsl_spdif_component,
+ &spdif_priv->cpu_dai_drv, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
+ return ret;
+ }
+
+ ret = imx_pcm_dma_init(pdev);
+ if (ret) {
+ dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret);
+ goto error_component;
+ }
+
+ return ret;
+
+error_component:
+ snd_soc_unregister_component(&pdev->dev);
+
+ return ret;
+}
+
+static int fsl_spdif_remove(struct platform_device *pdev)
+{
+ imx_pcm_dma_exit(pdev);
+ snd_soc_unregister_component(&pdev->dev);
+
+ return 0;
+}
+
+static const struct of_device_id fsl_spdif_dt_ids[] = {
+ { .compatible = "fsl,imx35-spdif", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, fsl_spdif_dt_ids);
+
+static struct platform_driver fsl_spdif_driver = {
+ .driver = {
+ .name = "fsl-spdif-dai",
+ .owner = THIS_MODULE,
+ .of_match_table = fsl_spdif_dt_ids,
+ },
+ .probe = fsl_spdif_probe,
+ .remove = fsl_spdif_remove,
+};
+
+module_platform_driver(fsl_spdif_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale S/PDIF CPU DAI Driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:fsl-spdif-dai");
diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h
new file mode 100644
index 0000000..b126679
--- /dev/null
+++ b/sound/soc/fsl/fsl_spdif.h
@@ -0,0 +1,191 @@
+/*
+ * fsl_spdif.h - ALSA S/PDIF interface for the Freescale i.MX SoC
+ *
+ * Copyright (C) 2013 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <b42378@freescale.com>
+ *
+ * Based on fsl_ssi.h
+ * Author: Timur Tabi <timur@freescale.com>
+ * Copyright 2007-2008 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#ifndef _FSL_SPDIF_DAI_H
+#define _FSL_SPDIF_DAI_H
+
+/* S/PDIF Register Map */
+#define REG_SPDIF_SCR 0x0 /* SPDIF Configuration Register */
+#define REG_SPDIF_SRCD 0x4 /* CDText Control Register */
+#define REG_SPDIF_SRPC 0x8 /* PhaseConfig Register */
+#define REG_SPDIF_SIE 0xc /* InterruptEn Register */
+#define REG_SPDIF_SIS 0x10 /* InterruptStat Register */
+#define REG_SPDIF_SIC 0x10 /* InterruptClear Register */
+#define REG_SPDIF_SRL 0x14 /* SPDIFRxLeft Register */
+#define REG_SPDIF_SRR 0x18 /* SPDIFRxRight Register */
+#define REG_SPDIF_SRCSH 0x1c /* SPDIFRxCChannel_h Register */
+#define REG_SPDIF_SRCSL 0x20 /* SPDIFRxCChannel_l Register */
+#define REG_SPDIF_SRU 0x24 /* UchannelRx Register */
+#define REG_SPDIF_SRQ 0x28 /* QchannelRx Register */
+#define REG_SPDIF_STL 0x2C /* SPDIFTxLeft Register */
+#define REG_SPDIF_STR 0x30 /* SPDIFTxRight Register */
+#define REG_SPDIF_STCSCH 0x34 /* SPDIFTxCChannelCons_h Register */
+#define REG_SPDIF_STCSCL 0x38 /* SPDIFTxCChannelCons_l Register */
+#define REG_SPDIF_SRFM 0x44 /* FreqMeas Register */
+#define REG_SPDIF_STC 0x50 /* SPDIFTxClk Register */
+
+
+/* SPDIF Configuration register */
+#define SCR_RXFIFO_CTL_OFFSET 23
+#define SCR_RXFIFO_CTL_MASK (1 << SCR_RXFIFO_CTL_OFFSET)
+#define SCR_RXFIFO_CTL_ZERO (1 << SCR_RXFIFO_CTL_OFFSET)
+#define SCR_RXFIFO_OFF_OFFSET 22
+#define SCR_RXFIFO_OFF_MASK (1 << SCR_RXFIFO_OFF_OFFSET)
+#define SCR_RXFIFO_OFF (1 << SCR_RXFIFO_OFF_OFFSET)
+#define SCR_RXFIFO_RST_OFFSET 21
+#define SCR_RXFIFO_RST_MASK (1 << SCR_RXFIFO_RST_OFFSET)
+#define SCR_RXFIFO_RST (1 << SCR_RXFIFO_RST_OFFSET)
+#define SCR_RXFIFO_FSEL_OFFSET 19
+#define SCR_RXFIFO_FSEL_MASK (0x3 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_FSEL_IF0 (0x0 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_FSEL_IF4 (0x1 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_FSEL_IF8 (0x2 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_FSEL_IF12 (0x3 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_AUTOSYNC_OFFSET 18
+#define SCR_RXFIFO_AUTOSYNC_MASK (1 << SCR_RXFIFO_AUTOSYNC_OFFSET)
+#define SCR_RXFIFO_AUTOSYNC (1 << SCR_RXFIFO_AUTOSYNC_OFFSET)
+#define SCR_TXFIFO_AUTOSYNC_OFFSET 17
+#define SCR_TXFIFO_AUTOSYNC_MASK (1 << SCR_TXFIFO_AUTOSYNC_OFFSET)
+#define SCR_TXFIFO_AUTOSYNC (1 << SCR_TXFIFO_AUTOSYNC_OFFSET)
+#define SCR_TXFIFO_FSEL_OFFSET 15
+#define SCR_TXFIFO_FSEL_MASK (0x3 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_TXFIFO_FSEL_IF0 (0x0 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_TXFIFO_FSEL_IF4 (0x1 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_TXFIFO_FSEL_IF8 (0x2 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_TXFIFO_FSEL_IF12 (0x3 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_LOW_POWER (1 << 13)
+#define SCR_SOFT_RESET (1 << 12)
+#define SCR_TXFIFO_CTRL_OFFSET 10
+#define SCR_TXFIFO_CTRL_MASK (0x3 << SCR_TXFIFO_CTRL_OFFSET)
+#define SCR_TXFIFO_CTRL_ZERO (0x0 << SCR_TXFIFO_CTRL_OFFSET)
+#define SCR_TXFIFO_CTRL_NORMAL (0x1 << SCR_TXFIFO_CTRL_OFFSET)
+#define SCR_TXFIFO_CTRL_ONESAMPLE (0x2 << SCR_TXFIFO_CTRL_OFFSET)
+#define SCR_DMA_RX_EN_OFFSET 9
+#define SCR_DMA_RX_EN_MASK (1 << SCR_DMA_RX_EN_OFFSET)
+#define SCR_DMA_RX_EN (1 << SCR_DMA_RX_EN_OFFSET)
+#define SCR_DMA_TX_EN_OFFSET 8
+#define SCR_DMA_TX_EN_MASK (1 << SCR_DMA_TX_EN_OFFSET)
+#define SCR_DMA_TX_EN (1 << SCR_DMA_TX_EN_OFFSET)
+#define SCR_VAL_OFFSET 5
+#define SCR_VAL_MASK (1 << SCR_VAL_OFFSET)
+#define SCR_VAL_CLEAR (1 << SCR_VAL_OFFSET)
+#define SCR_TXSEL_OFFSET 2
+#define SCR_TXSEL_MASK (0x7 << SCR_TXSEL_OFFSET)
+#define SCR_TXSEL_OFF (0 << SCR_TXSEL_OFFSET)
+#define SCR_TXSEL_RX (1 << SCR_TXSEL_OFFSET)
+#define SCR_TXSEL_NORMAL (0x5 << SCR_TXSEL_OFFSET)
+#define SCR_USRC_SEL_OFFSET 0x0
+#define SCR_USRC_SEL_MASK (0x3 << SCR_USRC_SEL_OFFSET)
+#define SCR_USRC_SEL_NONE (0x0 << SCR_USRC_SEL_OFFSET)
+#define SCR_USRC_SEL_RECV (0x1 << SCR_USRC_SEL_OFFSET)
+#define SCR_USRC_SEL_CHIP (0x3 << SCR_USRC_SEL_OFFSET)
+
+/* SPDIF CDText control */
+#define SRCD_CD_USER_OFFSET 1
+#define SRCD_CD_USER (1 << SRCD_CD_USER_OFFSET)
+
+/* SPDIF Phase Configuration register */
+#define SRPC_DPLL_LOCKED (1 << 6)
+#define SRPC_CLKSRC_SEL_OFFSET 7
+#define SRPC_CLKSRC_SEL_MASK (0xf << SRPC_CLKSRC_SEL_OFFSET)
+#define SRPC_CLKSRC_SEL_SET(x) ((x << SRPC_CLKSRC_SEL_OFFSET) & SRPC_CLKSRC_SEL_MASK)
+#define SRPC_CLKSRC_SEL_LOCKED_OFFSET1 5
+#define SRPC_CLKSRC_SEL_LOCKED_OFFSET2 2
+#define SRPC_GAINSEL_OFFSET 3
+#define SRPC_GAINSEL_MASK (0x7 << SRPC_GAINSEL_OFFSET)
+#define SRPC_GAINSEL_SET(x) ((x << SRPC_GAINSEL_OFFSET) & SRPC_GAINSEL_MASK)
+
+#define SRPC_CLKSRC_MAX 16
+
+enum spdif_gainsel {
+ GAINSEL_MULTI_24 = 0,
+ GAINSEL_MULTI_16,
+ GAINSEL_MULTI_12,
+ GAINSEL_MULTI_8,
+ GAINSEL_MULTI_6,
+ GAINSEL_MULTI_4,
+ GAINSEL_MULTI_3,
+};
+#define GAINSEL_MULTI_MAX (GAINSEL_MULTI_3 + 1)
+#define SPDIF_DEFAULT_GAINSEL GAINSEL_MULTI_8
+
+/* SPDIF interrupt mask define */
+#define INT_DPLL_LOCKED (1 << 20)
+#define INT_TXFIFO_UNOV (1 << 19)
+#define INT_TXFIFO_RESYNC (1 << 18)
+#define INT_CNEW (1 << 17)
+#define INT_VAL_NOGOOD (1 << 16)
+#define INT_SYM_ERR (1 << 15)
+#define INT_BIT_ERR (1 << 14)
+#define INT_URX_FUL (1 << 10)
+#define INT_URX_OV (1 << 9)
+#define INT_QRX_FUL (1 << 8)
+#define INT_QRX_OV (1 << 7)
+#define INT_UQ_SYNC (1 << 6)
+#define INT_UQ_ERR (1 << 5)
+#define INT_RXFIFO_UNOV (1 << 4)
+#define INT_RXFIFO_RESYNC (1 << 3)
+#define INT_LOSS_LOCK (1 << 2)
+#define INT_TX_EM (1 << 1)
+#define INT_RXFIFO_FUL (1 << 0)
+
+/* SPDIF Clock register */
+#define STC_SYSCLK_DIV_OFFSET 11
+#define STC_SYSCLK_DIV_MASK (0x1ff << STC_TXCLK_SRC_OFFSET)
+#define STC_SYSCLK_DIV(x) ((((x) - 1) << STC_TXCLK_DIV_OFFSET) & STC_SYSCLK_DIV_MASK)
+#define STC_TXCLK_SRC_OFFSET 8
+#define STC_TXCLK_SRC_MASK (0x7 << STC_TXCLK_SRC_OFFSET)
+#define STC_TXCLK_SRC_SET(x) ((x << STC_TXCLK_SRC_OFFSET) & STC_TXCLK_SRC_MASK)
+#define STC_TXCLK_ALL_EN_OFFSET 7
+#define STC_TXCLK_ALL_EN_MASK (1 << STC_TXCLK_ALL_EN_OFFSET)
+#define STC_TXCLK_ALL_EN (1 << STC_TXCLK_ALL_EN_OFFSET)
+#define STC_TXCLK_DIV_OFFSET 0
+#define STC_TXCLK_DIV_MASK (0x7ff << STC_TXCLK_DIV_OFFSET)
+#define STC_TXCLK_DIV(x) ((((x) - 1) << STC_TXCLK_DIV_OFFSET) & STC_TXCLK_DIV_MASK)
+#define STC_TXCLK_SRC_MAX 8
+
+/* SPDIF tx rate */
+enum spdif_txrate {
+ SPDIF_TXRATE_32000 = 0,
+ SPDIF_TXRATE_44100,
+ SPDIF_TXRATE_48000,
+};
+#define SPDIF_TXRATE_MAX (SPDIF_TXRATE_48000 + 1)
+
+
+#define SPDIF_CSTATUS_BYTE 6
+#define SPDIF_UBITS_SIZE 96
+#define SPDIF_QSUB_SIZE (SPDIF_UBITS_SIZE / 8)
+
+
+#define FSL_SPDIF_RATES_PLAYBACK (SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+#define FSL_SPDIF_RATES_CAPTURE (SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_64000 | \
+ SNDRV_PCM_RATE_96000)
+
+#define FSL_SPDIF_FORMATS_PLAYBACK (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+#define FSL_SPDIF_FORMATS_CAPTURE (SNDRV_PCM_FMTBIT_S24_LE)
+
+#endif /* _FSL_SPDIF_DAI_H */
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 2f2d837..c6b7439 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -8,6 +8,26 @@
* This file is licensed under the terms of the GNU General Public License
* version 2. This program is licensed "as is" without any warranty of any
* kind, whether express or implied.
+ *
+ *
+ * Some notes why imx-pcm-fiq is used instead of DMA on some boards:
+ *
+ * The i.MX SSI core has some nasty limitations in AC97 mode. While most
+ * sane processor vendors have a FIFO per AC97 slot, the i.MX has only
+ * one FIFO which combines all valid receive slots. We cannot even select
+ * which slots we want to receive. The WM9712 with which this driver
+ * was developed with always sends GPIO status data in slot 12 which
+ * we receive in our (PCM-) data stream. The only chance we have is to
+ * manually skip this data in the FIQ handler. With sampling rates different
+ * from 48000Hz not every frame has valid receive data, so the ratio
+ * between pcm data and GPIO status data changes. Our FIQ handler is not
+ * able to handle this, hence this driver only works with 48000Hz sampling
+ * rate.
+ * Reading and writing AC97 registers is another challenge. The core
+ * provides us status bits when the read register is updated with *another*
+ * value. When we read the same register two times (and the register still
+ * contains the same value) these status bits are not set. We work
+ * around this by not polling these bits but only wait a fixed delay.
*/
#include <linux/init.h>
@@ -36,7 +56,7 @@
#define read_ssi(addr) in_be32(addr)
#define write_ssi(val, addr) out_be32(addr, val)
#define write_ssi_mask(addr, clear, set) clrsetbits_be32(addr, clear, set)
-#elif defined ARM
+#else
#define read_ssi(addr) readl(addr)
#define write_ssi(val, addr) writel(val, addr)
/*
@@ -121,11 +141,14 @@
bool new_binding;
bool ssi_on_imx;
+ bool imx_ac97;
+ bool use_dma;
struct clk *clk;
struct snd_dmaengine_dai_dma_data dma_params_tx;
struct snd_dmaengine_dai_dma_data dma_params_rx;
struct imx_dma_data filter_data_tx;
struct imx_dma_data filter_data_rx;
+ struct imx_pcm_fiq_params fiq_params;
struct {
unsigned int rfrc;
@@ -298,6 +321,102 @@
return ret;
}
+static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private)
+{
+ struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+ u8 i2s_mode;
+ u8 wm;
+ int synchronous = ssi_private->cpu_dai_drv.symmetric_rates;
+
+ if (ssi_private->imx_ac97)
+ i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET;
+ else
+ i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE;
+
+ /*
+ * Section 16.5 of the MPC8610 reference manual says that the SSI needs
+ * to be disabled before updating the registers we set here.
+ */
+ write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0);
+
+ /*
+ * Program the SSI into I2S Slave Non-Network Synchronous mode. Also
+ * enable the transmit and receive FIFO.
+ *
+ * FIXME: Little-endian samples require a different shift dir
+ */
+ write_ssi_mask(&ssi->scr,
+ CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN,
+ CCSR_SSI_SCR_TFR_CLK_DIS |
+ i2s_mode |
+ (synchronous ? CCSR_SSI_SCR_SYN : 0));
+
+ write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
+ CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS |
+ CCSR_SSI_STCR_TSCKP, &ssi->stcr);
+
+ write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 |
+ CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS |
+ CCSR_SSI_SRCR_RSCKP, &ssi->srcr);
+ /*
+ * The DC and PM bits are only used if the SSI is the clock master.
+ */
+
+ /*
+ * Set the watermark for transmit FIFI 0 and receive FIFO 0. We don't
+ * use FIFO 1. We program the transmit water to signal a DMA transfer
+ * if there are only two (or fewer) elements left in the FIFO. Two
+ * elements equals one frame (left channel, right channel). This value,
+ * however, depends on the depth of the transmit buffer.
+ *
+ * We set the watermark on the same level as the DMA burstsize. For
+ * fiq it is probably better to use the biggest possible watermark
+ * size.
+ */
+ if (ssi_private->use_dma)
+ wm = ssi_private->fifo_depth - 2;
+ else
+ wm = ssi_private->fifo_depth;
+
+ write_ssi(CCSR_SSI_SFCSR_TFWM0(wm) | CCSR_SSI_SFCSR_RFWM0(wm) |
+ CCSR_SSI_SFCSR_TFWM1(wm) | CCSR_SSI_SFCSR_RFWM1(wm),
+ &ssi->sfcsr);
+
+ /*
+ * For ac97 interrupts are enabled with the startup of the substream
+ * because it is also running without an active substream. Normally SSI
+ * is only enabled when there is a substream.
+ */
+ if (ssi_private->imx_ac97) {
+ /*
+ * Setup the clock control register
+ */
+ write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13),
+ &ssi->stccr);
+ write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13),
+ &ssi->srccr);
+
+ /*
+ * Enable AC97 mode and startup the SSI
+ */
+ write_ssi(CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV,
+ &ssi->sacnt);
+ write_ssi(0xff, &ssi->saccdis);
+ write_ssi(0x300, &ssi->saccen);
+
+ /*
+ * Enable SSI, Transmit and Receive
+ */
+ write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN |
+ CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE);
+
+ write_ssi(CCSR_SSI_SOR_WAIT(3), &ssi->sor);
+ }
+
+ return 0;
+}
+
+
/**
* fsl_ssi_startup: create a new substream
*
@@ -319,70 +438,14 @@
* and initialize the SSI registers.
*/
if (!ssi_private->first_stream) {
- struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
-
ssi_private->first_stream = substream;
/*
- * Section 16.5 of the MPC8610 reference manual says that the
- * SSI needs to be disabled before updating the registers we set
- * here.
+ * fsl_ssi_setup was already called by ac97_init earlier if
+ * the driver is in ac97 mode.
*/
- write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0);
-
- /*
- * Program the SSI into I2S Slave Non-Network Synchronous mode.
- * Also enable the transmit and receive FIFO.
- *
- * FIXME: Little-endian samples require a different shift dir
- */
- write_ssi_mask(&ssi->scr,
- CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN,
- CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE
- | (synchronous ? CCSR_SSI_SCR_SYN : 0));
-
- write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
- CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS |
- CCSR_SSI_STCR_TSCKP, &ssi->stcr);
-
- write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 |
- CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS |
- CCSR_SSI_SRCR_RSCKP, &ssi->srcr);
-
- /*
- * The DC and PM bits are only used if the SSI is the clock
- * master.
- */
-
- /* Enable the interrupts and DMA requests */
- write_ssi(SIER_FLAGS, &ssi->sier);
-
- /*
- * Set the watermark for transmit FIFI 0 and receive FIFO 0. We
- * don't use FIFO 1. We program the transmit water to signal a
- * DMA transfer if there are only two (or fewer) elements left
- * in the FIFO. Two elements equals one frame (left channel,
- * right channel). This value, however, depends on the depth of
- * the transmit buffer.
- *
- * We program the receive FIFO to notify us if at least two
- * elements (one frame) have been written to the FIFO. We could
- * make this value larger (and maybe we should), but this way
- * data will be written to memory as soon as it's available.
- */
- write_ssi(CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) |
- CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2),
- &ssi->sfcsr);
-
- /*
- * We keep the SSI disabled because if we enable it, then the
- * DMA controller will start. It's not supposed to start until
- * the SCR.TE (or SCR.RE) bit is set, but it does anyway. The
- * DMA controller will transfer one "BWC" of data (i.e. the
- * amount of data that the MR.BWC bits are set to). The reason
- * this is bad is because at this point, the PCM driver has not
- * finished initializing the DMA controller.
- */
+ if (!ssi_private->imx_ac97)
+ fsl_ssi_setup(ssi_private);
} else {
if (synchronous) {
struct snd_pcm_runtime *first_runtime =
@@ -492,6 +555,27 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai);
struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+ unsigned int sier_bits;
+
+ /*
+ * Enable only the interrupts and DMA requests
+ * that are needed for the channel. As the fiq
+ * is polling for this bits, we have to ensure
+ * that this are aligned with the preallocated
+ * buffers
+ */
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (ssi_private->use_dma)
+ sier_bits = SIER_FLAGS;
+ else
+ sier_bits = CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TFE0_EN;
+ } else {
+ if (ssi_private->use_dma)
+ sier_bits = SIER_FLAGS;
+ else
+ sier_bits = CCSR_SSI_SIER_RIE | CCSR_SSI_SIER_RFF0_EN;
+ }
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
@@ -510,12 +594,18 @@
write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0);
else
write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0);
+
+ if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) &
+ (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0)
+ write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0);
break;
default:
return -EINVAL;
}
+ write_ssi(sier_bits, &ssi->sier);
+
return 0;
}
@@ -534,22 +624,13 @@
ssi_private->first_stream = ssi_private->second_stream;
ssi_private->second_stream = NULL;
-
- /*
- * If this is the last active substream, disable the SSI.
- */
- if (!ssi_private->first_stream) {
- struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
-
- write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0);
- }
}
static int fsl_ssi_dai_probe(struct snd_soc_dai *dai)
{
struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(dai);
- if (ssi_private->ssi_on_imx) {
+ if (ssi_private->ssi_on_imx && ssi_private->use_dma) {
dai->playback_dma_data = &ssi_private->dma_params_tx;
dai->capture_dma_data = &ssi_private->dma_params_rx;
}
@@ -587,6 +668,133 @@
.name = "fsl-ssi",
};
+/**
+ * fsl_ssi_ac97_trigger: start and stop the AC97 receive/transmit.
+ *
+ * This function is called by ALSA to start, stop, pause, and resume the
+ * transfer of data.
+ */
+static int fsl_ssi_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(
+ rtd->cpu_dai);
+ struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_TIE |
+ CCSR_SSI_SIER_TFE0_EN);
+ else
+ write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_RIE |
+ CCSR_SSI_SIER_RFF0_EN);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_TIE |
+ CCSR_SSI_SIER_TFE0_EN, 0);
+ else
+ write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_RIE |
+ CCSR_SSI_SIER_RFF0_EN, 0);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ write_ssi(CCSR_SSI_SOR_TX_CLR, &ssi->sor);
+ else
+ write_ssi(CCSR_SSI_SOR_RX_CLR, &ssi->sor);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops fsl_ssi_ac97_dai_ops = {
+ .startup = fsl_ssi_startup,
+ .shutdown = fsl_ssi_shutdown,
+ .trigger = fsl_ssi_ac97_trigger,
+};
+
+static struct snd_soc_dai_driver fsl_ssi_ac97_dai = {
+ .ac97_control = 1,
+ .playback = {
+ .stream_name = "AC97 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "AC97 Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &fsl_ssi_ac97_dai_ops,
+};
+
+
+static struct fsl_ssi_private *fsl_ac97_data;
+
+static void fsl_ssi_ac97_init(void)
+{
+ fsl_ssi_setup(fsl_ac97_data);
+}
+
+void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ struct ccsr_ssi *ssi = fsl_ac97_data->ssi;
+ unsigned int lreg;
+ unsigned int lval;
+
+ if (reg > 0x7f)
+ return;
+
+
+ lreg = reg << 12;
+ write_ssi(lreg, &ssi->sacadd);
+
+ lval = val << 4;
+ write_ssi(lval , &ssi->sacdat);
+
+ write_ssi_mask(&ssi->sacnt, CCSR_SSI_SACNT_RDWR_MASK,
+ CCSR_SSI_SACNT_WR);
+ udelay(100);
+}
+
+unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct ccsr_ssi *ssi = fsl_ac97_data->ssi;
+
+ unsigned short val = -1;
+ unsigned int lreg;
+
+ lreg = (reg & 0x7f) << 12;
+ write_ssi(lreg, &ssi->sacadd);
+ write_ssi_mask(&ssi->sacnt, CCSR_SSI_SACNT_RDWR_MASK,
+ CCSR_SSI_SACNT_RD);
+
+ udelay(100);
+
+ val = (read_ssi(&ssi->sacdat) >> 4) & 0xffff;
+
+ return val;
+}
+
+static struct snd_ac97_bus_ops fsl_ssi_ac97_ops = {
+ .read = fsl_ssi_ac97_read,
+ .write = fsl_ssi_ac97_write,
+};
+
/* Show the statistics of a flag only if its interrupt is enabled. The
* compiler will optimze this code to a no-op if the interrupt is not
* enabled.
@@ -663,6 +871,7 @@
struct resource res;
char name[64];
bool shared;
+ bool ac97 = false;
/* SSIs that are not connected on the board should have a
* status = "disabled"
@@ -673,14 +882,20 @@
/* We only support the SSI in "I2S Slave" mode */
sprop = of_get_property(np, "fsl,mode", NULL);
- if (!sprop || strcmp(sprop, "i2s-slave")) {
+ if (!sprop) {
+ dev_err(&pdev->dev, "fsl,mode property is necessary\n");
+ return -EINVAL;
+ }
+ if (!strcmp(sprop, "ac97-slave")) {
+ ac97 = true;
+ } else if (strcmp(sprop, "i2s-slave")) {
dev_notice(&pdev->dev, "mode %s is unsupported\n", sprop);
return -ENODEV;
}
/* The DAI name is the last part of the full name of the node. */
p = strrchr(np->full_name, '/') + 1;
- ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + strlen(p),
+ ssi_private = devm_kzalloc(&pdev->dev, sizeof(*ssi_private) + strlen(p),
GFP_KERNEL);
if (!ssi_private) {
dev_err(&pdev->dev, "could not allocate DAI object\n");
@@ -689,38 +904,41 @@
strcpy(ssi_private->name, p);
- /* Initialize this copy of the CPU DAI driver structure */
- memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template,
- sizeof(fsl_ssi_dai_template));
+ ssi_private->use_dma = !of_property_read_bool(np,
+ "fsl,fiq-stream-filter");
+
+ if (ac97) {
+ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_ac97_dai,
+ sizeof(fsl_ssi_ac97_dai));
+
+ fsl_ac97_data = ssi_private;
+ ssi_private->imx_ac97 = true;
+
+ snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev);
+ } else {
+ /* Initialize this copy of the CPU DAI driver structure */
+ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template,
+ sizeof(fsl_ssi_dai_template));
+ }
ssi_private->cpu_dai_drv.name = ssi_private->name;
/* Get the addresses and IRQ */
ret = of_address_to_resource(np, 0, &res);
if (ret) {
dev_err(&pdev->dev, "could not determine device resources\n");
- goto error_kmalloc;
+ return ret;
}
ssi_private->ssi = of_iomap(np, 0);
if (!ssi_private->ssi) {
dev_err(&pdev->dev, "could not map device resources\n");
- ret = -ENOMEM;
- goto error_kmalloc;
+ return -ENOMEM;
}
ssi_private->ssi_phys = res.start;
ssi_private->irq = irq_of_parse_and_map(np, 0);
if (ssi_private->irq == NO_IRQ) {
dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
- ret = -ENXIO;
- goto error_iomap;
- }
-
- /* The 'name' should not have any slashes in it. */
- ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name,
- ssi_private);
- if (ret < 0) {
- dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq);
- goto error_irqmap;
+ return -ENXIO;
}
/* Are the RX and the TX clocks locked? */
@@ -739,13 +957,18 @@
u32 dma_events[2];
ssi_private->ssi_on_imx = true;
- ssi_private->clk = clk_get(&pdev->dev, NULL);
+ ssi_private->clk = devm_clk_get(&pdev->dev, NULL);
if (IS_ERR(ssi_private->clk)) {
ret = PTR_ERR(ssi_private->clk);
dev_err(&pdev->dev, "could not get clock: %d\n", ret);
- goto error_irq;
+ goto error_irqmap;
}
- clk_prepare_enable(ssi_private->clk);
+ ret = clk_prepare_enable(ssi_private->clk);
+ if (ret) {
+ dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n",
+ ret);
+ goto error_irqmap;
+ }
/*
* We have burstsize be "fifo_depth - 2" to match the SSI
@@ -763,24 +986,38 @@
&ssi_private->filter_data_tx;
ssi_private->dma_params_rx.filter_data =
&ssi_private->filter_data_rx;
- /*
- * TODO: This is a temporary solution and should be changed
- * to use generic DMA binding later when the helplers get in.
- */
- ret = of_property_read_u32_array(pdev->dev.of_node,
+ if (!of_property_read_bool(pdev->dev.of_node, "dmas") &&
+ ssi_private->use_dma) {
+ /*
+ * FIXME: This is a temporary solution until all
+ * necessary dma drivers support the generic dma
+ * bindings.
+ */
+ ret = of_property_read_u32_array(pdev->dev.of_node,
"fsl,ssi-dma-events", dma_events, 2);
- if (ret) {
- dev_err(&pdev->dev, "could not get dma events\n");
- goto error_clk;
+ if (ret && ssi_private->use_dma) {
+ dev_err(&pdev->dev, "could not get dma events but fsl-ssi is configured to use DMA\n");
+ goto error_clk;
+ }
}
shared = of_device_is_compatible(of_get_parent(np),
"fsl,spba-bus");
imx_pcm_dma_params_init_data(&ssi_private->filter_data_tx,
- dma_events[0], shared);
+ dma_events[0], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI);
imx_pcm_dma_params_init_data(&ssi_private->filter_data_rx,
- dma_events[1], shared);
+ dma_events[1], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI);
+ } else if (ssi_private->use_dma) {
+ /* The 'name' should not have any slashes in it. */
+ ret = devm_request_irq(&pdev->dev, ssi_private->irq,
+ fsl_ssi_isr, 0, ssi_private->name,
+ ssi_private);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "could not claim irq %u\n",
+ ssi_private->irq);
+ goto error_irqmap;
+ }
}
/* Initialize the the device_attribute structure */
@@ -794,7 +1031,7 @@
if (ret) {
dev_err(&pdev->dev, "could not create sysfs %s file\n",
ssi_private->dev_attr.attr.name);
- goto error_irq;
+ goto error_clk;
}
/* Register with ASoC */
@@ -808,9 +1045,30 @@
}
if (ssi_private->ssi_on_imx) {
- ret = imx_pcm_dma_init(pdev);
- if (ret)
- goto error_dev;
+ if (!ssi_private->use_dma) {
+
+ /*
+ * Some boards use an incompatible codec. To get it
+ * working, we are using imx-fiq-pcm-audio, that
+ * can handle those codecs. DMA is not possible in this
+ * situation.
+ */
+
+ ssi_private->fiq_params.irq = ssi_private->irq;
+ ssi_private->fiq_params.base = ssi_private->ssi;
+ ssi_private->fiq_params.dma_params_rx =
+ &ssi_private->dma_params_rx;
+ ssi_private->fiq_params.dma_params_tx =
+ &ssi_private->dma_params_tx;
+
+ ret = imx_pcm_fiq_init(pdev, &ssi_private->fiq_params);
+ if (ret)
+ goto error_dev;
+ } else {
+ ret = imx_pcm_dma_init(pdev);
+ if (ret)
+ goto error_dev;
+ }
}
/*
@@ -845,6 +1103,9 @@
}
done:
+ if (ssi_private->imx_ac97)
+ fsl_ssi_ac97_init();
+
return 0;
error_dai:
@@ -853,27 +1114,15 @@
snd_soc_unregister_component(&pdev->dev);
error_dev:
- dev_set_drvdata(&pdev->dev, NULL);
device_remove_file(&pdev->dev, dev_attr);
error_clk:
- if (ssi_private->ssi_on_imx) {
+ if (ssi_private->ssi_on_imx)
clk_disable_unprepare(ssi_private->clk);
- clk_put(ssi_private->clk);
- }
-
-error_irq:
- free_irq(ssi_private->irq, ssi_private);
error_irqmap:
irq_dispose_mapping(ssi_private->irq);
-error_iomap:
- iounmap(ssi_private->ssi);
-
-error_kmalloc:
- kfree(ssi_private);
-
return ret;
}
@@ -883,19 +1132,14 @@
if (!ssi_private->new_binding)
platform_device_unregister(ssi_private->pdev);
- if (ssi_private->ssi_on_imx) {
+ if (ssi_private->ssi_on_imx)
imx_pcm_dma_exit(pdev);
- clk_disable_unprepare(ssi_private->clk);
- clk_put(ssi_private->clk);
- }
snd_soc_unregister_component(&pdev->dev);
- device_remove_file(&pdev->dev, &ssi_private->dev_attr);
-
- free_irq(ssi_private->irq, ssi_private);
- irq_dispose_mapping(ssi_private->irq);
-
- kfree(ssi_private);
dev_set_drvdata(&pdev->dev, NULL);
+ device_remove_file(&pdev->dev, &ssi_private->dev_attr);
+ if (ssi_private->ssi_on_imx)
+ clk_disable_unprepare(ssi_private->clk);
+ irq_dispose_mapping(ssi_private->irq);
return 0;
}
@@ -919,6 +1163,7 @@
module_platform_driver(fsl_ssi_driver);
+MODULE_ALIAS("platform:fsl-ssi-dai");
MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver");
MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index e260f1f..ab17381 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -73,8 +73,11 @@
if (!buf)
return -ENOMEM;
- if (audmux_clk)
- clk_prepare_enable(audmux_clk);
+ if (audmux_clk) {
+ ret = clk_prepare_enable(audmux_clk);
+ if (ret)
+ return ret;
+ }
ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port));
pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port));
@@ -224,14 +227,19 @@
int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr,
unsigned int pdcr)
{
+ int ret;
+
if (audmux_type != IMX31_AUDMUX)
return -EINVAL;
if (!audmux_base)
return -ENOSYS;
- if (audmux_clk)
- clk_prepare_enable(audmux_clk);
+ if (audmux_clk) {
+ ret = clk_prepare_enable(audmux_clk);
+ if (ret)
+ return ret;
+ }
writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port));
writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port));
@@ -243,6 +251,66 @@
}
EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port);
+static int imx_audmux_parse_dt_defaults(struct platform_device *pdev,
+ struct device_node *of_node)
+{
+ struct device_node *child;
+
+ for_each_available_child_of_node(of_node, child) {
+ unsigned int port;
+ unsigned int ptcr = 0;
+ unsigned int pdcr = 0;
+ unsigned int pcr = 0;
+ unsigned int val;
+ int ret;
+ int i = 0;
+
+ ret = of_property_read_u32(child, "fsl,audmux-port", &port);
+ if (ret) {
+ dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%s\"\n",
+ child->full_name);
+ continue;
+ }
+ if (!of_property_read_bool(child, "fsl,port-config")) {
+ dev_warn(&pdev->dev, "child node \"%s\" does not have property fsl,port-config\n",
+ child->full_name);
+ continue;
+ }
+
+ for (i = 0; (ret = of_property_read_u32_index(child,
+ "fsl,port-config", i, &val)) == 0;
+ ++i) {
+ if (audmux_type == IMX31_AUDMUX) {
+ if (i % 2)
+ pdcr |= val;
+ else
+ ptcr |= val;
+ } else {
+ pcr |= val;
+ }
+ }
+
+ if (ret != -EOVERFLOW) {
+ dev_err(&pdev->dev, "Failed to read u32 at index %d of child %s\n",
+ i, child->full_name);
+ continue;
+ }
+
+ if (audmux_type == IMX31_AUDMUX) {
+ if (i % 2) {
+ dev_err(&pdev->dev, "One pdcr value is missing in child node %s\n",
+ child->full_name);
+ continue;
+ }
+ imx_audmux_v2_configure_port(port, ptcr, pdcr);
+ } else {
+ imx_audmux_v1_configure_port(port, pcr);
+ }
+ }
+
+ return 0;
+}
+
static int imx_audmux_probe(struct platform_device *pdev)
{
struct resource *res;
@@ -267,6 +335,8 @@
if (audmux_type == IMX31_AUDMUX)
audmux_debugfs_init();
+ imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node);
+
return 0;
}
diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h
index b8ff44b..38a4209 100644
--- a/sound/soc/fsl/imx-audmux.h
+++ b/sound/soc/fsl/imx-audmux.h
@@ -1,57 +1,7 @@
#ifndef __IMX_AUDMUX_H
#define __IMX_AUDMUX_H
-#define MX27_AUDMUX_HPCR1_SSI0 0
-#define MX27_AUDMUX_HPCR2_SSI1 1
-#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2
-#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3
-#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4
-#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5
-
-#define MX31_AUDMUX_PORT1_SSI0 0
-#define MX31_AUDMUX_PORT2_SSI1 1
-#define MX31_AUDMUX_PORT3_SSI_PINS_3 2
-#define MX31_AUDMUX_PORT4_SSI_PINS_4 3
-#define MX31_AUDMUX_PORT5_SSI_PINS_5 4
-#define MX31_AUDMUX_PORT6_SSI_PINS_6 5
-#define MX31_AUDMUX_PORT7_SSI_PINS_7 6
-
-#define MX51_AUDMUX_PORT1_SSI0 0
-#define MX51_AUDMUX_PORT2_SSI1 1
-#define MX51_AUDMUX_PORT3 2
-#define MX51_AUDMUX_PORT4 3
-#define MX51_AUDMUX_PORT5 4
-#define MX51_AUDMUX_PORT6 5
-#define MX51_AUDMUX_PORT7 6
-
-/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */
-#define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff)
-#define IMX_AUDMUX_V1_PCR_INMEN (1 << 8)
-#define IMX_AUDMUX_V1_PCR_TXRXEN (1 << 10)
-#define IMX_AUDMUX_V1_PCR_SYN (1 << 12)
-#define IMX_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13)
-#define IMX_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20)
-#define IMX_AUDMUX_V1_PCR_RCLKDIR (1 << 24)
-#define IMX_AUDMUX_V1_PCR_RFSDIR (1 << 25)
-#define IMX_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26)
-#define IMX_AUDMUX_V1_PCR_TCLKDIR (1 << 30)
-#define IMX_AUDMUX_V1_PCR_TFSDIR (1 << 31)
-
-/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */
-#define IMX_AUDMUX_V2_PTCR_TFSDIR (1 << 31)
-#define IMX_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27)
-#define IMX_AUDMUX_V2_PTCR_TCLKDIR (1 << 26)
-#define IMX_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22)
-#define IMX_AUDMUX_V2_PTCR_RFSDIR (1 << 21)
-#define IMX_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17)
-#define IMX_AUDMUX_V2_PTCR_RCLKDIR (1 << 16)
-#define IMX_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12)
-#define IMX_AUDMUX_V2_PTCR_SYN (1 << 11)
-
-#define IMX_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13)
-#define IMX_AUDMUX_V2_PDCR_TXRXEN (1 << 12)
-#define IMX_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8)
-#define IMX_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff)
+#include <dt-bindings/sound/fsl-imx-audmux.h>
int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr);
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index 9df173c..a3d60d4 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -90,6 +90,7 @@
static struct snd_soc_card imx_mc13783 = {
.name = "imx_mc13783",
+ .owner = THIS_MODULE,
.dai_link = imx_mc13783_dai_mc13783,
.num_links = ARRAY_SIZE(imx_mc13783_dai_mc13783),
.dapm_widgets = imx_mc13783_widget,
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index fde4d2e..4dc1296 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -14,6 +14,7 @@
#include <linux/platform_device.h>
#include <linux/dmaengine.h>
#include <linux/types.h>
+#include <linux/module.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -64,7 +65,6 @@
{
return snd_dmaengine_pcm_register(&pdev->dev, &imx_dmaengine_pcm_config,
SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
- SND_DMAENGINE_PCM_FLAG_NO_DT |
SND_DMAENGINE_PCM_FLAG_COMPAT);
}
EXPORT_SYMBOL_GPL(imx_pcm_dma_init);
@@ -74,3 +74,5 @@
snd_dmaengine_pcm_unregister(&pdev->dev);
}
EXPORT_SYMBOL_GPL(imx_pcm_dma_exit);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index 310d902..34043c5 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -22,6 +22,7 @@
#include <linux/slab.h>
#include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
#include <sound/initval.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -32,6 +33,7 @@
#include <linux/platform_data/asoc-imx-ssi.h>
#include "imx-ssi.h"
+#include "imx-pcm.h"
struct imx_pcm_runtime_data {
unsigned int period;
@@ -366,9 +368,9 @@
.pcm_free = imx_pcm_fiq_free,
};
-int imx_pcm_fiq_init(struct platform_device *pdev)
+int imx_pcm_fiq_init(struct platform_device *pdev,
+ struct imx_pcm_fiq_params *params)
{
- struct imx_ssi *ssi = platform_get_drvdata(pdev);
int ret;
ret = claim_fiq(&fh);
@@ -377,15 +379,15 @@
return ret;
}
- mxc_set_irq_fiq(ssi->irq, 1);
- ssi_irq = ssi->irq;
+ mxc_set_irq_fiq(params->irq, 1);
+ ssi_irq = params->irq;
- imx_pcm_fiq = ssi->irq;
+ imx_pcm_fiq = params->irq;
- imx_ssi_fiq_base = (unsigned long)ssi->base;
+ imx_ssi_fiq_base = (unsigned long)params->base;
- ssi->dma_params_tx.maxburst = 4;
- ssi->dma_params_rx.maxburst = 6;
+ params->dma_params_tx->maxburst = 4;
+ params->dma_params_rx->maxburst = 6;
ret = snd_soc_register_platform(&pdev->dev, &imx_soc_platform_fiq);
if (ret)
@@ -406,3 +408,5 @@
snd_soc_unregister_platform(&pdev->dev);
}
EXPORT_SYMBOL_GPL(imx_pcm_fiq_exit);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h
index 67f656c..5d5b733 100644
--- a/sound/soc/fsl/imx-pcm.h
+++ b/sound/soc/fsl/imx-pcm.h
@@ -22,17 +22,23 @@
static inline void
imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data,
- int dma, bool shared)
+ int dma, enum sdma_peripheral_type peripheral_type)
{
dma_data->dma_request = dma;
dma_data->priority = DMA_PRIO_HIGH;
- if (shared)
- dma_data->peripheral_type = IMX_DMATYPE_SSI_SP;
- else
- dma_data->peripheral_type = IMX_DMATYPE_SSI;
+ dma_data->peripheral_type = peripheral_type;
}
-#ifdef CONFIG_SND_SOC_IMX_PCM_DMA
+struct imx_pcm_fiq_params {
+ int irq;
+ void __iomem *base;
+
+ /* Pointer to original ssi driver to setup tx rx sizes */
+ struct snd_dmaengine_dai_dma_data *dma_params_rx;
+ struct snd_dmaengine_dai_dma_data *dma_params_tx;
+};
+
+#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA)
int imx_pcm_dma_init(struct platform_device *pdev);
void imx_pcm_dma_exit(struct platform_device *pdev);
#else
@@ -46,11 +52,13 @@
}
#endif
-#ifdef CONFIG_SND_SOC_IMX_PCM_FIQ
-int imx_pcm_fiq_init(struct platform_device *pdev);
+#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_FIQ)
+int imx_pcm_fiq_init(struct platform_device *pdev,
+ struct imx_pcm_fiq_params *params);
void imx_pcm_fiq_exit(struct platform_device *pdev);
#else
-static inline int imx_pcm_fiq_init(struct platform_device *pdev)
+static inline int imx_pcm_fiq_init(struct platform_device *pdev,
+ struct imx_pcm_fiq_params *params)
{
return -ENODEV;
}
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index 3f726e4..389cbfa 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -129,8 +129,10 @@
}
data->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
- if (IS_ERR(data->codec_clk))
+ if (IS_ERR(data->codec_clk)) {
+ ret = PTR_ERR(data->codec_clk);
goto fail;
+ }
data->clk_frequency = clk_get_rate(data->codec_clk);
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
new file mode 100644
index 0000000..816013b
--- /dev/null
+++ b/sound/soc/fsl/imx-spdif.c
@@ -0,0 +1,148 @@
+/*
+ * Copyright (C) 2013 Freescale Semiconductor, Inc.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+
+struct imx_spdif_data {
+ struct snd_soc_dai_link dai[2];
+ struct snd_soc_card card;
+ struct platform_device *txdev;
+ struct platform_device *rxdev;
+};
+
+static int imx_spdif_audio_probe(struct platform_device *pdev)
+{
+ struct device_node *spdif_np, *np = pdev->dev.of_node;
+ struct imx_spdif_data *data;
+ int ret = 0, num_links = 0;
+
+ spdif_np = of_parse_phandle(np, "spdif-controller", 0);
+ if (!spdif_np) {
+ dev_err(&pdev->dev, "failed to find spdif-controller\n");
+ ret = -EINVAL;
+ goto end;
+ }
+
+ data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ dev_err(&pdev->dev, "failed to allocate memory\n");
+ ret = -ENOMEM;
+ goto end;
+ }
+
+ if (of_property_read_bool(np, "spdif-out")) {
+ data->dai[num_links].name = "S/PDIF TX";
+ data->dai[num_links].stream_name = "S/PDIF PCM Playback";
+ data->dai[num_links].codec_dai_name = "dit-hifi";
+ data->dai[num_links].codec_name = "spdif-dit";
+ data->dai[num_links].cpu_of_node = spdif_np;
+ data->dai[num_links].platform_of_node = spdif_np;
+ num_links++;
+
+ data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0);
+ if (IS_ERR(data->txdev)) {
+ ret = PTR_ERR(data->txdev);
+ dev_err(&pdev->dev, "register dit failed: %d\n", ret);
+ goto end;
+ }
+ }
+
+ if (of_property_read_bool(np, "spdif-in")) {
+ data->dai[num_links].name = "S/PDIF RX";
+ data->dai[num_links].stream_name = "S/PDIF PCM Capture";
+ data->dai[num_links].codec_dai_name = "dir-hifi";
+ data->dai[num_links].codec_name = "spdif-dir";
+ data->dai[num_links].cpu_of_node = spdif_np;
+ data->dai[num_links].platform_of_node = spdif_np;
+ num_links++;
+
+ data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0);
+ if (IS_ERR(data->rxdev)) {
+ ret = PTR_ERR(data->rxdev);
+ dev_err(&pdev->dev, "register dir failed: %d\n", ret);
+ goto error_dit;
+ }
+ }
+
+ if (!num_links) {
+ dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n");
+ goto error_dir;
+ }
+
+ data->card.dev = &pdev->dev;
+ data->card.num_links = num_links;
+ data->card.dai_link = data->dai;
+
+ ret = snd_soc_of_parse_card_name(&data->card, "model");
+ if (ret)
+ goto error_dir;
+
+ ret = snd_soc_register_card(&data->card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret);
+ goto error_dir;
+ }
+
+ platform_set_drvdata(pdev, data);
+
+ goto end;
+
+error_dir:
+ if (data->rxdev)
+ platform_device_unregister(data->rxdev);
+error_dit:
+ if (data->txdev)
+ platform_device_unregister(data->txdev);
+end:
+ if (spdif_np)
+ of_node_put(spdif_np);
+
+ return ret;
+}
+
+static int imx_spdif_audio_remove(struct platform_device *pdev)
+{
+ struct imx_spdif_data *data = platform_get_drvdata(pdev);
+
+ if (data->rxdev)
+ platform_device_unregister(data->rxdev);
+ if (data->txdev)
+ platform_device_unregister(data->txdev);
+
+ snd_soc_unregister_card(&data->card);
+
+ return 0;
+}
+
+static const struct of_device_id imx_spdif_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-spdif", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids);
+
+static struct platform_driver imx_spdif_driver = {
+ .driver = {
+ .name = "imx-spdif",
+ .owner = THIS_MODULE,
+ .of_match_table = imx_spdif_dt_ids,
+ },
+ .probe = imx_spdif_audio_probe,
+ .remove = imx_spdif_audio_remove,
+};
+
+module_platform_driver(imx_spdif_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-spdif");
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 51be377..f58bcd8 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -571,13 +571,13 @@
res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0");
if (res) {
imx_pcm_dma_params_init_data(&ssi->filter_data_tx, res->start,
- false);
+ IMX_DMATYPE_SSI);
}
res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0");
if (res) {
imx_pcm_dma_params_init_data(&ssi->filter_data_rx, res->start,
- false);
+ IMX_DMATYPE_SSI);
}
platform_set_drvdata(pdev, ssi);
@@ -595,7 +595,12 @@
goto failed_register;
}
- ret = imx_pcm_fiq_init(pdev);
+ ssi->fiq_params.irq = ssi->irq;
+ ssi->fiq_params.base = ssi->base;
+ ssi->fiq_params.dma_params_rx = &ssi->dma_params_rx;
+ ssi->fiq_params.dma_params_tx = &ssi->dma_params_tx;
+
+ ret = imx_pcm_fiq_init(pdev, &ssi->fiq_params);
if (ret)
goto failed_pcm_fiq;
diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h
index d5003ce..fb1616b 100644
--- a/sound/soc/fsl/imx-ssi.h
+++ b/sound/soc/fsl/imx-ssi.h
@@ -209,6 +209,7 @@
struct snd_dmaengine_dai_dma_data dma_params_tx;
struct imx_dma_data filter_data_tx;
struct imx_dma_data filter_data_rx;
+ struct imx_pcm_fiq_params fiq_params;
int enabled;
};
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 52a36a9..1d70e27 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -217,7 +217,8 @@
codec_dev = of_find_i2c_device_by_node(codec_np);
if (!codec_dev || !codec_dev->driver) {
dev_err(&pdev->dev, "failed to find codec platform device\n");
- return -EINVAL;
+ ret = -EINVAL;
+ goto fail;
}
data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index c62d715..78ed4a4 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -1,19 +1,15 @@
config SND_KIRKWOOD_SOC
- tristate "SoC Audio for the Marvell Kirkwood chip"
- depends on ARCH_KIRKWOOD
+ tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
+ depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST
help
Say Y or M if you want to add support for codecs attached to
the Kirkwood I2S interface. You will also need to select the
audio interfaces to support below.
-config SND_KIRKWOOD_SOC_I2S
- tristate
-
config SND_KIRKWOOD_SOC_OPENRD
tristate "SoC Audio support for Kirkwood Openrd Client"
- depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE)
+ depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST)
depends on I2C
- select SND_KIRKWOOD_SOC_I2S
select SND_SOC_CS42L51
help
Say Y if you want to add support for SoC audio on
@@ -21,8 +17,7 @@
config SND_KIRKWOOD_SOC_T5325
tristate "SoC Audio support for HP t5325"
- depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C
- select SND_KIRKWOOD_SOC_I2S
+ depends on SND_KIRKWOOD_SOC && (MACH_T5325 || COMPILE_TEST) && I2C
select SND_SOC_ALC5623
help
Say Y if you want to add support for SoC audio on
diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile
index 3e62ae9..9e78138 100644
--- a/sound/soc/kirkwood/Makefile
+++ b/sound/soc/kirkwood/Makefile
@@ -1,8 +1,6 @@
-snd-soc-kirkwood-objs := kirkwood-dma.o
-snd-soc-kirkwood-i2s-objs := kirkwood-i2s.o
+snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o
obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o
-obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o
snd-soc-openrd-objs := kirkwood-openrd.o
snd-soc-t5325-objs := kirkwood-t5325.o
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index a9f1453..b238434 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -33,11 +33,11 @@
SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE | \
SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE)
-struct kirkwood_dma_priv {
- struct snd_pcm_substream *play_stream;
- struct snd_pcm_substream *rec_stream;
- struct kirkwood_dma_data *data;
-};
+static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs)
+{
+ struct snd_soc_pcm_runtime *soc_runtime = subs->private_data;
+ return snd_soc_dai_get_drvdata(soc_runtime->cpu_dai);
+}
static struct snd_pcm_hardware kirkwood_dma_snd_hw = {
.info = (SNDRV_PCM_INFO_INTERLEAVED |
@@ -51,7 +51,7 @@
.rate_max = 384000,
.channels_min = 1,
.channels_max = 8,
- .buffer_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES * KIRKWOOD_SND_MAX_PERIODS,
+ .buffer_bytes_max = KIRKWOOD_SND_MAX_BUFFER_BYTES,
.period_bytes_min = KIRKWOOD_SND_MIN_PERIOD_BYTES,
.period_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES,
.periods_min = KIRKWOOD_SND_MIN_PERIODS,
@@ -63,8 +63,7 @@
static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id)
{
- struct kirkwood_dma_priv *prdata = dev_id;
- struct kirkwood_dma_data *priv = prdata->data;
+ struct kirkwood_dma_data *priv = dev_id;
unsigned long mask, status, cause;
mask = readl(priv->io + KIRKWOOD_INT_MASK);
@@ -89,10 +88,10 @@
writel(status, priv->io + KIRKWOOD_INT_CAUSE);
if (status & KIRKWOOD_INT_CAUSE_PLAY_BYTES)
- snd_pcm_period_elapsed(prdata->play_stream);
+ snd_pcm_period_elapsed(priv->substream_play);
if (status & KIRKWOOD_INT_CAUSE_REC_BYTES)
- snd_pcm_period_elapsed(prdata->rec_stream);
+ snd_pcm_period_elapsed(priv->substream_rec);
return IRQ_HANDLED;
}
@@ -126,15 +125,10 @@
{
int err;
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- struct snd_soc_platform *platform = soc_runtime->platform;
- struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai;
- struct kirkwood_dma_data *priv;
- struct kirkwood_dma_priv *prdata = snd_soc_platform_get_drvdata(platform);
+ struct kirkwood_dma_data *priv = kirkwood_priv(substream);
const struct mbus_dram_target_info *dram;
unsigned long addr;
- priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw);
/* Ensure that all constraints linked to dma burst are fulfilled */
@@ -157,21 +151,11 @@
if (err < 0)
return err;
- if (prdata == NULL) {
- prdata = kzalloc(sizeof(struct kirkwood_dma_priv), GFP_KERNEL);
- if (prdata == NULL)
- return -ENOMEM;
-
- prdata->data = priv;
-
+ if (!priv->substream_play && !priv->substream_rec) {
err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED,
- "kirkwood-i2s", prdata);
- if (err) {
- kfree(prdata);
+ "kirkwood-i2s", priv);
+ if (err)
return -EBUSY;
- }
-
- snd_soc_platform_set_drvdata(platform, prdata);
/*
* Enable Error interrupts. We're only ack'ing them but
@@ -183,11 +167,11 @@
dram = mv_mbus_dram_info();
addr = substream->dma_buffer.addr;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- prdata->play_stream = substream;
+ priv->substream_play = substream;
kirkwood_dma_conf_mbus_windows(priv->io,
KIRKWOOD_PLAYBACK_WIN, addr, dram);
} else {
- prdata->rec_stream = substream;
+ priv->substream_rec = substream;
kirkwood_dma_conf_mbus_windows(priv->io,
KIRKWOOD_RECORD_WIN, addr, dram);
}
@@ -197,27 +181,19 @@
static int kirkwood_dma_close(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai;
- struct snd_soc_platform *platform = soc_runtime->platform;
- struct kirkwood_dma_priv *prdata = snd_soc_platform_get_drvdata(platform);
- struct kirkwood_dma_data *priv;
+ struct kirkwood_dma_data *priv = kirkwood_priv(substream);
- priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
-
- if (!prdata || !priv)
+ if (!priv)
return 0;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- prdata->play_stream = NULL;
+ priv->substream_play = NULL;
else
- prdata->rec_stream = NULL;
+ priv->substream_rec = NULL;
- if (!prdata->play_stream && !prdata->rec_stream) {
+ if (!priv->substream_play && !priv->substream_rec) {
writel(0, priv->io + KIRKWOOD_ERR_MASK);
- free_irq(priv->irq, prdata);
- kfree(prdata);
- snd_soc_platform_set_drvdata(platform, NULL);
+ free_irq(priv->irq, priv);
}
return 0;
@@ -243,13 +219,9 @@
static int kirkwood_dma_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai;
- struct kirkwood_dma_data *priv;
+ struct kirkwood_dma_data *priv = kirkwood_priv(substream);
unsigned long size, count;
- priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
-
/* compute buffer size in term of "words" as requested in specs */
size = frames_to_bytes(runtime, runtime->buffer_size);
size = (size>>2)-1;
@@ -272,13 +244,9 @@
static snd_pcm_uframes_t kirkwood_dma_pointer(struct snd_pcm_substream
*substream)
{
- struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai;
- struct kirkwood_dma_data *priv;
+ struct kirkwood_dma_data *priv = kirkwood_priv(substream);
snd_pcm_uframes_t count;
- priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
-
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
count = bytes_to_frames(substream->runtime,
readl(priv->io + KIRKWOOD_PLAY_BYTE_COUNT));
@@ -366,36 +334,8 @@
}
}
-static struct snd_soc_platform_driver kirkwood_soc_platform = {
+struct snd_soc_platform_driver kirkwood_soc_platform = {
.ops = &kirkwood_dma_ops,
.pcm_new = kirkwood_dma_new,
.pcm_free = kirkwood_dma_free_dma_buffers,
};
-
-static int kirkwood_soc_platform_probe(struct platform_device *pdev)
-{
- return snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform);
-}
-
-static int kirkwood_soc_platform_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_platform(&pdev->dev);
- return 0;
-}
-
-static struct platform_driver kirkwood_pcm_driver = {
- .driver = {
- .name = "kirkwood-pcm-audio",
- .owner = THIS_MODULE,
- },
-
- .probe = kirkwood_soc_platform_probe,
- .remove = kirkwood_soc_platform_remove,
-};
-
-module_platform_driver(kirkwood_pcm_driver);
-
-MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
-MODULE_DESCRIPTION("Marvell Kirkwood Audio DMA module");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:kirkwood-pcm-audio");
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 4c9dad3..7fce340 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -22,13 +22,12 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <linux/platform_data/asoc-kirkwood.h>
+#include <linux/of.h>
+
#include "kirkwood.h"
-#define DRV_NAME "kirkwood-i2s"
+#define DRV_NAME "mvebu-audio"
-#define KIRKWOOD_I2S_RATES \
- (SNDRV_PCM_RATE_44100 | \
- SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
#define KIRKWOOD_I2S_FORMATS \
(SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE | \
@@ -105,14 +104,16 @@
uint32_t clks_ctrl;
if (rate == 44100 || rate == 48000 || rate == 96000) {
- /* use internal dco for supported rates */
+ /* use internal dco for the supported rates
+ * defined in kirkwood_i2s_dai */
dev_dbg(dai->dev, "%s: dco set rate = %lu\n",
__func__, rate);
kirkwood_set_dco(priv->io, rate);
clks_ctrl = KIRKWOOD_MCLK_SOURCE_DCO;
- } else if (!IS_ERR(priv->extclk)) {
- /* use optional external clk for other rates */
+ } else {
+ /* use the external clock for the other rates
+ * defined in kirkwood_i2s_dai_extclk */
dev_dbg(dai->dev, "%s: extclk set rate = %lu -> %lu\n",
__func__, rate, 256 * rate);
clk_set_rate(priv->extclk, 256 * rate);
@@ -199,8 +200,7 @@
ctl_play |= KIRKWOOD_PLAYCTL_MONO_OFF;
priv->ctl_play &= ~(KIRKWOOD_PLAYCTL_MONO_MASK |
- KIRKWOOD_PLAYCTL_I2S_EN |
- KIRKWOOD_PLAYCTL_SPDIF_EN |
+ KIRKWOOD_PLAYCTL_ENABLE_MASK |
KIRKWOOD_PLAYCTL_SIZE_MASK);
priv->ctl_play |= ctl_play;
} else {
@@ -244,8 +244,7 @@
case SNDRV_PCM_TRIGGER_START:
/* configure */
ctl = priv->ctl_play;
- value = ctl & ~(KIRKWOOD_PLAYCTL_I2S_EN |
- KIRKWOOD_PLAYCTL_SPDIF_EN);
+ value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK;
writel(value, priv->io + KIRKWOOD_PLAYCTL);
/* enable interrupts */
@@ -267,7 +266,7 @@
writel(value, priv->io + KIRKWOOD_INT_MASK);
/* disable all playbacks */
- ctl &= ~(KIRKWOOD_PLAYCTL_I2S_EN | KIRKWOOD_PLAYCTL_SPDIF_EN);
+ ctl &= ~KIRKWOOD_PLAYCTL_ENABLE_MASK;
writel(ctl, priv->io + KIRKWOOD_PLAYCTL);
break;
@@ -387,7 +386,7 @@
/* disable playback/record */
value = readl(priv->io + KIRKWOOD_PLAYCTL);
- value &= ~(KIRKWOOD_PLAYCTL_I2S_EN|KIRKWOOD_PLAYCTL_SPDIF_EN);
+ value &= ~KIRKWOOD_PLAYCTL_ENABLE_MASK;
writel(value, priv->io + KIRKWOOD_PLAYCTL);
value = readl(priv->io + KIRKWOOD_RECCTL);
@@ -398,11 +397,6 @@
}
-static int kirkwood_i2s_remove(struct snd_soc_dai *dai)
-{
- return 0;
-}
-
static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = {
.startup = kirkwood_i2s_startup,
.trigger = kirkwood_i2s_trigger,
@@ -413,17 +407,18 @@
static struct snd_soc_dai_driver kirkwood_i2s_dai = {
.probe = kirkwood_i2s_probe,
- .remove = kirkwood_i2s_remove,
.playback = {
.channels_min = 1,
.channels_max = 2,
- .rates = KIRKWOOD_I2S_RATES,
+ .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_96000,
.formats = KIRKWOOD_I2S_FORMATS,
},
.capture = {
.channels_min = 1,
.channels_max = 2,
- .rates = KIRKWOOD_I2S_RATES,
+ .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_96000,
.formats = KIRKWOOD_I2S_FORMATS,
},
.ops = &kirkwood_i2s_dai_ops,
@@ -431,7 +426,6 @@
static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = {
.probe = kirkwood_i2s_probe,
- .remove = kirkwood_i2s_remove,
.playback = {
.channels_min = 1,
.channels_max = 2,
@@ -461,6 +455,7 @@
struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai;
struct kirkwood_dma_data *priv;
struct resource *mem;
+ struct device_node *np = pdev->dev.of_node;
int err;
priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
@@ -481,14 +476,16 @@
return -ENXIO;
}
- if (!data) {
- dev_err(&pdev->dev, "no platform data ?!\n");
+ if (np) {
+ priv->burst = 128; /* might be 32 or 128 */
+ } else if (data) {
+ priv->burst = data->burst;
+ } else {
+ dev_err(&pdev->dev, "no DT nor platform data ?!\n");
return -EINVAL;
}
- priv->burst = data->burst;
-
- priv->clk = devm_clk_get(&pdev->dev, NULL);
+ priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL);
if (IS_ERR(priv->clk)) {
dev_err(&pdev->dev, "no clock\n");
return PTR_ERR(priv->clk);
@@ -498,10 +495,10 @@
if (err < 0)
return err;
- priv->extclk = clk_get(&pdev->dev, "extclk");
+ priv->extclk = devm_clk_get(&pdev->dev, "extclk");
if (!IS_ERR(priv->extclk)) {
if (priv->extclk == priv->clk) {
- clk_put(priv->extclk);
+ devm_clk_put(&pdev->dev, priv->extclk);
priv->extclk = ERR_PTR(-EINVAL);
} else {
dev_info(&pdev->dev, "found external clock\n");
@@ -515,7 +512,7 @@
priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24;
/* Select the burst size */
- if (data->burst == 32) {
+ if (priv->burst == 32) {
priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32;
priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32;
} else {
@@ -525,14 +522,22 @@
err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component,
soc_dai, 1);
- if (!err)
- return 0;
- dev_err(&pdev->dev, "snd_soc_register_component failed\n");
-
- if (!IS_ERR(priv->extclk)) {
- clk_disable_unprepare(priv->extclk);
- clk_put(priv->extclk);
+ if (err) {
+ dev_err(&pdev->dev, "snd_soc_register_component failed\n");
+ goto err_component;
}
+
+ err = snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform);
+ if (err) {
+ dev_err(&pdev->dev, "snd_soc_register_platform failed\n");
+ goto err_platform;
+ }
+ return 0;
+ err_platform:
+ snd_soc_unregister_component(&pdev->dev);
+ err_component:
+ if (!IS_ERR(priv->extclk))
+ clk_disable_unprepare(priv->extclk);
clk_disable_unprepare(priv->clk);
return err;
@@ -542,23 +547,31 @@
{
struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev);
+ snd_soc_unregister_platform(&pdev->dev);
snd_soc_unregister_component(&pdev->dev);
- if (!IS_ERR(priv->extclk)) {
+ if (!IS_ERR(priv->extclk))
clk_disable_unprepare(priv->extclk);
- clk_put(priv->extclk);
- }
clk_disable_unprepare(priv->clk);
return 0;
}
+#ifdef CONFIG_OF
+static struct of_device_id mvebu_audio_of_match[] = {
+ { .compatible = "marvell,mvebu-audio" },
+ { }
+};
+MODULE_DEVICE_TABLE(of, mvebu_audio_of_match);
+#endif
+
static struct platform_driver kirkwood_i2s_driver = {
.probe = kirkwood_i2s_dev_probe,
.remove = kirkwood_i2s_dev_remove,
.driver = {
.name = DRV_NAME,
.owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(mvebu_audio_of_match),
},
};
@@ -568,4 +581,4 @@
MODULE_AUTHOR("Arnaud Patard, <arnaud.patard@rtp-net.org>");
MODULE_DESCRIPTION("Kirkwood I2S SoC Interface");
MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:kirkwood-i2s");
+MODULE_ALIAS("platform:mvebu-audio");
diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c
index b979c71..025be0e 100644
--- a/sound/soc/kirkwood/kirkwood-openrd.c
+++ b/sound/soc/kirkwood/kirkwood-openrd.c
@@ -16,9 +16,7 @@
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/soc.h>
-#include <mach/kirkwood.h>
#include <linux/platform_data/asoc-kirkwood.h>
-#include <asm/mach-types.h>
#include "../codecs/cs42l51.h"
static int openrd_client_hw_params(struct snd_pcm_substream *substream,
@@ -54,8 +52,8 @@
{
.name = "CS42L51",
.stream_name = "CS42L51 HiFi",
- .cpu_dai_name = "kirkwood-i2s",
- .platform_name = "kirkwood-pcm-audio",
+ .cpu_dai_name = "mvebu-audio",
+ .platform_name = "mvebu-audio",
.codec_dai_name = "cs42l51-hifi",
.codec_name = "cs42l51-codec.0-004a",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c
index 1d0ed6f..27545b0 100644
--- a/sound/soc/kirkwood/kirkwood-t5325.c
+++ b/sound/soc/kirkwood/kirkwood-t5325.c
@@ -15,9 +15,7 @@
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/soc.h>
-#include <mach/kirkwood.h>
#include <linux/platform_data/asoc-kirkwood.h>
-#include <asm/mach-types.h>
#include "../codecs/alc5623.h"
static int t5325_hw_params(struct snd_pcm_substream *substream,
@@ -70,8 +68,8 @@
{
.name = "ALC5621",
.stream_name = "ALC5621 HiFi",
- .cpu_dai_name = "kirkwood-i2s",
- .platform_name = "kirkwood-pcm-audio",
+ .cpu_dai_name = "mvebu-audio",
+ .platform_name = "mvebu-audio",
.codec_dai_name = "alc5621-hifi",
.codec_name = "alc562x-codec.0-001a",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h
index 4d92637..f8e1ccc 100644
--- a/sound/soc/kirkwood/kirkwood.h
+++ b/sound/soc/kirkwood/kirkwood.h
@@ -54,7 +54,7 @@
#define KIRKWOOD_PLAYCTL_MONO_OFF (0<<5)
#define KIRKWOOD_PLAYCTL_I2S_MUTE (1<<7)
#define KIRKWOOD_PLAYCTL_SPDIF_EN (1<<4)
-#define KIRKWOOD_PLAYCTL_I2S_EN (1<<3)
+#define KIRKWOOD_PLAYCTL_I2S_EN (1<<3)
#define KIRKWOOD_PLAYCTL_SIZE_MASK (7<<0)
#define KIRKWOOD_PLAYCTL_SIZE_16 (7<<0)
#define KIRKWOOD_PLAYCTL_SIZE_16_C (3<<0)
@@ -62,6 +62,9 @@
#define KIRKWOOD_PLAYCTL_SIZE_24 (1<<0)
#define KIRKWOOD_PLAYCTL_SIZE_32 (0<<0)
+#define KIRKWOOD_PLAYCTL_ENABLE_MASK (KIRKWOOD_PLAYCTL_SPDIF_EN | \
+ KIRKWOOD_PLAYCTL_I2S_EN)
+
#define KIRKWOOD_PLAY_BUF_ADDR 0x1104
#define KIRKWOOD_PLAY_BUF_SIZE 0x1108
#define KIRKWOOD_PLAY_BYTE_COUNT 0x110C
@@ -122,6 +125,8 @@
#define KIRKWOOD_SND_MAX_PERIODS 16
#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x4000
#define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x4000
+#define KIRKWOOD_SND_MAX_BUFFER_BYTES (KIRKWOOD_SND_MAX_PERIOD_BYTES \
+ * KIRKWOOD_SND_MAX_PERIODS)
struct kirkwood_dma_data {
void __iomem *io;
@@ -129,8 +134,12 @@
struct clk *extclk;
uint32_t ctl_play;
uint32_t ctl_rec;
+ struct snd_pcm_substream *substream_play;
+ struct snd_pcm_substream *substream_rec;
int irq;
int burst;
};
+extern struct snd_soc_platform_driver kirkwood_soc_platform;
+
#endif
diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig
index 78d321c..219235c 100644
--- a/sound/soc/mxs/Kconfig
+++ b/sound/soc/mxs/Kconfig
@@ -1,6 +1,7 @@
menuconfig SND_MXS_SOC
tristate "SoC Audio for Freescale MXS CPUs"
- depends on ARCH_MXS
+ depends on ARCH_MXS || COMPILE_TEST
+ depends on COMMON_CLK
select SND_SOC_GENERIC_DMAENGINE_PCM
help
Say Y or M if you want to add support for codecs attached to
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index 54511c5..b56b8a0 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -31,7 +31,6 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
-#include <asm/mach-types.h>
#include "mxs-saif.h"
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 1b134d7..ce084eb 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -25,7 +25,6 @@
#include <sound/soc.h>
#include <sound/jack.h>
#include <sound/soc-dapm.h>
-#include <asm/mach-types.h>
#include "../codecs/sgtl5000.h"
#include "mxs-saif.h"
@@ -51,18 +50,27 @@
}
/* Sgtl5000 sysclk should be >= 8MHz and <= 27M */
- if (mclk < 8000000 || mclk > 27000000)
+ if (mclk < 8000000 || mclk > 27000000) {
+ dev_err(codec_dai->dev, "Invalid mclk frequency: %u.%03uMHz\n",
+ mclk / 1000000, mclk / 1000 % 1000);
return -EINVAL;
+ }
/* Set SGTL5000's SYSCLK (provided by SAIF MCLK) */
ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk, 0);
- if (ret)
+ if (ret) {
+ dev_err(codec_dai->dev, "Failed to set sysclk to %u.%03uMHz\n",
+ mclk / 1000000, mclk / 1000 % 1000);
return ret;
+ }
/* The SAIF MCLK should be the same as SGTL5000_SYSCLK */
ret = snd_soc_dai_set_sysclk(cpu_dai, MXS_SAIF_MCLK, mclk, 0);
- if (ret)
+ if (ret) {
+ dev_err(cpu_dai->dev, "Failed to set sysclk to %u.%03uMHz\n",
+ mclk / 1000000, mclk / 1000 % 1000);
return ret;
+ }
/* set codec to slave mode */
dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
@@ -70,13 +78,19 @@
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, dai_format);
- if (ret)
+ if (ret) {
+ dev_err(codec_dai->dev, "Failed to set dai format to %08x\n",
+ dai_format);
return ret;
+ }
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, dai_format);
- if (ret)
+ if (ret) {
+ dev_err(cpu_dai->dev, "Failed to set dai format to %08x\n",
+ dai_format);
return ret;
+ }
return 0;
}
@@ -154,8 +168,10 @@
* should be >= 8MHz and <= 27M.
*/
ret = mxs_saif_get_mclk(0, 44100 * 256, 44100);
- if (ret)
+ if (ret) {
+ dev_err(&pdev->dev, "failed to get mclk\n");
return ret;
+ }
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index f4c2417..8987bf9 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -333,9 +333,6 @@
spin_lock_init(&nuc900_audio->lock);
nuc900_audio->res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!nuc900_audio->res)
- return ret;
-
nuc900_audio->mmio = devm_ioremap_resource(&pdev->dev,
nuc900_audio->res);
if (IS_ERR(nuc900_audio->mmio))
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 9f5d55e..daa78a0 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -1,7 +1,7 @@
config SND_OMAP_SOC
tristate "SoC Audio for the Texas Instruments OMAP chips"
- depends on ARCH_OMAP && DMA_OMAP
- select SND_SOC_DMAENGINE_PCM
+ depends on (ARCH_OMAP && DMA_OMAP) || (ARCH_ARM && COMPILE_TEST)
+ select SND_DMAENGINE_PCM
config SND_OMAP_SOC_DMIC
tristate
@@ -26,7 +26,7 @@
config SND_OMAP_SOC_RX51
tristate "SoC Audio support for Nokia RX-51"
- depends on SND_OMAP_SOC && MACH_NOKIA_RX51
+ depends on SND_OMAP_SOC && ARCH_ARM && (MACH_NOKIA_RX51 || COMPILE_TEST)
select SND_OMAP_SOC_MCBSP
select SND_SOC_TLV320AIC3X
select SND_SOC_TPA6130A2
@@ -87,7 +87,7 @@
config SND_OMAP_SOC_OMAP_ABE_TWL6040
tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec"
- depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4
+ depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST)
select SND_OMAP_SOC_DMIC
select SND_OMAP_SOC_MCPDM
select SND_SOC_TWL6040
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index 361e4c0..83433fd 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -781,7 +781,7 @@
unsigned long val; \
int status; \
\
- status = strict_strtoul(buf, 0, &val); \
+ status = kstrtoul(buf, 0, &val); \
if (status) \
return status; \
\
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 70cd5c7..ebb1390 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -23,7 +23,6 @@
#include <linux/clk.h>
#include <linux/platform_device.h>
#include <linux/mfd/twl6040.h>
-#include <linux/platform_data/omap-abe-twl6040.h>
#include <linux/module.h>
#include <linux/of.h>
@@ -166,19 +165,10 @@
{"AFMR", NULL, "Line In"},
};
-static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm,
- int connected, char *pin)
-{
- if (!connected)
- snd_soc_dapm_disable_pin(dapm, pin);
-}
-
static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_card *card = codec->card;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
int hs_trim;
int ret = 0;
@@ -203,24 +193,6 @@
twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
}
- /*
- * NULL pdata means we booted with DT. In this case the routing is
- * provided and the card is fully routed, no need to mark pins.
- */
- if (!pdata)
- return ret;
-
- /* Disable not connected paths if not used */
- twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone");
- twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
- twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
- twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
- twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator");
- twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
- twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
- twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
- twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In");
-
return ret;
}
@@ -274,13 +246,18 @@
static int omap_abe_probe(struct platform_device *pdev)
{
- struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev);
struct device_node *node = pdev->dev.of_node;
struct snd_soc_card *card = &omap_abe_card;
+ struct device_node *dai_node;
struct abe_twl6040 *priv;
int num_links = 0;
int ret = 0;
+ if (!node) {
+ dev_err(&pdev->dev, "of node is missing.\n");
+ return -ENODEV;
+ }
+
card->dev = &pdev->dev;
priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
@@ -289,78 +266,50 @@
priv->dmic_codec_dev = ERR_PTR(-EINVAL);
- if (node) {
- struct device_node *dai_node;
-
- if (snd_soc_of_parse_card_name(card, "ti,model")) {
- dev_err(&pdev->dev, "Card name is not provided\n");
- return -ENODEV;
- }
-
- ret = snd_soc_of_parse_audio_routing(card,
- "ti,audio-routing");
- if (ret) {
- dev_err(&pdev->dev,
- "Error while parsing DAPM routing\n");
- return ret;
- }
-
- dai_node = of_parse_phandle(node, "ti,mcpdm", 0);
- if (!dai_node) {
- dev_err(&pdev->dev, "McPDM node is not provided\n");
- return -EINVAL;
- }
- abe_twl6040_dai_links[0].cpu_dai_name = NULL;
- abe_twl6040_dai_links[0].cpu_of_node = dai_node;
-
- dai_node = of_parse_phandle(node, "ti,dmic", 0);
- if (dai_node) {
- num_links = 2;
- abe_twl6040_dai_links[1].cpu_dai_name = NULL;
- abe_twl6040_dai_links[1].cpu_of_node = dai_node;
-
- priv->dmic_codec_dev = platform_device_register_simple(
- "dmic-codec", -1, NULL, 0);
- if (IS_ERR(priv->dmic_codec_dev)) {
- dev_err(&pdev->dev,
- "Can't instantiate dmic-codec\n");
- return PTR_ERR(priv->dmic_codec_dev);
- }
- } else {
- num_links = 1;
- }
-
- priv->jack_detection = of_property_read_bool(node,
- "ti,jack-detection");
- of_property_read_u32(node, "ti,mclk-freq",
- &priv->mclk_freq);
- if (!priv->mclk_freq) {
- dev_err(&pdev->dev, "MCLK frequency not provided\n");
- ret = -EINVAL;
- goto err_unregister;
- }
-
- omap_abe_card.fully_routed = 1;
- } else if (pdata) {
- if (pdata->card_name) {
- card->name = pdata->card_name;
- } else {
- dev_err(&pdev->dev, "Card name is not provided\n");
- return -ENODEV;
- }
-
- if (pdata->has_dmic)
- num_links = 2;
- else
- num_links = 1;
-
- priv->jack_detection = pdata->jack_detection;
- priv->mclk_freq = pdata->mclk_freq;
- } else {
- dev_err(&pdev->dev, "Missing pdata\n");
+ if (snd_soc_of_parse_card_name(card, "ti,model")) {
+ dev_err(&pdev->dev, "Card name is not provided\n");
return -ENODEV;
}
+ ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing");
+ if (ret) {
+ dev_err(&pdev->dev, "Error while parsing DAPM routing\n");
+ return ret;
+ }
+
+ dai_node = of_parse_phandle(node, "ti,mcpdm", 0);
+ if (!dai_node) {
+ dev_err(&pdev->dev, "McPDM node is not provided\n");
+ return -EINVAL;
+ }
+ abe_twl6040_dai_links[0].cpu_dai_name = NULL;
+ abe_twl6040_dai_links[0].cpu_of_node = dai_node;
+
+ dai_node = of_parse_phandle(node, "ti,dmic", 0);
+ if (dai_node) {
+ num_links = 2;
+ abe_twl6040_dai_links[1].cpu_dai_name = NULL;
+ abe_twl6040_dai_links[1].cpu_of_node = dai_node;
+
+ priv->dmic_codec_dev = platform_device_register_simple(
+ "dmic-codec", -1, NULL, 0);
+ if (IS_ERR(priv->dmic_codec_dev)) {
+ dev_err(&pdev->dev, "Can't instantiate dmic-codec\n");
+ return PTR_ERR(priv->dmic_codec_dev);
+ }
+ } else {
+ num_links = 1;
+ }
+
+ priv->jack_detection = of_property_read_bool(node, "ti,jack-detection");
+ of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq);
+ if (!priv->mclk_freq) {
+ dev_err(&pdev->dev, "MCLK frequency not provided\n");
+ ret = -EINVAL;
+ goto err_unregister;
+ }
+
+ card->fully_routed = 1;
if (!priv->mclk_freq) {
dev_err(&pdev->dev, "MCLK frequency missing\n");
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 4db1f8e..12e566b 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -480,15 +480,12 @@
dmic->dma_data.filter_data = "up_link";
res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
- if (!res) {
- dev_err(dmic->dev, "invalid memory resource\n");
- ret = -ENODEV;
+ dmic->io_base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(dmic->io_base)) {
+ ret = PTR_ERR(dmic->io_base);
goto err_put_clk;
}
- dmic->io_base = devm_ioremap_resource(&pdev->dev, res);
- if (IS_ERR(dmic->io_base))
- return PTR_ERR(dmic->io_base);
ret = snd_soc_register_component(&pdev->dev, &omap_dmic_component,
&omap_dmic_dai, 1);
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 7483efb..6c19bba 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -433,6 +433,11 @@
/* Sample rate generator drives the FS */
regs->srgr2 |= FSGM;
break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ /* McBSP slave. FS clock as output */
+ regs->srgr2 |= FSGM;
+ regs->pcr0 |= FSXM;
+ break;
case SND_SOC_DAIFMT_CBM_CFM:
/* McBSP slave */
break;
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index a49dc52..90d2a7c 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -480,9 +480,6 @@
mcpdm->dma_data[1].filter_data = "up_link";
res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
- if (res == NULL)
- return -ENOMEM;
-
mcpdm->io_base = devm_ioremap_resource(&pdev->dev, res);
if (IS_ERR(mcpdm->io_base))
return PTR_ERR(mcpdm->io_base);
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index b358094..4db74a0 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -11,7 +11,7 @@
config SND_MMP_SOC
bool "Soc Audio for Marvell MMP chips"
depends on ARCH_MMP
- select SND_SOC_DMAENGINE_PCM
+ select SND_DMAENGINE_PCM
select SND_ARM
help
Say Y if you want to add support for codecs attached to
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
index 4ad7609..5b7d969 100644
--- a/sound/soc/pxa/brownstone.c
+++ b/sound/soc/pxa/brownstone.c
@@ -129,6 +129,7 @@
/* audio machine driver */
static struct snd_soc_card brownstone = {
.name = "brownstone",
+ .owner = THIS_MODULE,
.dai_link = brownstone_wm8994_dai,
.num_links = ARRAY_SIZE(brownstone_wm8994_dai),
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 97b711e..bbea778 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -56,8 +56,6 @@
#include "pxa2xx-ac97.h"
#include "../codecs/wm9713.h"
-#define ARRAY_AND_SIZE(x) (x), ARRAY_SIZE(x)
-
#define AC97_GPIO_PULL 0x58
/* Use GPIO8 for rear speaker amplifier */
@@ -133,10 +131,11 @@
unsigned short reg;
/* Add mioa701 specific widgets */
- snd_soc_dapm_new_controls(dapm, ARRAY_AND_SIZE(mioa701_dapm_widgets));
+ snd_soc_dapm_new_controls(dapm, mioa701_dapm_widgets,
+ ARRAY_SIZE(mioa701_dapm_widgets));
/* Set up mioa701 specific audio path audio_mapnects */
- snd_soc_dapm_add_routes(dapm, ARRAY_AND_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
/* Prepare GPIO8 for rear speaker amplifier */
reg = codec->driver->read(codec, AC97_GPIO_CFG);
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
index 5d57e07..8235e23 100644
--- a/sound/soc/pxa/mmp-pcm.c
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -17,6 +17,7 @@
#include <linux/dmaengine.h>
#include <linux/platform_data/dma-mmp_tdma.h>
#include <linux/platform_data/mmp_audio.h>
+
#include <sound/pxa2xx-lib.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -67,7 +68,7 @@
{
struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct pxa2xx_pcm_dma_params *dma_params;
+ struct snd_dmaengine_dai_dma_data *dma_params;
struct dma_slave_config slave_config;
int ret;
@@ -80,10 +81,10 @@
return ret;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- slave_config.dst_addr = dma_params->dev_addr;
+ slave_config.dst_addr = dma_params->addr;
slave_config.dst_maxburst = 4;
} else {
- slave_config.src_addr = dma_params->dev_addr;
+ slave_config.src_addr = dma_params->addr;
slave_config.src_maxburst = 4;
}
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
index 62142ce..41752a5 100644
--- a/sound/soc/pxa/mmp-sspa.c
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -27,12 +27,15 @@
#include <linux/slab.h>
#include <linux/pxa2xx_ssp.h>
#include <linux/io.h>
+#include <linux/dmaengine.h>
+
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/initval.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
#include "mmp-sspa.h"
/*
@@ -40,7 +43,7 @@
*/
struct sspa_priv {
struct ssp_device *sspa;
- struct pxa2xx_pcm_dma_params *dma_params;
+ struct snd_dmaengine_dai_dma_data *dma_params;
struct clk *audio_clk;
struct clk *sysclk;
int dai_fmt;
@@ -266,7 +269,7 @@
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
struct ssp_device *sspa = sspa_priv->sspa;
- struct pxa2xx_pcm_dma_params *dma_params;
+ struct snd_dmaengine_dai_dma_data *dma_params;
u32 sspa_ctrl;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -309,7 +312,7 @@
}
dma_params = &sspa_priv->dma_params[substream->stream];
- dma_params->dev_addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ dma_params->addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
(sspa->phys_base + SSPA_TXD) :
(sspa->phys_base + SSPA_RXD);
snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params);
@@ -425,14 +428,12 @@
return -ENOMEM;
priv->dma_params = devm_kzalloc(&pdev->dev,
- 2 * sizeof(struct pxa2xx_pcm_dma_params), GFP_KERNEL);
+ 2 * sizeof(struct snd_dmaengine_dai_dma_data),
+ GFP_KERNEL);
if (priv->dma_params == NULL)
return -ENOMEM;
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (res == NULL)
- return -ENOMEM;
-
priv->sspa->mmio_base = devm_ioremap_resource(&pdev->dev, res);
if (IS_ERR(priv->sspa->mmio_base))
return PTR_ERR(priv->sspa->mmio_base);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 6f4dd75..a3119a0 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -21,6 +21,8 @@
#include <linux/clk.h>
#include <linux/io.h>
#include <linux/pxa2xx_ssp.h>
+#include <linux/of.h>
+#include <linux/dmaengine.h>
#include <asm/irq.h>
@@ -30,9 +32,9 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
#include <mach/hardware.h>
-#include <mach/dma.h>
#include "../../arm/pxa2xx-pcm.h"
#include "pxa-ssp.h"
@@ -79,27 +81,13 @@
__raw_writel(sscr0, ssp->mmio_base + SSCR0);
}
-struct pxa2xx_pcm_dma_data {
- struct pxa2xx_pcm_dma_params params;
- char name[20];
-};
-
static void pxa_ssp_set_dma_params(struct ssp_device *ssp, int width4,
- int out, struct pxa2xx_pcm_dma_params *dma_data)
+ int out, struct snd_dmaengine_dai_dma_data *dma)
{
- struct pxa2xx_pcm_dma_data *dma;
-
- dma = container_of(dma_data, struct pxa2xx_pcm_dma_data, params);
-
- snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id,
- width4 ? "32-bit" : "16-bit", out ? "out" : "in");
-
- dma->params.name = dma->name;
- dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx);
- dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) :
- (DCMD_INCTRGADDR | DCMD_FLOWSRC)) |
- (width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16;
- dma->params.dev_addr = ssp->phys_base + SSDR;
+ dma->addr_width = width4 ? DMA_SLAVE_BUSWIDTH_4_BYTES :
+ DMA_SLAVE_BUSWIDTH_2_BYTES;
+ dma->maxburst = 16;
+ dma->addr = ssp->phys_base + SSDR;
}
static int pxa_ssp_startup(struct snd_pcm_substream *substream,
@@ -107,7 +95,7 @@
{
struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
struct ssp_device *ssp = priv->ssp;
- struct pxa2xx_pcm_dma_data *dma;
+ struct snd_dmaengine_dai_dma_data *dma;
int ret = 0;
if (!cpu_dai->active) {
@@ -115,10 +103,14 @@
pxa_ssp_disable(ssp);
}
- dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL);
+ dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL);
if (!dma)
return -ENOMEM;
- snd_soc_dai_set_dma_data(cpu_dai, substream, &dma->params);
+
+ dma->filter_data = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ &ssp->drcmr_tx : &ssp->drcmr_rx;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma);
return ret;
}
@@ -559,7 +551,7 @@
u32 sspsp;
int width = snd_pcm_format_physical_width(params_format(params));
int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf;
- struct pxa2xx_pcm_dma_params *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream);
@@ -719,6 +711,7 @@
static int pxa_ssp_probe(struct snd_soc_dai *dai)
{
+ struct device *dev = dai->dev;
struct ssp_priv *priv;
int ret;
@@ -726,10 +719,26 @@
if (!priv)
return -ENOMEM;
- priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio");
- if (priv->ssp == NULL) {
- ret = -ENODEV;
- goto err_priv;
+ if (dev->of_node) {
+ struct device_node *ssp_handle;
+
+ ssp_handle = of_parse_phandle(dev->of_node, "port", 0);
+ if (!ssp_handle) {
+ dev_err(dev, "unable to get 'port' phandle\n");
+ return -ENODEV;
+ }
+
+ priv->ssp = pxa_ssp_request_of(ssp_handle, "SoC audio");
+ if (priv->ssp == NULL) {
+ ret = -ENODEV;
+ goto err_priv;
+ }
+ } else {
+ priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio");
+ if (priv->ssp == NULL) {
+ ret = -ENODEV;
+ goto err_priv;
+ }
}
priv->dai_fmt = (unsigned int) -1;
@@ -798,6 +807,12 @@
.name = "pxa-ssp",
};
+#ifdef CONFIG_OF
+static const struct of_device_id pxa_ssp_of_ids[] = {
+ { .compatible = "mrvl,pxa-ssp-dai" },
+};
+#endif
+
static int asoc_ssp_probe(struct platform_device *pdev)
{
return snd_soc_register_component(&pdev->dev, &pxa_ssp_component,
@@ -812,8 +827,9 @@
static struct platform_driver asoc_ssp_driver = {
.driver = {
- .name = "pxa-ssp-dai",
- .owner = THIS_MODULE,
+ .name = "pxa-ssp-dai",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(pxa_ssp_of_ids),
},
.probe = asoc_ssp_probe,
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 1475515..f1059d9 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -14,15 +14,16 @@
#include <linux/io.h>
#include <linux/module.h>
#include <linux/platform_device.h>
+#include <linux/dmaengine.h>
#include <sound/core.h>
#include <sound/ac97_codec.h>
#include <sound/soc.h>
#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
#include <mach/hardware.h>
#include <mach/regs-ac97.h>
-#include <mach/dma.h>
#include <mach/audio.h>
#include "pxa2xx-ac97.h"
@@ -48,44 +49,44 @@
.reset = pxa2xx_ac97_cold_reset,
};
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = {
- .name = "AC97 PCM Stereo out",
- .dev_addr = __PREG(PCDR),
- .drcmr = &DRCMR(12),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
+ .addr = __PREG(PCDR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .maxburst = 32,
+ .filter_data = &pxa2xx_ac97_pcm_stereo_in_req,
};
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = {
- .name = "AC97 PCM Stereo in",
- .dev_addr = __PREG(PCDR),
- .drcmr = &DRCMR(11),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
+ .addr = __PREG(PCDR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .maxburst = 32,
+ .filter_data = &pxa2xx_ac97_pcm_stereo_out_req,
};
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = {
- .name = "AC97 Aux PCM (Slot 5) Mono out",
- .dev_addr = __PREG(MODR),
- .drcmr = &DRCMR(10),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST16 | DCMD_WIDTH2,
+static unsigned long pxa2xx_ac97_pcm_aux_mono_out_req = 10;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = {
+ .addr = __PREG(MODR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
+ .maxburst = 16,
+ .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req,
};
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = {
- .name = "AC97 Aux PCM (Slot 5) Mono in",
- .dev_addr = __PREG(MODR),
- .drcmr = &DRCMR(9),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST16 | DCMD_WIDTH2,
+static unsigned long pxa2xx_ac97_pcm_aux_mono_in_req = 9;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = {
+ .addr = __PREG(MODR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
+ .maxburst = 16,
+ .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req,
};
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
- .name = "AC97 Mic PCM (Slot 6) Mono in",
- .dev_addr = __PREG(MCDR),
- .drcmr = &DRCMR(8),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST16 | DCMD_WIDTH2,
+static unsigned long pxa2xx_ac97_pcm_aux_mic_mono_req = 8;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = {
+ .addr = __PREG(MCDR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
+ .maxburst = 16,
+ .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req,
};
#ifdef CONFIG_PM
@@ -119,7 +120,7 @@
struct snd_pcm_hw_params *params,
struct snd_soc_dai *cpu_dai)
{
- struct pxa2xx_pcm_dma_params *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
dma_data = &pxa2xx_ac97_pcm_stereo_out;
@@ -135,7 +136,7 @@
struct snd_pcm_hw_params *params,
struct snd_soc_dai *cpu_dai)
{
- struct pxa2xx_pcm_dma_params *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index f7ca716..d5340a0 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -23,9 +23,9 @@
#include <sound/initval.h>
#include <sound/soc.h>
#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
#include <mach/hardware.h>
-#include <mach/dma.h>
#include <mach/audio.h>
#include "pxa2xx-i2s.h"
@@ -82,20 +82,20 @@
static struct clk *clk_i2s;
static int clk_ena = 0;
-static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = {
- .name = "I2S PCM Stereo out",
- .dev_addr = __PREG(SADR),
- .drcmr = &DRCMR(3),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3;
+static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = {
+ .addr = __PREG(SADR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .maxburst = 32,
+ .filter_data = &pxa2xx_i2s_pcm_stereo_out_req,
};
-static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = {
- .name = "I2S PCM Stereo in",
- .dev_addr = __PREG(SADR),
- .drcmr = &DRCMR(2),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2;
+static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = {
+ .addr = __PREG(SADR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .maxburst = 32,
+ .filter_data = &pxa2xx_i2s_pcm_stereo_in_req,
};
static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
@@ -163,7 +163,7 @@
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct pxa2xx_pcm_dma_params *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
BUG_ON(IS_ERR(clk_i2s));
clk_prepare_enable(clk_i2s);
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index ecff116..806da27 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -12,10 +12,13 @@
#include <linux/dma-mapping.h>
#include <linux/module.h>
+#include <linux/dmaengine.h>
+#include <linux/of.h>
#include <sound/core.h>
#include <sound/soc.h>
#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
#include "../../arm/pxa2xx-pcm.h"
@@ -25,7 +28,7 @@
struct snd_pcm_runtime *runtime = substream->runtime;
struct pxa2xx_runtime_data *prtd = runtime->private_data;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct pxa2xx_pcm_dma_params *dma;
+ struct snd_dmaengine_dai_dma_data *dma;
int ret;
dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
@@ -39,7 +42,7 @@
* with different params */
if (prtd->params == NULL) {
prtd->params = dma;
- ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW,
+ ret = pxa_request_dma("name", DMA_PRIO_LOW,
pxa2xx_pcm_dma_irq, substream);
if (ret < 0)
return ret;
@@ -47,7 +50,7 @@
} else if (prtd->params != dma) {
pxa_free_dma(prtd->dma_ch);
prtd->params = dma;
- ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW,
+ ret = pxa_request_dma("name", DMA_PRIO_LOW,
pxa2xx_pcm_dma_irq, substream);
if (ret < 0)
return ret;
@@ -131,10 +134,18 @@
return 0;
}
+#ifdef CONFIG_OF
+static const struct of_device_id snd_soc_pxa_audio_match[] = {
+ { .compatible = "mrvl,pxa-pcm-audio" },
+ { }
+};
+#endif
+
static struct platform_driver pxa_pcm_driver = {
.driver = {
- .name = "pxa-pcm-audio",
- .owner = THIS_MODULE,
+ .name = "pxa-pcm-audio",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(snd_soc_pxa_audio_match),
},
.probe = pxa2xx_soc_platform_probe,
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
index f4ea4f6..13c9ee0 100644
--- a/sound/soc/pxa/ttc-dkb.c
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -122,6 +122,7 @@
/* ttc/td audio machine driver */
static struct snd_soc_card ttc_dkb_card = {
.name = "ttc-dkb-hifi",
+ .owner = THIS_MODULE,
.dai_link = ttc_pm860x_hifi_dai,
.num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai),
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
index 58cfb1e..945e8ab 100644
--- a/sound/soc/s6000/s6105-ipcam.c
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -192,7 +192,7 @@
.num_links = 1,
};
-static struct s6000_snd_platform_data __initdata s6105_snd_data = {
+static struct s6000_snd_platform_data s6105_snd_data __initdata = {
.wide = 0,
.channel_in = 0,
.channel_out = 1,
diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c
index 2dd623f..2acf987 100644
--- a/sound/soc/samsung/ac97.c
+++ b/sound/soc/samsung/ac97.c
@@ -404,18 +404,13 @@
return -ENXIO;
}
- mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!mem_res) {
- dev_err(&pdev->dev, "Unable to get register resource\n");
- return -ENXIO;
- }
-
irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
if (!irq_res) {
dev_err(&pdev->dev, "AC97 IRQ not provided!\n");
return -ENXIO;
}
+ mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
s3c_ac97.regs = devm_ioremap_resource(&pdev->dev, mem_res);
if (IS_ERR(s3c_ac97.regs))
return PTR_ERR(s3c_ac97.regs);
@@ -462,7 +457,7 @@
if (ret)
goto err5;
- ret = asoc_dma_platform_register(&pdev->dev);
+ ret = samsung_asoc_dma_platform_register(&pdev->dev);
if (ret) {
dev_err(&pdev->dev, "failed to get register DMA: %d\n", ret);
goto err6;
@@ -485,7 +480,7 @@
{
struct resource *irq_res;
- asoc_dma_platform_unregister(&pdev->dev);
+ samsung_asoc_dma_platform_unregister(&pdev->dev);
snd_soc_unregister_component(&pdev->dev);
irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index 21b7926..9338d11 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -90,6 +90,13 @@
dma_info.period = prtd->dma_period;
dma_info.len = prtd->dma_period*limit;
+ if (dma_info.cap == DMA_CYCLIC) {
+ dma_info.buf = pos;
+ prtd->params->ops->prepare(prtd->params->ch, &dma_info);
+ prtd->dma_loaded += limit;
+ return;
+ }
+
while (prtd->dma_loaded < limit) {
pr_debug("dma_loaded: %d\n", prtd->dma_loaded);
@@ -176,6 +183,10 @@
prtd->params->ch = prtd->params->ops->request(
prtd->params->channel, &req, rtd->cpu_dai->dev,
prtd->params->ch_name);
+ if (!prtd->params->ch) {
+ pr_err("Failed to allocate DMA channel\n");
+ return -ENXIO;
+ }
prtd->params->ops->config(prtd->params->ch, &config);
}
@@ -433,17 +444,17 @@
.pcm_free = dma_free_dma_buffers,
};
-int asoc_dma_platform_register(struct device *dev)
+int samsung_asoc_dma_platform_register(struct device *dev)
{
return snd_soc_register_platform(dev, &samsung_asoc_platform);
}
-EXPORT_SYMBOL_GPL(asoc_dma_platform_register);
+EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register);
-void asoc_dma_platform_unregister(struct device *dev)
+void samsung_asoc_dma_platform_unregister(struct device *dev)
{
snd_soc_unregister_platform(dev);
}
-EXPORT_SYMBOL_GPL(asoc_dma_platform_unregister);
+EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_unregister);
MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
MODULE_DESCRIPTION("Samsung ASoC DMA Driver");
diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h
index 189a7a6..0e86315 100644
--- a/sound/soc/samsung/dma.h
+++ b/sound/soc/samsung/dma.h
@@ -22,7 +22,7 @@
char *ch_name;
};
-int asoc_dma_platform_register(struct device *dev);
-void asoc_dma_platform_unregister(struct device *dev);
+int samsung_asoc_dma_platform_register(struct device *dev);
+void samsung_asoc_dma_platform_unregister(struct device *dev);
#endif
diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h
index c0e6d9a..821a502 100644
--- a/sound/soc/samsung/i2s-regs.h
+++ b/sound/soc/samsung/i2s-regs.h
@@ -31,6 +31,10 @@
#define I2SLVL1ADDR 0x34
#define I2SLVL2ADDR 0x38
#define I2SLVL3ADDR 0x3c
+#define I2SSTR1 0x40
+#define I2SVER 0x44
+#define I2SFIC2 0x48
+#define I2STDM 0x4c
#define CON_RSTCLR (1 << 31)
#define CON_FRXOFSTATUS (1 << 26)
@@ -95,24 +99,39 @@
#define MOD_RXONLY (1 << 8)
#define MOD_TXRX (2 << 8)
#define MOD_MASK (3 << 8)
-#define MOD_LR_LLOW (0 << 7)
-#define MOD_LR_RLOW (1 << 7)
-#define MOD_SDF_IIS (0 << 5)
-#define MOD_SDF_MSB (1 << 5)
-#define MOD_SDF_LSB (2 << 5)
-#define MOD_SDF_MASK (3 << 5)
-#define MOD_RCLK_256FS (0 << 3)
-#define MOD_RCLK_512FS (1 << 3)
-#define MOD_RCLK_384FS (2 << 3)
-#define MOD_RCLK_768FS (3 << 3)
-#define MOD_RCLK_MASK (3 << 3)
-#define MOD_BCLK_32FS (0 << 1)
-#define MOD_BCLK_48FS (1 << 1)
-#define MOD_BCLK_16FS (2 << 1)
-#define MOD_BCLK_24FS (3 << 1)
-#define MOD_BCLK_MASK (3 << 1)
+#define MOD_LRP_SHIFT 7
+#define MOD_LR_LLOW 0
+#define MOD_LR_RLOW 1
+#define MOD_SDF_SHIFT 5
+#define MOD_SDF_IIS 0
+#define MOD_SDF_MSB 1
+#define MOD_SDF_LSB 2
+#define MOD_SDF_MASK 3
+#define MOD_RCLK_SHIFT 3
+#define MOD_RCLK_256FS 0
+#define MOD_RCLK_512FS 1
+#define MOD_RCLK_384FS 2
+#define MOD_RCLK_768FS 3
+#define MOD_RCLK_MASK 3
+#define MOD_BCLK_SHIFT 1
+#define MOD_BCLK_32FS 0
+#define MOD_BCLK_48FS 1
+#define MOD_BCLK_16FS 2
+#define MOD_BCLK_24FS 3
+#define MOD_BCLK_MASK 3
#define MOD_8BIT (1 << 0)
+#define EXYNOS5420_MOD_LRP_SHIFT 15
+#define EXYNOS5420_MOD_SDF_SHIFT 6
+#define EXYNOS5420_MOD_RCLK_SHIFT 4
+#define EXYNOS5420_MOD_BCLK_SHIFT 0
+#define EXYNOS5420_MOD_BCLK_64FS 4
+#define EXYNOS5420_MOD_BCLK_96FS 5
+#define EXYNOS5420_MOD_BCLK_128FS 6
+#define EXYNOS5420_MOD_BCLK_192FS 7
+#define EXYNOS5420_MOD_BCLK_256FS 8
+#define EXYNOS5420_MOD_BCLK_MASK 0xf
+
#define MOD_CDCLKCON (1 << 12)
#define PSR_PSREN (1 << 15)
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 959c702..b302f3b 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -40,6 +40,7 @@
struct samsung_i2s_dai_data {
int dai_type;
+ u32 quirks;
};
struct i2s_dai {
@@ -198,7 +199,13 @@
/* Read RCLK of I2S (in multiples of LRCLK) */
static inline unsigned get_rfs(struct i2s_dai *i2s)
{
- u32 rfs = (readl(i2s->addr + I2SMOD) >> 3) & 0x3;
+ u32 rfs;
+
+ if (i2s->quirks & QUIRK_SUPPORTS_TDM)
+ rfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_RCLK_SHIFT;
+ else
+ rfs = (readl(i2s->addr + I2SMOD) >> MOD_RCLK_SHIFT);
+ rfs &= MOD_RCLK_MASK;
switch (rfs) {
case 3: return 768;
@@ -212,21 +219,26 @@
static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs)
{
u32 mod = readl(i2s->addr + I2SMOD);
+ int rfs_shift;
- mod &= ~MOD_RCLK_MASK;
+ if (i2s->quirks & QUIRK_SUPPORTS_TDM)
+ rfs_shift = EXYNOS5420_MOD_RCLK_SHIFT;
+ else
+ rfs_shift = MOD_RCLK_SHIFT;
+ mod &= ~(MOD_RCLK_MASK << rfs_shift);
switch (rfs) {
case 768:
- mod |= MOD_RCLK_768FS;
+ mod |= (MOD_RCLK_768FS << rfs_shift);
break;
case 512:
- mod |= MOD_RCLK_512FS;
+ mod |= (MOD_RCLK_512FS << rfs_shift);
break;
case 384:
- mod |= MOD_RCLK_384FS;
+ mod |= (MOD_RCLK_384FS << rfs_shift);
break;
default:
- mod |= MOD_RCLK_256FS;
+ mod |= (MOD_RCLK_256FS << rfs_shift);
break;
}
@@ -236,9 +248,22 @@
/* Read Bit-Clock of I2S (in multiples of LRCLK) */
static inline unsigned get_bfs(struct i2s_dai *i2s)
{
- u32 bfs = (readl(i2s->addr + I2SMOD) >> 1) & 0x3;
+ u32 bfs;
+
+ if (i2s->quirks & QUIRK_SUPPORTS_TDM) {
+ bfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_BCLK_SHIFT;
+ bfs &= EXYNOS5420_MOD_BCLK_MASK;
+ } else {
+ bfs = readl(i2s->addr + I2SMOD) >> MOD_BCLK_SHIFT;
+ bfs &= MOD_BCLK_MASK;
+ }
switch (bfs) {
+ case 8: return 256;
+ case 7: return 192;
+ case 6: return 128;
+ case 5: return 96;
+ case 4: return 64;
case 3: return 24;
case 2: return 16;
case 1: return 48;
@@ -250,21 +275,50 @@
static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs)
{
u32 mod = readl(i2s->addr + I2SMOD);
+ int bfs_shift;
+ int tdm = i2s->quirks & QUIRK_SUPPORTS_TDM;
- mod &= ~MOD_BCLK_MASK;
+ if (i2s->quirks & QUIRK_SUPPORTS_TDM) {
+ bfs_shift = EXYNOS5420_MOD_BCLK_SHIFT;
+ mod &= ~(EXYNOS5420_MOD_BCLK_MASK << bfs_shift);
+ } else {
+ bfs_shift = MOD_BCLK_SHIFT;
+ mod &= ~(MOD_BCLK_MASK << bfs_shift);
+ }
+
+ /* Non-TDM I2S controllers do not support BCLK > 48 * FS */
+ if (!tdm && bfs > 48) {
+ dev_err(&i2s->pdev->dev, "Unsupported BCLK divider\n");
+ return;
+ }
switch (bfs) {
case 48:
- mod |= MOD_BCLK_48FS;
+ mod |= (MOD_BCLK_48FS << bfs_shift);
break;
case 32:
- mod |= MOD_BCLK_32FS;
+ mod |= (MOD_BCLK_32FS << bfs_shift);
break;
case 24:
- mod |= MOD_BCLK_24FS;
+ mod |= (MOD_BCLK_24FS << bfs_shift);
break;
case 16:
- mod |= MOD_BCLK_16FS;
+ mod |= (MOD_BCLK_16FS << bfs_shift);
+ break;
+ case 64:
+ mod |= (EXYNOS5420_MOD_BCLK_64FS << bfs_shift);
+ break;
+ case 96:
+ mod |= (EXYNOS5420_MOD_BCLK_96FS << bfs_shift);
+ break;
+ case 128:
+ mod |= (EXYNOS5420_MOD_BCLK_128FS << bfs_shift);
+ break;
+ case 192:
+ mod |= (EXYNOS5420_MOD_BCLK_192FS << bfs_shift);
+ break;
+ case 256:
+ mod |= (EXYNOS5420_MOD_BCLK_256FS << bfs_shift);
break;
default:
dev_err(&i2s->pdev->dev, "Wrong BCLK Divider!\n");
@@ -491,20 +545,32 @@
{
struct i2s_dai *i2s = to_info(dai);
u32 mod = readl(i2s->addr + I2SMOD);
+ int lrp_shift, sdf_shift, sdf_mask, lrp_rlow;
u32 tmp = 0;
+ if (i2s->quirks & QUIRK_SUPPORTS_TDM) {
+ lrp_shift = EXYNOS5420_MOD_LRP_SHIFT;
+ sdf_shift = EXYNOS5420_MOD_SDF_SHIFT;
+ } else {
+ lrp_shift = MOD_LRP_SHIFT;
+ sdf_shift = MOD_SDF_SHIFT;
+ }
+
+ sdf_mask = MOD_SDF_MASK << sdf_shift;
+ lrp_rlow = MOD_LR_RLOW << lrp_shift;
+
/* Format is priority */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_RIGHT_J:
- tmp |= MOD_LR_RLOW;
- tmp |= MOD_SDF_MSB;
+ tmp |= lrp_rlow;
+ tmp |= (MOD_SDF_MSB << sdf_shift);
break;
case SND_SOC_DAIFMT_LEFT_J:
- tmp |= MOD_LR_RLOW;
- tmp |= MOD_SDF_LSB;
+ tmp |= lrp_rlow;
+ tmp |= (MOD_SDF_LSB << sdf_shift);
break;
case SND_SOC_DAIFMT_I2S:
- tmp |= MOD_SDF_IIS;
+ tmp |= (MOD_SDF_IIS << sdf_shift);
break;
default:
dev_err(&i2s->pdev->dev, "Format not supported\n");
@@ -519,10 +585,10 @@
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_NB_IF:
- if (tmp & MOD_LR_RLOW)
- tmp &= ~MOD_LR_RLOW;
+ if (tmp & lrp_rlow)
+ tmp &= ~lrp_rlow;
else
- tmp |= MOD_LR_RLOW;
+ tmp |= lrp_rlow;
break;
default:
dev_err(&i2s->pdev->dev, "Polarity not supported\n");
@@ -544,15 +610,18 @@
return -EINVAL;
}
+ /*
+ * Don't change the I2S mode if any controller is active on this
+ * channel.
+ */
if (any_active(i2s) &&
- ((mod & (MOD_SDF_MASK | MOD_LR_RLOW
- | MOD_SLAVE)) != tmp)) {
+ ((mod & (sdf_mask | lrp_rlow | MOD_SLAVE)) != tmp)) {
dev_err(&i2s->pdev->dev,
"%s:%d Other DAI busy\n", __func__, __LINE__);
return -EAGAIN;
}
- mod &= ~(MOD_SDF_MASK | MOD_LR_RLOW | MOD_SLAVE);
+ mod &= ~(sdf_mask | lrp_rlow | MOD_SLAVE);
mod |= tmp;
writel(mod, i2s->addr + I2SMOD);
@@ -1007,6 +1076,8 @@
if (IS_ERR(i2s->pdev))
return NULL;
+ i2s->pdev->dev.parent = &pdev->dev;
+
platform_set_drvdata(i2s->pdev, i2s);
ret = platform_device_add(i2s->pdev);
if (ret < 0)
@@ -1018,18 +1089,18 @@
static const struct of_device_id exynos_i2s_match[];
-static inline int samsung_i2s_get_driver_data(struct platform_device *pdev)
+static inline const struct samsung_i2s_dai_data *samsung_i2s_get_driver_data(
+ struct platform_device *pdev)
{
#ifdef CONFIG_OF
- struct samsung_i2s_dai_data *data;
if (pdev->dev.of_node) {
const struct of_device_id *match;
match = of_match_node(exynos_i2s_match, pdev->dev.of_node);
- data = (struct samsung_i2s_dai_data *) match->data;
- return data->dai_type;
+ return match->data;
} else
#endif
- return platform_get_device_id(pdev)->driver_data;
+ return (struct samsung_i2s_dai_data *)
+ platform_get_device_id(pdev)->driver_data;
}
#ifdef CONFIG_PM_RUNTIME
@@ -1060,13 +1131,13 @@
struct resource *res;
u32 regs_base, quirks = 0, idma_addr = 0;
struct device_node *np = pdev->dev.of_node;
- enum samsung_dai_type samsung_dai_type;
+ const struct samsung_i2s_dai_data *i2s_dai_data;
int ret = 0;
/* Call during Seconday interface registration */
- samsung_dai_type = samsung_i2s_get_driver_data(pdev);
+ i2s_dai_data = samsung_i2s_get_driver_data(pdev);
- if (samsung_dai_type == TYPE_SEC) {
+ if (i2s_dai_data->dai_type == TYPE_SEC) {
sec_dai = dev_get_drvdata(&pdev->dev);
if (!sec_dai) {
dev_err(&pdev->dev, "Unable to get drvdata\n");
@@ -1075,7 +1146,7 @@
snd_soc_register_component(&sec_dai->pdev->dev,
&samsung_i2s_component,
&sec_dai->i2s_dai_drv, 1);
- asoc_dma_platform_register(&pdev->dev);
+ samsung_asoc_dma_platform_register(&pdev->dev);
return 0;
}
@@ -1115,15 +1186,7 @@
idma_addr = i2s_cfg->idma_addr;
}
} else {
- if (of_find_property(np, "samsung,supports-6ch", NULL))
- quirks |= QUIRK_PRI_6CHAN;
-
- if (of_find_property(np, "samsung,supports-secdai", NULL))
- quirks |= QUIRK_SEC_DAI;
-
- if (of_find_property(np, "samsung,supports-rstclr", NULL))
- quirks |= QUIRK_NEED_RSTCLR;
-
+ quirks = i2s_dai_data->quirks;
if (of_property_read_u32(np, "samsung,idma-addr",
&idma_addr)) {
if (quirks & QUIRK_SEC_DAI) {
@@ -1200,7 +1263,7 @@
pm_runtime_enable(&pdev->dev);
- asoc_dma_platform_register(&pdev->dev);
+ samsung_asoc_dma_platform_register(&pdev->dev);
return 0;
err:
@@ -1230,33 +1293,59 @@
i2s->pri_dai = NULL;
i2s->sec_dai = NULL;
- asoc_dma_platform_unregister(&pdev->dev);
+ samsung_asoc_dma_platform_unregister(&pdev->dev);
snd_soc_unregister_component(&pdev->dev);
return 0;
}
+static const struct samsung_i2s_dai_data i2sv3_dai_type = {
+ .dai_type = TYPE_PRI,
+ .quirks = QUIRK_NO_MUXPSR,
+};
+
+static const struct samsung_i2s_dai_data i2sv5_dai_type = {
+ .dai_type = TYPE_PRI,
+ .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR,
+};
+
+static const struct samsung_i2s_dai_data i2sv6_dai_type = {
+ .dai_type = TYPE_PRI,
+ .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR |
+ QUIRK_SUPPORTS_TDM,
+};
+
+static const struct samsung_i2s_dai_data samsung_dai_type_pri = {
+ .dai_type = TYPE_PRI,
+};
+
+static const struct samsung_i2s_dai_data samsung_dai_type_sec = {
+ .dai_type = TYPE_SEC,
+};
+
static struct platform_device_id samsung_i2s_driver_ids[] = {
{
.name = "samsung-i2s",
- .driver_data = TYPE_PRI,
+ .driver_data = (kernel_ulong_t)&samsung_dai_type_pri,
}, {
.name = "samsung-i2s-sec",
- .driver_data = TYPE_SEC,
+ .driver_data = (kernel_ulong_t)&samsung_dai_type_sec,
},
{},
};
MODULE_DEVICE_TABLE(platform, samsung_i2s_driver_ids);
#ifdef CONFIG_OF
-static struct samsung_i2s_dai_data samsung_i2s_dai_data_array[] = {
- [TYPE_PRI] = { TYPE_PRI },
- [TYPE_SEC] = { TYPE_SEC },
-};
-
static const struct of_device_id exynos_i2s_match[] = {
- { .compatible = "samsung,i2s-v5",
- .data = &samsung_i2s_dai_data_array[TYPE_PRI],
+ {
+ .compatible = "samsung,s3c6410-i2s",
+ .data = &i2sv3_dai_type,
+ }, {
+ .compatible = "samsung,s5pv210-i2s",
+ .data = &i2sv5_dai_type,
+ }, {
+ .compatible = "samsung,exynos5420-i2s",
+ .data = &i2sv6_dai_type,
},
{},
};
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index 1566afe..e54256f 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -594,7 +594,7 @@
goto err5;
}
- ret = asoc_dma_platform_register(&pdev->dev);
+ ret = samsung_asoc_dma_platform_register(&pdev->dev);
if (ret) {
dev_err(&pdev->dev, "failed to get register DMA: %d\n", ret);
goto err6;
@@ -623,7 +623,7 @@
struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id];
struct resource *mem_res;
- asoc_dma_platform_unregister(&pdev->dev);
+ samsung_asoc_dma_platform_unregister(&pdev->dev);
snd_soc_unregister_component(&pdev->dev);
pm_runtime_disable(&pdev->dev);
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 47e2386..ea885cb 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -176,7 +176,7 @@
return ret;
}
- ret = asoc_dma_platform_register(&pdev->dev);
+ ret = samsung_asoc_dma_platform_register(&pdev->dev);
if (ret) {
pr_err("failed to register the DMA: %d\n", ret);
goto err;
@@ -190,7 +190,7 @@
static int s3c2412_iis_dev_remove(struct platform_device *pdev)
{
- asoc_dma_platform_unregister(&pdev->dev);
+ samsung_asoc_dma_platform_unregister(&pdev->dev);
snd_soc_unregister_component(&pdev->dev);
return 0;
}
diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c
index 8b34145..9c8ebd8 100644
--- a/sound/soc/samsung/s3c24xx-i2s.c
+++ b/sound/soc/samsung/s3c24xx-i2s.c
@@ -480,7 +480,7 @@
return ret;
}
- ret = asoc_dma_platform_register(&pdev->dev);
+ ret = samsung_asoc_dma_platform_register(&pdev->dev);
if (ret) {
pr_err("failed to register the dma: %d\n", ret);
goto err;
@@ -494,7 +494,7 @@
static int s3c24xx_iis_dev_remove(struct platform_device *pdev)
{
- asoc_dma_platform_unregister(&pdev->dev);
+ samsung_asoc_dma_platform_unregister(&pdev->dev);
snd_soc_unregister_component(&pdev->dev);
return 0;
}
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index 581ea4a..5fd7a05 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -11,6 +11,7 @@
#include <sound/pcm_params.h>
#include <linux/module.h>
#include <linux/of.h>
+#include <linux/of_device.h>
/*
* Default CFG switch settings to use this driver:
@@ -37,11 +38,19 @@
/* SMDK has a 16.934MHZ crystal attached to WM8994 */
#define SMDK_WM8994_FREQ 16934000
+struct smdk_wm8994_data {
+ int mclk1_rate;
+};
+
+/* Default SMDKs */
+static struct smdk_wm8994_data smdk_board_data = {
+ .mclk1_rate = SMDK_WM8994_FREQ,
+};
+
static int smdk_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
unsigned int pll_out;
int ret;
@@ -54,18 +63,6 @@
else
pll_out = params_rate(params) * 256;
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
- | SND_SOC_DAIFMT_NB_NF
- | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
- | SND_SOC_DAIFMT_NB_NF
- | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1,
SMDK_WM8994_FREQ, pll_out);
if (ret < 0)
@@ -131,6 +128,8 @@
.platform_name = "samsung-i2s.0",
.codec_name = "wm8994-codec",
.init = smdk_wm8994_init_paiftx,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.ops = &smdk_ops,
}, { /* Sec_Fifo Playback i/f */
.name = "Sec_FIFO TX",
@@ -139,6 +138,8 @@
.codec_dai_name = "wm8994-aif1",
.platform_name = "samsung-i2s-sec",
.codec_name = "wm8994-codec",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.ops = &smdk_ops,
},
};
@@ -150,15 +151,28 @@
.num_links = ARRAY_SIZE(smdk_dai),
};
+#ifdef CONFIG_OF
+static const struct of_device_id samsung_wm8994_of_match[] = {
+ { .compatible = "samsung,smdk-wm8994", .data = &smdk_board_data },
+ {},
+};
+MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match);
+#endif /* CONFIG_OF */
static int smdk_audio_probe(struct platform_device *pdev)
{
int ret;
struct device_node *np = pdev->dev.of_node;
struct snd_soc_card *card = &smdk;
+ struct smdk_wm8994_data *board;
+ const struct of_device_id *id;
card->dev = &pdev->dev;
+ board = devm_kzalloc(&pdev->dev, sizeof(*board), GFP_KERNEL);
+ if (!board)
+ return -ENOMEM;
+
if (np) {
smdk_dai[0].cpu_dai_name = NULL;
smdk_dai[0].cpu_of_node = of_parse_phandle(np,
@@ -173,6 +187,12 @@
smdk_dai[0].platform_of_node = smdk_dai[0].cpu_of_node;
}
+ id = of_match_device(samsung_wm8994_of_match, &pdev->dev);
+ if (id)
+ *board = *((struct smdk_wm8994_data *)id->data);
+
+ platform_set_drvdata(pdev, board);
+
ret = snd_soc_register_card(card);
if (ret)
@@ -190,17 +210,9 @@
return 0;
}
-#ifdef CONFIG_OF
-static const struct of_device_id samsung_wm8994_of_match[] = {
- { .compatible = "samsung,smdk-wm8994", },
- {},
-};
-MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match);
-#endif /* CONFIG_OF */
-
static struct platform_driver smdk_audio_driver = {
.driver = {
- .name = "smdk-audio",
+ .name = "smdk-audio-wm8894",
.owner = THIS_MODULE,
.of_match_table = of_match_ptr(samsung_wm8994_of_match),
},
@@ -212,4 +224,4 @@
MODULE_DESCRIPTION("ALSA SoC SMDK WM8994");
MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:smdk-audio");
+MODULE_ALIAS("platform:smdk-audio-wm8994");
diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c
index 2e5ebb2..28487dc 100644
--- a/sound/soc/samsung/spdif.c
+++ b/sound/soc/samsung/spdif.c
@@ -395,7 +395,7 @@
spin_lock_init(&spdif->lock);
- spdif->pclk = clk_get(&pdev->dev, "spdif");
+ spdif->pclk = devm_clk_get(&pdev->dev, "spdif");
if (IS_ERR(spdif->pclk)) {
dev_err(&pdev->dev, "failed to get peri-clock\n");
ret = -ENOENT;
@@ -403,7 +403,7 @@
}
clk_prepare_enable(spdif->pclk);
- spdif->sclk = clk_get(&pdev->dev, "sclk_spdif");
+ spdif->sclk = devm_clk_get(&pdev->dev, "sclk_spdif");
if (IS_ERR(spdif->sclk)) {
dev_err(&pdev->dev, "failed to get internal source clock\n");
ret = -ENOENT;
@@ -442,7 +442,7 @@
spdif->dma_playback = &spdif_stereo_out;
- ret = asoc_dma_platform_register(&pdev->dev);
+ ret = samsung_asoc_dma_platform_register(&pdev->dev);
if (ret) {
dev_err(&pdev->dev, "failed to register DMA: %d\n", ret);
goto err5;
@@ -457,10 +457,8 @@
release_mem_region(mem_res->start, resource_size(mem_res));
err2:
clk_disable_unprepare(spdif->sclk);
- clk_put(spdif->sclk);
err1:
clk_disable_unprepare(spdif->pclk);
- clk_put(spdif->pclk);
err0:
return ret;
}
@@ -470,7 +468,7 @@
struct samsung_spdif_info *spdif = &spdif_info;
struct resource *mem_res;
- asoc_dma_platform_unregister(&pdev->dev);
+ samsung_asoc_dma_platform_unregister(&pdev->dev);
snd_soc_unregister_component(&pdev->dev);
iounmap(spdif->regs);
@@ -480,9 +478,7 @@
release_mem_region(mem_res->start, resource_size(mem_res));
clk_disable_unprepare(spdif->sclk);
- clk_put(spdif->sclk);
clk_disable_unprepare(spdif->pclk);
- clk_put(spdif->pclk);
return 0;
}
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 6bcb116..56d8ff6 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -34,6 +34,13 @@
select SH_DMAE
select FW_LOADER
+config SND_SOC_RCAR
+ tristate "R-Car series SRU/SCU/SSIU/SSI support"
+ select SND_SIMPLE_CARD
+ select RCAR_CLK_ADG
+ help
+ This option enables R-Car SUR/SCU/SSIU/SSI sound support
+
##
## Boards
##
diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile
index 849b387..aaf3dcd 100644
--- a/sound/soc/sh/Makefile
+++ b/sound/soc/sh/Makefile
@@ -12,6 +12,9 @@
obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o
obj-$(CONFIG_SND_SOC_SH4_SIU) += snd-soc-siu.o
+## audio units for R-Car
+obj-$(CONFIG_SND_SOC_RCAR) += rcar/
+
## boards
snd-soc-sh7760-ac97-objs := sh7760-ac97.o
snd-soc-migor-objs := migor.o
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 3039026..b33ca7c 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -235,6 +235,8 @@
struct sh_dmae_slave slave; /* see fsi_handler_init() */
struct work_struct work;
dma_addr_t dma;
+ int loop_cnt;
+ int additional_pos;
};
struct fsi_clk {
@@ -1289,6 +1291,8 @@
io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
BUSOP_SET(16, PACKAGE_16BITBUS_STREAM);
+ io->loop_cnt = 2; /* push 1st, 2nd period first, then 3rd, 4th... */
+ io->additional_pos = 0;
io->dma = dma_map_single(dai->dev, runtime->dma_area,
snd_pcm_lib_buffer_bytes(io->substream), dir);
return 0;
@@ -1305,11 +1309,15 @@
return 0;
}
-static dma_addr_t fsi_dma_get_area(struct fsi_stream *io)
+static dma_addr_t fsi_dma_get_area(struct fsi_stream *io, int additional)
{
struct snd_pcm_runtime *runtime = io->substream->runtime;
+ int period = io->period_pos + additional;
- return io->dma + samples_to_bytes(runtime, io->buff_sample_pos);
+ if (period >= runtime->periods)
+ period = 0;
+
+ return io->dma + samples_to_bytes(runtime, period * io->period_samples);
}
static void fsi_dma_complete(void *data)
@@ -1321,7 +1329,7 @@
enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
DMA_TO_DEVICE : DMA_FROM_DEVICE;
- dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io),
+ dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io, 0),
samples_to_bytes(runtime, io->period_samples), dir);
io->buff_sample_pos += io->period_samples;
@@ -1347,7 +1355,7 @@
struct snd_pcm_runtime *runtime;
enum dma_data_direction dir;
int is_play = fsi_stream_is_play(fsi, io);
- int len;
+ int len, i;
dma_addr_t buf;
if (!fsi_stream_is_working(fsi, io))
@@ -1357,26 +1365,33 @@
runtime = io->substream->runtime;
dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
len = samples_to_bytes(runtime, io->period_samples);
- buf = fsi_dma_get_area(io);
- dma_sync_single_for_device(dai->dev, buf, len, dir);
+ for (i = 0; i < io->loop_cnt; i++) {
+ buf = fsi_dma_get_area(io, io->additional_pos);
- desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
- DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
- if (!desc) {
- dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
- return;
+ dma_sync_single_for_device(dai->dev, buf, len, dir);
+
+ desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
+ DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+ if (!desc) {
+ dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
+ return;
+ }
+
+ desc->callback = fsi_dma_complete;
+ desc->callback_param = io;
+
+ if (dmaengine_submit(desc) < 0) {
+ dev_err(dai->dev, "tx_submit() fail\n");
+ return;
+ }
+
+ dma_async_issue_pending(io->chan);
+
+ io->additional_pos = 1;
}
- desc->callback = fsi_dma_complete;
- desc->callback_param = io;
-
- if (dmaengine_submit(desc) < 0) {
- dev_err(dai->dev, "tx_submit() fail\n");
- return;
- }
-
- dma_async_issue_pending(io->chan);
+ io->loop_cnt = 1;
/*
* FIXME
diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile
new file mode 100644
index 0000000..0ff492d
--- /dev/null
+++ b/sound/soc/sh/rcar/Makefile
@@ -0,0 +1,2 @@
+snd-soc-rcar-objs := core.o gen.o scu.o adg.o ssi.o
+obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o
\ No newline at end of file
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
new file mode 100644
index 0000000..d80deb7
--- /dev/null
+++ b/sound/soc/sh/rcar/adg.c
@@ -0,0 +1,234 @@
+/*
+ * Helper routines for R-Car sound ADG.
+ *
+ * Copyright (C) 2013 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This file is subject to the terms and conditions of the GNU General Public
+ * License. See the file "COPYING" in the main directory of this archive
+ * for more details.
+ */
+#include <linux/sh_clk.h>
+#include <mach/clock.h>
+#include "rsnd.h"
+
+#define CLKA 0
+#define CLKB 1
+#define CLKC 2
+#define CLKI 3
+#define CLKMAX 4
+
+struct rsnd_adg {
+ struct clk *clk[CLKMAX];
+
+ int rate_of_441khz_div_6;
+ int rate_of_48khz_div_6;
+};
+
+#define for_each_rsnd_clk(pos, adg, i) \
+ for (i = 0, (pos) = adg->clk[i]; \
+ i < CLKMAX; \
+ i++, (pos) = adg->clk[i])
+#define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg)
+
+static enum rsnd_reg rsnd_adg_ssi_reg_get(int id)
+{
+ enum rsnd_reg reg;
+
+ /*
+ * SSI 8 is not connected to ADG.
+ * it works with SSI 7
+ */
+ if (id == 8)
+ return RSND_REG_MAX;
+
+ if (0 <= id && id <= 3)
+ reg = RSND_REG_AUDIO_CLK_SEL0;
+ else if (4 <= id && id <= 7)
+ reg = RSND_REG_AUDIO_CLK_SEL1;
+ else
+ reg = RSND_REG_AUDIO_CLK_SEL2;
+
+ return reg;
+}
+
+int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ enum rsnd_reg reg;
+ int id;
+
+ /*
+ * "mod" = "ssi" here.
+ * we can get "ssi id" from mod
+ */
+ id = rsnd_mod_id(mod);
+ reg = rsnd_adg_ssi_reg_get(id);
+
+ rsnd_write(priv, mod, reg, 0);
+
+ return 0;
+}
+
+int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct clk *clk;
+ enum rsnd_reg reg;
+ int id, shift, i;
+ u32 data;
+ int sel_table[] = {
+ [CLKA] = 0x1,
+ [CLKB] = 0x2,
+ [CLKC] = 0x3,
+ [CLKI] = 0x0,
+ };
+
+ dev_dbg(dev, "request clock = %d\n", rate);
+
+ /*
+ * find suitable clock from
+ * AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI.
+ */
+ data = 0;
+ for_each_rsnd_clk(clk, adg, i) {
+ if (rate == clk_get_rate(clk)) {
+ data = sel_table[i];
+ goto found_clock;
+ }
+ }
+
+ /*
+ * find 1/6 clock from BRGA/BRGB
+ */
+ if (rate == adg->rate_of_441khz_div_6) {
+ data = 0x10;
+ goto found_clock;
+ }
+
+ if (rate == adg->rate_of_48khz_div_6) {
+ data = 0x20;
+ goto found_clock;
+ }
+
+ return -EIO;
+
+found_clock:
+
+ /*
+ * This "mod" = "ssi" here.
+ * we can get "ssi id" from mod
+ */
+ id = rsnd_mod_id(mod);
+ reg = rsnd_adg_ssi_reg_get(id);
+
+ dev_dbg(dev, "ADG: ssi%d selects clk%d = %d", id, i, rate);
+
+ /*
+ * Enable SSIx clock
+ */
+ shift = (id % 4) * 8;
+
+ rsnd_bset(priv, mod, reg,
+ 0xFF << shift,
+ data << shift);
+
+ return 0;
+}
+
+static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
+{
+ struct clk *clk;
+ unsigned long rate;
+ u32 ckr;
+ int i;
+ int brg_table[] = {
+ [CLKA] = 0x0,
+ [CLKB] = 0x1,
+ [CLKC] = 0x4,
+ [CLKI] = 0x2,
+ };
+
+ /*
+ * This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC
+ * have 44.1kHz or 48kHz base clocks for now.
+ *
+ * SSI itself can divide parent clock by 1/1 - 1/16
+ * So, BRGA outputs 44.1kHz base parent clock 1/32,
+ * and, BRGB outputs 48.0kHz base parent clock 1/32 here.
+ * see
+ * rsnd_adg_ssi_clk_try_start()
+ */
+ ckr = 0;
+ adg->rate_of_441khz_div_6 = 0;
+ adg->rate_of_48khz_div_6 = 0;
+ for_each_rsnd_clk(clk, adg, i) {
+ rate = clk_get_rate(clk);
+
+ if (0 == rate) /* not used */
+ continue;
+
+ /* RBGA */
+ if (!adg->rate_of_441khz_div_6 && (0 == rate % 44100)) {
+ adg->rate_of_441khz_div_6 = rate / 6;
+ ckr |= brg_table[i] << 20;
+ }
+
+ /* RBGB */
+ if (!adg->rate_of_48khz_div_6 && (0 == rate % 48000)) {
+ adg->rate_of_48khz_div_6 = rate / 6;
+ ckr |= brg_table[i] << 16;
+ }
+ }
+
+ rsnd_priv_bset(priv, SSICKR, 0x00FF0000, ckr);
+ rsnd_priv_write(priv, BRRA, 0x00000002); /* 1/6 */
+ rsnd_priv_write(priv, BRRB, 0x00000002); /* 1/6 */
+}
+
+int rsnd_adg_probe(struct platform_device *pdev,
+ struct rcar_snd_info *info,
+ struct rsnd_priv *priv)
+{
+ struct rsnd_adg *adg;
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct clk *clk;
+ int i;
+
+ adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL);
+ if (!adg) {
+ dev_err(dev, "ADG allocate failed\n");
+ return -ENOMEM;
+ }
+
+ adg->clk[CLKA] = clk_get(NULL, "audio_clk_a");
+ adg->clk[CLKB] = clk_get(NULL, "audio_clk_b");
+ adg->clk[CLKC] = clk_get(NULL, "audio_clk_c");
+ adg->clk[CLKI] = clk_get(NULL, "audio_clk_internal");
+ for_each_rsnd_clk(clk, adg, i) {
+ if (IS_ERR(clk)) {
+ dev_err(dev, "Audio clock failed\n");
+ return -EIO;
+ }
+ }
+
+ rsnd_adg_ssi_clk_init(priv, adg);
+
+ priv->adg = adg;
+
+ dev_dbg(dev, "adg probed\n");
+
+ return 0;
+}
+
+void rsnd_adg_remove(struct platform_device *pdev,
+ struct rsnd_priv *priv)
+{
+ struct rsnd_adg *adg = priv->adg;
+ struct clk *clk;
+ int i;
+
+ for_each_rsnd_clk(clk, adg, i)
+ clk_put(clk);
+}
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
new file mode 100644
index 0000000..a357060
--- /dev/null
+++ b/sound/soc/sh/rcar/core.c
@@ -0,0 +1,861 @@
+/*
+ * Renesas R-Car SRU/SCU/SSIU/SSI support
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * Based on fsi.c
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+/*
+ * Renesas R-Car sound device structure
+ *
+ * Gen1
+ *
+ * SRU : Sound Routing Unit
+ * - SRC : Sampling Rate Converter
+ * - CMD
+ * - CTU : Channel Count Conversion Unit
+ * - MIX : Mixer
+ * - DVC : Digital Volume and Mute Function
+ * - SSI : Serial Sound Interface
+ *
+ * Gen2
+ *
+ * SCU : Sampling Rate Converter Unit
+ * - SRC : Sampling Rate Converter
+ * - CMD
+ * - CTU : Channel Count Conversion Unit
+ * - MIX : Mixer
+ * - DVC : Digital Volume and Mute Function
+ * SSIU : Serial Sound Interface Unit
+ * - SSI : Serial Sound Interface
+ */
+
+/*
+ * driver data Image
+ *
+ * rsnd_priv
+ * |
+ * | ** this depends on Gen1/Gen2
+ * |
+ * +- gen
+ * |
+ * | ** these depend on data path
+ * | ** gen and platform data control it
+ * |
+ * +- rdai[0]
+ * | | sru ssiu ssi
+ * | +- playback -> [mod] -> [mod] -> [mod] -> ...
+ * | |
+ * | | sru ssiu ssi
+ * | +- capture -> [mod] -> [mod] -> [mod] -> ...
+ * |
+ * +- rdai[1]
+ * | | sru ssiu ssi
+ * | +- playback -> [mod] -> [mod] -> [mod] -> ...
+ * | |
+ * | | sru ssiu ssi
+ * | +- capture -> [mod] -> [mod] -> [mod] -> ...
+ * ...
+ * |
+ * | ** these control ssi
+ * |
+ * +- ssi
+ * | |
+ * | +- ssi[0]
+ * | +- ssi[1]
+ * | +- ssi[2]
+ * | ...
+ * |
+ * | ** these control scu
+ * |
+ * +- scu
+ * |
+ * +- scu[0]
+ * +- scu[1]
+ * +- scu[2]
+ * ...
+ *
+ *
+ * for_each_rsnd_dai(xx, priv, xx)
+ * rdai[0] => rdai[1] => rdai[2] => ...
+ *
+ * for_each_rsnd_mod(xx, rdai, xx)
+ * [mod] => [mod] => [mod] => ...
+ *
+ * rsnd_dai_call(xxx, fn )
+ * [mod]->fn() -> [mod]->fn() -> [mod]->fn()...
+ *
+ */
+#include <linux/pm_runtime.h>
+#include "rsnd.h"
+
+#define RSND_RATES SNDRV_PCM_RATE_8000_96000
+#define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
+
+/*
+ * rsnd_platform functions
+ */
+#define rsnd_platform_call(priv, dai, func, param...) \
+ (!(priv->info->func) ? -ENODEV : \
+ priv->info->func(param))
+
+
+/*
+ * basic function
+ */
+u32 rsnd_read(struct rsnd_priv *priv,
+ struct rsnd_mod *mod, enum rsnd_reg reg)
+{
+ void __iomem *base = rsnd_gen_reg_get(priv, mod, reg);
+
+ BUG_ON(!base);
+
+ return ioread32(base);
+}
+
+void rsnd_write(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ enum rsnd_reg reg, u32 data)
+{
+ void __iomem *base = rsnd_gen_reg_get(priv, mod, reg);
+ struct device *dev = rsnd_priv_to_dev(priv);
+
+ BUG_ON(!base);
+
+ dev_dbg(dev, "w %p : %08x\n", base, data);
+
+ iowrite32(data, base);
+}
+
+void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod,
+ enum rsnd_reg reg, u32 mask, u32 data)
+{
+ void __iomem *base = rsnd_gen_reg_get(priv, mod, reg);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ u32 val;
+
+ BUG_ON(!base);
+
+ val = ioread32(base);
+ val &= ~mask;
+ val |= data & mask;
+ iowrite32(val, base);
+
+ dev_dbg(dev, "s %p : %08x\n", base, val);
+}
+
+/*
+ * rsnd_mod functions
+ */
+char *rsnd_mod_name(struct rsnd_mod *mod)
+{
+ if (!mod || !mod->ops)
+ return "unknown";
+
+ return mod->ops->name;
+}
+
+void rsnd_mod_init(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ struct rsnd_mod_ops *ops,
+ int id)
+{
+ mod->priv = priv;
+ mod->id = id;
+ mod->ops = ops;
+ INIT_LIST_HEAD(&mod->list);
+}
+
+/*
+ * rsnd_dma functions
+ */
+static void rsnd_dma_continue(struct rsnd_dma *dma)
+{
+ /* push next A or B plane */
+ dma->submit_loop = 1;
+ schedule_work(&dma->work);
+}
+
+void rsnd_dma_start(struct rsnd_dma *dma)
+{
+ /* push both A and B plane*/
+ dma->submit_loop = 2;
+ schedule_work(&dma->work);
+}
+
+void rsnd_dma_stop(struct rsnd_dma *dma)
+{
+ dma->submit_loop = 0;
+ cancel_work_sync(&dma->work);
+ dmaengine_terminate_all(dma->chan);
+}
+
+static void rsnd_dma_complete(void *data)
+{
+ struct rsnd_dma *dma = (struct rsnd_dma *)data;
+ struct rsnd_priv *priv = dma->priv;
+ unsigned long flags;
+
+ rsnd_lock(priv, flags);
+
+ dma->complete(dma);
+
+ if (dma->submit_loop)
+ rsnd_dma_continue(dma);
+
+ rsnd_unlock(priv, flags);
+}
+
+static void rsnd_dma_do_work(struct work_struct *work)
+{
+ struct rsnd_dma *dma = container_of(work, struct rsnd_dma, work);
+ struct rsnd_priv *priv = dma->priv;
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct dma_async_tx_descriptor *desc;
+ dma_addr_t buf;
+ size_t len;
+ int i;
+
+ for (i = 0; i < dma->submit_loop; i++) {
+
+ if (dma->inquiry(dma, &buf, &len) < 0)
+ return;
+
+ desc = dmaengine_prep_slave_single(
+ dma->chan, buf, len, dma->dir,
+ DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+ if (!desc) {
+ dev_err(dev, "dmaengine_prep_slave_sg() fail\n");
+ return;
+ }
+
+ desc->callback = rsnd_dma_complete;
+ desc->callback_param = dma;
+
+ if (dmaengine_submit(desc) < 0) {
+ dev_err(dev, "dmaengine_submit() fail\n");
+ return;
+ }
+
+ }
+
+ dma_async_issue_pending(dma->chan);
+}
+
+int rsnd_dma_available(struct rsnd_dma *dma)
+{
+ return !!dma->chan;
+}
+
+static bool rsnd_dma_filter(struct dma_chan *chan, void *param)
+{
+ chan->private = param;
+
+ return true;
+}
+
+int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
+ int is_play, int id,
+ int (*inquiry)(struct rsnd_dma *dma,
+ dma_addr_t *buf, int *len),
+ int (*complete)(struct rsnd_dma *dma))
+{
+ struct device *dev = rsnd_priv_to_dev(priv);
+ dma_cap_mask_t mask;
+
+ if (dma->chan) {
+ dev_err(dev, "it already has dma channel\n");
+ return -EIO;
+ }
+
+ dma_cap_zero(mask);
+ dma_cap_set(DMA_SLAVE, mask);
+
+ dma->slave.shdma_slave.slave_id = id;
+
+ dma->chan = dma_request_channel(mask, rsnd_dma_filter,
+ &dma->slave.shdma_slave);
+ if (!dma->chan) {
+ dev_err(dev, "can't get dma channel\n");
+ return -EIO;
+ }
+
+ dma->dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
+ dma->priv = priv;
+ dma->inquiry = inquiry;
+ dma->complete = complete;
+ INIT_WORK(&dma->work, rsnd_dma_do_work);
+
+ return 0;
+}
+
+void rsnd_dma_quit(struct rsnd_priv *priv,
+ struct rsnd_dma *dma)
+{
+ if (dma->chan)
+ dma_release_channel(dma->chan);
+
+ dma->chan = NULL;
+}
+
+/*
+ * rsnd_dai functions
+ */
+#define rsnd_dai_call(rdai, io, fn) \
+({ \
+ struct rsnd_mod *mod, *n; \
+ int ret = 0; \
+ for_each_rsnd_mod(mod, n, io) { \
+ ret = rsnd_mod_call(mod, fn, rdai, io); \
+ if (ret < 0) \
+ break; \
+ } \
+ ret; \
+})
+
+int rsnd_dai_connect(struct rsnd_dai *rdai,
+ struct rsnd_mod *mod,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+
+ if (!mod) {
+ dev_err(dev, "NULL mod\n");
+ return -EIO;
+ }
+
+ if (!list_empty(&mod->list)) {
+ dev_err(dev, "%s%d is not empty\n",
+ rsnd_mod_name(mod),
+ rsnd_mod_id(mod));
+ return -EIO;
+ }
+
+ list_add_tail(&mod->list, &io->head);
+
+ return 0;
+}
+
+int rsnd_dai_disconnect(struct rsnd_mod *mod)
+{
+ list_del_init(&mod->list);
+
+ return 0;
+}
+
+int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai)
+{
+ int id = rdai - priv->rdai;
+
+ if ((id < 0) || (id >= rsnd_dai_nr(priv)))
+ return -EINVAL;
+
+ return id;
+}
+
+struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id)
+{
+ return priv->rdai + id;
+}
+
+static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai)
+{
+ struct rsnd_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+ return rsnd_dai_get(priv, dai->id);
+}
+
+int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io)
+{
+ return &rdai->playback == io;
+}
+
+/*
+ * rsnd_soc_dai functions
+ */
+int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional)
+{
+ struct snd_pcm_substream *substream = io->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int pos = io->byte_pos + additional;
+
+ pos %= (runtime->periods * io->byte_per_period);
+
+ return pos;
+}
+
+void rsnd_dai_pointer_update(struct rsnd_dai_stream *io, int byte)
+{
+ io->byte_pos += byte;
+
+ if (io->byte_pos >= io->next_period_byte) {
+ struct snd_pcm_substream *substream = io->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ io->period_pos++;
+ io->next_period_byte += io->byte_per_period;
+
+ if (io->period_pos >= runtime->periods) {
+ io->byte_pos = 0;
+ io->period_pos = 0;
+ io->next_period_byte = io->byte_per_period;
+ }
+
+ snd_pcm_period_elapsed(substream);
+ }
+}
+
+static int rsnd_dai_stream_init(struct rsnd_dai_stream *io,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (!list_empty(&io->head))
+ return -EIO;
+
+ INIT_LIST_HEAD(&io->head);
+ io->substream = substream;
+ io->byte_pos = 0;
+ io->period_pos = 0;
+ io->byte_per_period = runtime->period_size *
+ runtime->channels *
+ samples_to_bytes(runtime, 1);
+ io->next_period_byte = io->byte_per_period;
+
+ return 0;
+}
+
+static
+struct snd_soc_dai *rsnd_substream_to_dai(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ return rtd->cpu_dai;
+}
+
+static
+struct rsnd_dai_stream *rsnd_rdai_to_io(struct rsnd_dai *rdai,
+ struct snd_pcm_substream *substream)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return &rdai->playback;
+ else
+ return &rdai->capture;
+}
+
+static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct rsnd_priv *priv = snd_soc_dai_get_drvdata(dai);
+ struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
+ struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream);
+ struct rsnd_mod *mod = rsnd_ssi_mod_get_frm_dai(priv,
+ rsnd_dai_id(priv, rdai),
+ rsnd_dai_is_play(rdai, io));
+ int ssi_id = rsnd_mod_id(mod);
+ int ret;
+ unsigned long flags;
+
+ rsnd_lock(priv, flags);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ ret = rsnd_dai_stream_init(io, substream);
+ if (ret < 0)
+ goto dai_trigger_end;
+
+ ret = rsnd_platform_call(priv, dai, start, ssi_id);
+ if (ret < 0)
+ goto dai_trigger_end;
+
+ ret = rsnd_gen_path_init(priv, rdai, io);
+ if (ret < 0)
+ goto dai_trigger_end;
+
+ ret = rsnd_dai_call(rdai, io, init);
+ if (ret < 0)
+ goto dai_trigger_end;
+
+ ret = rsnd_dai_call(rdai, io, start);
+ if (ret < 0)
+ goto dai_trigger_end;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ ret = rsnd_dai_call(rdai, io, stop);
+ if (ret < 0)
+ goto dai_trigger_end;
+
+ ret = rsnd_dai_call(rdai, io, quit);
+ if (ret < 0)
+ goto dai_trigger_end;
+
+ ret = rsnd_gen_path_exit(priv, rdai, io);
+ if (ret < 0)
+ goto dai_trigger_end;
+
+ ret = rsnd_platform_call(priv, dai, stop, ssi_id);
+ if (ret < 0)
+ goto dai_trigger_end;
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+dai_trigger_end:
+ rsnd_unlock(priv, flags);
+
+ return ret;
+}
+
+static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ rdai->clk_master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ rdai->clk_master = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_IF:
+ rdai->bit_clk_inv = 0;
+ rdai->frm_clk_inv = 1;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ rdai->bit_clk_inv = 1;
+ rdai->frm_clk_inv = 0;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ rdai->bit_clk_inv = 1;
+ rdai->frm_clk_inv = 1;
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ default:
+ rdai->bit_clk_inv = 0;
+ rdai->frm_clk_inv = 0;
+ break;
+ }
+
+ /* set format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ rdai->sys_delay = 0;
+ rdai->data_alignment = 0;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ rdai->sys_delay = 1;
+ rdai->data_alignment = 0;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ rdai->sys_delay = 1;
+ rdai->data_alignment = 1;
+ break;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops rsnd_soc_dai_ops = {
+ .trigger = rsnd_soc_dai_trigger,
+ .set_fmt = rsnd_soc_dai_set_fmt,
+};
+
+static int rsnd_dai_probe(struct platform_device *pdev,
+ struct rcar_snd_info *info,
+ struct rsnd_priv *priv)
+{
+ struct snd_soc_dai_driver *drv;
+ struct rsnd_dai *rdai;
+ struct rsnd_mod *pmod, *cmod;
+ struct device *dev = rsnd_priv_to_dev(priv);
+ int dai_nr;
+ int i;
+
+ /* get max dai nr */
+ for (dai_nr = 0; dai_nr < 32; dai_nr++) {
+ pmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 1);
+ cmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 0);
+
+ if (!pmod && !cmod)
+ break;
+ }
+
+ if (!dai_nr) {
+ dev_err(dev, "no dai\n");
+ return -EIO;
+ }
+
+ drv = devm_kzalloc(dev, sizeof(*drv) * dai_nr, GFP_KERNEL);
+ rdai = devm_kzalloc(dev, sizeof(*rdai) * dai_nr, GFP_KERNEL);
+ if (!drv || !rdai) {
+ dev_err(dev, "dai allocate failed\n");
+ return -ENOMEM;
+ }
+
+ for (i = 0; i < dai_nr; i++) {
+
+ pmod = rsnd_ssi_mod_get_frm_dai(priv, i, 1);
+ cmod = rsnd_ssi_mod_get_frm_dai(priv, i, 0);
+
+ /*
+ * init rsnd_dai
+ */
+ INIT_LIST_HEAD(&rdai[i].playback.head);
+ INIT_LIST_HEAD(&rdai[i].capture.head);
+
+ snprintf(rdai[i].name, RSND_DAI_NAME_SIZE, "rsnd-dai.%d", i);
+
+ /*
+ * init snd_soc_dai_driver
+ */
+ drv[i].name = rdai[i].name;
+ drv[i].ops = &rsnd_soc_dai_ops;
+ if (pmod) {
+ drv[i].playback.rates = RSND_RATES;
+ drv[i].playback.formats = RSND_FMTS;
+ drv[i].playback.channels_min = 2;
+ drv[i].playback.channels_max = 2;
+ }
+ if (cmod) {
+ drv[i].capture.rates = RSND_RATES;
+ drv[i].capture.formats = RSND_FMTS;
+ drv[i].capture.channels_min = 2;
+ drv[i].capture.channels_max = 2;
+ }
+
+ dev_dbg(dev, "%s (%s/%s)\n", rdai[i].name,
+ pmod ? "play" : " -- ",
+ cmod ? "capture" : " -- ");
+ }
+
+ priv->dai_nr = dai_nr;
+ priv->daidrv = drv;
+ priv->rdai = rdai;
+
+ return 0;
+}
+
+static void rsnd_dai_remove(struct platform_device *pdev,
+ struct rsnd_priv *priv)
+{
+}
+
+/*
+ * pcm ops
+ */
+static struct snd_pcm_hardware rsnd_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE,
+ .formats = RSND_FMTS,
+ .rates = RSND_RATES,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 64 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8192,
+ .periods_min = 1,
+ .periods_max = 32,
+ .fifo_size = 256,
+};
+
+static int rsnd_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ret = 0;
+
+ snd_soc_set_runtime_hwparams(substream, &rsnd_pcm_hardware);
+
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+
+ return ret;
+}
+
+static int rsnd_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static snd_pcm_uframes_t rsnd_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_dai *dai = rsnd_substream_to_dai(substream);
+ struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
+ struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream);
+
+ return bytes_to_frames(runtime, io->byte_pos);
+}
+
+static struct snd_pcm_ops rsnd_pcm_ops = {
+ .open = rsnd_pcm_open,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = rsnd_hw_params,
+ .hw_free = snd_pcm_lib_free_pages,
+ .pointer = rsnd_pointer,
+};
+
+/*
+ * snd_soc_platform
+ */
+
+#define PREALLOC_BUFFER (32 * 1024)
+#define PREALLOC_BUFFER_MAX (32 * 1024)
+
+static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ return snd_pcm_lib_preallocate_pages_for_all(
+ rtd->pcm,
+ SNDRV_DMA_TYPE_DEV,
+ rtd->card->snd_card->dev,
+ PREALLOC_BUFFER, PREALLOC_BUFFER_MAX);
+}
+
+static void rsnd_pcm_free(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static struct snd_soc_platform_driver rsnd_soc_platform = {
+ .ops = &rsnd_pcm_ops,
+ .pcm_new = rsnd_pcm_new,
+ .pcm_free = rsnd_pcm_free,
+};
+
+static const struct snd_soc_component_driver rsnd_soc_component = {
+ .name = "rsnd",
+};
+
+/*
+ * rsnd probe
+ */
+static int rsnd_probe(struct platform_device *pdev)
+{
+ struct rcar_snd_info *info;
+ struct rsnd_priv *priv;
+ struct device *dev = &pdev->dev;
+ int ret;
+
+ info = pdev->dev.platform_data;
+ if (!info) {
+ dev_err(dev, "driver needs R-Car sound information\n");
+ return -ENODEV;
+ }
+
+ /*
+ * init priv data
+ */
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv) {
+ dev_err(dev, "priv allocate failed\n");
+ return -ENODEV;
+ }
+
+ priv->dev = dev;
+ priv->info = info;
+ spin_lock_init(&priv->lock);
+
+ /*
+ * init each module
+ */
+ ret = rsnd_gen_probe(pdev, info, priv);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_scu_probe(pdev, info, priv);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_adg_probe(pdev, info, priv);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_ssi_probe(pdev, info, priv);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_dai_probe(pdev, info, priv);
+ if (ret < 0)
+ return ret;
+
+ /*
+ * asoc register
+ */
+ ret = snd_soc_register_platform(dev, &rsnd_soc_platform);
+ if (ret < 0) {
+ dev_err(dev, "cannot snd soc register\n");
+ return ret;
+ }
+
+ ret = snd_soc_register_component(dev, &rsnd_soc_component,
+ priv->daidrv, rsnd_dai_nr(priv));
+ if (ret < 0) {
+ dev_err(dev, "cannot snd dai register\n");
+ goto exit_snd_soc;
+ }
+
+ dev_set_drvdata(dev, priv);
+
+ pm_runtime_enable(dev);
+
+ dev_info(dev, "probed\n");
+ return ret;
+
+exit_snd_soc:
+ snd_soc_unregister_platform(dev);
+
+ return ret;
+}
+
+static int rsnd_remove(struct platform_device *pdev)
+{
+ struct rsnd_priv *priv = dev_get_drvdata(&pdev->dev);
+
+ pm_runtime_disable(&pdev->dev);
+
+ /*
+ * remove each module
+ */
+ rsnd_ssi_remove(pdev, priv);
+ rsnd_adg_remove(pdev, priv);
+ rsnd_scu_remove(pdev, priv);
+ rsnd_dai_remove(pdev, priv);
+ rsnd_gen_remove(pdev, priv);
+
+ return 0;
+}
+
+static struct platform_driver rsnd_driver = {
+ .driver = {
+ .name = "rcar_sound",
+ },
+ .probe = rsnd_probe,
+ .remove = rsnd_remove,
+};
+module_platform_driver(rsnd_driver);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Renesas R-Car audio driver");
+MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
+MODULE_ALIAS("platform:rcar-pcm-audio");
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
new file mode 100644
index 0000000..babb203
--- /dev/null
+++ b/sound/soc/sh/rcar/gen.c
@@ -0,0 +1,280 @@
+/*
+ * Renesas R-Car Gen1 SRU/SSI support
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#include "rsnd.h"
+
+struct rsnd_gen_ops {
+ int (*path_init)(struct rsnd_priv *priv,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
+ int (*path_exit)(struct rsnd_priv *priv,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
+};
+
+struct rsnd_gen_reg_map {
+ int index; /* -1 : not supported */
+ u32 offset_id; /* offset of ssi0, ssi1, ssi2... */
+ u32 offset_adr; /* offset of SSICR, SSISR, ... */
+};
+
+struct rsnd_gen {
+ void __iomem *base[RSND_BASE_MAX];
+
+ struct rsnd_gen_reg_map reg_map[RSND_REG_MAX];
+ struct rsnd_gen_ops *ops;
+};
+
+#define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen)
+
+/*
+ * Gen2
+ * will be filled in the future
+ */
+
+/*
+ * Gen1
+ */
+static int rsnd_gen1_path_init(struct rsnd_priv *priv,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_mod *mod;
+ int ret;
+ int id;
+
+ /*
+ * Gen1 is created by SRU/SSI, and this SRU is base module of
+ * Gen2's SCU/SSIU/SSI. (Gen2 SCU/SSIU came from SRU)
+ *
+ * Easy image is..
+ * Gen1 SRU = Gen2 SCU + SSIU + etc
+ *
+ * Gen2 SCU path is very flexible, but, Gen1 SRU (SCU parts) is
+ * using fixed path.
+ *
+ * Then, SSI id = SCU id here
+ */
+
+ /* get SSI's ID */
+ mod = rsnd_ssi_mod_get_frm_dai(priv,
+ rsnd_dai_id(priv, rdai),
+ rsnd_dai_is_play(rdai, io));
+ id = rsnd_mod_id(mod);
+
+ /* SSI */
+ mod = rsnd_ssi_mod_get(priv, id);
+ ret = rsnd_dai_connect(rdai, mod, io);
+ if (ret < 0)
+ return ret;
+
+ /* SCU */
+ mod = rsnd_scu_mod_get(priv, id);
+ ret = rsnd_dai_connect(rdai, mod, io);
+
+ return ret;
+}
+
+static int rsnd_gen1_path_exit(struct rsnd_priv *priv,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_mod *mod, *n;
+ int ret = 0;
+
+ /*
+ * remove all mod from rdai
+ */
+ for_each_rsnd_mod(mod, n, io)
+ ret |= rsnd_dai_disconnect(mod);
+
+ return ret;
+}
+
+static struct rsnd_gen_ops rsnd_gen1_ops = {
+ .path_init = rsnd_gen1_path_init,
+ .path_exit = rsnd_gen1_path_exit,
+};
+
+#define RSND_GEN1_REG_MAP(g, s, i, oi, oa) \
+ do { \
+ (g)->reg_map[RSND_REG_##i].index = RSND_GEN1_##s; \
+ (g)->reg_map[RSND_REG_##i].offset_id = oi; \
+ (g)->reg_map[RSND_REG_##i].offset_adr = oa; \
+ } while (0)
+
+static void rsnd_gen1_reg_map_init(struct rsnd_gen *gen)
+{
+ RSND_GEN1_REG_MAP(gen, SRU, SRC_ROUTE_SEL, 0x0, 0x00);
+ RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL0, 0x0, 0x08);
+ RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL1, 0x0, 0x0c);
+ RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL2, 0x0, 0x10);
+ RSND_GEN1_REG_MAP(gen, SRU, SRC_CTRL, 0x0, 0xc0);
+ RSND_GEN1_REG_MAP(gen, SRU, SSI_MODE0, 0x0, 0xD0);
+ RSND_GEN1_REG_MAP(gen, SRU, SSI_MODE1, 0x0, 0xD4);
+ RSND_GEN1_REG_MAP(gen, SRU, BUSIF_MODE, 0x4, 0x20);
+ RSND_GEN1_REG_MAP(gen, SRU, BUSIF_ADINR, 0x40, 0x214);
+
+ RSND_GEN1_REG_MAP(gen, ADG, BRRA, 0x0, 0x00);
+ RSND_GEN1_REG_MAP(gen, ADG, BRRB, 0x0, 0x04);
+ RSND_GEN1_REG_MAP(gen, ADG, SSICKR, 0x0, 0x08);
+ RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL0, 0x0, 0x0c);
+ RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL1, 0x0, 0x10);
+ RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL3, 0x0, 0x18);
+ RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL4, 0x0, 0x1c);
+ RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL5, 0x0, 0x20);
+
+ RSND_GEN1_REG_MAP(gen, SSI, SSICR, 0x40, 0x00);
+ RSND_GEN1_REG_MAP(gen, SSI, SSISR, 0x40, 0x04);
+ RSND_GEN1_REG_MAP(gen, SSI, SSITDR, 0x40, 0x08);
+ RSND_GEN1_REG_MAP(gen, SSI, SSIRDR, 0x40, 0x0c);
+ RSND_GEN1_REG_MAP(gen, SSI, SSIWSR, 0x40, 0x20);
+}
+
+static int rsnd_gen1_probe(struct platform_device *pdev,
+ struct rcar_snd_info *info,
+ struct rsnd_priv *priv)
+{
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
+ struct resource *sru_res;
+ struct resource *adg_res;
+ struct resource *ssi_res;
+
+ /*
+ * map address
+ */
+ sru_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SRU);
+ adg_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_ADG);
+ ssi_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SSI);
+
+ gen->base[RSND_GEN1_SRU] = devm_ioremap_resource(dev, sru_res);
+ gen->base[RSND_GEN1_ADG] = devm_ioremap_resource(dev, adg_res);
+ gen->base[RSND_GEN1_SSI] = devm_ioremap_resource(dev, ssi_res);
+ if (IS_ERR(gen->base[RSND_GEN1_SRU]) ||
+ IS_ERR(gen->base[RSND_GEN1_ADG]) ||
+ IS_ERR(gen->base[RSND_GEN1_SSI]))
+ return -ENODEV;
+
+ gen->ops = &rsnd_gen1_ops;
+ rsnd_gen1_reg_map_init(gen);
+
+ dev_dbg(dev, "Gen1 device probed\n");
+ dev_dbg(dev, "SRU : %08x => %p\n", sru_res->start,
+ gen->base[RSND_GEN1_SRU]);
+ dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start,
+ gen->base[RSND_GEN1_ADG]);
+ dev_dbg(dev, "SSI : %08x => %p\n", ssi_res->start,
+ gen->base[RSND_GEN1_SSI]);
+
+ return 0;
+
+}
+
+static void rsnd_gen1_remove(struct platform_device *pdev,
+ struct rsnd_priv *priv)
+{
+}
+
+/*
+ * Gen
+ */
+int rsnd_gen_path_init(struct rsnd_priv *priv,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
+
+ return gen->ops->path_init(priv, rdai, io);
+}
+
+int rsnd_gen_path_exit(struct rsnd_priv *priv,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
+
+ return gen->ops->path_exit(priv, rdai, io);
+}
+
+void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ enum rsnd_reg reg)
+{
+ struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ int index;
+ u32 offset_id, offset_adr;
+
+ if (reg >= RSND_REG_MAX) {
+ dev_err(dev, "rsnd_reg reg error\n");
+ return NULL;
+ }
+
+ index = gen->reg_map[reg].index;
+ offset_id = gen->reg_map[reg].offset_id;
+ offset_adr = gen->reg_map[reg].offset_adr;
+
+ if (index < 0) {
+ dev_err(dev, "unsupported reg access %d\n", reg);
+ return NULL;
+ }
+
+ if (offset_id && mod)
+ offset_id *= rsnd_mod_id(mod);
+
+ /*
+ * index/offset were set on gen1/gen2
+ */
+
+ return gen->base[index] + offset_id + offset_adr;
+}
+
+int rsnd_gen_probe(struct platform_device *pdev,
+ struct rcar_snd_info *info,
+ struct rsnd_priv *priv)
+{
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct rsnd_gen *gen;
+ int i;
+
+ gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL);
+ if (!gen) {
+ dev_err(dev, "GEN allocate failed\n");
+ return -ENOMEM;
+ }
+
+ priv->gen = gen;
+
+ /*
+ * see
+ * rsnd_reg_get()
+ * rsnd_gen_probe()
+ */
+ for (i = 0; i < RSND_REG_MAX; i++)
+ gen->reg_map[i].index = -1;
+
+ /*
+ * init each module
+ */
+ if (rsnd_is_gen1(priv))
+ return rsnd_gen1_probe(pdev, info, priv);
+
+ dev_err(dev, "unknown generation R-Car sound device\n");
+
+ return -ENODEV;
+}
+
+void rsnd_gen_remove(struct platform_device *pdev,
+ struct rsnd_priv *priv)
+{
+ if (rsnd_is_gen1(priv))
+ rsnd_gen1_remove(pdev, priv);
+}
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
new file mode 100644
index 0000000..9cc6986
--- /dev/null
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -0,0 +1,302 @@
+/*
+ * Renesas R-Car
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#ifndef RSND_H
+#define RSND_H
+
+#include <linux/clk.h>
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+#include <linux/io.h>
+#include <linux/list.h>
+#include <linux/module.h>
+#include <linux/sh_dma.h>
+#include <linux/workqueue.h>
+#include <sound/rcar_snd.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+/*
+ * pseudo register
+ *
+ * The register address offsets SRU/SCU/SSIU on Gen1/Gen2 are very different.
+ * This driver uses pseudo register in order to hide it.
+ * see gen1/gen2 for detail
+ */
+enum rsnd_reg {
+ /* SRU/SCU */
+ RSND_REG_SRC_ROUTE_SEL,
+ RSND_REG_SRC_TMG_SEL0,
+ RSND_REG_SRC_TMG_SEL1,
+ RSND_REG_SRC_TMG_SEL2,
+ RSND_REG_SRC_CTRL,
+ RSND_REG_SSI_MODE0,
+ RSND_REG_SSI_MODE1,
+ RSND_REG_BUSIF_MODE,
+ RSND_REG_BUSIF_ADINR,
+
+ /* ADG */
+ RSND_REG_BRRA,
+ RSND_REG_BRRB,
+ RSND_REG_SSICKR,
+ RSND_REG_AUDIO_CLK_SEL0,
+ RSND_REG_AUDIO_CLK_SEL1,
+ RSND_REG_AUDIO_CLK_SEL2,
+ RSND_REG_AUDIO_CLK_SEL3,
+ RSND_REG_AUDIO_CLK_SEL4,
+ RSND_REG_AUDIO_CLK_SEL5,
+
+ /* SSI */
+ RSND_REG_SSICR,
+ RSND_REG_SSISR,
+ RSND_REG_SSITDR,
+ RSND_REG_SSIRDR,
+ RSND_REG_SSIWSR,
+
+ RSND_REG_MAX,
+};
+
+struct rsnd_priv;
+struct rsnd_mod;
+struct rsnd_dai;
+struct rsnd_dai_stream;
+
+/*
+ * R-Car basic functions
+ */
+#define rsnd_mod_read(m, r) \
+ rsnd_read(rsnd_mod_to_priv(m), m, RSND_REG_##r)
+#define rsnd_mod_write(m, r, d) \
+ rsnd_write(rsnd_mod_to_priv(m), m, RSND_REG_##r, d)
+#define rsnd_mod_bset(m, r, s, d) \
+ rsnd_bset(rsnd_mod_to_priv(m), m, RSND_REG_##r, s, d)
+
+#define rsnd_priv_read(p, r) rsnd_read(p, NULL, RSND_REG_##r)
+#define rsnd_priv_write(p, r, d) rsnd_write(p, NULL, RSND_REG_##r, d)
+#define rsnd_priv_bset(p, r, s, d) rsnd_bset(p, NULL, RSND_REG_##r, s, d)
+
+u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg);
+void rsnd_write(struct rsnd_priv *priv, struct rsnd_mod *mod,
+ enum rsnd_reg reg, u32 data);
+void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg,
+ u32 mask, u32 data);
+
+/*
+ * R-Car DMA
+ */
+struct rsnd_dma {
+ struct rsnd_priv *priv;
+ struct sh_dmae_slave slave;
+ struct work_struct work;
+ struct dma_chan *chan;
+ enum dma_data_direction dir;
+ int (*inquiry)(struct rsnd_dma *dma, dma_addr_t *buf, int *len);
+ int (*complete)(struct rsnd_dma *dma);
+
+ int submit_loop;
+};
+
+void rsnd_dma_start(struct rsnd_dma *dma);
+void rsnd_dma_stop(struct rsnd_dma *dma);
+int rsnd_dma_available(struct rsnd_dma *dma);
+int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
+ int is_play, int id,
+ int (*inquiry)(struct rsnd_dma *dma, dma_addr_t *buf, int *len),
+ int (*complete)(struct rsnd_dma *dma));
+void rsnd_dma_quit(struct rsnd_priv *priv,
+ struct rsnd_dma *dma);
+
+
+/*
+ * R-Car sound mod
+ */
+
+struct rsnd_mod_ops {
+ char *name;
+ int (*init)(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
+ int (*quit)(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
+ int (*start)(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
+ int (*stop)(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
+};
+
+struct rsnd_mod {
+ int id;
+ struct rsnd_priv *priv;
+ struct rsnd_mod_ops *ops;
+ struct list_head list; /* connect to rsnd_dai playback/capture */
+ struct rsnd_dma dma;
+};
+
+#define rsnd_mod_to_priv(mod) ((mod)->priv)
+#define rsnd_mod_to_dma(mod) (&(mod)->dma)
+#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma)
+#define rsnd_mod_id(mod) ((mod)->id)
+#define for_each_rsnd_mod(pos, n, io) \
+ list_for_each_entry_safe(pos, n, &(io)->head, list)
+#define rsnd_mod_call(mod, func, rdai, io) \
+ (!(mod) ? -ENODEV : \
+ !((mod)->ops->func) ? 0 : \
+ (mod)->ops->func(mod, rdai, io))
+
+void rsnd_mod_init(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ struct rsnd_mod_ops *ops,
+ int id);
+char *rsnd_mod_name(struct rsnd_mod *mod);
+
+/*
+ * R-Car sound DAI
+ */
+#define RSND_DAI_NAME_SIZE 16
+struct rsnd_dai_stream {
+ struct list_head head; /* head of rsnd_mod list */
+ struct snd_pcm_substream *substream;
+ int byte_pos;
+ int period_pos;
+ int byte_per_period;
+ int next_period_byte;
+};
+
+struct rsnd_dai {
+ char name[RSND_DAI_NAME_SIZE];
+ struct rsnd_dai_platform_info *info; /* rcar_snd.h */
+ struct rsnd_dai_stream playback;
+ struct rsnd_dai_stream capture;
+
+ int clk_master:1;
+ int bit_clk_inv:1;
+ int frm_clk_inv:1;
+ int sys_delay:1;
+ int data_alignment:1;
+};
+
+#define rsnd_dai_nr(priv) ((priv)->dai_nr)
+#define for_each_rsnd_dai(rdai, priv, i) \
+ for (i = 0, (rdai) = rsnd_dai_get(priv, i); \
+ i < rsnd_dai_nr(priv); \
+ i++, (rdai) = rsnd_dai_get(priv, i))
+
+struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id);
+int rsnd_dai_disconnect(struct rsnd_mod *mod);
+int rsnd_dai_connect(struct rsnd_dai *rdai, struct rsnd_mod *mod,
+ struct rsnd_dai_stream *io);
+int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io);
+int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai);
+#define rsnd_dai_get_platform_info(rdai) ((rdai)->info)
+#define rsnd_io_to_runtime(io) ((io)->substream->runtime)
+
+void rsnd_dai_pointer_update(struct rsnd_dai_stream *io, int cnt);
+int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional);
+
+/*
+ * R-Car Gen1/Gen2
+ */
+int rsnd_gen_probe(struct platform_device *pdev,
+ struct rcar_snd_info *info,
+ struct rsnd_priv *priv);
+void rsnd_gen_remove(struct platform_device *pdev,
+ struct rsnd_priv *priv);
+int rsnd_gen_path_init(struct rsnd_priv *priv,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
+int rsnd_gen_path_exit(struct rsnd_priv *priv,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
+void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ enum rsnd_reg reg);
+#define rsnd_is_gen1(s) ((s)->info->flags & RSND_GEN1)
+#define rsnd_is_gen2(s) ((s)->info->flags & RSND_GEN2)
+
+/*
+ * R-Car ADG
+ */
+int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod);
+int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate);
+int rsnd_adg_probe(struct platform_device *pdev,
+ struct rcar_snd_info *info,
+ struct rsnd_priv *priv);
+void rsnd_adg_remove(struct platform_device *pdev,
+ struct rsnd_priv *priv);
+
+/*
+ * R-Car sound priv
+ */
+struct rsnd_priv {
+
+ struct device *dev;
+ struct rcar_snd_info *info;
+ spinlock_t lock;
+
+ /*
+ * below value will be filled on rsnd_gen_probe()
+ */
+ void *gen;
+
+ /*
+ * below value will be filled on rsnd_scu_probe()
+ */
+ void *scu;
+ int scu_nr;
+
+ /*
+ * below value will be filled on rsnd_adg_probe()
+ */
+ void *adg;
+
+ /*
+ * below value will be filled on rsnd_ssi_probe()
+ */
+ void *ssiu;
+
+ /*
+ * below value will be filled on rsnd_dai_probe()
+ */
+ struct snd_soc_dai_driver *daidrv;
+ struct rsnd_dai *rdai;
+ int dai_nr;
+};
+
+#define rsnd_priv_to_dev(priv) ((priv)->dev)
+#define rsnd_lock(priv, flags) spin_lock_irqsave(&priv->lock, flags)
+#define rsnd_unlock(priv, flags) spin_unlock_irqrestore(&priv->lock, flags)
+
+/*
+ * R-Car SCU
+ */
+int rsnd_scu_probe(struct platform_device *pdev,
+ struct rcar_snd_info *info,
+ struct rsnd_priv *priv);
+void rsnd_scu_remove(struct platform_device *pdev,
+ struct rsnd_priv *priv);
+struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id);
+#define rsnd_scu_nr(priv) ((priv)->scu_nr)
+
+/*
+ * R-Car SSI
+ */
+int rsnd_ssi_probe(struct platform_device *pdev,
+ struct rcar_snd_info *info,
+ struct rsnd_priv *priv);
+void rsnd_ssi_remove(struct platform_device *pdev,
+ struct rsnd_priv *priv);
+struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id);
+struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv,
+ int dai_id, int is_play);
+
+#endif
diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c
new file mode 100644
index 0000000..184d9008
--- /dev/null
+++ b/sound/soc/sh/rcar/scu.c
@@ -0,0 +1,236 @@
+/*
+ * Renesas R-Car SCU support
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#include "rsnd.h"
+
+struct rsnd_scu {
+ struct rsnd_scu_platform_info *info; /* rcar_snd.h */
+ struct rsnd_mod mod;
+};
+
+#define rsnd_scu_mode_flags(p) ((p)->info->flags)
+
+/*
+ * ADINR
+ */
+#define OTBL_24 (0 << 16)
+#define OTBL_22 (2 << 16)
+#define OTBL_20 (4 << 16)
+#define OTBL_18 (6 << 16)
+#define OTBL_16 (8 << 16)
+
+
+#define rsnd_mod_to_scu(_mod) \
+ container_of((_mod), struct rsnd_scu, mod)
+
+#define for_each_rsnd_scu(pos, priv, i) \
+ for ((i) = 0; \
+ ((i) < rsnd_scu_nr(priv)) && \
+ ((pos) = (struct rsnd_scu *)(priv)->scu + i); \
+ i++)
+
+static int rsnd_scu_set_route(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct scu_route_config {
+ u32 mask;
+ int shift;
+ } routes[] = {
+ { 0xF, 0, }, /* 0 */
+ { 0xF, 4, }, /* 1 */
+ { 0xF, 8, }, /* 2 */
+ { 0x7, 12, }, /* 3 */
+ { 0x7, 16, }, /* 4 */
+ { 0x7, 20, }, /* 5 */
+ { 0x7, 24, }, /* 6 */
+ { 0x3, 28, }, /* 7 */
+ { 0x3, 30, }, /* 8 */
+ };
+
+ u32 mask;
+ u32 val;
+ int shift;
+ int id;
+
+ /*
+ * Gen1 only
+ */
+ if (!rsnd_is_gen1(priv))
+ return 0;
+
+ id = rsnd_mod_id(mod);
+ if (id < 0 || id > ARRAY_SIZE(routes))
+ return -EIO;
+
+ /*
+ * SRC_ROUTE_SELECT
+ */
+ val = rsnd_dai_is_play(rdai, io) ? 0x1 : 0x2;
+ val = val << routes[id].shift;
+ mask = routes[id].mask << routes[id].shift;
+
+ rsnd_mod_bset(mod, SRC_ROUTE_SEL, mask, val);
+
+ /*
+ * SRC_TIMING_SELECT
+ */
+ shift = (id % 4) * 8;
+ mask = 0x1F << shift;
+ if (8 == id) /* SRU8 is very special */
+ val = id << shift;
+ else
+ val = (id + 1) << shift;
+
+ switch (id / 4) {
+ case 0:
+ rsnd_mod_bset(mod, SRC_TMG_SEL0, mask, val);
+ break;
+ case 1:
+ rsnd_mod_bset(mod, SRC_TMG_SEL1, mask, val);
+ break;
+ case 2:
+ rsnd_mod_bset(mod, SRC_TMG_SEL2, mask, val);
+ break;
+ }
+
+ return 0;
+}
+
+static int rsnd_scu_set_mode(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ int id = rsnd_mod_id(mod);
+ u32 val;
+
+ if (rsnd_is_gen1(priv)) {
+ val = (1 << id);
+ rsnd_mod_bset(mod, SRC_CTRL, val, val);
+ }
+
+ return 0;
+}
+
+static int rsnd_scu_set_hpbif(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+ u32 adinr = runtime->channels;
+
+ switch (runtime->sample_bits) {
+ case 16:
+ adinr |= OTBL_16;
+ break;
+ case 32:
+ adinr |= OTBL_24;
+ break;
+ default:
+ return -EIO;
+ }
+
+ rsnd_mod_write(mod, BUSIF_MODE, 1);
+ rsnd_mod_write(mod, BUSIF_ADINR, adinr);
+
+ return 0;
+}
+
+static int rsnd_scu_start(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ u32 flags = rsnd_scu_mode_flags(scu);
+ int ret;
+
+ /*
+ * SCU will be used if it has RSND_SCU_USB_HPBIF flags
+ */
+ if (!(flags & RSND_SCU_USB_HPBIF)) {
+ /* it use PIO transter */
+ dev_dbg(dev, "%s%d is not used\n",
+ rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+ return 0;
+ }
+
+ /* it use DMA transter */
+ ret = rsnd_scu_set_route(priv, mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_scu_set_mode(priv, mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_scu_set_hpbif(priv, mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ dev_dbg(dev, "%s%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+ return 0;
+}
+
+static struct rsnd_mod_ops rsnd_scu_ops = {
+ .name = "scu",
+ .start = rsnd_scu_start,
+};
+
+struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id)
+{
+ BUG_ON(id < 0 || id >= rsnd_scu_nr(priv));
+
+ return &((struct rsnd_scu *)(priv->scu) + id)->mod;
+}
+
+int rsnd_scu_probe(struct platform_device *pdev,
+ struct rcar_snd_info *info,
+ struct rsnd_priv *priv)
+{
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct rsnd_scu *scu;
+ int i, nr;
+
+ /*
+ * init SCU
+ */
+ nr = info->scu_info_nr;
+ scu = devm_kzalloc(dev, sizeof(*scu) * nr, GFP_KERNEL);
+ if (!scu) {
+ dev_err(dev, "SCU allocate failed\n");
+ return -ENOMEM;
+ }
+
+ priv->scu_nr = nr;
+ priv->scu = scu;
+
+ for_each_rsnd_scu(scu, priv, i) {
+ rsnd_mod_init(priv, &scu->mod,
+ &rsnd_scu_ops, i);
+ scu->info = &info->scu_info[i];
+
+ dev_dbg(dev, "SCU%d probed\n", i);
+ }
+ dev_dbg(dev, "scu probed\n");
+
+ return 0;
+}
+
+void rsnd_scu_remove(struct platform_device *pdev,
+ struct rsnd_priv *priv)
+{
+}
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
new file mode 100644
index 0000000..fae26d3
--- /dev/null
+++ b/sound/soc/sh/rcar/ssi.c
@@ -0,0 +1,728 @@
+/*
+ * Renesas R-Car SSIU/SSI support
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * Based on fsi.c
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#include <linux/delay.h>
+#include "rsnd.h"
+#define RSND_SSI_NAME_SIZE 16
+
+/*
+ * SSICR
+ */
+#define FORCE (1 << 31) /* Fixed */
+#define DMEN (1 << 28) /* DMA Enable */
+#define UIEN (1 << 27) /* Underflow Interrupt Enable */
+#define OIEN (1 << 26) /* Overflow Interrupt Enable */
+#define IIEN (1 << 25) /* Idle Mode Interrupt Enable */
+#define DIEN (1 << 24) /* Data Interrupt Enable */
+
+#define DWL_8 (0 << 19) /* Data Word Length */
+#define DWL_16 (1 << 19) /* Data Word Length */
+#define DWL_18 (2 << 19) /* Data Word Length */
+#define DWL_20 (3 << 19) /* Data Word Length */
+#define DWL_22 (4 << 19) /* Data Word Length */
+#define DWL_24 (5 << 19) /* Data Word Length */
+#define DWL_32 (6 << 19) /* Data Word Length */
+
+#define SWL_32 (3 << 16) /* R/W System Word Length */
+#define SCKD (1 << 15) /* Serial Bit Clock Direction */
+#define SWSD (1 << 14) /* Serial WS Direction */
+#define SCKP (1 << 13) /* Serial Bit Clock Polarity */
+#define SWSP (1 << 12) /* Serial WS Polarity */
+#define SDTA (1 << 10) /* Serial Data Alignment */
+#define DEL (1 << 8) /* Serial Data Delay */
+#define CKDV(v) (v << 4) /* Serial Clock Division Ratio */
+#define TRMD (1 << 1) /* Transmit/Receive Mode Select */
+#define EN (1 << 0) /* SSI Module Enable */
+
+/*
+ * SSISR
+ */
+#define UIRQ (1 << 27) /* Underflow Error Interrupt Status */
+#define OIRQ (1 << 26) /* Overflow Error Interrupt Status */
+#define IIRQ (1 << 25) /* Idle Mode Interrupt Status */
+#define DIRQ (1 << 24) /* Data Interrupt Status Flag */
+
+/*
+ * SSIWSR
+ */
+#define CONT (1 << 8) /* WS Continue Function */
+
+struct rsnd_ssi {
+ struct clk *clk;
+ struct rsnd_ssi_platform_info *info; /* rcar_snd.h */
+ struct rsnd_ssi *parent;
+ struct rsnd_mod mod;
+
+ struct rsnd_dai *rdai;
+ struct rsnd_dai_stream *io;
+ u32 cr_own;
+ u32 cr_clk;
+ u32 cr_etc;
+ int err;
+ int dma_offset;
+ unsigned int usrcnt;
+ unsigned int rate;
+};
+
+struct rsnd_ssiu {
+ u32 ssi_mode0;
+ u32 ssi_mode1;
+
+ int ssi_nr;
+ struct rsnd_ssi *ssi;
+};
+
+#define for_each_rsnd_ssi(pos, priv, i) \
+ for (i = 0; \
+ (i < rsnd_ssi_nr(priv)) && \
+ ((pos) = ((struct rsnd_ssiu *)((priv)->ssiu))->ssi + i); \
+ i++)
+
+#define rsnd_ssi_nr(priv) (((struct rsnd_ssiu *)((priv)->ssiu))->ssi_nr)
+#define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod)
+#define rsnd_dma_to_ssi(dma) rsnd_mod_to_ssi(rsnd_dma_to_mod(dma))
+#define rsnd_ssi_pio_available(ssi) ((ssi)->info->pio_irq > 0)
+#define rsnd_ssi_dma_available(ssi) \
+ rsnd_dma_available(rsnd_mod_to_dma(&(ssi)->mod))
+#define rsnd_ssi_clk_from_parent(ssi) ((ssi)->parent)
+#define rsnd_rdai_is_clk_master(rdai) ((rdai)->clk_master)
+#define rsnd_ssi_mode_flags(p) ((p)->info->flags)
+#define rsnd_ssi_dai_id(ssi) ((ssi)->info->dai_id)
+#define rsnd_ssi_to_ssiu(ssi)\
+ (((struct rsnd_ssiu *)((ssi) - rsnd_mod_id(&(ssi)->mod))) - 1)
+
+static void rsnd_ssi_mode_init(struct rsnd_priv *priv,
+ struct rsnd_ssiu *ssiu)
+{
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct rsnd_ssi *ssi;
+ u32 flags;
+ u32 val;
+ int i;
+
+ /*
+ * SSI_MODE0
+ */
+ ssiu->ssi_mode0 = 0;
+ for_each_rsnd_ssi(ssi, priv, i) {
+ flags = rsnd_ssi_mode_flags(ssi);
+
+ /* see also BUSIF_MODE */
+ if (!(flags & RSND_SSI_DEPENDENT)) {
+ ssiu->ssi_mode0 |= (1 << i);
+ dev_dbg(dev, "SSI%d uses INDEPENDENT mode\n", i);
+ } else {
+ dev_dbg(dev, "SSI%d uses DEPENDENT mode\n", i);
+ }
+ }
+
+ /*
+ * SSI_MODE1
+ */
+#define ssi_parent_set(p, sync, adg, ext) \
+ do { \
+ ssi->parent = ssiu->ssi + p; \
+ if (flags & RSND_SSI_CLK_FROM_ADG) \
+ val = adg; \
+ else \
+ val = ext; \
+ if (flags & RSND_SSI_SYNC) \
+ val |= sync; \
+ } while (0)
+
+ ssiu->ssi_mode1 = 0;
+ for_each_rsnd_ssi(ssi, priv, i) {
+ flags = rsnd_ssi_mode_flags(ssi);
+
+ if (!(flags & RSND_SSI_CLK_PIN_SHARE))
+ continue;
+
+ val = 0;
+ switch (i) {
+ case 1:
+ ssi_parent_set(0, (1 << 4), (0x2 << 0), (0x1 << 0));
+ break;
+ case 2:
+ ssi_parent_set(0, (1 << 4), (0x2 << 2), (0x1 << 2));
+ break;
+ case 4:
+ ssi_parent_set(3, (1 << 20), (0x2 << 16), (0x1 << 16));
+ break;
+ case 8:
+ ssi_parent_set(7, 0, 0, 0);
+ break;
+ }
+
+ ssiu->ssi_mode1 |= val;
+ }
+}
+
+static void rsnd_ssi_mode_set(struct rsnd_ssi *ssi)
+{
+ struct rsnd_ssiu *ssiu = rsnd_ssi_to_ssiu(ssi);
+
+ rsnd_mod_write(&ssi->mod, SSI_MODE0, ssiu->ssi_mode0);
+ rsnd_mod_write(&ssi->mod, SSI_MODE1, ssiu->ssi_mode1);
+}
+
+static void rsnd_ssi_status_check(struct rsnd_mod *mod,
+ u32 bit)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ u32 status;
+ int i;
+
+ for (i = 0; i < 1024; i++) {
+ status = rsnd_mod_read(mod, SSISR);
+ if (status & bit)
+ return;
+
+ udelay(50);
+ }
+
+ dev_warn(dev, "status check failed\n");
+}
+
+static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
+ unsigned int rate)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ int i, j, ret;
+ int adg_clk_div_table[] = {
+ 1, 6, /* see adg.c */
+ };
+ int ssi_clk_mul_table[] = {
+ 1, 2, 4, 8, 16, 6, 12,
+ };
+ unsigned int main_rate;
+
+ /*
+ * Find best clock, and try to start ADG
+ */
+ for (i = 0; i < ARRAY_SIZE(adg_clk_div_table); i++) {
+ for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) {
+
+ /*
+ * this driver is assuming that
+ * system word is 64fs (= 2 x 32bit)
+ * see rsnd_ssi_start()
+ */
+ main_rate = rate / adg_clk_div_table[i]
+ * 32 * 2 * ssi_clk_mul_table[j];
+
+ ret = rsnd_adg_ssi_clk_try_start(&ssi->mod, main_rate);
+ if (0 == ret) {
+ ssi->rate = rate;
+ ssi->cr_clk = FORCE | SWL_32 |
+ SCKD | SWSD | CKDV(j);
+
+ dev_dbg(dev, "ssi%d outputs %u Hz\n",
+ rsnd_mod_id(&ssi->mod), rate);
+
+ return 0;
+ }
+ }
+ }
+
+ dev_err(dev, "unsupported clock rate\n");
+ return -EIO;
+}
+
+static void rsnd_ssi_master_clk_stop(struct rsnd_ssi *ssi)
+{
+ ssi->rate = 0;
+ ssi->cr_clk = 0;
+ rsnd_adg_ssi_clk_stop(&ssi->mod);
+}
+
+static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ u32 cr;
+
+ if (0 == ssi->usrcnt) {
+ clk_enable(ssi->clk);
+
+ if (rsnd_rdai_is_clk_master(rdai)) {
+ struct snd_pcm_runtime *runtime;
+
+ runtime = rsnd_io_to_runtime(io);
+
+ if (rsnd_ssi_clk_from_parent(ssi))
+ rsnd_ssi_hw_start(ssi->parent, rdai, io);
+ else
+ rsnd_ssi_master_clk_start(ssi, runtime->rate);
+ }
+ }
+
+ cr = ssi->cr_own |
+ ssi->cr_clk |
+ ssi->cr_etc |
+ EN;
+
+ rsnd_mod_write(&ssi->mod, SSICR, cr);
+
+ ssi->usrcnt++;
+
+ dev_dbg(dev, "ssi%d hw started\n", rsnd_mod_id(&ssi->mod));
+}
+
+static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi,
+ struct rsnd_dai *rdai)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ u32 cr;
+
+ if (0 == ssi->usrcnt) /* stop might be called without start */
+ return;
+
+ ssi->usrcnt--;
+
+ if (0 == ssi->usrcnt) {
+ /*
+ * disable all IRQ,
+ * and, wait all data was sent
+ */
+ cr = ssi->cr_own |
+ ssi->cr_clk;
+
+ rsnd_mod_write(&ssi->mod, SSICR, cr | EN);
+ rsnd_ssi_status_check(&ssi->mod, DIRQ);
+
+ /*
+ * disable SSI,
+ * and, wait idle state
+ */
+ rsnd_mod_write(&ssi->mod, SSICR, cr); /* disabled all */
+ rsnd_ssi_status_check(&ssi->mod, IIRQ);
+
+ if (rsnd_rdai_is_clk_master(rdai)) {
+ if (rsnd_ssi_clk_from_parent(ssi))
+ rsnd_ssi_hw_stop(ssi->parent, rdai);
+ else
+ rsnd_ssi_master_clk_stop(ssi);
+ }
+
+ clk_disable(ssi->clk);
+ }
+
+ dev_dbg(dev, "ssi%d hw stopped\n", rsnd_mod_id(&ssi->mod));
+}
+
+/*
+ * SSI mod common functions
+ */
+static int rsnd_ssi_init(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+ u32 cr;
+
+ cr = FORCE;
+
+ /*
+ * always use 32bit system word for easy clock calculation.
+ * see also rsnd_ssi_master_clk_enable()
+ */
+ cr |= SWL_32;
+
+ /*
+ * init clock settings for SSICR
+ */
+ switch (runtime->sample_bits) {
+ case 16:
+ cr |= DWL_16;
+ break;
+ case 32:
+ cr |= DWL_24;
+ break;
+ default:
+ return -EIO;
+ }
+
+ if (rdai->bit_clk_inv)
+ cr |= SCKP;
+ if (rdai->frm_clk_inv)
+ cr |= SWSP;
+ if (rdai->data_alignment)
+ cr |= SDTA;
+ if (rdai->sys_delay)
+ cr |= DEL;
+ if (rsnd_dai_is_play(rdai, io))
+ cr |= TRMD;
+
+ /*
+ * set ssi parameter
+ */
+ ssi->rdai = rdai;
+ ssi->io = io;
+ ssi->cr_own = cr;
+ ssi->err = -1; /* ignore 1st error */
+
+ rsnd_ssi_mode_set(ssi);
+
+ dev_dbg(dev, "%s.%d init\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+ return 0;
+}
+
+static int rsnd_ssi_quit(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+
+ dev_dbg(dev, "%s.%d quit\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+ if (ssi->err > 0)
+ dev_warn(dev, "ssi under/over flow err = %d\n", ssi->err);
+
+ ssi->rdai = NULL;
+ ssi->io = NULL;
+ ssi->cr_own = 0;
+ ssi->err = 0;
+
+ return 0;
+}
+
+static void rsnd_ssi_record_error(struct rsnd_ssi *ssi, u32 status)
+{
+ /* under/over flow error */
+ if (status & (UIRQ | OIRQ)) {
+ ssi->err++;
+
+ /* clear error status */
+ rsnd_mod_write(&ssi->mod, SSISR, 0);
+ }
+}
+
+/*
+ * SSI PIO
+ */
+static irqreturn_t rsnd_ssi_pio_interrupt(int irq, void *data)
+{
+ struct rsnd_ssi *ssi = data;
+ struct rsnd_dai_stream *io = ssi->io;
+ u32 status = rsnd_mod_read(&ssi->mod, SSISR);
+ irqreturn_t ret = IRQ_NONE;
+
+ if (io && (status & DIRQ)) {
+ struct rsnd_dai *rdai = ssi->rdai;
+ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+ u32 *buf = (u32 *)(runtime->dma_area +
+ rsnd_dai_pointer_offset(io, 0));
+
+ rsnd_ssi_record_error(ssi, status);
+
+ /*
+ * 8/16/32 data can be assesse to TDR/RDR register
+ * directly as 32bit data
+ * see rsnd_ssi_init()
+ */
+ if (rsnd_dai_is_play(rdai, io))
+ rsnd_mod_write(&ssi->mod, SSITDR, *buf);
+ else
+ *buf = rsnd_mod_read(&ssi->mod, SSIRDR);
+
+ rsnd_dai_pointer_update(io, sizeof(*buf));
+
+ ret = IRQ_HANDLED;
+ }
+
+ return ret;
+}
+
+static int rsnd_ssi_pio_start(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+
+ /* enable PIO IRQ */
+ ssi->cr_etc = UIEN | OIEN | DIEN;
+
+ rsnd_ssi_hw_start(ssi, rdai, io);
+
+ dev_dbg(dev, "%s.%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+ return 0;
+}
+
+static int rsnd_ssi_pio_stop(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+
+ dev_dbg(dev, "%s.%d stop\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+ ssi->cr_etc = 0;
+
+ rsnd_ssi_hw_stop(ssi, rdai);
+
+ return 0;
+}
+
+static struct rsnd_mod_ops rsnd_ssi_pio_ops = {
+ .name = "ssi (pio)",
+ .init = rsnd_ssi_init,
+ .quit = rsnd_ssi_quit,
+ .start = rsnd_ssi_pio_start,
+ .stop = rsnd_ssi_pio_stop,
+};
+
+static int rsnd_ssi_dma_inquiry(struct rsnd_dma *dma, dma_addr_t *buf, int *len)
+{
+ struct rsnd_ssi *ssi = rsnd_dma_to_ssi(dma);
+ struct rsnd_dai_stream *io = ssi->io;
+ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+
+ *len = io->byte_per_period;
+ *buf = runtime->dma_addr +
+ rsnd_dai_pointer_offset(io, ssi->dma_offset + *len);
+ ssi->dma_offset = *len; /* it cares A/B plane */
+
+ return 0;
+}
+
+static int rsnd_ssi_dma_complete(struct rsnd_dma *dma)
+{
+ struct rsnd_ssi *ssi = rsnd_dma_to_ssi(dma);
+ struct rsnd_dai_stream *io = ssi->io;
+ u32 status = rsnd_mod_read(&ssi->mod, SSISR);
+
+ rsnd_ssi_record_error(ssi, status);
+
+ rsnd_dai_pointer_update(ssi->io, io->byte_per_period);
+
+ return 0;
+}
+
+static int rsnd_ssi_dma_start(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+ struct rsnd_dma *dma = rsnd_mod_to_dma(&ssi->mod);
+
+ /* enable DMA transfer */
+ ssi->cr_etc = DMEN;
+ ssi->dma_offset = 0;
+
+ rsnd_dma_start(dma);
+
+ rsnd_ssi_hw_start(ssi, ssi->rdai, io);
+
+ /* enable WS continue */
+ if (rsnd_rdai_is_clk_master(rdai))
+ rsnd_mod_write(&ssi->mod, SSIWSR, CONT);
+
+ return 0;
+}
+
+static int rsnd_ssi_dma_stop(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+ struct rsnd_dma *dma = rsnd_mod_to_dma(&ssi->mod);
+
+ ssi->cr_etc = 0;
+
+ rsnd_ssi_hw_stop(ssi, rdai);
+
+ rsnd_dma_stop(dma);
+
+ return 0;
+}
+
+static struct rsnd_mod_ops rsnd_ssi_dma_ops = {
+ .name = "ssi (dma)",
+ .init = rsnd_ssi_init,
+ .quit = rsnd_ssi_quit,
+ .start = rsnd_ssi_dma_start,
+ .stop = rsnd_ssi_dma_stop,
+};
+
+/*
+ * Non SSI
+ */
+static int rsnd_ssi_non(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+
+ dev_dbg(dev, "%s\n", __func__);
+
+ return 0;
+}
+
+static struct rsnd_mod_ops rsnd_ssi_non_ops = {
+ .name = "ssi (non)",
+ .init = rsnd_ssi_non,
+ .quit = rsnd_ssi_non,
+ .start = rsnd_ssi_non,
+ .stop = rsnd_ssi_non,
+};
+
+/*
+ * ssi mod function
+ */
+struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv,
+ int dai_id, int is_play)
+{
+ struct rsnd_ssi *ssi;
+ int i, has_play;
+
+ is_play = !!is_play;
+
+ for_each_rsnd_ssi(ssi, priv, i) {
+ if (rsnd_ssi_dai_id(ssi) != dai_id)
+ continue;
+
+ has_play = !!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_PLAY);
+
+ if (is_play == has_play)
+ return &ssi->mod;
+ }
+
+ return NULL;
+}
+
+struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id)
+{
+ BUG_ON(id < 0 || id >= rsnd_ssi_nr(priv));
+
+ return &(((struct rsnd_ssiu *)(priv->ssiu))->ssi + id)->mod;
+}
+
+int rsnd_ssi_probe(struct platform_device *pdev,
+ struct rcar_snd_info *info,
+ struct rsnd_priv *priv)
+{
+ struct rsnd_ssi_platform_info *pinfo;
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct rsnd_mod_ops *ops;
+ struct clk *clk;
+ struct rsnd_ssiu *ssiu;
+ struct rsnd_ssi *ssi;
+ char name[RSND_SSI_NAME_SIZE];
+ int i, nr, ret;
+
+ /*
+ * init SSI
+ */
+ nr = info->ssi_info_nr;
+ ssiu = devm_kzalloc(dev, sizeof(*ssiu) + (sizeof(*ssi) * nr),
+ GFP_KERNEL);
+ if (!ssiu) {
+ dev_err(dev, "SSI allocate failed\n");
+ return -ENOMEM;
+ }
+
+ priv->ssiu = ssiu;
+ ssiu->ssi = (struct rsnd_ssi *)(ssiu + 1);
+ ssiu->ssi_nr = nr;
+
+ for_each_rsnd_ssi(ssi, priv, i) {
+ pinfo = &info->ssi_info[i];
+
+ snprintf(name, RSND_SSI_NAME_SIZE, "ssi.%d", i);
+
+ clk = clk_get(dev, name);
+ if (IS_ERR(clk))
+ return PTR_ERR(clk);
+
+ ssi->info = pinfo;
+ ssi->clk = clk;
+
+ ops = &rsnd_ssi_non_ops;
+
+ /*
+ * SSI DMA case
+ */
+ if (pinfo->dma_id > 0) {
+ ret = rsnd_dma_init(
+ priv, rsnd_mod_to_dma(&ssi->mod),
+ (rsnd_ssi_mode_flags(ssi) & RSND_SSI_PLAY),
+ pinfo->dma_id,
+ rsnd_ssi_dma_inquiry,
+ rsnd_ssi_dma_complete);
+ if (ret < 0)
+ dev_info(dev, "SSI DMA failed. try PIO transter\n");
+ else
+ ops = &rsnd_ssi_dma_ops;
+
+ dev_dbg(dev, "SSI%d use DMA transfer\n", i);
+ }
+
+ /*
+ * SSI PIO case
+ */
+ if (!rsnd_ssi_dma_available(ssi) &&
+ rsnd_ssi_pio_available(ssi)) {
+ ret = devm_request_irq(dev, pinfo->pio_irq,
+ &rsnd_ssi_pio_interrupt,
+ IRQF_SHARED,
+ dev_name(dev), ssi);
+ if (ret) {
+ dev_err(dev, "SSI request interrupt failed\n");
+ return ret;
+ }
+
+ ops = &rsnd_ssi_pio_ops;
+
+ dev_dbg(dev, "SSI%d use PIO transfer\n", i);
+ }
+
+ rsnd_mod_init(priv, &ssi->mod, ops, i);
+ }
+
+ rsnd_ssi_mode_init(priv, ssiu);
+
+ dev_dbg(dev, "ssi probed\n");
+
+ return 0;
+}
+
+void rsnd_ssi_remove(struct platform_device *pdev,
+ struct rsnd_priv *priv)
+{
+ struct rsnd_ssi *ssi;
+ int i;
+
+ for_each_rsnd_ssi(ssi, priv, i) {
+ clk_put(ssi->clk);
+ if (rsnd_ssi_dma_available(ssi))
+ rsnd_dma_quit(priv, rsnd_mod_to_dma(&ssi->mod));
+ }
+
+}
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 06a8000..53c9ecd 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -149,8 +149,9 @@
SND_SOC_DAPM_STREAM_STOP);
} else {
rtd->pop_wait = 1;
- schedule_delayed_work(&rtd->delayed_work,
- msecs_to_jiffies(rtd->pmdown_time));
+ queue_delayed_work(system_power_efficient_wq,
+ &rtd->delayed_work,
+ msecs_to_jiffies(rtd->pmdown_time));
}
} else {
/* capture streams can be powered down now */
@@ -334,7 +335,7 @@
return ret;
}
-static int sst_compr_set_metadata(struct snd_compr_stream *cstream,
+static int soc_compr_set_metadata(struct snd_compr_stream *cstream,
struct snd_compr_metadata *metadata)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
@@ -347,7 +348,7 @@
return ret;
}
-static int sst_compr_get_metadata(struct snd_compr_stream *cstream,
+static int soc_compr_get_metadata(struct snd_compr_stream *cstream,
struct snd_compr_metadata *metadata)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
@@ -364,8 +365,8 @@
.open = soc_compr_open,
.free = soc_compr_free,
.set_params = soc_compr_set_params,
- .set_metadata = sst_compr_set_metadata,
- .get_metadata = sst_compr_get_metadata,
+ .set_metadata = soc_compr_set_metadata,
+ .get_metadata = soc_compr_get_metadata,
.get_params = soc_compr_get_params,
.trigger = soc_compr_trigger,
.pointer = soc_compr_pointer,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ed3c253..4d05613 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -30,9 +30,12 @@
#include <linux/bitops.h>
#include <linux/debugfs.h>
#include <linux/platform_device.h>
+#include <linux/pinctrl/consumer.h>
#include <linux/ctype.h>
#include <linux/slab.h>
#include <linux/of.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
#include <sound/ac97_codec.h>
#include <sound/core.h>
#include <sound/jack.h>
@@ -67,6 +70,16 @@
module_param(pmdown_time, int, 0);
MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
+struct snd_ac97_reset_cfg {
+ struct pinctrl *pctl;
+ struct pinctrl_state *pstate_reset;
+ struct pinctrl_state *pstate_warm_reset;
+ struct pinctrl_state *pstate_run;
+ int gpio_sdata;
+ int gpio_sync;
+ int gpio_reset;
+};
+
/* returns the minimum number of bytes needed to represent
* a particular given value */
static int min_bytes_needed(unsigned long val)
@@ -190,7 +203,7 @@
struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
int ret;
- ret = strict_strtol(buf, 10, &rtd->pmdown_time);
+ ret = kstrtol(buf, 10, &rtd->pmdown_time);
if (ret)
return ret;
@@ -235,6 +248,7 @@
char *start = buf;
unsigned long reg, value;
struct snd_soc_codec *codec = file->private_data;
+ int ret;
buf_size = min(count, (sizeof(buf)-1));
if (copy_from_user(buf, user_buf, buf_size))
@@ -246,8 +260,9 @@
reg = simple_strtoul(start, &start, 16);
while (*start == ' ')
start++;
- if (strict_strtoul(start, 16, &value))
- return -EINVAL;
+ ret = kstrtoul(start, 16, &value);
+ if (ret)
+ return ret;
/* Userspace has been fiddling around behind the kernel's back */
add_taint(TAINT_USER, LOCKDEP_NOW_UNRELIABLE);
@@ -2085,6 +2100,117 @@
}
EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec);
+static struct snd_ac97_reset_cfg snd_ac97_rst_cfg;
+
+static void snd_soc_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ struct pinctrl *pctl = snd_ac97_rst_cfg.pctl;
+
+ pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_warm_reset);
+
+ gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 1);
+
+ udelay(10);
+
+ gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 0);
+
+ pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_run);
+ msleep(2);
+}
+
+static void snd_soc_ac97_reset(struct snd_ac97 *ac97)
+{
+ struct pinctrl *pctl = snd_ac97_rst_cfg.pctl;
+
+ pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_reset);
+
+ gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 0);
+ gpio_direction_output(snd_ac97_rst_cfg.gpio_sdata, 0);
+ gpio_direction_output(snd_ac97_rst_cfg.gpio_reset, 0);
+
+ udelay(10);
+
+ gpio_direction_output(snd_ac97_rst_cfg.gpio_reset, 1);
+
+ pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_run);
+ msleep(2);
+}
+
+static int snd_soc_ac97_parse_pinctl(struct device *dev,
+ struct snd_ac97_reset_cfg *cfg)
+{
+ struct pinctrl *p;
+ struct pinctrl_state *state;
+ int gpio;
+ int ret;
+
+ p = devm_pinctrl_get(dev);
+ if (IS_ERR(p)) {
+ dev_err(dev, "Failed to get pinctrl\n");
+ return PTR_RET(p);
+ }
+ cfg->pctl = p;
+
+ state = pinctrl_lookup_state(p, "ac97-reset");
+ if (IS_ERR(state)) {
+ dev_err(dev, "Can't find pinctrl state ac97-reset\n");
+ return PTR_RET(state);
+ }
+ cfg->pstate_reset = state;
+
+ state = pinctrl_lookup_state(p, "ac97-warm-reset");
+ if (IS_ERR(state)) {
+ dev_err(dev, "Can't find pinctrl state ac97-warm-reset\n");
+ return PTR_RET(state);
+ }
+ cfg->pstate_warm_reset = state;
+
+ state = pinctrl_lookup_state(p, "ac97-running");
+ if (IS_ERR(state)) {
+ dev_err(dev, "Can't find pinctrl state ac97-running\n");
+ return PTR_RET(state);
+ }
+ cfg->pstate_run = state;
+
+ gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 0);
+ if (gpio < 0) {
+ dev_err(dev, "Can't find ac97-sync gpio\n");
+ return gpio;
+ }
+ ret = devm_gpio_request(dev, gpio, "AC97 link sync");
+ if (ret) {
+ dev_err(dev, "Failed requesting ac97-sync gpio\n");
+ return ret;
+ }
+ cfg->gpio_sync = gpio;
+
+ gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 1);
+ if (gpio < 0) {
+ dev_err(dev, "Can't find ac97-sdata gpio %d\n", gpio);
+ return gpio;
+ }
+ ret = devm_gpio_request(dev, gpio, "AC97 link sdata");
+ if (ret) {
+ dev_err(dev, "Failed requesting ac97-sdata gpio\n");
+ return ret;
+ }
+ cfg->gpio_sdata = gpio;
+
+ gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 2);
+ if (gpio < 0) {
+ dev_err(dev, "Can't find ac97-reset gpio\n");
+ return gpio;
+ }
+ ret = devm_gpio_request(dev, gpio, "AC97 link reset");
+ if (ret) {
+ dev_err(dev, "Failed requesting ac97-reset gpio\n");
+ return ret;
+ }
+ cfg->gpio_reset = gpio;
+
+ return 0;
+}
+
struct snd_ac97_bus_ops *soc_ac97_ops;
EXPORT_SYMBOL_GPL(soc_ac97_ops);
@@ -2103,6 +2229,35 @@
EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops);
/**
+ * snd_soc_set_ac97_ops_of_reset - Set ac97 ops with generic ac97 reset functions
+ *
+ * This function sets the reset and warm_reset properties of ops and parses
+ * the device node of pdev to get pinctrl states and gpio numbers to use.
+ */
+int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops,
+ struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct snd_ac97_reset_cfg cfg;
+ int ret;
+
+ ret = snd_soc_ac97_parse_pinctl(dev, &cfg);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_set_ac97_ops(ops);
+ if (ret)
+ return ret;
+
+ ops->warm_reset = snd_soc_ac97_warm_reset;
+ ops->reset = snd_soc_ac97_reset;
+
+ snd_ac97_rst_cfg = cfg;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops_of_reset);
+
+/**
* snd_soc_free_ac97_codec - free AC97 codec device
* @codec: audio codec
*
@@ -2304,6 +2459,22 @@
return 0;
}
+struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card,
+ const char *name)
+{
+ struct snd_card *card = soc_card->snd_card;
+ struct snd_kcontrol *kctl;
+
+ if (unlikely(!name))
+ return NULL;
+
+ list_for_each_entry(kctl, &card->controls, list)
+ if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name)))
+ return kctl;
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol);
+
/**
* snd_soc_add_codec_controls - add an array of controls to a codec.
* Convenience function to add a list of controls. Many codecs were
diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig
index 3567d73..0a53053 100644
--- a/sound/soc/spear/Kconfig
+++ b/sound/soc/spear/Kconfig
@@ -1,6 +1,6 @@
config SND_SPEAR_SOC
tristate
- select SND_SOC_DMAENGINE_PCM
+ select SND_DMAENGINE_PCM
config SND_SPEAR_SPDIF_OUT
tristate
diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index 995b120..8fc653c 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -1,8 +1,8 @@
config SND_SOC_TEGRA
tristate "SoC Audio for the Tegra System-on-Chip"
- depends on ARCH_TEGRA && TEGRA20_APB_DMA
+ depends on (ARCH_TEGRA && TEGRA20_APB_DMA) || COMPILE_TEST
select REGMAP_MMIO
- select SND_SOC_GENERIC_DMAENGINE_PCM if TEGRA20_APB_DMA
+ select SND_SOC_GENERIC_DMAENGINE_PCM
help
Say Y or M here if you want support for SoC audio on Tegra.
@@ -61,7 +61,7 @@
config SND_SOC_TEGRA_RT5640
tristate "SoC Audio support for Tegra boards using an RT5640 codec"
- depends on SND_SOC_TEGRA && I2C
+ depends on SND_SOC_TEGRA && I2C && GPIOLIB
select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
select SND_SOC_RT5640
@@ -71,7 +71,7 @@
config SND_SOC_TEGRA_WM8753
tristate "SoC Audio support for Tegra boards using a WM8753 codec"
- depends on SND_SOC_TEGRA && I2C
+ depends on SND_SOC_TEGRA && I2C && GPIOLIB
select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
select SND_SOC_WM8753
@@ -81,7 +81,7 @@
config SND_SOC_TEGRA_WM8903
tristate "SoC Audio support for Tegra boards using a WM8903 codec"
- depends on SND_SOC_TEGRA && I2C
+ depends on SND_SOC_TEGRA && I2C && GPIOLIB
select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
select SND_SOC_WM8903
@@ -92,7 +92,7 @@
config SND_SOC_TEGRA_WM9712
tristate "SoC Audio support for Tegra boards using a WM9712 codec"
- depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC
+ depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC && GPIOLIB
select SND_SOC_TEGRA20_AC97
select SND_SOC_WM9712
help
@@ -110,7 +110,7 @@
config SND_SOC_TEGRA_ALC5632
tristate "SoC Audio support for Tegra boards using an ALC5632 codec"
- depends on SND_SOC_TEGRA && I2C
+ depends on SND_SOC_TEGRA && I2C && GPIOLIB
select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
select SND_SOC_ALC5632
help
diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c
index 6c48662..ae27bcd 100644
--- a/sound/soc/tegra/tegra20_ac97.c
+++ b/sound/soc/tegra/tegra20_ac97.c
@@ -334,12 +334,6 @@
}
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!mem) {
- dev_err(&pdev->dev, "No memory resource\n");
- ret = -ENODEV;
- goto err_clk_put;
- }
-
regs = devm_ioremap_resource(&pdev->dev, mem);
if (IS_ERR(regs)) {
ret = PTR_ERR(regs);
@@ -432,8 +426,6 @@
return 0;
-err_unregister_pcm:
- tegra_pcm_platform_unregister(&pdev->dev);
err_unregister_component:
snd_soc_unregister_component(&pdev->dev);
err_asoc_utils_fini:
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 48d05d9..c61ea3a 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -13,8 +13,6 @@
* published by the Free Software Foundation.
*/
-#include <asm/mach-types.h>
-
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c
index 08794f9..4511c5a 100644
--- a/sound/soc/tegra/tegra_rt5640.c
+++ b/sound/soc/tegra/tegra_rt5640.c
@@ -99,6 +99,7 @@
static const struct snd_soc_dapm_widget tegra_rt5640_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphones", NULL),
SND_SOC_DAPM_SPK("Speakers", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
};
static const struct snd_kcontrol_new tegra_rt5640_controls[] = {
diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c
index f87fc53..8e774d1 100644
--- a/sound/soc/tegra/tegra_wm8753.c
+++ b/sound/soc/tegra/tegra_wm8753.c
@@ -28,8 +28,6 @@
*
*/
-#include <asm/mach-types.h>
-
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 05c68aa..734bfcd 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -24,8 +24,6 @@
*
*/
-#include <asm/mach-types.h>
-
#include <linux/module.h>
#include <linux/of.h>
#include <linux/platform_device.h>
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
index 4bcce8a..e0305a1 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -184,9 +184,6 @@
if (irq < 0)
return irq;
r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!r)
- return -EBUSY;
-
drvdata->base = devm_ioremap_resource(&pdev->dev, r);
if (IS_ERR(drvdata->base))
return PTR_ERR(drvdata->base);
diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
index 8f5cd00..178d1ba 100644
--- a/sound/soc/ux500/mop500.c
+++ b/sound/soc/ux500/mop500.c
@@ -52,6 +52,7 @@
static struct snd_soc_card mop500_card = {
.name = "MOP500-card",
+ .owner = THIS_MODULE,
.probe = NULL,
.dai_link = mop500_dai_links,
.num_links = ARRAY_SIZE(mop500_dai_links),
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index b9defcd..780bf3f 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -346,10 +346,10 @@
if (!memcmp(version, known_fw_versions + i, 2))
return 0;
- snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. "
+ snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %4ph. "
"please reconnect to power. if this failure "
"still happens, check your firmware installation.",
- 4, version);
+ version);
return -EINVAL;
}
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 659950e..93e970f 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -418,6 +418,9 @@
struct snd_usb_endpoint *ep;
int is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK;
+ if (WARN_ON(!alts))
+ return NULL;
+
mutex_lock(&chip->mutex);
list_for_each_entry(ep, &chip->ep_list, list) {
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 15b151e..b375d58 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -327,6 +327,137 @@
return 0;
}
+static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
+ struct usb_device *dev,
+ struct usb_interface_descriptor *altsd,
+ unsigned int attr)
+{
+ struct usb_host_interface *alts;
+ struct usb_interface *iface;
+ unsigned int ep;
+
+ /* Implicit feedback sync EPs consumers are always playback EPs */
+ if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
+ switch (subs->stream->chip->usb_id) {
+ case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
+ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
+ ep = 0x81;
+ iface = usb_ifnum_to_if(dev, 3);
+
+ if (!iface || iface->num_altsetting == 0)
+ return -EINVAL;
+
+ alts = &iface->altsetting[1];
+ goto add_sync_ep;
+ break;
+ case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
+ case USB_ID(0x0763, 0x2081):
+ ep = 0x81;
+ iface = usb_ifnum_to_if(dev, 2);
+
+ if (!iface || iface->num_altsetting == 0)
+ return -EINVAL;
+
+ alts = &iface->altsetting[1];
+ goto add_sync_ep;
+ }
+ if (attr == USB_ENDPOINT_SYNC_ASYNC &&
+ altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
+ altsd->bInterfaceProtocol == 2 &&
+ altsd->bNumEndpoints == 1 &&
+ USB_ID_VENDOR(subs->stream->chip->usb_id) == 0x0582 /* Roland */ &&
+ search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1,
+ altsd->bAlternateSetting,
+ &alts, &ep) >= 0) {
+ goto add_sync_ep;
+ }
+
+ /* No quirk */
+ return 0;
+
+add_sync_ep:
+ subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
+ alts, ep, !subs->direction,
+ SND_USB_ENDPOINT_TYPE_DATA);
+ if (!subs->sync_endpoint)
+ return -EINVAL;
+
+ subs->data_endpoint->sync_master = subs->sync_endpoint;
+
+ return 0;
+}
+
+static int set_sync_endpoint(struct snd_usb_substream *subs,
+ struct audioformat *fmt,
+ struct usb_device *dev,
+ struct usb_host_interface *alts,
+ struct usb_interface_descriptor *altsd)
+{
+ int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
+ unsigned int ep, attr;
+ bool implicit_fb;
+ int err;
+
+ /* we need a sync pipe in async OUT or adaptive IN mode */
+ /* check the number of EP, since some devices have broken
+ * descriptors which fool us. if it has only one EP,
+ * assume it as adaptive-out or sync-in.
+ */
+ attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE;
+
+ err = set_sync_ep_implicit_fb_quirk(subs, dev, altsd, attr);
+ if (err < 0)
+ return err;
+
+ if (altsd->bNumEndpoints < 2)
+ return 0;
+
+ if ((is_playback && attr != USB_ENDPOINT_SYNC_ASYNC) ||
+ (!is_playback && attr != USB_ENDPOINT_SYNC_ADAPTIVE))
+ return 0;
+
+ /* check sync-pipe endpoint */
+ /* ... and check descriptor size before accessing bSynchAddress
+ because there is a version of the SB Audigy 2 NX firmware lacking
+ the audio fields in the endpoint descriptors */
+ if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC ||
+ (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+ get_endpoint(alts, 1)->bSynchAddress != 0)) {
+ snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
+ dev->devnum, fmt->iface, fmt->altsetting,
+ get_endpoint(alts, 1)->bmAttributes,
+ get_endpoint(alts, 1)->bLength,
+ get_endpoint(alts, 1)->bSynchAddress);
+ return -EINVAL;
+ }
+ ep = get_endpoint(alts, 1)->bEndpointAddress;
+ if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+ ((is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
+ (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
+ snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
+ dev->devnum, fmt->iface, fmt->altsetting,
+ is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
+ return -EINVAL;
+ }
+
+ implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK)
+ == USB_ENDPOINT_USAGE_IMPLICIT_FB;
+
+ subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
+ alts, ep, !subs->direction,
+ implicit_fb ?
+ SND_USB_ENDPOINT_TYPE_DATA :
+ SND_USB_ENDPOINT_TYPE_SYNC);
+ if (!subs->sync_endpoint)
+ return -EINVAL;
+
+ subs->data_endpoint->sync_master = subs->sync_endpoint;
+
+ return 0;
+}
+
/*
* find a matching format and set up the interface
*/
@@ -336,9 +467,7 @@
struct usb_host_interface *alts;
struct usb_interface_descriptor *altsd;
struct usb_interface *iface;
- unsigned int ep, attr;
- int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
- int err, implicit_fb = 0;
+ int err;
iface = usb_ifnum_to_if(dev, fmt->iface);
if (WARN_ON(!iface))
@@ -383,118 +512,22 @@
subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip,
alts, fmt->endpoint, subs->direction,
SND_USB_ENDPOINT_TYPE_DATA);
+
if (!subs->data_endpoint)
return -EINVAL;
- /* we need a sync pipe in async OUT or adaptive IN mode */
- /* check the number of EP, since some devices have broken
- * descriptors which fool us. if it has only one EP,
- * assume it as adaptive-out or sync-in.
- */
- attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE;
+ err = set_sync_endpoint(subs, fmt, dev, alts, altsd);
+ if (err < 0)
+ return err;
- switch (subs->stream->chip->usb_id) {
- case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
- case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
- if (is_playback) {
- implicit_fb = 1;
- ep = 0x81;
- iface = usb_ifnum_to_if(dev, 3);
-
- if (!iface || iface->num_altsetting == 0)
- return -EINVAL;
-
- alts = &iface->altsetting[1];
- goto add_sync_ep;
- }
- break;
- case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
- case USB_ID(0x0763, 0x2081):
- if (is_playback) {
- implicit_fb = 1;
- ep = 0x81;
- iface = usb_ifnum_to_if(dev, 2);
-
- if (!iface || iface->num_altsetting == 0)
- return -EINVAL;
-
- alts = &iface->altsetting[1];
- goto add_sync_ep;
- }
- }
- if (is_playback &&
- attr == USB_ENDPOINT_SYNC_ASYNC &&
- altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
- altsd->bInterfaceProtocol == 2 &&
- altsd->bNumEndpoints == 1 &&
- USB_ID_VENDOR(subs->stream->chip->usb_id) == 0x0582 /* Roland */ &&
- search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1,
- altsd->bAlternateSetting,
- &alts, &ep) >= 0) {
- implicit_fb = 1;
- goto add_sync_ep;
- }
-
- if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) ||
- (!is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) &&
- altsd->bNumEndpoints >= 2) {
- /* check sync-pipe endpoint */
- /* ... and check descriptor size before accessing bSynchAddress
- because there is a version of the SB Audigy 2 NX firmware lacking
- the audio fields in the endpoint descriptors */
- if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC ||
- (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
- get_endpoint(alts, 1)->bSynchAddress != 0 &&
- !implicit_fb)) {
- snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
- dev->devnum, fmt->iface, fmt->altsetting,
- get_endpoint(alts, 1)->bmAttributes,
- get_endpoint(alts, 1)->bLength,
- get_endpoint(alts, 1)->bSynchAddress);
- return -EINVAL;
- }
- ep = get_endpoint(alts, 1)->bEndpointAddress;
- if (!implicit_fb &&
- get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
- (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
- (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
- snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
- dev->devnum, fmt->iface, fmt->altsetting,
- is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
- return -EINVAL;
- }
-
- implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK)
- == USB_ENDPOINT_USAGE_IMPLICIT_FB;
-
-add_sync_ep:
- subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
- alts, ep, !subs->direction,
- implicit_fb ?
- SND_USB_ENDPOINT_TYPE_DATA :
- SND_USB_ENDPOINT_TYPE_SYNC);
- if (!subs->sync_endpoint)
- return -EINVAL;
-
- subs->data_endpoint->sync_master = subs->sync_endpoint;
- }
-
- if ((err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt)) < 0)
+ err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt);
+ if (err < 0)
return err;
subs->cur_audiofmt = fmt;
snd_usb_set_format_quirk(subs, fmt);
-#if 0
- printk(KERN_DEBUG
- "setting done: format = %d, rate = %d..%d, channels = %d\n",
- fmt->format, fmt->rate_min, fmt->rate_max, fmt->channels);
- printk(KERN_DEBUG
- " datapipe = 0x%0x, syncpipe = 0x%0x\n",
- subs->datapipe, subs->syncpipe);
-#endif
-
return 0;
}
diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c
index 1f9bbd5..5a51b18 100644
--- a/sound/usb/usx2y/usbusx2y.c
+++ b/sound/usb/usx2y/usbusx2y.c
@@ -305,11 +305,9 @@
{
int i;
for (i = 0; i < URBS_AsyncSeq; ++i) {
- if (S[i].urb) {
- usb_kill_urb(S->urb[i]);
- usb_free_urb(S->urb[i]);
- S->urb[i] = NULL;
- }
+ usb_kill_urb(S->urb[i]);
+ usb_free_urb(S->urb[i]);
+ S->urb[i] = NULL;
}
kfree(S->buffer);
}