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-rw-r--r--Documentation/devicetree/bindings/sound/ak5386.txt4
-rw-r--r--Documentation/devicetree/bindings/sound/cs4265.txt29
-rw-r--r--Documentation/devicetree/bindings/sound/fsl,asrc.txt60
-rw-r--r--Documentation/devicetree/bindings/sound/max98090.txt2
-rw-r--r--Documentation/devicetree/bindings/sound/renesas,rsnd.txt9
-rw-r--r--Documentation/devicetree/bindings/sound/rockchip-i2s.txt37
-rw-r--r--Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt35
-rw-r--r--Documentation/devicetree/bindings/sound/sirf-usp.txt27
-rw-r--r--Documentation/devicetree/bindings/sound/snow.txt5
-rw-r--r--Documentation/devicetree/bindings/sound/tas2552.txt26
-rw-r--r--Documentation/devicetree/bindings/sound/ti,tas5086.txt5
-rw-r--r--Documentation/devicetree/bindings/sound/wm8904.txt33
-rw-r--r--Documentation/sound/alsa/ALSA-Configuration.txt4
-rw-r--r--Documentation/sound/alsa/HD-Audio-Models.txt5
-rw-r--r--MAINTAINERS7
-rw-r--r--arch/arm/mach-shmobile/board-armadillo800eva.c4
-rw-r--r--arch/arm/mach-shmobile/board-kzm9g.c2
-rw-r--r--arch/arm/mach-shmobile/board-mackerel.c4
-rw-r--r--arch/sh/boards/mach-ecovec24/setup.c2
-rw-r--r--arch/x86/include/asm/platform_sst_audio.h78
-rw-r--r--drivers/dma/edma.c1
-rw-r--r--drivers/misc/atmel-ssc.c13
-rw-r--r--include/linux/atmel-ssc.h13
-rw-r--r--include/linux/dmaengine.h1
-rw-r--r--include/linux/mfd/arizona/core.h6
-rw-r--r--include/linux/platform_data/asoc-s3c.h9
-rw-r--r--include/linux/platform_data/dma-imx.h1
-rw-r--r--include/sound/control.h7
-rw-r--r--include/sound/pcm.h11
-rw-r--r--include/sound/rcar_snd.h1
-rw-r--r--include/sound/rt286.h19
-rw-r--r--include/sound/rt5670.h27
-rw-r--r--include/sound/soc-dai.h5
-rw-r--r--include/sound/soc-dapm.h8
-rw-r--r--include/sound/soc.h94
-rw-r--r--include/sound/tas2552-plat.h25
-rw-r--r--include/sound/wm8962.h1
-rw-r--r--include/trace/events/asoc.h6
-rw-r--r--include/uapi/sound/asound.h9
-rw-r--r--sound/aoa/soundbus/i2sbus/core.c12
-rw-r--r--sound/arm/pxa2xx-ac97-lib.c12
-rw-r--r--sound/core/compress_offload.c2
-rw-r--r--sound/core/control.c6
-rw-r--r--sound/core/pcm_compat.c8
-rw-r--r--sound/core/pcm_dmaengine.c4
-rw-r--r--sound/core/pcm_native.c9
-rw-r--r--sound/core/seq/seq_memory.c4
-rw-r--r--sound/firewire/Kconfig14
-rw-r--r--sound/firewire/fireworks/fireworks_proc.c4
-rw-r--r--sound/oss/mpu401.c2
-rw-r--r--sound/oss/opl3.c4
-rw-r--r--sound/oss/pss.c46
-rw-r--r--sound/pci/Kconfig4
-rw-r--r--sound/pci/echoaudio/echoaudio.c6
-rw-r--r--sound/pci/hda/dell_wmi_helper.c76
-rw-r--r--sound/pci/hda/hda_auto_parser.c17
-rw-r--r--sound/pci/hda/hda_codec.c45
-rw-r--r--sound/pci/hda/hda_codec.h4
-rw-r--r--sound/pci/hda/hda_controller.c203
-rw-r--r--sound/pci/hda/hda_controller.h9
-rw-r--r--sound/pci/hda/hda_eld.c46
-rw-r--r--sound/pci/hda/hda_generic.c22
-rw-r--r--sound/pci/hda/hda_i915.c4
-rw-r--r--sound/pci/hda/hda_intel.c372
-rw-r--r--sound/pci/hda/hda_local.h9
-rw-r--r--sound/pci/hda/hda_priv.h253
-rw-r--r--sound/pci/hda/hda_tegra.c36
-rw-r--r--sound/pci/hda/patch_ca0132.c6
-rw-r--r--sound/pci/hda/patch_cirrus.c4
-rw-r--r--sound/pci/hda/patch_cmedia.c624
-rw-r--r--sound/pci/hda/patch_conexant.c2631
-rw-r--r--sound/pci/hda/patch_hdmi.c13
-rw-r--r--sound/pci/hda/patch_realtek.c170
-rw-r--r--sound/pci/hda/patch_sigmatel.c29
-rw-r--r--sound/pci/ice1712/ice1712.h15
-rw-r--r--sound/pci/mixart/mixart_core.c4
-rw-r--r--sound/pci/oxygen/virtuoso.c1
-rw-r--r--sound/pci/oxygen/xonar_pcm179x.c12
-rw-r--r--sound/pci/riptide/riptide.c4
-rw-r--r--sound/pci/trident/trident_main.c2
-rw-r--r--sound/pci/trident/trident_memory.c3
-rw-r--r--sound/soc/Kconfig1
-rw-r--r--sound/soc/Makefile1
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c34
-rw-r--r--sound/soc/atmel/atmel_wm8904.c50
-rw-r--r--sound/soc/blackfin/bf5xx-i2s-pcm.c8
-rw-r--r--sound/soc/codecs/88pm860x-codec.c12
-rw-r--r--sound/soc/codecs/Kconfig27
-rw-r--r--sound/soc/codecs/Makefile8
-rw-r--r--sound/soc/codecs/ac97.c4
-rw-r--r--sound/soc/codecs/adau1701.c6
-rw-r--r--sound/soc/codecs/adau17x1.c8
-rw-r--r--sound/soc/codecs/adau1977.c2
-rw-r--r--sound/soc/codecs/ak4642.c4
-rw-r--r--sound/soc/codecs/ak5386.c50
-rw-r--r--sound/soc/codecs/arizona.c288
-rw-r--r--sound/soc/codecs/arizona.h1
-rw-r--r--sound/soc/codecs/cs4265.c682
-rw-r--r--sound/soc/codecs/cs4265.h64
-rw-r--r--sound/soc/codecs/cs4270.c4
-rw-r--r--sound/soc/codecs/cs42l52.c14
-rw-r--r--sound/soc/codecs/cs42l56.c76
-rw-r--r--sound/soc/codecs/cs42l73.c6
-rw-r--r--sound/soc/codecs/cs42xx8.c5
-rw-r--r--sound/soc/codecs/cs42xx8.h8
-rw-r--r--sound/soc/codecs/cx20442.c10
-rw-r--r--sound/soc/codecs/max98088.c6
-rw-r--r--sound/soc/codecs/max98090.c44
-rw-r--r--sound/soc/codecs/max98095.c12
-rw-r--r--sound/soc/codecs/mc13783.c6
-rw-r--r--sound/soc/codecs/pcm1792a.c3
-rw-r--r--sound/soc/codecs/pcm1792a.h3
-rw-r--r--sound/soc/codecs/rl6231.c19
-rw-r--r--sound/soc/codecs/rt286.c1222
-rw-r--r--sound/soc/codecs/rt286.h198
-rw-r--r--sound/soc/codecs/rt5631.c10
-rw-r--r--sound/soc/codecs/rt5640.c10
-rw-r--r--sound/soc/codecs/rt5645.c10
-rw-r--r--sound/soc/codecs/rt5651.c10
-rw-r--r--sound/soc/codecs/rt5670-dsp.h54
-rw-r--r--sound/soc/codecs/rt5670.c2657
-rw-r--r--sound/soc/codecs/rt5670.h2000
-rw-r--r--sound/soc/codecs/rt5677.c272
-rw-r--r--sound/soc/codecs/rt5677.h15
-rw-r--r--sound/soc/codecs/sgtl5000.c25
-rw-r--r--sound/soc/codecs/si476x.c10
-rw-r--r--sound/soc/codecs/sirf-audio-codec.c4
-rw-r--r--sound/soc/codecs/sn95031.c6
-rw-r--r--sound/soc/codecs/spdif_transmitter.c2
-rw-r--r--sound/soc/codecs/ssm2518.c6
-rw-r--r--sound/soc/codecs/ssm2602.c10
-rw-r--r--sound/soc/codecs/sta32x.c19
-rw-r--r--sound/soc/codecs/sta529.c12
-rw-r--r--sound/soc/codecs/tas2552.c544
-rw-r--r--sound/soc/codecs/tas2552.h129
-rw-r--r--sound/soc/codecs/tas5086.c75
-rw-r--r--sound/soc/codecs/tlv320aic23.c10
-rw-r--r--sound/soc/codecs/tlv320aic26.c14
-rw-r--r--sound/soc/codecs/tlv320aic31xx.c40
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c31
-rw-r--r--sound/soc/codecs/tlv320aic3x.c21
-rw-r--r--sound/soc/codecs/tlv320dac33.c12
-rw-r--r--sound/soc/codecs/tpa6130a2.c4
-rw-r--r--sound/soc/codecs/twl4030.c19
-rw-r--r--sound/soc/codecs/uda134x.c10
-rw-r--r--sound/soc/codecs/wl1273.c9
-rw-r--r--sound/soc/codecs/wm0010.c14
-rw-r--r--sound/soc/codecs/wm1250-ev1.c1
-rw-r--r--sound/soc/codecs/wm2000.c4
-rw-r--r--sound/soc/codecs/wm5100.c3
-rw-r--r--sound/soc/codecs/wm5102.c65
-rw-r--r--sound/soc/codecs/wm5110.c4
-rw-r--r--sound/soc/codecs/wm8350.c13
-rw-r--r--sound/soc/codecs/wm8400.c10
-rw-r--r--sound/soc/codecs/wm8510.c10
-rw-r--r--sound/soc/codecs/wm8523.c10
-rw-r--r--sound/soc/codecs/wm8580.c10
-rw-r--r--sound/soc/codecs/wm8711.c8
-rw-r--r--sound/soc/codecs/wm8728.c8
-rw-r--r--sound/soc/codecs/wm8731.c8
-rw-r--r--sound/soc/codecs/wm8737.c10
-rw-r--r--sound/soc/codecs/wm8741.c14
-rw-r--r--sound/soc/codecs/wm8750.c10
-rw-r--r--sound/soc/codecs/wm8753.c20
-rw-r--r--sound/soc/codecs/wm8770.c10
-rw-r--r--sound/soc/codecs/wm8804.c10
-rw-r--r--sound/soc/codecs/wm8900.c10
-rw-r--r--sound/soc/codecs/wm8903.c13
-rw-r--r--sound/soc/codecs/wm8904.c27
-rw-r--r--sound/soc/codecs/wm8940.c12
-rw-r--r--sound/soc/codecs/wm8955.c10
-rw-r--r--sound/soc/codecs/wm8958-dsp2.c24
-rw-r--r--sound/soc/codecs/wm8960.c17
-rw-r--r--sound/soc/codecs/wm8961.c10
-rw-r--r--sound/soc/codecs/wm8962.c29
-rw-r--r--sound/soc/codecs/wm8971.c10
-rw-r--r--sound/soc/codecs/wm8974.c10
-rw-r--r--sound/soc/codecs/wm8978.c14
-rw-r--r--sound/soc/codecs/wm8983.c12
-rw-r--r--sound/soc/codecs/wm8985.c15
-rw-r--r--sound/soc/codecs/wm8988.c10
-rw-r--r--sound/soc/codecs/wm8990.c10
-rw-r--r--sound/soc/codecs/wm8991.c10
-rw-r--r--sound/soc/codecs/wm8993.c10
-rw-r--r--sound/soc/codecs/wm8994.c35
-rw-r--r--sound/soc/codecs/wm8995.c12
-rw-r--r--sound/soc/codecs/wm8996.c6
-rw-r--r--sound/soc/codecs/wm8997.c2
-rw-r--r--sound/soc/codecs/wm9081.c10
-rw-r--r--sound/soc/codecs/wm9090.c4
-rw-r--r--sound/soc/codecs/wm9713.c10
-rw-r--r--sound/soc/codecs/wm_adsp.c6
-rw-r--r--sound/soc/codecs/wm_hubs.c4
-rw-r--r--sound/soc/davinci/Kconfig25
-rw-r--r--sound/soc/davinci/Makefile2
-rw-r--r--sound/soc/davinci/davinci-mcasp.c93
-rw-r--r--sound/soc/davinci/edma-pcm.c2
-rw-r--r--sound/soc/davinci/edma-pcm.h7
-rw-r--r--sound/soc/fsl/Kconfig16
-rw-r--r--sound/soc/fsl/Makefile2
-rw-r--r--sound/soc/fsl/fsl_asrc.c995
-rw-r--r--sound/soc/fsl/fsl_asrc.h461
-rw-r--r--sound/soc/fsl/fsl_asrc_dma.c391
-rw-r--r--sound/soc/fsl/fsl_esai.c2
-rw-r--r--sound/soc/fsl/fsl_sai.c39
-rw-r--r--sound/soc/fsl/fsl_spdif.c88
-rw-r--r--sound/soc/fsl/fsl_spdif.h10
-rw-r--r--sound/soc/fsl/fsl_ssi.c6
-rw-r--r--sound/soc/fsl/imx-audmux.c8
-rw-r--r--sound/soc/generic/simple-card.c13
-rw-r--r--sound/soc/intel/Kconfig12
-rw-r--r--sound/soc/intel/Makefile2
-rw-r--r--sound/soc/intel/broadwell.c251
-rw-r--r--sound/soc/intel/byt-max98090.c27
-rw-r--r--sound/soc/intel/byt-rt5640.c1
-rw-r--r--sound/soc/intel/sst-atom-controls.h30
-rw-r--r--sound/soc/intel/sst-baytrail-ipc.c30
-rw-r--r--sound/soc/intel/sst-baytrail-pcm.c2
-rw-r--r--sound/soc/intel/sst-dsp.c10
-rw-r--r--sound/soc/intel/sst-dsp.h39
-rw-r--r--sound/soc/intel/sst-haswell-dsp.c70
-rw-r--r--sound/soc/intel/sst-haswell-ipc.c40
-rw-r--r--sound/soc/intel/sst-haswell-pcm.c27
-rw-r--r--sound/soc/intel/sst-mfld-dsp.h429
-rw-r--r--sound/soc/intel/sst-mfld-platform-compress.c11
-rw-r--r--sound/soc/intel/sst-mfld-platform-pcm.c319
-rw-r--r--sound/soc/intel/sst-mfld-platform.h29
-rw-r--r--sound/soc/kirkwood/Kconfig19
-rw-r--r--sound/soc/kirkwood/Makefile4
-rw-r--r--sound/soc/kirkwood/kirkwood-dma.c11
-rw-r--r--sound/soc/kirkwood/kirkwood-i2s.c33
-rw-r--r--sound/soc/kirkwood/kirkwood-openrd.c109
-rw-r--r--sound/soc/kirkwood/kirkwood-t5325.c116
-rw-r--r--sound/soc/kirkwood/kirkwood.h7
-rw-r--r--sound/soc/omap/ams-delta.c2
-rw-r--r--sound/soc/omap/omap-dmic.c35
-rw-r--r--sound/soc/omap/omap-mcbsp.c7
-rw-r--r--sound/soc/omap/omap-pcm.c1
-rw-r--r--sound/soc/pxa/pxa-ssp.c3
-rw-r--r--sound/soc/rockchip/Kconfig12
-rw-r--r--sound/soc/rockchip/Makefile4
-rw-r--r--sound/soc/rockchip/rockchip_i2s.c529
-rw-r--r--sound/soc/rockchip/rockchip_i2s.h223
-rw-r--r--sound/soc/s6000/Kconfig13
-rw-r--r--sound/soc/s6000/Makefile2
-rw-r--r--sound/soc/s6000/s6000-i2s.c4
-rw-r--r--sound/soc/s6000/s6105-ipcam.c17
-rw-r--r--sound/soc/samsung/Kconfig40
-rw-r--r--sound/soc/samsung/Makefile6
-rw-r--r--sound/soc/samsung/ac97.c32
-rw-r--r--sound/soc/samsung/dma.c454
-rw-r--r--sound/soc/samsung/dma.h7
-rw-r--r--sound/soc/samsung/dmaengine.c3
-rw-r--r--sound/soc/samsung/i2s.c35
-rw-r--r--sound/soc/samsung/idma.c3
-rw-r--r--sound/soc/samsung/odroidx2_max98090.c177
-rw-r--r--sound/soc/samsung/pcm.c12
-rw-r--r--sound/soc/samsung/s3c-i2s-v2.c19
-rw-r--r--sound/soc/samsung/s3c2412-i2s.c43
-rw-r--r--sound/soc/samsung/s3c24xx-i2s.c58
-rw-r--r--sound/soc/samsung/smdk_wm8580pcm.c2
-rw-r--r--sound/soc/samsung/snow.c4
-rw-r--r--sound/soc/samsung/spdif.c5
-rw-r--r--sound/soc/sh/Kconfig2
-rw-r--r--sound/soc/sh/fsi.c201
-rw-r--r--sound/soc/sh/rcar/core.c247
-rw-r--r--sound/soc/sh/rcar/dvc.c135
-rw-r--r--sound/soc/sh/rcar/gen.c554
-rw-r--r--sound/soc/sh/rcar/rsnd.h26
-rw-r--r--sound/soc/sh/rcar/src.c86
-rw-r--r--sound/soc/sh/rcar/ssi.c33
-rw-r--r--sound/soc/sirf/Kconfig6
-rw-r--r--sound/soc/sirf/Makefile2
-rw-r--r--sound/soc/sirf/sirf-usp.c415
-rw-r--r--sound/soc/sirf/sirf-usp.h293
-rw-r--r--sound/soc/soc-cache.c7
-rw-r--r--sound/soc/soc-compress.c13
-rw-r--r--sound/soc/soc-core.c900
-rw-r--r--sound/soc/soc-dapm.c279
-rw-r--r--sound/soc/soc-generic-dmaengine-pcm.c37
-rw-r--r--sound/soc/soc-jack.c4
-rw-r--r--sound/soc/soc-pcm.c581
-rw-r--r--sound/soc/tegra/tegra_alc5632.c5
-rw-r--r--sound/soc/tegra/tegra_max98090.c5
-rw-r--r--sound/soc/tegra/tegra_rt5640.c5
-rw-r--r--sound/soc/tegra/tegra_wm8753.c3
-rw-r--r--sound/soc/tegra/tegra_wm8903.c5
-rw-r--r--sound/soc/tegra/trimslice.c3
-rw-r--r--sound/sparc/dbri.c6
-rw-r--r--sound/usb/card.c9
-rw-r--r--sound/usb/midi.c401
-rw-r--r--sound/usb/midi.h6
-rw-r--r--sound/usb/mixer.c9
-rw-r--r--sound/usb/quirks.c2
294 files changed, 17264 insertions, 7462 deletions
diff --git a/Documentation/devicetree/bindings/sound/ak5386.txt b/Documentation/devicetree/bindings/sound/ak5386.txt
index dc3914fe6ce8..ec3df3abba0c 100644
--- a/Documentation/devicetree/bindings/sound/ak5386.txt
+++ b/Documentation/devicetree/bindings/sound/ak5386.txt
@@ -10,10 +10,14 @@ Optional properties:
- reset-gpio : a GPIO spec for the reset/power down pin.
If specified, it will be deasserted at probe time.
+ - va-supply : a regulator spec, providing 5.0V
+ - vd-supply : a regulator spec, providing 3.3V
Example:
spdif: ak5386@0 {
compatible = "asahi-kasei,ak5386";
reset-gpio = <&gpio0 23>;
+ va-supply = <&vdd_5v0_reg>;
+ vd-supply = <&vdd_3v3_reg>;
};
diff --git a/Documentation/devicetree/bindings/sound/cs4265.txt b/Documentation/devicetree/bindings/sound/cs4265.txt
new file mode 100644
index 000000000000..380fff8e4e83
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs4265.txt
@@ -0,0 +1,29 @@
+CS4265 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+ - compatible : "cirrus,cs4265"
+
+ - reg : the I2C address of the device for I2C. The I2C address depends on
+ the state of the AD0 pin. If AD0 is high, the i2c address is 0x4f.
+ If it is low, the i2c address is 0x4e.
+
+Optional properties:
+
+ - reset-gpios : a GPIO spec for the reset pin. If specified, it will be
+ deasserted before communication to the codec starts.
+
+Examples:
+
+codec_ad0_high: cs4265@4f { /* AD0 Pin is high */
+ compatible = "cirrus,cs4265";
+ reg = <0x4f>;
+};
+
+
+codec_ad0_low: cs4265@4e { /* AD0 Pin is low */
+ compatible = "cirrus,cs4265";
+ reg = <0x4e>;
+};
diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt
new file mode 100644
index 000000000000..b93362a570be
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt
@@ -0,0 +1,60 @@
+Freescale Asynchronous Sample Rate Converter (ASRC) Controller
+
+The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of a
+signal associated with an input clock into a signal associated with a different
+output clock. The driver currently works as a Front End of DPCM with other Back
+Ends Audio controller such as ESAI, SSI and SAI. It has three pairs to support
+three substreams within totally 10 channels.
+
+Required properties:
+
+ - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc".
+
+ - 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 : Contains "rxa", "rxb", "rxc", "txa", "txb" and "txc".
+
+ - clocks : Contains an entry for each entry in clock-names.
+
+ - clock-names : Contains the following entries
+ "mem" Peripheral access clock to access registers.
+ "ipg" Peripheral clock to driver module.
+ "asrck_<0-f>" Clock sources for input and output clock.
+
+ - big-endian : If this property is absent, the little endian mode
+ will be in use as default. Otherwise, the big endian
+ mode will be in use for all the device registers.
+
+ - fsl,asrc-rate : Defines a mutual sample rate used by DPCM Back Ends.
+
+ - fsl,asrc-width : Defines a mutual sample width used by DPCM Back Ends.
+
+Example:
+
+asrc: asrc@02034000 {
+ compatible = "fsl,imx53-asrc";
+ reg = <0x02034000 0x4000>;
+ interrupts = <0 50 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&clks 107>, <&clks 107>, <&clks 0>,
+ <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>,
+ <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>,
+ <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>,
+ <&clks 107>, <&clks 0>, <&clks 0>;
+ clock-names = "mem", "ipg", "asrck0",
+ "asrck_1", "asrck_2", "asrck_3", "asrck_4",
+ "asrck_5", "asrck_6", "asrck_7", "asrck_8",
+ "asrck_9", "asrck_a", "asrck_b", "asrck_c",
+ "asrck_d", "asrck_e", "asrck_f";
+ dmas = <&sdma 17 23 1>, <&sdma 18 23 1>, <&sdma 19 23 1>,
+ <&sdma 20 23 1>, <&sdma 21 23 1>, <&sdma 22 23 1>;
+ dma-names = "rxa", "rxb", "rxc",
+ "txa", "txb", "txc";
+ fsl,asrc-rate = <48000>;
+ fsl,asrc-width = <16>;
+ status = "okay";
+};
diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt
index a5e63fa47dc5..c454e67f54bb 100644
--- a/Documentation/devicetree/bindings/sound/max98090.txt
+++ b/Documentation/devicetree/bindings/sound/max98090.txt
@@ -4,7 +4,7 @@ This device supports I2C only.
Required properties:
-- compatible : "maxim,max98090".
+- compatible : "maxim,max98090" or "maxim,max98091".
- reg : The I2C address of the device.
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index 8346cab046cd..aa697abf337e 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -13,6 +13,9 @@ Required properties:
- rcar_sound,src : Should contain SRC feature.
The number of SRC subnode should be same as HW.
see below for detail.
+- rcar_sound,dvc : Should contain DVC feature.
+ The number of DVC subnode should be same as HW.
+ see below for detail.
- rcar_sound,dai : DAI contents.
The number of DAI subnode should be same as HW.
see below for detail.
@@ -21,6 +24,7 @@ SSI subnode properties:
- interrupts : Should contain SSI interrupt for PIO transfer
- shared-pin : if shared clock pin
- pio-transfer : use PIO transfer mode
+- no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case
SRC subnode properties:
no properties at this point
@@ -39,6 +43,11 @@ rcar_sound: rcar_sound@0xffd90000 {
<0 0xec540000 0 0x1000>, /* SSIU */
<0 0xec541000 0 0x1280>; /* SSI */
+ rcar_sound,dvc {
+ dvc0: dvc@0 { };
+ dvc1: dvc@1 { };
+ };
+
rcar_sound,src {
src0: src@0 { };
src1: src@1 { };
diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
new file mode 100644
index 000000000000..6c55fcfe5e1d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
@@ -0,0 +1,37 @@
+* Rockchip I2S controller
+
+The I2S bus (Inter-IC sound bus) is a serial link for digital
+audio data transfer between devices in the system.
+
+Required properties:
+
+- compatible: should be one of the followings
+ - "rockchip,rk3066-i2s": for rk3066
+ - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188
+ - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288
+- reg: physical base address of the controller and length of memory mapped
+ region.
+- interrupts: should contain the I2S interrupt.
+- #address-cells: should be 1.
+- #size-cells: should be 0.
+- dmas: DMA specifiers for tx and rx dma. See the DMA client binding,
+ Documentation/devicetree/bindings/dma/dma.txt
+- dma-names: should include "tx" and "rx".
+- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names.
+- clock-names: should contain followings:
+ - "i2s_hclk": clock for I2S BUS
+ - "i2s_clk" : clock for I2S controller
+
+Example for rk3288 I2S controller:
+
+i2s@ff890000 {
+ compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s";
+ reg = <0xff890000 0x10000>;
+ interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>;
+ #address-cells = <1>;
+ #size-cells = <0>;
+ dmas = <&pdma1 0>, <&pdma1 1>;
+ dma-names = "rx", "tx";
+ clock-names = "i2s_hclk", "i2s_clk";
+ clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt
new file mode 100644
index 000000000000..9148f72319e1
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt
@@ -0,0 +1,35 @@
+Samsung Exynos Odroid X2/U3 audio complex with MAX98090 codec
+
+Required properties:
+ - compatible : "samsung,odroidx2-audio" - for Odroid X2 board,
+ "samsung,odroidu3-audio" - for Odroid U3 board
+ - samsung,model : the user-visible name of this sound complex
+ - samsung,i2s-controller : the phandle of the I2S controller
+ - samsung,audio-codec : the phandle of the MAX98090 audio codec
+ - samsung,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 MAX98090's pins (as
+ documented in its binding), and the jacks on the board
+ For Odroid X2:
+ * Headphone Jack
+ * Mic Jack
+ * DMIC
+
+ For Odroid U3:
+ * Headphone Jack
+ * Speakers
+
+Example:
+
+sound {
+ compatible = "samsung,odroidu3-audio";
+ samsung,i2s-controller = <&i2s0>;
+ samsung,audio-codec = <&max98090>;
+ samsung,model = "Odroid-X2";
+ samsung,audio-routing =
+ "Headphone Jack", "HPL",
+ "Headphone Jack", "HPR",
+ "IN1", "Mic Jack",
+ "Mic Jack", "MICBIAS";
+};
diff --git a/Documentation/devicetree/bindings/sound/sirf-usp.txt b/Documentation/devicetree/bindings/sound/sirf-usp.txt
new file mode 100644
index 000000000000..02f85b32d359
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sirf-usp.txt
@@ -0,0 +1,27 @@
+* SiRF SoC USP module
+
+Required properties:
+- compatible: "sirf,prima2-usp-pcm"
+- reg: Base address and size entries:
+- dmas: List of DMA controller phandle and DMA request line ordered pairs.
+- dma-names: Identifier string for each DMA request line in the dmas property.
+ These strings correspond 1:1 with the ordered pairs in dmas.
+
+ One of the DMA channels will be responsible for transmission (should be
+ named "tx") and one for reception (should be named "rx").
+
+- clocks: USP controller clock source
+- pinctrl-names: Must contain a "default" entry.
+- pinctrl-NNN: One property must exist for each entry in pinctrl-names.
+
+Example:
+usp0: usp@b0080000 {
+ compatible = "sirf,prima2-usp-pcm";
+ reg = <0xb0080000 0x10000>;
+ clocks = <&clks 28>;
+ dmas = <&dmac1 1>, <&dmac1 2>;
+ dma-names = "rx", "tx";
+ pinctrl-names = "default";
+ pinctrl-0 = <&usp0_only_utfs_pins_a>;
+};
+
diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt
index 678b191c37b8..6df74f15687f 100644
--- a/Documentation/devicetree/bindings/sound/snow.txt
+++ b/Documentation/devicetree/bindings/sound/snow.txt
@@ -3,15 +3,20 @@ Audio Binding for Snow boards
Required properties:
- compatible : Can be one of the following,
"google,snow-audio-max98090" or
+ "google,snow-audio-max98091" or
"google,snow-audio-max98095"
- samsung,i2s-controller: The phandle of the Samsung I2S controller
- samsung,audio-codec: The phandle of the audio codec
+Optional:
+- samsung,model: The name of the sound-card
+
Example:
sound {
compatible = "google,snow-audio-max98095";
+ samsung,model = "Snow-I2S-MAX98095";
samsung,i2s-controller = <&i2s0>;
samsung,audio-codec = <&max98095>;
};
diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt
new file mode 100644
index 000000000000..55e2a0af5645
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tas2552.txt
@@ -0,0 +1,26 @@
+Texas Instruments - tas2552 Codec module
+
+The tas2552 serial control bus communicates through I2C protocols
+
+Required properties:
+ - compatible - One of:
+ "ti,tas2552" - TAS2552
+ - reg - I2C slave address
+ - supply-*: Required supply regulators are:
+ "vbat" battery voltage
+ "iovdd" I/O Voltage
+ "avdd" Analog DAC Voltage
+
+Optional properties:
+ - enable-gpio - gpio pin to enable/disable the device
+
+Example:
+
+tas2552: tas2552@41 {
+ compatible = "ti,tas2552";
+ reg = <0x41>;
+ enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>;
+};
+
+For more product information please see the link below:
+http://www.ti.com/product/TAS2552
diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt
index d2866a0d6a26..234dad296da7 100644
--- a/Documentation/devicetree/bindings/sound/ti,tas5086.txt
+++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt
@@ -31,6 +31,9 @@ Optional properties:
Most systems should not set any of these properties.
+ - avdd-supply: Power supply for AVDD, providing 3.3V
+ - dvdd-supply: Power supply for DVDD, providing 3.3V
+
Examples:
i2c_bus {
@@ -39,5 +42,7 @@ Examples:
reg = <0x1b>;
reset-gpio = <&gpio 23 0>;
ti,charge-period = <156000>;
+ avdd-supply = <&vdd_3v3_reg>;
+ dvdd-supply = <&vdd_3v3_reg>;
};
};
diff --git a/Documentation/devicetree/bindings/sound/wm8904.txt b/Documentation/devicetree/bindings/sound/wm8904.txt
new file mode 100644
index 000000000000..e99f4097c83c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8904.txt
@@ -0,0 +1,33 @@
+WM8904 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+ - compatible: "wlf,wm8904"
+ - reg: the I2C address of the device.
+ - clock-names: "mclk"
+ - clocks: reference to
+ <Documentation/devicetree/bindings/clock/clock-bindings.txt>
+
+Pins on the device (for linking into audio routes):
+
+ * IN1L
+ * IN1R
+ * IN2L
+ * IN2R
+ * IN3L
+ * IN3R
+ * HPOUTL
+ * HPOUTR
+ * LINEOUTL
+ * LINEOUTR
+ * MICBIAS
+
+Examples:
+
+codec: wm8904@1a {
+ compatible = "wlf,wm8904";
+ reg = <0x1a>;
+ clocks = <&pck0>;
+ clock-names = "mclk";
+};
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 7ccf933bfbe0..48148d6d9307 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -2026,8 +2026,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
-------------------
Module for sound cards based on the Asus AV66/AV100/AV200 chips,
- i.e., Xonar D1, DX, D2, D2X, DS, Essence ST (Deluxe), Essence STX,
- HDAV1.3 (Deluxe), and HDAV1.3 Slim.
+ i.e., Xonar D1, DX, D2, D2X, DS, DSX, Essence ST (Deluxe),
+ Essence STX (II), HDAV1.3 (Deluxe), and HDAV1.3 Slim.
This module supports autoprobe and multiple cards.
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index d1ab5e17eb13..a5e754714344 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -284,6 +284,11 @@ STAC92HD83*
hp-zephyr HP Zephyr
hp-led HP with broken BIOS for mute LED
hp-inv-led HP with broken BIOS for inverted mute LED
+ hp-mic-led HP with mic-mute LED
+ headset-jack Dell Latitude with a 4-pin headset jack
+ hp-envy-bass Pin fixup for HP Envy bass speaker (NID 0x0f)
+ hp-envy-ts-bass Pin fixup for HP Envy TS bass speaker (NID 0x10)
+ hp-bnb13-eq Hardware equalizer setup for HP laptops
auto BIOS setup (default)
STAC92HD95
diff --git a/MAINTAINERS b/MAINTAINERS
index 12fee4ef936b..731c8a48e19c 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -7526,6 +7526,13 @@ F: drivers/rtc/
F: include/linux/rtc.h
F: include/uapi/linux/rtc.h
+REALTEK AUDIO CODECS
+M: Bard Liao <bardliao@realtek.com>
+M: Oder Chiou <oder_chiou@realtek.com>
+S: Maintained
+F: sound/soc/codecs/rt*
+F: include/sound/rt*.h
+
REISERFS FILE SYSTEM
L: reiserfs-devel@vger.kernel.org
S: Supported
diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c
index 30fcac73a540..689c121157ec 100644
--- a/arch/arm/mach-shmobile/board-armadillo800eva.c
+++ b/arch/arm/mach-shmobile/board-armadillo800eva.c
@@ -998,6 +998,8 @@ static struct platform_device fsi_wm8978_device = {
.id = 0,
.dev = {
.platform_data = &fsi_wm8978_info,
+ .coherent_dma_mask = DMA_BIT_MASK(32),
+ .dma_mask = &fsi_wm8978_device.dev.coherent_dma_mask,
},
};
@@ -1021,6 +1023,8 @@ static struct platform_device fsi_hdmi_device = {
.id = 1,
.dev = {
.platform_data = &fsi2_hdmi_info,
+ .coherent_dma_mask = DMA_BIT_MASK(32),
+ .dma_mask = &fsi_hdmi_device.dev.coherent_dma_mask,
},
};
diff --git a/arch/arm/mach-shmobile/board-kzm9g.c b/arch/arm/mach-shmobile/board-kzm9g.c
index f94ec8ca42c1..01e0d1386db7 100644
--- a/arch/arm/mach-shmobile/board-kzm9g.c
+++ b/arch/arm/mach-shmobile/board-kzm9g.c
@@ -603,6 +603,8 @@ static struct platform_device fsi_ak4648_device = {
.name = "asoc-simple-card",
.dev = {
.platform_data = &fsi2_ak4648_info,
+ .coherent_dma_mask = DMA_BIT_MASK(32),
+ .dma_mask = &fsi_ak4648_device.dev.coherent_dma_mask,
},
};
diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c
index 0ff4d8e45cf7..112553f0f9bf 100644
--- a/arch/arm/mach-shmobile/board-mackerel.c
+++ b/arch/arm/mach-shmobile/board-mackerel.c
@@ -523,6 +523,8 @@ static struct platform_device fsi_hdmi_device = {
.id = 1,
.dev = {
.platform_data = &fsi2_hdmi_info,
+ .coherent_dma_mask = DMA_BIT_MASK(32),
+ .dma_mask = &fsi_hdmi_device.dev.coherent_dma_mask,
},
};
@@ -919,6 +921,8 @@ static struct platform_device fsi_ak4643_device = {
.name = "asoc-simple-card",
.dev = {
.platform_data = &fsi2_ak4643_info,
+ .coherent_dma_mask = DMA_BIT_MASK(32),
+ .dma_mask = &fsi_ak4643_device.dev.coherent_dma_mask,
},
};
diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c
index 85d5255d259f..0d3049244cd3 100644
--- a/arch/sh/boards/mach-ecovec24/setup.c
+++ b/arch/sh/boards/mach-ecovec24/setup.c
@@ -874,6 +874,8 @@ static struct platform_device fsi_da7210_device = {
.name = "asoc-simple-card",
.dev = {
.platform_data = &fsi_da7210_info,
+ .coherent_dma_mask = DMA_BIT_MASK(32),
+ .dma_mask = &fsi_da7210_device.dev.coherent_dma_mask,
},
};
diff --git a/arch/x86/include/asm/platform_sst_audio.h b/arch/x86/include/asm/platform_sst_audio.h
new file mode 100644
index 000000000000..0a4e140315b6
--- /dev/null
+++ b/arch/x86/include/asm/platform_sst_audio.h
@@ -0,0 +1,78 @@
+/*
+ * platform_sst_audio.h: sst audio platform data header file
+ *
+ * Copyright (C) 2012-14 Intel Corporation
+ * Author: Jeeja KP <jeeja.kp@intel.com>
+ * Omair Mohammed Abdullah <omair.m.abdullah@intel.com>
+ * Vinod Koul ,vinod.koul@intel.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; version 2
+ * of the License.
+ */
+#ifndef _PLATFORM_SST_AUDIO_H_
+#define _PLATFORM_SST_AUDIO_H_
+
+#include <linux/sfi.h>
+
+enum sst_audio_task_id_mrfld {
+ SST_TASK_ID_NONE = 0,
+ SST_TASK_ID_SBA = 1,
+ SST_TASK_ID_MEDIA = 3,
+ SST_TASK_ID_MAX = SST_TASK_ID_MEDIA,
+};
+
+/* Device IDs for Merrifield are Pipe IDs,
+ * ref: DSP spec v0.75 */
+enum sst_audio_device_id_mrfld {
+ /* Output pipeline IDs */
+ PIPE_ID_OUT_START = 0x0,
+ PIPE_CODEC_OUT0 = 0x2,
+ PIPE_CODEC_OUT1 = 0x3,
+ PIPE_SPROT_LOOP_OUT = 0x4,
+ PIPE_MEDIA_LOOP1_OUT = 0x5,
+ PIPE_MEDIA_LOOP2_OUT = 0x6,
+ PIPE_VOIP_OUT = 0xC,
+ PIPE_PCM0_OUT = 0xD,
+ PIPE_PCM1_OUT = 0xE,
+ PIPE_PCM2_OUT = 0xF,
+ PIPE_MEDIA0_OUT = 0x12,
+ PIPE_MEDIA1_OUT = 0x13,
+/* Input Pipeline IDs */
+ PIPE_ID_IN_START = 0x80,
+ PIPE_CODEC_IN0 = 0x82,
+ PIPE_CODEC_IN1 = 0x83,
+ PIPE_SPROT_LOOP_IN = 0x84,
+ PIPE_MEDIA_LOOP1_IN = 0x85,
+ PIPE_MEDIA_LOOP2_IN = 0x86,
+ PIPE_VOIP_IN = 0x8C,
+ PIPE_PCM0_IN = 0x8D,
+ PIPE_PCM1_IN = 0x8E,
+ PIPE_MEDIA0_IN = 0x8F,
+ PIPE_MEDIA1_IN = 0x90,
+ PIPE_MEDIA2_IN = 0x91,
+ PIPE_RSVD = 0xFF,
+};
+
+/* The stream map for each platform consists of an array of the below
+ * stream map structure.
+ */
+struct sst_dev_stream_map {
+ u8 dev_num; /* device id */
+ u8 subdev_num; /* substream */
+ u8 direction;
+ u8 device_id; /* fw id */
+ u8 task_id; /* fw task */
+ u8 status;
+};
+
+struct sst_platform_data {
+ /* Intel software platform id*/
+ struct sst_dev_stream_map *pdev_strm_map;
+ unsigned int strm_map_size;
+};
+
+int add_sst_platform_device(void);
+#endif
+
diff --git a/drivers/dma/edma.c b/drivers/dma/edma.c
index d08c4dedef35..b512caf46944 100644
--- a/drivers/dma/edma.c
+++ b/drivers/dma/edma.c
@@ -982,6 +982,7 @@ static void __init edma_chan_init(struct edma_cc *ecc,
#define EDMA_DMA_BUSWIDTHS (BIT(DMA_SLAVE_BUSWIDTH_1_BYTE) | \
BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | \
+ BIT(DMA_SLAVE_BUSWIDTH_3_BYTES) | \
BIT(DMA_SLAVE_BUSWIDTH_4_BYTES))
static int edma_dma_device_slave_caps(struct dma_chan *dchan,
diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c
index 22de13727641..60843a275abd 100644
--- a/drivers/misc/atmel-ssc.c
+++ b/drivers/misc/atmel-ssc.c
@@ -83,10 +83,17 @@ EXPORT_SYMBOL(ssc_free);
static struct atmel_ssc_platform_data at91rm9200_config = {
.use_dma = 0,
+ .has_fslen_ext = 0,
+};
+
+static struct atmel_ssc_platform_data at91sam9rl_config = {
+ .use_dma = 0,
+ .has_fslen_ext = 1,
};
static struct atmel_ssc_platform_data at91sam9g45_config = {
.use_dma = 1,
+ .has_fslen_ext = 1,
};
static const struct platform_device_id atmel_ssc_devtypes[] = {
@@ -94,6 +101,9 @@ static const struct platform_device_id atmel_ssc_devtypes[] = {
.name = "at91rm9200_ssc",
.driver_data = (unsigned long) &at91rm9200_config,
}, {
+ .name = "at91sam9rl_ssc",
+ .driver_data = (unsigned long) &at91sam9rl_config,
+ }, {
.name = "at91sam9g45_ssc",
.driver_data = (unsigned long) &at91sam9g45_config,
}, {
@@ -107,6 +117,9 @@ static const struct of_device_id atmel_ssc_dt_ids[] = {
.compatible = "atmel,at91rm9200-ssc",
.data = &at91rm9200_config,
}, {
+ .compatible = "atmel,at91sam9rl-ssc",
+ .data = &at91sam9rl_config,
+ }, {
.compatible = "atmel,at91sam9g45-ssc",
.data = &at91sam9g45_config,
}, {
diff --git a/include/linux/atmel-ssc.h b/include/linux/atmel-ssc.h
index 571a12ebb018..7c0f6549898b 100644
--- a/include/linux/atmel-ssc.h
+++ b/include/linux/atmel-ssc.h
@@ -7,6 +7,7 @@
struct atmel_ssc_platform_data {
int use_dma;
+ int has_fslen_ext;
};
struct ssc_device {
@@ -71,6 +72,12 @@ void ssc_free(struct ssc_device *ssc);
#define SSC_RFMR_DATNB_OFFSET 8
#define SSC_RFMR_FSEDGE_SIZE 1
#define SSC_RFMR_FSEDGE_OFFSET 24
+/*
+ * The FSLEN_EXT exist on at91sam9rl, at91sam9g10,
+ * at91sam9g20, and at91sam9g45 and newer SoCs
+ */
+#define SSC_RFMR_FSLEN_EXT_SIZE 4
+#define SSC_RFMR_FSLEN_EXT_OFFSET 28
#define SSC_RFMR_FSLEN_SIZE 4
#define SSC_RFMR_FSLEN_OFFSET 16
#define SSC_RFMR_FSOS_SIZE 4
@@ -109,6 +116,12 @@ void ssc_free(struct ssc_device *ssc);
#define SSC_TFMR_FSDEN_OFFSET 23
#define SSC_TFMR_FSEDGE_SIZE 1
#define SSC_TFMR_FSEDGE_OFFSET 24
+/*
+ * The FSLEN_EXT exist on at91sam9rl, at91sam9g10,
+ * at91sam9g20, and at91sam9g45 and newer SoCs
+ */
+#define SSC_TFMR_FSLEN_EXT_SIZE 4
+#define SSC_TFMR_FSLEN_EXT_OFFSET 28
#define SSC_TFMR_FSLEN_SIZE 4
#define SSC_TFMR_FSLEN_OFFSET 16
#define SSC_TFMR_FSOS_SIZE 3
diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h
index d2c5cc7c583c..3d1c2aa51530 100644
--- a/include/linux/dmaengine.h
+++ b/include/linux/dmaengine.h
@@ -299,6 +299,7 @@ enum dma_slave_buswidth {
DMA_SLAVE_BUSWIDTH_UNDEFINED = 0,
DMA_SLAVE_BUSWIDTH_1_BYTE = 1,
DMA_SLAVE_BUSWIDTH_2_BYTES = 2,
+ DMA_SLAVE_BUSWIDTH_3_BYTES = 3,
DMA_SLAVE_BUSWIDTH_4_BYTES = 4,
DMA_SLAVE_BUSWIDTH_8_BYTES = 8,
};
diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h
index 6d9371f88875..a614b33d0a39 100644
--- a/include/linux/mfd/arizona/core.h
+++ b/include/linux/mfd/arizona/core.h
@@ -110,6 +110,12 @@ struct arizona {
int clk32k_ref;
struct snd_soc_dapm_context *dapm;
+
+ int tdm_width[ARIZONA_MAX_AIF];
+ int tdm_slots[ARIZONA_MAX_AIF];
+
+ uint16_t dac_comp_coeff;
+ uint8_t dac_comp_enabled;
};
int arizona_clk32k_enable(struct arizona *arizona);
diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h
index 709c6f7e2f8c..a6591c693ebb 100644
--- a/include/linux/platform_data/asoc-s3c.h
+++ b/include/linux/platform_data/asoc-s3c.h
@@ -15,15 +15,6 @@
#define S3C64XX_AC97_GPE 1
extern void s3c64xx_ac97_setup_gpio(int);
-/*
- * The machine init code calls s5p*_spdif_setup_gpio with
- * one of these defines in order to select appropriate bank
- * of GPIO for S/PDIF pins
- */
-#define S5PC100_SPDIF_GPD 0
-#define S5PC100_SPDIF_GPG3 1
-extern void s5pc100_spdif_setup_gpio(int);
-
struct samsung_i2s {
/* If the Primary DAI has 5.1 Channels */
#define QUIRK_PRI_6CHAN (1 << 0)
diff --git a/include/linux/platform_data/dma-imx.h b/include/linux/platform_data/dma-imx.h
index bcbc6c3c14c0..d05542aafa3e 100644
--- a/include/linux/platform_data/dma-imx.h
+++ b/include/linux/platform_data/dma-imx.h
@@ -50,6 +50,7 @@ enum imx_dma_prio {
struct imx_dma_data {
int dma_request; /* DMA request line */
+ int dma_request2; /* secondary DMA request line */
enum sdma_peripheral_type peripheral_type;
int priority;
};
diff --git a/include/sound/control.h b/include/sound/control.h
index 5358892b1b39..042613938a1d 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -31,10 +31,15 @@ typedef int (snd_kcontrol_info_t) (struct snd_kcontrol * kcontrol, struct snd_ct
typedef int (snd_kcontrol_get_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol);
typedef int (snd_kcontrol_put_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol);
typedef int (snd_kcontrol_tlv_rw_t)(struct snd_kcontrol *kcontrol,
- int op_flag, /* 0=read,1=write,-1=command */
+ int op_flag, /* SNDRV_CTL_TLV_OP_XXX */
unsigned int size,
unsigned int __user *tlv);
+enum {
+ SNDRV_CTL_TLV_OP_READ = 0,
+ SNDRV_CTL_TLV_OP_WRITE = 1,
+ SNDRV_CTL_TLV_OP_CMD = -1,
+};
struct snd_kcontrol_new {
snd_ctl_elem_iface_t iface; /* interface identifier */
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index d854fb31c000..6f3e10ca0e32 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -931,10 +931,17 @@ void snd_pcm_timer_done(struct snd_pcm_substream *substream);
static inline void snd_pcm_gettime(struct snd_pcm_runtime *runtime,
struct timespec *tv)
{
- if (runtime->tstamp_type == SNDRV_PCM_TSTAMP_TYPE_MONOTONIC)
+ switch (runtime->tstamp_type) {
+ case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC:
ktime_get_ts(tv);
- else
+ break;
+ case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW:
+ getrawmonotonic(tv);
+ break;
+ default:
getnstimeofday(tv);
+ break;
+ }
}
/*
diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h
index f4a706f82cb7..d76412b84b48 100644
--- a/include/sound/rcar_snd.h
+++ b/include/sound/rcar_snd.h
@@ -34,6 +34,7 @@
* B : SSI direction
*/
#define RSND_SSI_CLK_PIN_SHARE (1 << 31)
+#define RSND_SSI_NO_BUSIF (1 << 30) /* SSI+DMA without BUSIF */
#define RSND_SSI(_dma_id, _pio_irq, _flags) \
{ .dma_id = _dma_id, .pio_irq = _pio_irq, .flags = _flags }
diff --git a/include/sound/rt286.h b/include/sound/rt286.h
new file mode 100644
index 000000000000..eb773d1485f2
--- /dev/null
+++ b/include/sound/rt286.h
@@ -0,0 +1,19 @@
+/*
+ * linux/sound/rt286.h -- Platform data for RT286
+ *
+ * Copyright 2013 Realtek Microelectronics
+ *
+ * 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 __LINUX_SND_RT286_H
+#define __LINUX_SND_RT286_H
+
+struct rt286_platform_data {
+ bool cbj_en; /*combo jack enable*/
+ bool gpio2_en; /*GPIO2 enable*/
+};
+
+#endif
diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h
new file mode 100644
index 000000000000..bd311197a3b5
--- /dev/null
+++ b/include/sound/rt5670.h
@@ -0,0 +1,27 @@
+/*
+ * linux/sound/rt5670.h -- Platform data for RT5670
+ *
+ * Copyright 2014 Realtek Microelectronics
+ *
+ * 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 __LINUX_SND_RT5670_H
+#define __LINUX_SND_RT5670_H
+
+struct rt5670_platform_data {
+ int jd_mode;
+ bool in2_diff;
+
+ bool dmic_en;
+ unsigned int dmic1_data_pin;
+ /* 0 = GPIO6; 1 = IN2P; 3 = GPIO7*/
+ unsigned int dmic2_data_pin;
+ /* 0 = GPIO8; 1 = IN3N; */
+ unsigned int dmic3_data_pin;
+ /* 0 = GPIO9; 1 = GPIO10; 2 = GPIO5*/
+};
+
+#endif
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 688f2ba8009f..e8b3080d196a 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -257,7 +257,6 @@ struct snd_soc_dai {
struct snd_soc_dapm_widget *playback_widget;
struct snd_soc_dapm_widget *capture_widget;
- struct snd_soc_dapm_context dapm;
/* DAI DMA data */
void *playback_dma_data;
@@ -273,6 +272,10 @@ struct snd_soc_dai {
struct snd_soc_codec *codec;
struct snd_soc_component *component;
+ /* CODEC TDM slot masks and params (for fixup) */
+ unsigned int tx_mask;
+ unsigned int rx_mask;
+
struct snd_soc_card *card;
struct list_head list;
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 6b59471cdf44..aac04ff84eea 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -431,7 +431,7 @@ int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm,
const char *pin);
int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
const char *pin);
-void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec);
+void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card);
/* Mostly internal - should not normally be used */
void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm);
@@ -441,6 +441,8 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
struct snd_soc_dapm_widget_list **list);
struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol);
+struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm(
+ struct snd_kcontrol *kcontrol);
/* dapm widget types */
enum snd_soc_dapm_type {
@@ -524,7 +526,6 @@ struct snd_soc_dapm_widget {
const char *name; /* widget name */
const char *sname; /* stream name */
struct snd_soc_codec *codec;
- struct snd_soc_platform *platform;
struct list_head list;
struct snd_soc_dapm_context *dapm;
@@ -593,7 +594,6 @@ struct snd_soc_dapm_context {
struct device *dev; /* from parent - for debug */
struct snd_soc_component *component; /* parent component */
struct snd_soc_codec *codec; /* parent codec */
- struct snd_soc_platform *platform; /* parent platform */
struct snd_soc_card *card; /* parent card */
/* used during DAPM updates */
@@ -601,6 +601,8 @@ struct snd_soc_dapm_context {
struct list_head list;
int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
+ int (*set_bias_level)(struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level);
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_dapm;
diff --git a/include/sound/soc.h b/include/sound/soc.h
index ed9e2d7e5fdc..be6ecae247b0 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -248,6 +248,8 @@
.info = snd_soc_info_enum_double, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = (unsigned long)&xenum }
+#define SOC_VALUE_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \
+ SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put)
#define SND_SOC_BYTES(xname, xbase, xregs) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -270,7 +272,14 @@
.get = xhandler_get, .put = xhandler_put, \
.private_value = (unsigned long)&(struct soc_bytes_ext) \
{.max = xcount} }
-
+#define SND_SOC_BYTES_TLV(xname, xcount, xhandler_get, xhandler_put) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
+ .tlv.c = (snd_soc_bytes_tlv_callback), \
+ .info = snd_soc_info_bytes_ext, \
+ .private_value = (unsigned long)&(struct soc_bytes_ext) \
+ {.max = xcount, .get = xhandler_get, .put = xhandler_put, } }
#define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \
xmin, xmax, xinvert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
@@ -436,6 +445,10 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
int snd_soc_platform_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_platform *platform);
+int soc_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai);
+
/* Jack reporting */
int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type,
struct snd_soc_jack *jack);
@@ -503,10 +516,12 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
const char *prefix);
struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card,
const char *name);
+int snd_soc_add_component_controls(struct snd_soc_component *component,
+ const struct snd_kcontrol_new *controls, unsigned int num_controls);
int snd_soc_add_codec_controls(struct snd_soc_codec *codec,
- const struct snd_kcontrol_new *controls, int num_controls);
+ const struct snd_kcontrol_new *controls, unsigned int num_controls);
int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
- const struct snd_kcontrol_new *controls, int num_controls);
+ const struct snd_kcontrol_new *controls, unsigned int num_controls);
int snd_soc_add_card_controls(struct snd_soc_card *soc_card,
const struct snd_kcontrol_new *controls, int num_controls);
int snd_soc_add_dai_controls(struct snd_soc_dai *dai,
@@ -552,6 +567,8 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *ucontrol);
+int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *tlv);
int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol,
@@ -677,12 +694,17 @@ struct snd_soc_component_driver {
int (*of_xlate_dai_name)(struct snd_soc_component *component,
struct of_phandle_args *args,
const char **dai_name);
+ void (*seq_notifier)(struct snd_soc_component *, enum snd_soc_dapm_type,
+ int subseq);
+ int (*stream_event)(struct snd_soc_component *, int event);
};
struct snd_soc_component {
const char *name;
int id;
+ const char *name_prefix;
struct device *dev;
+ struct snd_soc_card *card;
unsigned int active;
@@ -705,18 +727,18 @@ struct snd_soc_component {
int val_bytes;
struct mutex io_mutex;
+
+ /* Don't use these, use snd_soc_component_get_dapm() */
+ struct snd_soc_dapm_context dapm;
+ struct snd_soc_dapm_context *dapm_ptr;
};
/* SoC Audio Codec device */
struct snd_soc_codec {
- const char *name;
- const char *name_prefix;
- int id;
struct device *dev;
const struct snd_soc_codec_driver *driver;
struct mutex mutex;
- struct snd_soc_card *card;
struct list_head list;
struct list_head card_list;
@@ -790,9 +812,6 @@ struct snd_soc_codec_driver {
void (*seq_notifier)(struct snd_soc_dapm_context *,
enum snd_soc_dapm_type, int);
- /* codec stream completion event */
- int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
-
bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */
/* probe ordering - for components with runtime dependencies */
@@ -834,9 +853,6 @@ struct snd_soc_platform_driver {
/* platform stream compress ops */
const struct snd_compr_ops *compr_ops;
- /* platform stream completion event */
- int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
-
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
@@ -847,23 +863,23 @@ struct snd_soc_platform_driver {
int (*bespoke_trigger)(struct snd_pcm_substream *, int);
};
-struct snd_soc_platform {
+struct snd_soc_dai_link_component {
const char *name;
- int id;
+ const struct device_node *of_node;
+ const char *dai_name;
+};
+
+struct snd_soc_platform {
struct device *dev;
const struct snd_soc_platform_driver *driver;
unsigned int suspended:1; /* platform is suspended */
unsigned int probed:1;
- struct snd_soc_card *card;
struct list_head list;
- struct list_head card_list;
struct snd_soc_component component;
- struct snd_soc_dapm_context dapm;
-
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_platform_root;
#endif
@@ -896,6 +912,10 @@ struct snd_soc_dai_link {
const struct device_node *codec_of_node;
/* You MUST specify the DAI name within the codec */
const char *codec_dai_name;
+
+ struct snd_soc_dai_link_component *codecs;
+ unsigned int num_codecs;
+
/*
* You MAY specify the link's platform/PCM/DMA driver, either by
* device name, or by DT/OF node, but not both. Some forms of link
@@ -1047,7 +1067,6 @@ struct snd_soc_card {
/* lists of probed devices belonging to this card */
struct list_head codec_dev_list;
- struct list_head platform_dev_list;
struct list_head widgets;
struct list_head paths;
@@ -1094,6 +1113,9 @@ struct snd_soc_pcm_runtime {
struct snd_soc_dai *codec_dai;
struct snd_soc_dai *cpu_dai;
+ struct snd_soc_dai **codec_dais;
+ unsigned int num_codecs;
+
struct delayed_work delayed_work;
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_dpcm_root;
@@ -1119,6 +1141,9 @@ struct soc_bytes {
struct soc_bytes_ext {
int max;
+ /* used for TLV byte control */
+ int (*get)(unsigned int __user *bytes, unsigned int size);
+ int (*put)(const unsigned int __user *bytes, unsigned int size);
};
/* multi register control */
@@ -1165,6 +1190,21 @@ static inline struct snd_soc_platform *snd_soc_component_to_platform(
}
/**
+ * snd_soc_dapm_to_component() - Casts a DAPM context to the component it is
+ * embedded in
+ * @dapm: The DAPM context to cast to the component
+ *
+ * This function must only be used on DAPM contexts that are known to be part of
+ * a component (e.g. in a component driver). Otherwise the behavior is
+ * undefined.
+ */
+static inline struct snd_soc_component *snd_soc_dapm_to_component(
+ struct snd_soc_dapm_context *dapm)
+{
+ return container_of(dapm, struct snd_soc_component, dapm);
+}
+
+/**
* snd_soc_dapm_to_codec() - Casts a DAPM context to the CODEC it is embedded in
* @dapm: The DAPM context to cast to the CODEC
*
@@ -1188,7 +1228,18 @@ static inline struct snd_soc_codec *snd_soc_dapm_to_codec(
static inline struct snd_soc_platform *snd_soc_dapm_to_platform(
struct snd_soc_dapm_context *dapm)
{
- return container_of(dapm, struct snd_soc_platform, dapm);
+ return snd_soc_component_to_platform(snd_soc_dapm_to_component(dapm));
+}
+
+/**
+ * snd_soc_component_get_dapm() - Returns the DAPM context associated with a
+ * component
+ * @component: The component for which to get the DAPM context
+ */
+static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm(
+ struct snd_soc_component *component)
+{
+ return component->dapm_ptr;
}
/* codec IO */
@@ -1261,7 +1312,6 @@ static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd)
static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
{
INIT_LIST_HEAD(&card->codec_dev_list);
- INIT_LIST_HEAD(&card->platform_dev_list);
INIT_LIST_HEAD(&card->widgets);
INIT_LIST_HEAD(&card->paths);
INIT_LIST_HEAD(&card->dapm_list);
diff --git a/include/sound/tas2552-plat.h b/include/sound/tas2552-plat.h
new file mode 100644
index 000000000000..65e7627ba38e
--- /dev/null
+++ b/include/sound/tas2552-plat.h
@@ -0,0 +1,25 @@
+/*
+ * TAS2552 driver platform header
+ *
+ * Copyright (C) 2014 Texas Instruments Inc.
+ *
+ * Author: Dan Murphy <dmurphy@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.
+ */
+
+#ifndef TAS2552_PLAT_H
+#define TAS2552_PLAT_H
+
+struct tas2552_platform_data {
+ int enable_gpio;
+};
+
+#endif
diff --git a/include/sound/wm8962.h b/include/sound/wm8962.h
index 79e6d427b858..0af7c1674cbf 100644
--- a/include/sound/wm8962.h
+++ b/include/sound/wm8962.h
@@ -37,6 +37,7 @@
#define WM8962_GPIO_FN_MICSCD 22
struct wm8962_pdata {
+ struct clk *mclk;
int gpio_base;
u32 gpio_init[WM8962_MAX_GPIO];
diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h
index c75c795a377b..0194a641e4e2 100644
--- a/include/trace/events/asoc.h
+++ b/include/trace/events/asoc.h
@@ -296,17 +296,17 @@ TRACE_EVENT(snd_soc_cache_sync,
TP_ARGS(codec, type, status),
TP_STRUCT__entry(
- __string( name, codec->name )
+ __string( name, codec->component.name)
__string( status, status )
__string( type, type )
__field( int, id )
),
TP_fast_assign(
- __assign_str(name, codec->name);
+ __assign_str(name, codec->component.name);
__assign_str(status, status);
__assign_str(type, type);
- __entry->id = codec->id;
+ __entry->id = codec->component.id;
),
TP_printk("codec=%s.%d type=%s status=%s", __get_str(name),
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index 224948342f14..32168f7ffce3 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -139,7 +139,7 @@ struct snd_hwdep_dsp_image {
* *
*****************************************************************************/
-#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 11)
+#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 12)
typedef unsigned long snd_pcm_uframes_t;
typedef signed long snd_pcm_sframes_t;
@@ -391,7 +391,9 @@ struct snd_pcm_sw_params {
snd_pcm_uframes_t silence_threshold; /* min distance from noise for silence filling */
snd_pcm_uframes_t silence_size; /* silence block size */
snd_pcm_uframes_t boundary; /* pointers wrap point */
- unsigned char reserved[64]; /* reserved for future */
+ unsigned int proto; /* protocol version */
+ unsigned int tstamp_type; /* timestamp type (req. proto >= 2.0.12) */
+ unsigned char reserved[56]; /* reserved for future */
};
struct snd_pcm_channel_info {
@@ -462,7 +464,8 @@ struct snd_xfern {
enum {
SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */
SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */
- SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC,
+ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */
+ SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW,
};
/* channel positions */
diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c
index 467836057ee5..a80d5ea87ccd 100644
--- a/sound/aoa/soundbus/i2sbus/core.c
+++ b/sound/aoa/soundbus/i2sbus/core.c
@@ -47,15 +47,11 @@ static int alloc_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev,
/* We use the PCI APIs for now until the generic one gets fixed
* enough or until we get some macio-specific versions
*/
- r->space = dma_alloc_coherent(
- &macio_get_pci_dev(i2sdev->macio)->dev,
- r->size,
- &r->bus_addr,
- GFP_KERNEL);
+ r->space = dma_zalloc_coherent(&macio_get_pci_dev(i2sdev->macio)->dev,
+ r->size, &r->bus_addr, GFP_KERNEL);
+ if (!r->space)
+ return -ENOMEM;
- if (!r->space) return -ENOMEM;
-
- memset(r->space, 0, r->size);
r->cmds = (void*)DBDMA_ALIGN(r->space);
r->bus_cmd_start = r->bus_addr +
(dma_addr_t)((char*)r->cmds - (char*)r->space);
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 66de90ed30ca..39c3969ac1c7 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -152,9 +152,9 @@ static inline void pxa_ac97_cold_pxa27x(void)
gsr_bits = 0;
/* PXA27x Developers Manual section 13.5.2.2.1 */
- clk_enable(ac97conf_clk);
+ clk_prepare_enable(ac97conf_clk);
udelay(5);
- clk_disable(ac97conf_clk);
+ clk_disable_unprepare(ac97conf_clk);
GCR = GCR_COLD_RST | GCR_WARM_RST;
}
#endif
@@ -299,14 +299,14 @@ static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id)
int pxa2xx_ac97_hw_suspend(void)
{
GCR |= GCR_ACLINK_OFF;
- clk_disable(ac97_clk);
+ clk_disable_unprepare(ac97_clk);
return 0;
}
EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_suspend);
int pxa2xx_ac97_hw_resume(void)
{
- clk_enable(ac97_clk);
+ clk_prepare_enable(ac97_clk);
return 0;
}
EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume);
@@ -368,7 +368,7 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev)
goto err_clk;
}
- ret = clk_enable(ac97_clk);
+ ret = clk_prepare_enable(ac97_clk);
if (ret)
goto err_clk2;
@@ -403,7 +403,7 @@ void pxa2xx_ac97_hw_remove(struct platform_device *dev)
clk_put(ac97conf_clk);
ac97conf_clk = NULL;
}
- clk_disable(ac97_clk);
+ clk_disable_unprepare(ac97_clk);
clk_put(ac97_clk);
ac97_clk = NULL;
}
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 7403f348ed14..89028fab64fd 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -491,7 +491,7 @@ static int snd_compress_check_input(struct snd_compr_params *params)
{
/* first let's check the buffer parameter's */
if (params->buffer.fragment_size == 0 ||
- params->buffer.fragments > SIZE_MAX / params->buffer.fragment_size)
+ params->buffer.fragments > INT_MAX / params->buffer.fragment_size)
return -EINVAL;
/* now codec parameters */
diff --git a/sound/core/control.c b/sound/core/control.c
index f0b0e14497a5..b9611344ff9e 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -1406,11 +1406,11 @@ static long snd_ctl_ioctl(struct file *file, unsigned int cmd, unsigned long arg
case SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS:
return snd_ctl_subscribe_events(ctl, ip);
case SNDRV_CTL_IOCTL_TLV_READ:
- return snd_ctl_tlv_ioctl(ctl, argp, 0);
+ return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_READ);
case SNDRV_CTL_IOCTL_TLV_WRITE:
- return snd_ctl_tlv_ioctl(ctl, argp, 1);
+ return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_WRITE);
case SNDRV_CTL_IOCTL_TLV_COMMAND:
- return snd_ctl_tlv_ioctl(ctl, argp, -1);
+ return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_CMD);
case SNDRV_CTL_IOCTL_POWER:
return -ENOPROTOOPT;
case SNDRV_CTL_IOCTL_POWER_STATE:
diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c
index af49721ba0e3..102e8fd1d450 100644
--- a/sound/core/pcm_compat.c
+++ b/sound/core/pcm_compat.c
@@ -101,7 +101,9 @@ struct snd_pcm_sw_params32 {
u32 silence_threshold;
u32 silence_size;
u32 boundary;
- unsigned char reserved[64];
+ u32 proto;
+ u32 tstamp_type;
+ unsigned char reserved[56];
};
/* recalcuate the boundary within 32bit */
@@ -133,7 +135,9 @@ static int snd_pcm_ioctl_sw_params_compat(struct snd_pcm_substream *substream,
get_user(params.start_threshold, &src->start_threshold) ||
get_user(params.stop_threshold, &src->stop_threshold) ||
get_user(params.silence_threshold, &src->silence_threshold) ||
- get_user(params.silence_size, &src->silence_size))
+ get_user(params.silence_size, &src->silence_size) ||
+ get_user(params.tstamp_type, &src->tstamp_type) ||
+ get_user(params.proto, &src->proto))
return -EFAULT;
/*
* Check silent_size parameter. Since we have 64bit boundary,
diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c
index 76cbb9ec953a..6542c4083594 100644
--- a/sound/core/pcm_dmaengine.c
+++ b/sound/core/pcm_dmaengine.c
@@ -65,13 +65,15 @@ int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream,
enum dma_slave_buswidth buswidth;
int bits;
- bits = snd_pcm_format_physical_width(params_format(params));
+ bits = params_physical_width(params);
if (bits < 8 || bits > 64)
return -EINVAL;
else if (bits == 8)
buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE;
else if (bits == 16)
buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
+ else if (bits == 24)
+ buswidth = DMA_SLAVE_BUSWIDTH_3_BYTES;
else if (bits <= 32)
buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES;
else
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index b653ab001fba..8cd2f930ad0b 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -543,6 +543,9 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream,
if (params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST)
return -EINVAL;
+ if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12) &&
+ params->tstamp_type > SNDRV_PCM_TSTAMP_TYPE_LAST)
+ return -EINVAL;
if (params->avail_min == 0)
return -EINVAL;
if (params->silence_size >= runtime->boundary) {
@@ -557,6 +560,8 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream,
err = 0;
snd_pcm_stream_lock_irq(substream);
runtime->tstamp_mode = params->tstamp_mode;
+ if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12))
+ runtime->tstamp_type = params->tstamp_type;
runtime->period_step = params->period_step;
runtime->control->avail_min = params->avail_min;
runtime->start_threshold = params->start_threshold;
@@ -2540,9 +2545,7 @@ static int snd_pcm_tstamp(struct snd_pcm_substream *substream, int __user *_arg)
return -EFAULT;
if (arg < 0 || arg > SNDRV_PCM_TSTAMP_TYPE_LAST)
return -EINVAL;
- runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY;
- if (arg == SNDRV_PCM_TSTAMP_TYPE_MONOTONIC)
- runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC;
+ runtime->tstamp_type = arg;
return 0;
}
diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c
index 1e206de0c2dd..ba8e4a64e13e 100644
--- a/sound/core/seq/seq_memory.c
+++ b/sound/core/seq/seq_memory.c
@@ -101,9 +101,9 @@ int snd_seq_dump_var_event(const struct snd_seq_event *event,
len -= size;
}
return 0;
- } if (! (event->data.ext.len & SNDRV_SEQ_EXT_CHAINED)) {
- return func(private_data, event->data.ext.ptr, len);
}
+ if (!(event->data.ext.len & SNDRV_SEQ_EXT_CHAINED))
+ return func(private_data, event->data.ext.ptr, len);
cell = (struct snd_seq_event_cell *)event->data.ext.ptr;
for (; len > 0 && cell; cell = cell->next) {
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index 775ef2efc296..46dff64908c8 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -83,8 +83,8 @@ config SND_BEBOB
* Edirol FA-66/FA-101
* PreSonus FIREBOX/FIREPOD/FP10/Inspire1394
* BridgeCo RDAudio1/Audio5
- * Mackie Onyx 1220/1620/1640 (Firewire I/O Card)
- * Mackie d.2 (Firewire Option)
+ * Mackie Onyx 1220/1620/1640 (FireWire I/O Card)
+ * Mackie d.2 (FireWire Option)
* Stanton FinalScratch 2 (ScratchAmp)
* Tascam IF-FW/DM
* Behringer XENIX UFX 1204/1604
@@ -92,7 +92,7 @@ config SND_BEBOB
* Apogee Rosetta 200/400 (X-FireWire card)
* Apogee DA/AD/DD-16X (X-FireWire card)
* Apogee Ensemble
- * ESI Quotafire610
+ * ESI QuataFire 610
* AcousticReality eARMasterOne
* CME MatrixKFW
* Phonic Helix Board 12 MkII/18 MkII/24 MkII
@@ -101,13 +101,13 @@ config SND_BEBOB
* ICON FireXon
* PrismSound Orpheus/ADA-8XR
* TerraTec PHASE 24 FW/PHASE X24 FW/PHASE 88 Rack FW
- * Terratec EWS MIC2/EWS MIC4
- * Terratec Aureon 7.1 Firewire
+ * TerraTec EWS MIC2/EWS MIC8
+ * TerraTec Aureon 7.1 FireWire
* Yamaha GO44/GO46
* Focusrite Saffire/Saffire LE/SaffirePro10 IO/SaffirePro26 IO
- * M-Audio Firewire410/AudioPhile/Solo
+ * M-Audio FireWire410/AudioPhile/Solo
* M-Audio Ozonic/NRV10/ProfireLightBridge
- * M-Audio Firewire 1814/ProjectMix IO
+ * M-Audio FireWire 1814/ProjectMix IO
To compile this driver as a module, choose M here: the module
will be called snd-bebob.
diff --git a/sound/firewire/fireworks/fireworks_proc.c b/sound/firewire/fireworks/fireworks_proc.c
index f29d4aaf56a1..0639dcb13f7d 100644
--- a/sound/firewire/fireworks/fireworks_proc.c
+++ b/sound/firewire/fireworks/fireworks_proc.c
@@ -64,7 +64,7 @@ proc_read_hwinfo(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
hwinfo->phys_in_grp_count);
for (i = 0; i < hwinfo->phys_in_grp_count; i++) {
snd_iprintf(buffer,
- "phys in grp[0x%d]: type 0x%d, count 0x%d\n",
+ "phys in grp[%d]: type 0x%X, count 0x%X\n",
i, hwinfo->phys_out_grps[i].type,
hwinfo->phys_out_grps[i].count);
}
@@ -73,7 +73,7 @@ proc_read_hwinfo(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
hwinfo->phys_out_grp_count);
for (i = 0; i < hwinfo->phys_out_grp_count; i++) {
snd_iprintf(buffer,
- "phys out grps[0x%d]: type 0x%d, count 0x%d\n",
+ "phys out grps[%d]: type 0x%X, count 0x%X\n",
i, hwinfo->phys_out_grps[i].type,
hwinfo->phys_out_grps[i].count);
}
diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c
index 3bbc3ec5be82..862735005b43 100644
--- a/sound/oss/mpu401.c
+++ b/sound/oss/mpu401.c
@@ -316,6 +316,7 @@ static int mpu_input_scanner(struct mpu_config *devc, unsigned char midic)
case 0xf6:
/* printk( "tune_request\n"); */
devc->m_state = ST_INIT;
+ break;
/*
* Real time messages
@@ -972,7 +973,6 @@ int attach_mpu401(struct address_info *hw_config, struct module *owner)
devc->m_busy = 0;
devc->m_state = ST_INIT;
devc->shared_irq = hw_config->always_detect;
- devc->irq = hw_config->irq;
spin_lock_init(&devc->lock);
if (devc->irq < 0)
diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c
index 4709e592e2cc..607cee4d545e 100644
--- a/sound/oss/opl3.c
+++ b/sound/oss/opl3.c
@@ -52,7 +52,7 @@ struct voice_info
int panning; /* 0xffff means not set */
};
-typedef struct opl_devinfo
+struct opl_devinfo
{
int base;
int left_io, right_io;
@@ -73,7 +73,7 @@ typedef struct opl_devinfo
unsigned char cmask;
int is_opl4;
-} opl_devinfo;
+};
static struct opl_devinfo *devc = NULL;
diff --git a/sound/oss/pss.c b/sound/oss/pss.c
index 145e36b2cfd0..ca0d6e9f49f5 100644
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -123,25 +123,25 @@ static bool pss_mixer;
#endif
-typedef struct pss_mixerdata {
+struct pss_mixerdata {
unsigned int volume_l;
unsigned int volume_r;
unsigned int bass;
unsigned int treble;
unsigned int synth;
-} pss_mixerdata;
+};
-typedef struct pss_confdata {
+struct pss_confdata {
int base;
int irq;
int dma;
int *osp;
- pss_mixerdata mixer;
+ struct pss_mixerdata mixer;
int ad_mixer_dev;
-} pss_confdata;
+};
-static pss_confdata pss_data;
-static pss_confdata *devc = &pss_data;
+static struct pss_confdata pss_data;
+static struct pss_confdata *devc = &pss_data;
static DEFINE_SPINLOCK(lock);
static int pss_initialized;
@@ -150,7 +150,7 @@ static int pss_cdrom_port = -1; /* Parameter for the PSS cdrom port */
static bool pss_enable_joystick; /* Parameter for enabling the joystick */
static coproc_operations pss_coproc_operations;
-static void pss_write(pss_confdata *devc, int data)
+static void pss_write(struct pss_confdata *devc, int data)
{
unsigned long i, limit;
@@ -206,7 +206,7 @@ static int __init probe_pss(struct address_info *hw_config)
return 1;
}
-static int set_irq(pss_confdata * devc, int dev, int irq)
+static int set_irq(struct pss_confdata *devc, int dev, int irq)
{
static unsigned short irq_bits[16] =
{
@@ -232,7 +232,7 @@ static int set_irq(pss_confdata * devc, int dev, int irq)
return 1;
}
-static void set_io_base(pss_confdata * devc, int dev, int base)
+static void set_io_base(struct pss_confdata *devc, int dev, int base)
{
unsigned short tmp = inw(REG(dev)) & 0x003f;
unsigned short bits = (base & 0x0ffc) << 4;
@@ -240,7 +240,7 @@ static void set_io_base(pss_confdata * devc, int dev, int base)
outw(bits | tmp, REG(dev));
}
-static int set_dma(pss_confdata * devc, int dev, int dma)
+static int set_dma(struct pss_confdata *devc, int dev, int dma)
{
static unsigned short dma_bits[8] =
{
@@ -264,7 +264,7 @@ static int set_dma(pss_confdata * devc, int dev, int dma)
return 1;
}
-static int pss_reset_dsp(pss_confdata * devc)
+static int pss_reset_dsp(struct pss_confdata *devc)
{
unsigned long i, limit = jiffies + HZ/10;
@@ -275,7 +275,7 @@ static int pss_reset_dsp(pss_confdata * devc)
return 1;
}
-static int pss_put_dspword(pss_confdata * devc, unsigned short word)
+static int pss_put_dspword(struct pss_confdata *devc, unsigned short word)
{
int i, val;
@@ -291,7 +291,7 @@ static int pss_put_dspword(pss_confdata * devc, unsigned short word)
return 0;
}
-static int pss_get_dspword(pss_confdata * devc, unsigned short *word)
+static int pss_get_dspword(struct pss_confdata *devc, unsigned short *word)
{
int i, val;
@@ -307,7 +307,8 @@ static int pss_get_dspword(pss_confdata * devc, unsigned short *word)
return 0;
}
-static int pss_download_boot(pss_confdata * devc, unsigned char *block, int size, int flags)
+static int pss_download_boot(struct pss_confdata *devc, unsigned char *block,
+ int size, int flags)
{
int i, val, count;
unsigned long limit;
@@ -397,7 +398,7 @@ static int pss_download_boot(pss_confdata * devc, unsigned char *block, int size
}
/* Mixer */
-static void set_master_volume(pss_confdata *devc, int left, int right)
+static void set_master_volume(struct pss_confdata *devc, int left, int right)
{
static unsigned char log_scale[101] = {
0xdb, 0xe0, 0xe3, 0xe5, 0xe7, 0xe9, 0xea, 0xeb, 0xec, 0xed, 0xed, 0xee,
@@ -416,7 +417,7 @@ static void set_master_volume(pss_confdata *devc, int left, int right)
pss_write(devc, log_scale[right] | 0x0100);
}
-static void set_synth_volume(pss_confdata *devc, int volume)
+static void set_synth_volume(struct pss_confdata *devc, int volume)
{
int vol = ((0x8000*volume)/100L);
pss_write(devc, 0x0080);
@@ -425,21 +426,21 @@ static void set_synth_volume(pss_confdata *devc, int volume)
pss_write(devc, vol);
}
-static void set_bass(pss_confdata *devc, int level)
+static void set_bass(struct pss_confdata *devc, int level)
{
int vol = (int)(((0xfd - 0xf0) * level)/100L) + 0xf0;
pss_write(devc, 0x0010);
pss_write(devc, vol | 0x0200);
};
-static void set_treble(pss_confdata *devc, int level)
+static void set_treble(struct pss_confdata *devc, int level)
{
int vol = (((0xfd - 0xf0) * level)/100L) + 0xf0;
pss_write(devc, 0x0010);
pss_write(devc, vol | 0x0300);
};
-static void pss_mixer_reset(pss_confdata *devc)
+static void pss_mixer_reset(struct pss_confdata *devc)
{
set_master_volume(devc, 33, 33);
set_bass(devc, 50);
@@ -499,7 +500,8 @@ static int ret_vol_stereo(int left, int right)
return ((right << 8) | left);
}
-static int call_ad_mixer(pss_confdata *devc,unsigned int cmd, void __user *arg)
+static int call_ad_mixer(struct pss_confdata *devc, unsigned int cmd,
+ void __user *arg)
{
if (devc->ad_mixer_dev != NO_WSS_MIXER)
return mixer_devs[devc->ad_mixer_dev]->ioctl(devc->ad_mixer_dev, cmd, arg);
@@ -509,7 +511,7 @@ static int call_ad_mixer(pss_confdata *devc,unsigned int cmd, void __user *arg)
static int pss_mixer_ioctl (int dev, unsigned int cmd, void __user *arg)
{
- pss_confdata *devc = mixer_devs[dev]->devc;
+ struct pss_confdata *devc = mixer_devs[dev]->devc;
int cmdf = cmd & 0xff;
if ((cmdf != SOUND_MIXER_VOLUME) && (cmdf != SOUND_MIXER_BASS) &&
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 3a3a3a71088b..50dd0086cfb1 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -858,8 +858,8 @@ config SND_VIRTUOSO
select SND_JACK if INPUT=y || INPUT=SND
help
Say Y here to include support for sound cards based on the
- Asus AV66/AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, DS,
- Essence ST (Deluxe), and Essence STX.
+ Asus AV66/AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, DS, DSX,
+ Essence ST (Deluxe), and Essence STX (II).
Support for the HDAV1.3 (Deluxe) and HDAV1.3 Slim is experimental;
for the Xense, missing.
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 9f10c9e0df5e..631aaa4046ad 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -1754,9 +1754,6 @@ static struct snd_kcontrol_new snd_echo_vumeters_switch = {
static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- struct echoaudio *chip;
-
- chip = snd_kcontrol_chip(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 96;
uinfo->value.integer.min = ECHOGAIN_MINOUT;
@@ -1798,9 +1795,6 @@ static struct snd_kcontrol_new snd_echo_vumeters = {
static int snd_echo_channels_info_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- struct echoaudio *chip;
-
- chip = snd_kcontrol_chip(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 6;
uinfo->value.integer.min = 0;
diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c
new file mode 100644
index 000000000000..9c22f95838ef
--- /dev/null
+++ b/sound/pci/hda/dell_wmi_helper.c
@@ -0,0 +1,76 @@
+/* Helper functions for Dell Mic Mute LED control;
+ * to be included from codec driver
+ */
+
+#if IS_ENABLED(CONFIG_LEDS_DELL_NETBOOKS)
+#include <linux/dell-led.h>
+
+static int dell_led_value;
+static int (*dell_led_set_func)(int, int);
+static void (*dell_old_cap_hook)(struct hda_codec *,
+ struct snd_kcontrol *,
+ struct snd_ctl_elem_value *);
+
+static void update_dell_wmi_micmute_led(struct hda_codec *codec,
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ if (dell_old_cap_hook)
+ dell_old_cap_hook(codec, kcontrol, ucontrol);
+
+ if (!ucontrol || !dell_led_set_func)
+ return;
+ if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) {
+ /* TODO: How do I verify if it's a mono or stereo here? */
+ int val = (ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]) ? 0 : 1;
+ if (val == dell_led_value)
+ return;
+ dell_led_value = val;
+ if (dell_led_set_func)
+ dell_led_set_func(DELL_LED_MICMUTE, dell_led_value);
+ }
+}
+
+
+static void alc_fixup_dell_wmi(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ bool removefunc = false;
+
+ if (action == HDA_FIXUP_ACT_PROBE) {
+ if (!dell_led_set_func)
+ dell_led_set_func = symbol_request(dell_app_wmi_led_set);
+ if (!dell_led_set_func) {
+ codec_warn(codec, "Failed to find dell wmi symbol dell_app_wmi_led_set\n");
+ return;
+ }
+
+ removefunc = true;
+ if (dell_led_set_func(DELL_LED_MICMUTE, false) >= 0) {
+ dell_led_value = 0;
+ if (spec->gen.num_adc_nids > 1)
+ codec_dbg(codec, "Skipping micmute LED control due to several ADCs");
+ else {
+ dell_old_cap_hook = spec->gen.cap_sync_hook;
+ spec->gen.cap_sync_hook = update_dell_wmi_micmute_led;
+ removefunc = false;
+ }
+ }
+
+ }
+
+ if (dell_led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) {
+ symbol_put(dell_app_wmi_led_set);
+ dell_led_set_func = NULL;
+ dell_old_cap_hook = NULL;
+ }
+}
+
+#else /* CONFIG_LEDS_DELL_NETBOOKS */
+static void alc_fixup_dell_wmi(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+}
+
+#endif /* CONFIG_LEDS_DELL_NETBOOKS */
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index dabe41975a9d..51dea49aadd4 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -17,8 +17,6 @@
#include "hda_local.h"
#include "hda_auto_parser.h"
-#define SFX "hda_codec: "
-
/*
* Helper for automatic pin configuration
*/
@@ -856,7 +854,7 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec,
{
const struct snd_hda_pin_quirk *pq;
- if (codec->fixup_forced)
+ if (codec->fixup_id != HDA_FIXUP_ID_NOT_SET)
return;
for (pq = pin_quirk; pq->subvendor; pq++) {
@@ -882,14 +880,17 @@ void snd_hda_pick_fixup(struct hda_codec *codec,
const struct hda_fixup *fixlist)
{
const struct snd_pci_quirk *q;
- int id = -1;
+ int id = HDA_FIXUP_ID_NOT_SET;
const char *name = NULL;
+ if (codec->fixup_id != HDA_FIXUP_ID_NOT_SET)
+ return;
+
/* when model=nofixup is given, don't pick up any fixups */
if (codec->modelname && !strcmp(codec->modelname, "nofixup")) {
codec->fixup_list = NULL;
- codec->fixup_id = -1;
- codec->fixup_forced = 1;
+ codec->fixup_name = NULL;
+ codec->fixup_id = HDA_FIXUP_ID_NO_FIXUP;
return;
}
@@ -899,13 +900,12 @@ void snd_hda_pick_fixup(struct hda_codec *codec,
codec->fixup_id = models->id;
codec->fixup_name = models->name;
codec->fixup_list = fixlist;
- codec->fixup_forced = 1;
return;
}
models++;
}
}
- if (id < 0 && quirk) {
+ if (quirk) {
q = snd_pci_quirk_lookup(codec->bus->pci, quirk);
if (q) {
id = q->value;
@@ -929,7 +929,6 @@ void snd_hda_pick_fixup(struct hda_codec *codec,
}
}
- codec->fixup_forced = 0;
codec->fixup_id = id;
if (id >= 0) {
codec->fixup_list = fixlist;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 4c20277a6835..ec6a7d0d1886 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1476,6 +1476,7 @@ int snd_hda_codec_new(struct hda_bus *bus,
INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work);
codec->depop_delay = -1;
+ codec->fixup_id = HDA_FIXUP_ID_NOT_SET;
#ifdef CONFIG_PM
spin_lock_init(&codec->power_lock);
@@ -2727,7 +2728,7 @@ int snd_hda_codec_reset(struct hda_codec *codec)
return 0;
}
-typedef int (*map_slave_func_t)(void *, struct snd_kcontrol *);
+typedef int (*map_slave_func_t)(struct hda_codec *, void *, struct snd_kcontrol *);
/* apply the function to all matching slave ctls in the mixer list */
static int map_slaves(struct hda_codec *codec, const char * const *slaves,
@@ -2751,7 +2752,7 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves,
name = tmpname;
}
if (!strcmp(sctl->id.name, name)) {
- err = func(data, sctl);
+ err = func(codec, data, sctl);
if (err)
return err;
break;
@@ -2761,13 +2762,15 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves,
return 0;
}
-static int check_slave_present(void *data, struct snd_kcontrol *sctl)
+static int check_slave_present(struct hda_codec *codec,
+ void *data, struct snd_kcontrol *sctl)
{
return 1;
}
/* guess the value corresponding to 0dB */
-static int get_kctl_0dB_offset(struct snd_kcontrol *kctl, int *step_to_check)
+static int get_kctl_0dB_offset(struct hda_codec *codec,
+ struct snd_kcontrol *kctl, int *step_to_check)
{
int _tlv[4];
const int *tlv = NULL;
@@ -2788,7 +2791,7 @@ static int get_kctl_0dB_offset(struct snd_kcontrol *kctl, int *step_to_check)
if (!step)
return -1;
if (*step_to_check && *step_to_check != step) {
- snd_printk(KERN_ERR "hda_codec: Mismatching dB step for vmaster slave (%d!=%d)\n",
+ codec_err(codec, "Mismatching dB step for vmaster slave (%d!=%d)\n",
- *step_to_check, step);
return -1;
}
@@ -2813,20 +2816,28 @@ static int put_kctl_with_value(struct snd_kcontrol *kctl, int val)
}
/* initialize the slave volume with 0dB */
-static int init_slave_0dB(void *data, struct snd_kcontrol *slave)
+static int init_slave_0dB(struct hda_codec *codec,
+ void *data, struct snd_kcontrol *slave)
{
- int offset = get_kctl_0dB_offset(slave, data);
+ int offset = get_kctl_0dB_offset(codec, slave, data);
if (offset > 0)
put_kctl_with_value(slave, offset);
return 0;
}
/* unmute the slave */
-static int init_slave_unmute(void *data, struct snd_kcontrol *slave)
+static int init_slave_unmute(struct hda_codec *codec,
+ void *data, struct snd_kcontrol *slave)
{
return put_kctl_with_value(slave, 1);
}
+static int add_slave(struct hda_codec *codec,
+ void *data, struct snd_kcontrol *slave)
+{
+ return snd_ctl_add_slave(data, slave);
+}
+
/**
* snd_hda_add_vmaster - create a virtual master control and add slaves
* @codec: HD-audio codec
@@ -2869,8 +2880,7 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
if (err < 0)
return err;
- err = map_slaves(codec, slaves, suffix,
- (map_slave_func_t)snd_ctl_add_slave, kctl);
+ err = map_slaves(codec, slaves, suffix, add_slave, kctl);
if (err < 0)
return err;
@@ -4280,6 +4290,7 @@ static struct hda_rate_tbl rate_bits[] = {
/**
* snd_hda_calc_stream_format - calculate format bitset
+ * @codec: HD-audio codec
* @rate: the sample rate
* @channels: the number of channels
* @format: the PCM format (SNDRV_PCM_FORMAT_XXX)
@@ -4289,7 +4300,8 @@ static struct hda_rate_tbl rate_bits[] = {
*
* Return zero if invalid.
*/
-unsigned int snd_hda_calc_stream_format(unsigned int rate,
+unsigned int snd_hda_calc_stream_format(struct hda_codec *codec,
+ unsigned int rate,
unsigned int channels,
unsigned int format,
unsigned int maxbps,
@@ -4304,12 +4316,12 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
break;
}
if (!rate_bits[i].hz) {
- snd_printdd("invalid rate %d\n", rate);
+ codec_dbg(codec, "invalid rate %d\n", rate);
return 0;
}
if (channels == 0 || channels > 8) {
- snd_printdd("invalid channels %d\n", channels);
+ codec_dbg(codec, "invalid channels %d\n", channels);
return 0;
}
val |= channels - 1;
@@ -4332,7 +4344,7 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
val |= AC_FMT_BITS_20;
break;
default:
- snd_printdd("invalid format width %d\n",
+ codec_dbg(codec, "invalid format width %d\n",
snd_pcm_format_width(format));
return 0;
}
@@ -5670,12 +5682,13 @@ EXPORT_SYMBOL_GPL(_snd_hda_set_pin_ctl);
* suffix is appended to the label. This label index number is stored
* to type_idx when non-NULL pointer is given.
*/
-int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label,
+int snd_hda_add_imux_item(struct hda_codec *codec,
+ struct hda_input_mux *imux, const char *label,
int index, int *type_idx)
{
int i, label_idx = 0;
if (imux->num_items >= HDA_MAX_NUM_INPUTS) {
- snd_printd(KERN_ERR "hda_codec: Too many imux items!\n");
+ codec_err(codec, "hda_codec: Too many imux items!\n");
return -EINVAL;
}
for (i = 0; i < imux->num_items; i++) {
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 5825aa17d8e3..bbc5a1392c75 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -402,7 +402,6 @@ struct hda_codec {
/* fix-up list */
int fixup_id;
- unsigned int fixup_forced:1; /* fixup explicitly set by user */
const struct hda_fixup *fixup_list;
const char *fixup_name;
@@ -538,7 +537,8 @@ void __snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid,
int do_now);
#define snd_hda_codec_cleanup_stream(codec, nid) \
__snd_hda_codec_cleanup_stream(codec, nid, 0)
-unsigned int snd_hda_calc_stream_format(unsigned int rate,
+unsigned int snd_hda_calc_stream_format(struct hda_codec *codec,
+ unsigned int rate,
unsigned int channels,
unsigned int format,
unsigned int maxbps,
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 6df04d91c93c..8337645aa7a5 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -27,6 +27,7 @@
#include <linux/module.h>
#include <linux/pm_runtime.h>
#include <linux/slab.h>
+#include <linux/reboot.h>
#include <sound/core.h>
#include <sound/initval.h>
#include "hda_priv.h"
@@ -152,11 +153,11 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
upper_32_bits(azx_dev->bdl.addr));
/* enable the position buffer */
- if (chip->position_fix[0] != POS_FIX_LPIB ||
- chip->position_fix[1] != POS_FIX_LPIB) {
- if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE))
+ if (chip->get_position[0] != azx_get_pos_lpib ||
+ chip->get_position[1] != azx_get_pos_lpib) {
+ if (!(azx_readl(chip, DPLBASE) & AZX_DPLBASE_ENABLE))
azx_writel(chip, DPLBASE,
- (u32)chip->posbuf.addr | ICH6_DPLBASE_ENABLE);
+ (u32)chip->posbuf.addr | AZX_DPLBASE_ENABLE);
}
/* set the interrupt enable bits in the descriptor control register */
@@ -482,7 +483,8 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
}
azx_stream_reset(chip, azx_dev);
- format_val = snd_hda_calc_stream_format(runtime->rate,
+ format_val = snd_hda_calc_stream_format(apcm->codec,
+ runtime->rate,
runtime->channels,
runtime->format,
hinfo->maxbps,
@@ -673,125 +675,40 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
return 0;
}
-/* get the current DMA position with correction on VIA chips */
-static unsigned int azx_via_get_position(struct azx *chip,
- struct azx_dev *azx_dev)
+unsigned int azx_get_pos_lpib(struct azx *chip, struct azx_dev *azx_dev)
{
- unsigned int link_pos, mini_pos, bound_pos;
- unsigned int mod_link_pos, mod_dma_pos, mod_mini_pos;
- unsigned int fifo_size;
-
- link_pos = azx_sd_readl(chip, azx_dev, SD_LPIB);
- if (azx_dev->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- /* Playback, no problem using link position */
- return link_pos;
- }
-
- /* Capture */
- /* For new chipset,
- * use mod to get the DMA position just like old chipset
- */
- mod_dma_pos = le32_to_cpu(*azx_dev->posbuf);
- mod_dma_pos %= azx_dev->period_bytes;
-
- /* azx_dev->fifo_size can't get FIFO size of in stream.
- * Get from base address + offset.
- */
- fifo_size = readw(chip->remap_addr + VIA_IN_STREAM0_FIFO_SIZE_OFFSET);
-
- if (azx_dev->insufficient) {
- /* Link position never gather than FIFO size */
- if (link_pos <= fifo_size)
- return 0;
-
- azx_dev->insufficient = 0;
- }
-
- if (link_pos <= fifo_size)
- mini_pos = azx_dev->bufsize + link_pos - fifo_size;
- else
- mini_pos = link_pos - fifo_size;
-
- /* Find nearest previous boudary */
- mod_mini_pos = mini_pos % azx_dev->period_bytes;
- mod_link_pos = link_pos % azx_dev->period_bytes;
- if (mod_link_pos >= fifo_size)
- bound_pos = link_pos - mod_link_pos;
- else if (mod_dma_pos >= mod_mini_pos)
- bound_pos = mini_pos - mod_mini_pos;
- else {
- bound_pos = mini_pos - mod_mini_pos + azx_dev->period_bytes;
- if (bound_pos >= azx_dev->bufsize)
- bound_pos = 0;
- }
+ return azx_sd_readl(chip, azx_dev, SD_LPIB);
+}
+EXPORT_SYMBOL_GPL(azx_get_pos_lpib);
- /* Calculate real DMA position we want */
- return bound_pos + mod_dma_pos;
+unsigned int azx_get_pos_posbuf(struct azx *chip, struct azx_dev *azx_dev)
+{
+ return le32_to_cpu(*azx_dev->posbuf);
}
+EXPORT_SYMBOL_GPL(azx_get_pos_posbuf);
unsigned int azx_get_position(struct azx *chip,
- struct azx_dev *azx_dev,
- bool with_check)
+ struct azx_dev *azx_dev)
{
struct snd_pcm_substream *substream = azx_dev->substream;
- struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
unsigned int pos;
int stream = substream->stream;
- struct hda_pcm_stream *hinfo = apcm->hinfo[stream];
int delay = 0;
- switch (chip->position_fix[stream]) {
- case POS_FIX_LPIB:
- /* read LPIB */
- pos = azx_sd_readl(chip, azx_dev, SD_LPIB);
- break;
- case POS_FIX_VIACOMBO:
- pos = azx_via_get_position(chip, azx_dev);
- break;
- default:
- /* use the position buffer */
- pos = le32_to_cpu(*azx_dev->posbuf);
- if (with_check && chip->position_fix[stream] == POS_FIX_AUTO) {
- if (!pos || pos == (u32)-1) {
- dev_info(chip->card->dev,
- "Invalid position buffer, using LPIB read method instead.\n");
- chip->position_fix[stream] = POS_FIX_LPIB;
- pos = azx_sd_readl(chip, azx_dev, SD_LPIB);
- } else
- chip->position_fix[stream] = POS_FIX_POSBUF;
- }
- break;
- }
+ if (chip->get_position[stream])
+ pos = chip->get_position[stream](chip, azx_dev);
+ else /* use the position buffer as default */
+ pos = azx_get_pos_posbuf(chip, azx_dev);
if (pos >= azx_dev->bufsize)
pos = 0;
- /* calculate runtime delay from LPIB */
- if (substream->runtime &&
- chip->position_fix[stream] == POS_FIX_POSBUF &&
- (chip->driver_caps & AZX_DCAPS_COUNT_LPIB_DELAY)) {
- unsigned int lpib_pos = azx_sd_readl(chip, azx_dev, SD_LPIB);
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
- delay = pos - lpib_pos;
- else
- delay = lpib_pos - pos;
- if (delay < 0) {
- if (delay >= azx_dev->delay_negative_threshold)
- delay = 0;
- else
- delay += azx_dev->bufsize;
- }
- if (delay >= azx_dev->period_bytes) {
- dev_info(chip->card->dev,
- "Unstable LPIB (%d >= %d); disabling LPIB delay counting\n",
- delay, azx_dev->period_bytes);
- delay = 0;
- chip->driver_caps &= ~AZX_DCAPS_COUNT_LPIB_DELAY;
- }
- delay = bytes_to_frames(substream->runtime, delay);
- }
-
if (substream->runtime) {
+ struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+ struct hda_pcm_stream *hinfo = apcm->hinfo[stream];
+
+ if (chip->get_delay[stream])
+ delay += chip->get_delay[stream](chip, azx_dev, pos);
if (hinfo->ops.get_delay)
delay += hinfo->ops.get_delay(hinfo, apcm->codec,
substream);
@@ -809,7 +726,7 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
struct azx *chip = apcm->chip;
struct azx_dev *azx_dev = get_azx_dev(substream);
return bytes_to_frames(substream->runtime,
- azx_get_position(chip, azx_dev, false));
+ azx_get_position(chip, azx_dev));
}
static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream,
@@ -1059,10 +976,10 @@ static void azx_init_cmd_io(struct azx *chip)
azx_writew(chip, CORBWP, 0);
/* reset the corb hw read pointer */
- azx_writew(chip, CORBRP, ICH6_CORBRP_RST);
+ azx_writew(chip, CORBRP, AZX_CORBRP_RST);
if (!(chip->driver_caps & AZX_DCAPS_CORBRP_SELF_CLEAR)) {
for (timeout = 1000; timeout > 0; timeout--) {
- if ((azx_readw(chip, CORBRP) & ICH6_CORBRP_RST) == ICH6_CORBRP_RST)
+ if ((azx_readw(chip, CORBRP) & AZX_CORBRP_RST) == AZX_CORBRP_RST)
break;
udelay(1);
}
@@ -1082,7 +999,7 @@ static void azx_init_cmd_io(struct azx *chip)
}
/* enable corb dma */
- azx_writeb(chip, CORBCTL, ICH6_CORBCTL_RUN);
+ azx_writeb(chip, CORBCTL, AZX_CORBCTL_RUN);
/* RIRB set up */
chip->rirb.addr = chip->rb.addr + 2048;
@@ -1095,14 +1012,14 @@ static void azx_init_cmd_io(struct azx *chip)
/* set the rirb size to 256 entries (ULI requires explicitly) */
azx_writeb(chip, RIRBSIZE, 0x02);
/* reset the rirb hw write pointer */
- azx_writew(chip, RIRBWP, ICH6_RIRBWP_RST);
+ azx_writew(chip, RIRBWP, AZX_RIRBWP_RST);
/* set N=1, get RIRB response interrupt for new entry */
if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND)
azx_writew(chip, RINTCNT, 0xc0);
else
azx_writew(chip, RINTCNT, 1);
/* enable rirb dma and response irq */
- azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN | ICH6_RBCTL_IRQ_EN);
+ azx_writeb(chip, RIRBCTL, AZX_RBCTL_DMA_EN | AZX_RBCTL_IRQ_EN);
spin_unlock_irq(&chip->reg_lock);
}
EXPORT_SYMBOL_GPL(azx_init_cmd_io);
@@ -1146,7 +1063,7 @@ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val)
return -EIO;
}
wp++;
- wp %= ICH6_MAX_CORB_ENTRIES;
+ wp %= AZX_MAX_CORB_ENTRIES;
rp = azx_readw(chip, CORBRP);
if (wp == rp) {
@@ -1164,7 +1081,7 @@ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val)
return 0;
}
-#define ICH6_RIRB_EX_UNSOL_EV (1<<4)
+#define AZX_RIRB_EX_UNSOL_EV (1<<4)
/* retrieve RIRB entry - called from interrupt handler */
static void azx_update_rirb(struct azx *chip)
@@ -1185,7 +1102,7 @@ static void azx_update_rirb(struct azx *chip)
while (chip->rirb.rp != wp) {
chip->rirb.rp++;
- chip->rirb.rp %= ICH6_MAX_RIRB_ENTRIES;
+ chip->rirb.rp %= AZX_MAX_RIRB_ENTRIES;
rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */
res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]);
@@ -1196,8 +1113,7 @@ static void azx_update_rirb(struct azx *chip)
res, res_ex,
chip->rirb.rp, wp);
snd_BUG();
- }
- else if (res_ex & ICH6_RIRB_EX_UNSOL_EV)
+ } else if (res_ex & AZX_RIRB_EX_UNSOL_EV)
snd_hda_queue_unsol_event(chip->bus, res, res_ex);
else if (chip->rirb.cmds[addr]) {
chip->rirb.res[addr] = res;
@@ -1305,7 +1221,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
/* release CORB/RIRB */
azx_free_cmd_io(chip);
/* disable unsolicited responses */
- azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_UNSOL);
+ azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~AZX_GCTL_UNSOL);
return -1;
}
@@ -1326,7 +1242,7 @@ static int azx_single_wait_for_response(struct azx *chip, unsigned int addr)
while (timeout--) {
/* check IRV busy bit */
- if (azx_readw(chip, IRS) & ICH6_IRS_VALID) {
+ if (azx_readw(chip, IRS) & AZX_IRS_VALID) {
/* reuse rirb.res as the response return value */
chip->rirb.res[addr] = azx_readl(chip, IR);
return 0;
@@ -1350,13 +1266,13 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val)
bus->rirb_error = 0;
while (timeout--) {
/* check ICB busy bit */
- if (!((azx_readw(chip, IRS) & ICH6_IRS_BUSY))) {
+ if (!((azx_readw(chip, IRS) & AZX_IRS_BUSY))) {
/* Clear IRV valid bit */
azx_writew(chip, IRS, azx_readw(chip, IRS) |
- ICH6_IRS_VALID);
+ AZX_IRS_VALID);
azx_writel(chip, IC, val);
azx_writew(chip, IRS, azx_readw(chip, IRS) |
- ICH6_IRS_BUSY);
+ AZX_IRS_BUSY);
return azx_single_wait_for_response(chip, addr);
}
udelay(1);
@@ -1585,10 +1501,10 @@ void azx_enter_link_reset(struct azx *chip)
unsigned long timeout;
/* reset controller */
- azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET);
+ azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~AZX_GCTL_RESET);
timeout = jiffies + msecs_to_jiffies(100);
- while ((azx_readb(chip, GCTL) & ICH6_GCTL_RESET) &&
+ while ((azx_readb(chip, GCTL) & AZX_GCTL_RESET) &&
time_before(jiffies, timeout))
usleep_range(500, 1000);
}
@@ -1599,7 +1515,7 @@ static void azx_exit_link_reset(struct azx *chip)
{
unsigned long timeout;
- azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET);
+ azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | AZX_GCTL_RESET);
timeout = jiffies + msecs_to_jiffies(100);
while (!azx_readb(chip, GCTL) &&
@@ -1640,7 +1556,7 @@ static int azx_reset(struct azx *chip, bool full_reset)
/* Accept unsolicited responses */
if (!chip->single_cmd)
azx_writel(chip, GCTL, azx_readl(chip, GCTL) |
- ICH6_GCTL_UNSOL);
+ AZX_GCTL_UNSOL);
/* detect codecs */
if (!chip->codec_mask) {
@@ -1657,7 +1573,7 @@ static void azx_int_enable(struct azx *chip)
{
/* enable controller CIE and GIE */
azx_writel(chip, INTCTL, azx_readl(chip, INTCTL) |
- ICH6_INT_CTRL_EN | ICH6_INT_GLOBAL_EN);
+ AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN);
}
/* disable interrupts */
@@ -1678,7 +1594,7 @@ static void azx_int_disable(struct azx *chip)
/* disable controller CIE and GIE */
azx_writel(chip, INTCTL, azx_readl(chip, INTCTL) &
- ~(ICH6_INT_CTRL_EN | ICH6_INT_GLOBAL_EN));
+ ~(AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN));
}
/* clear interrupts */
@@ -1699,7 +1615,7 @@ static void azx_int_clear(struct azx *chip)
azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
/* clear int status */
- azx_writel(chip, INTSTS, ICH6_INT_CTRL_EN | ICH6_INT_ALL_STREAM);
+ azx_writel(chip, INTSTS, AZX_INT_CTRL_EN | AZX_INT_ALL_STREAM);
}
/*
@@ -2031,5 +1947,30 @@ int azx_init_stream(struct azx *chip)
}
EXPORT_SYMBOL_GPL(azx_init_stream);
+/*
+ * reboot notifier for hang-up problem at power-down
+ */
+static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf)
+{
+ struct azx *chip = container_of(nb, struct azx, reboot_notifier);
+ snd_hda_bus_reboot_notify(chip->bus);
+ azx_stop_chip(chip);
+ return NOTIFY_OK;
+}
+
+void azx_notifier_register(struct azx *chip)
+{
+ chip->reboot_notifier.notifier_call = azx_halt;
+ register_reboot_notifier(&chip->reboot_notifier);
+}
+EXPORT_SYMBOL_GPL(azx_notifier_register);
+
+void azx_notifier_unregister(struct azx *chip)
+{
+ if (chip->reboot_notifier.notifier_call)
+ unregister_reboot_notifier(&chip->reboot_notifier);
+}
+EXPORT_SYMBOL_GPL(azx_notifier_unregister);
+
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Common HDA driver funcitons");
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index baf0e77330af..c90d10fd4d8f 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -25,9 +25,9 @@ static inline struct azx_dev *get_azx_dev(struct snd_pcm_substream *substream)
{
return substream->runtime->private_data;
}
-unsigned int azx_get_position(struct azx *chip,
- struct azx_dev *azx_dev,
- bool with_check);
+unsigned int azx_get_position(struct azx *chip, struct azx_dev *azx_dev);
+unsigned int azx_get_pos_lpib(struct azx *chip, struct azx_dev *azx_dev);
+unsigned int azx_get_pos_posbuf(struct azx *chip, struct azx_dev *azx_dev);
/* Stream control. */
void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev);
@@ -50,4 +50,7 @@ int azx_codec_configure(struct azx *chip);
int azx_mixer_create(struct azx *chip);
int azx_init_stream(struct azx *chip);
+void azx_notifier_register(struct azx *chip);
+void azx_notifier_unregister(struct azx *chip);
+
#endif /* __SOUND_HDA_CONTROLLER_H */
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 46690a7f48f6..e1cd34d9011d 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -167,7 +167,8 @@ static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid,
(buf[byte] >> (lowbit)) & ((1 << (bits)) - 1); \
})
-static void hdmi_update_short_audio_desc(struct cea_sad *a,
+static void hdmi_update_short_audio_desc(struct hda_codec *codec,
+ struct cea_sad *a,
const unsigned char *buf)
{
int i;
@@ -188,8 +189,7 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a,
a->format = GRAB_BITS(buf, 0, 3, 4);
switch (a->format) {
case AUDIO_CODING_TYPE_REF_STREAM_HEADER:
- snd_printd(KERN_INFO
- "HDMI: audio coding type 0 not expected\n");
+ codec_info(codec, "HDMI: audio coding type 0 not expected\n");
break;
case AUDIO_CODING_TYPE_LPCM:
@@ -233,9 +233,9 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a,
a->format = GRAB_BITS(buf, 2, 3, 5);
if (a->format == AUDIO_CODING_XTYPE_HE_REF_CT ||
a->format >= AUDIO_CODING_XTYPE_FIRST_RESERVED) {
- snd_printd(KERN_INFO
- "HDMI: audio coding xtype %d not expected\n",
- a->format);
+ codec_info(codec,
+ "HDMI: audio coding xtype %d not expected\n",
+ a->format);
a->format = 0;
} else
a->format += AUDIO_CODING_TYPE_HE_AAC -
@@ -247,7 +247,7 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a,
/*
* Be careful, ELD buf could be totally rubbish!
*/
-int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e,
+int snd_hdmi_parse_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e,
const unsigned char *buf, int size)
{
int mnl;
@@ -256,8 +256,7 @@ int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e,
e->eld_ver = GRAB_BITS(buf, 0, 3, 5);
if (e->eld_ver != ELD_VER_CEA_861D &&
e->eld_ver != ELD_VER_PARTIAL) {
- snd_printd(KERN_INFO "HDMI: Unknown ELD version %d\n",
- e->eld_ver);
+ codec_info(codec, "HDMI: Unknown ELD version %d\n", e->eld_ver);
goto out_fail;
}
@@ -280,20 +279,20 @@ int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e,
e->product_id = get_unaligned_le16(buf + 18);
if (mnl > ELD_MAX_MNL) {
- snd_printd(KERN_INFO "HDMI: MNL is reserved value %d\n", mnl);
+ codec_info(codec, "HDMI: MNL is reserved value %d\n", mnl);
goto out_fail;
} else if (ELD_FIXED_BYTES + mnl > size) {
- snd_printd(KERN_INFO "HDMI: out of range MNL %d\n", mnl);
+ codec_info(codec, "HDMI: out of range MNL %d\n", mnl);
goto out_fail;
} else
strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl + 1);
for (i = 0; i < e->sad_count; i++) {
if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) {
- snd_printd(KERN_INFO "HDMI: out of range SAD %d\n", i);
+ codec_info(codec, "HDMI: out of range SAD %d\n", i);
goto out_fail;
}
- hdmi_update_short_audio_desc(e->sad + i,
+ hdmi_update_short_audio_desc(codec, e->sad + i,
buf + ELD_FIXED_BYTES + mnl + 3 * i);
}
@@ -394,7 +393,8 @@ static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen)
#define SND_PRINT_RATES_ADVISED_BUFSIZE 80
-static void hdmi_show_short_audio_desc(struct cea_sad *a)
+static void hdmi_show_short_audio_desc(struct hda_codec *codec,
+ struct cea_sad *a)
{
char buf[SND_PRINT_RATES_ADVISED_BUFSIZE];
char buf2[8 + SND_PRINT_BITS_ADVISED_BUFSIZE] = ", bits =";
@@ -412,12 +412,10 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a)
else
buf2[0] = '\0';
- _snd_printd(SND_PR_VERBOSE, "HDMI: supports coding type %s:"
- " channels = %d, rates =%s%s\n",
- cea_audio_coding_type_names[a->format],
- a->channels,
- buf,
- buf2);
+ codec_dbg(codec,
+ "HDMI: supports coding type %s: channels = %d, rates =%s%s\n",
+ cea_audio_coding_type_names[a->format],
+ a->channels, buf, buf2);
}
void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen)
@@ -432,22 +430,22 @@ void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen)
buf[j] = '\0'; /* necessary when j == 0 */
}
-void snd_hdmi_show_eld(struct parsed_hdmi_eld *e)
+void snd_hdmi_show_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e)
{
int i;
- _snd_printd(SND_PR_VERBOSE, "HDMI: detected monitor %s at connection type %s\n",
+ codec_dbg(codec, "HDMI: detected monitor %s at connection type %s\n",
e->monitor_name,
eld_connection_type_names[e->conn_type]);
if (e->spk_alloc) {
char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE];
snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf));
- _snd_printd(SND_PR_VERBOSE, "HDMI: available speakers:%s\n", buf);
+ codec_dbg(codec, "HDMI: available speakers:%s\n", buf);
}
for (i = 0; i < e->sad_count; i++)
- hdmi_show_short_audio_desc(e->sad + i);
+ hdmi_show_short_audio_desc(codec, e->sad + i);
}
#ifdef CONFIG_PROC_FS
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 589e47c5aeb3..b956449ddada 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -350,16 +350,16 @@ static void print_nid_path(struct hda_codec *codec,
const char *pfx, struct nid_path *path)
{
char buf[40];
+ char *pos = buf;
int i;
+ *pos = 0;
+ for (i = 0; i < path->depth; i++)
+ pos += scnprintf(pos, sizeof(buf) - (pos - buf), "%s%02x",
+ pos != buf ? ":" : "",
+ path->path[i]);
- buf[0] = 0;
- for (i = 0; i < path->depth; i++) {
- char tmp[4];
- sprintf(tmp, ":%02x", path->path[i]);
- strlcat(buf, tmp, sizeof(buf));
- }
- codec_dbg(codec, "%s path: depth=%d %s\n", pfx, path->depth, buf);
+ codec_dbg(codec, "%s path: depth=%d '%s'\n", pfx, path->depth, buf);
}
/* called recursively */
@@ -1700,9 +1700,11 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
#define DEBUG_BADNESS
#ifdef DEBUG_BADNESS
-#define debug_badness(fmt, args...) codec_dbg(codec, fmt, ##args)
+#define debug_badness(fmt, ...) \
+ codec_dbg(codec, fmt, ##__VA_ARGS__)
#else
-#define debug_badness(...)
+#define debug_badness(fmt, ...) \
+ do { if (0) codec_dbg(codec, fmt, ##__VA_ARGS__); } while (0)
#endif
#ifdef DEBUG_BADNESS
@@ -3054,7 +3056,7 @@ static int parse_capture_source(struct hda_codec *codec, hda_nid_t pin,
if (spec->hp_mic_pin == pin)
spec->hp_mic_mux_idx = imux->num_items;
spec->imux_pins[imux->num_items] = pin;
- snd_hda_add_imux_item(imux, label, cfg_idx, NULL);
+ snd_hda_add_imux_item(codec, imux, label, cfg_idx, NULL);
imux_added = true;
if (spec->dyn_adc_switch)
spec->dyn_adc_idx[imux_idx] = c;
diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c
index 8b4940ba33d6..d4d0375ac181 100644
--- a/sound/pci/hda/hda_i915.c
+++ b/sound/pci/hda/hda_i915.c
@@ -28,8 +28,8 @@
* Clock) to 24MHz BCLK: BCLK = CDCLK * M / N
* The values will be lost when the display power well is disabled.
*/
-#define ICH6_REG_EM4 0x100c
-#define ICH6_REG_EM5 0x1010
+#define AZX_REG_EM4 0x100c
+#define AZX_REG_EM5 0x1010
static int (*get_power)(void);
static int (*put_power)(void);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 83cd19017cf3..5db1948699d8 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -44,7 +44,6 @@
#include <linux/slab.h>
#include <linux/pci.h>
#include <linux/mutex.h>
-#include <linux/reboot.h>
#include <linux/io.h>
#include <linux/pm_runtime.h>
#include <linux/clocksource.h>
@@ -66,6 +65,52 @@
#include "hda_priv.h"
#include "hda_i915.h"
+/* position fix mode */
+enum {
+ POS_FIX_AUTO,
+ POS_FIX_LPIB,
+ POS_FIX_POSBUF,
+ POS_FIX_VIACOMBO,
+ POS_FIX_COMBO,
+};
+
+/* Defines for ATI HD Audio support in SB450 south bridge */
+#define ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR 0x42
+#define ATI_SB450_HDAUDIO_ENABLE_SNOOP 0x02
+
+/* Defines for Nvidia HDA support */
+#define NVIDIA_HDA_TRANSREG_ADDR 0x4e
+#define NVIDIA_HDA_ENABLE_COHBITS 0x0f
+#define NVIDIA_HDA_ISTRM_COH 0x4d
+#define NVIDIA_HDA_OSTRM_COH 0x4c
+#define NVIDIA_HDA_ENABLE_COHBIT 0x01
+
+/* Defines for Intel SCH HDA snoop control */
+#define INTEL_SCH_HDA_DEVC 0x78
+#define INTEL_SCH_HDA_DEVC_NOSNOOP (0x1<<11)
+
+/* Define IN stream 0 FIFO size offset in VIA controller */
+#define VIA_IN_STREAM0_FIFO_SIZE_OFFSET 0x90
+/* Define VIA HD Audio Device ID*/
+#define VIA_HDAC_DEVICE_ID 0x3288
+
+/* max number of SDs */
+/* ICH, ATI and VIA have 4 playback and 4 capture */
+#define ICH6_NUM_CAPTURE 4
+#define ICH6_NUM_PLAYBACK 4
+
+/* ULI has 6 playback and 5 capture */
+#define ULI_NUM_CAPTURE 5
+#define ULI_NUM_PLAYBACK 6
+
+/* ATI HDMI may have up to 8 playbacks and 0 capture */
+#define ATIHDMI_NUM_CAPTURE 0
+#define ATIHDMI_NUM_PLAYBACK 8
+
+/* TERA has 4 playback and 3 capture */
+#define TERA_NUM_CAPTURE 3
+#define TERA_NUM_PLAYBACK 4
+
static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
@@ -290,8 +335,28 @@ static char *driver_short_names[] = {
struct hda_intel {
struct azx chip;
-};
+ /* for pending irqs */
+ struct work_struct irq_pending_work;
+
+ /* sync probing */
+ struct completion probe_wait;
+ struct work_struct probe_work;
+
+ /* card list (for power_save trigger) */
+ struct list_head list;
+
+ /* extra flags */
+ unsigned int irq_pending_warned:1;
+
+ /* VGA-switcheroo setup */
+ unsigned int use_vga_switcheroo:1;
+ unsigned int vga_switcheroo_registered:1;
+ unsigned int init_failed:1; /* delayed init failed */
+
+ /* secondary power domain for hdmi audio under vga device */
+ struct dev_pm_domain hdmi_pm_domain;
+};
#ifdef CONFIG_X86
static void __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, bool on)
@@ -373,7 +438,7 @@ static void azx_init_pci(struct azx *chip)
*/
if (!(chip->driver_caps & AZX_DCAPS_NO_TCSEL)) {
dev_dbg(chip->card->dev, "Clearing TCSEL\n");
- update_pci_byte(chip->pci, ICH6_PCIREG_TCSEL, 0x07, 0);
+ update_pci_byte(chip->pci, AZX_PCIREG_TCSEL, 0x07, 0);
}
/* For ATI SB450/600/700/800/900 and AMD Hudson azalia HD audio,
@@ -421,11 +486,44 @@ static void azx_init_pci(struct azx *chip)
}
}
+/* calculate runtime delay from LPIB */
+static int azx_get_delay_from_lpib(struct azx *chip, struct azx_dev *azx_dev,
+ unsigned int pos)
+{
+ struct snd_pcm_substream *substream = azx_dev->substream;
+ int stream = substream->stream;
+ unsigned int lpib_pos = azx_get_pos_lpib(chip, azx_dev);
+ int delay;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ delay = pos - lpib_pos;
+ else
+ delay = lpib_pos - pos;
+ if (delay < 0) {
+ if (delay >= azx_dev->delay_negative_threshold)
+ delay = 0;
+ else
+ delay += azx_dev->bufsize;
+ }
+
+ if (delay >= azx_dev->period_bytes) {
+ dev_info(chip->card->dev,
+ "Unstable LPIB (%d >= %d); disabling LPIB delay counting\n",
+ delay, azx_dev->period_bytes);
+ delay = 0;
+ chip->driver_caps &= ~AZX_DCAPS_COUNT_LPIB_DELAY;
+ chip->get_delay[stream] = NULL;
+ }
+
+ return bytes_to_frames(substream->runtime, delay);
+}
+
static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev);
/* called from IRQ */
static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev)
{
+ struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
int ok;
ok = azx_position_ok(chip, azx_dev);
@@ -435,7 +533,7 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev)
} else if (ok == 0 && chip->bus && chip->bus->workq) {
/* bogus IRQ, process it later */
azx_dev->irq_pending = 1;
- queue_work(chip->bus->workq, &chip->irq_pending_work);
+ queue_work(chip->bus->workq, &hda->irq_pending_work);
}
return 0;
}
@@ -451,6 +549,8 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev)
*/
static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
{
+ struct snd_pcm_substream *substream = azx_dev->substream;
+ int stream = substream->stream;
u32 wallclk;
unsigned int pos;
@@ -458,7 +558,25 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
if (wallclk < (azx_dev->period_wallclk * 2) / 3)
return -1; /* bogus (too early) interrupt */
- pos = azx_get_position(chip, azx_dev, true);
+ if (chip->get_position[stream])
+ pos = chip->get_position[stream](chip, azx_dev);
+ else { /* use the position buffer as default */
+ pos = azx_get_pos_posbuf(chip, azx_dev);
+ if (!pos || pos == (u32)-1) {
+ dev_info(chip->card->dev,
+ "Invalid position buffer, using LPIB read method instead.\n");
+ chip->get_position[stream] = azx_get_pos_lpib;
+ pos = azx_get_pos_lpib(chip, azx_dev);
+ chip->get_delay[stream] = NULL;
+ } else {
+ chip->get_position[stream] = azx_get_pos_posbuf;
+ if (chip->driver_caps & AZX_DCAPS_COUNT_LPIB_DELAY)
+ chip->get_delay[stream] = azx_get_delay_from_lpib;
+ }
+ }
+
+ if (pos >= azx_dev->bufsize)
+ pos = 0;
if (WARN_ONCE(!azx_dev->period_bytes,
"hda-intel: zero azx_dev->period_bytes"))
@@ -476,14 +594,15 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
*/
static void azx_irq_pending_work(struct work_struct *work)
{
- struct azx *chip = container_of(work, struct azx, irq_pending_work);
+ struct hda_intel *hda = container_of(work, struct hda_intel, irq_pending_work);
+ struct azx *chip = &hda->chip;
int i, pending, ok;
- if (!chip->irq_pending_warned) {
+ if (!hda->irq_pending_warned) {
dev_info(chip->card->dev,
"IRQ timing workaround is activated for card #%d. Suggest a bigger bdl_pos_adj.\n",
chip->card->number);
- chip->irq_pending_warned = 1;
+ hda->irq_pending_warned = 1;
}
for (;;) {
@@ -541,27 +660,86 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect)
return 0;
}
+/* get the current DMA position with correction on VIA chips */
+static unsigned int azx_via_get_position(struct azx *chip,
+ struct azx_dev *azx_dev)
+{
+ unsigned int link_pos, mini_pos, bound_pos;
+ unsigned int mod_link_pos, mod_dma_pos, mod_mini_pos;
+ unsigned int fifo_size;
+
+ link_pos = azx_sd_readl(chip, azx_dev, SD_LPIB);
+ if (azx_dev->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* Playback, no problem using link position */
+ return link_pos;
+ }
+
+ /* Capture */
+ /* For new chipset,
+ * use mod to get the DMA position just like old chipset
+ */
+ mod_dma_pos = le32_to_cpu(*azx_dev->posbuf);
+ mod_dma_pos %= azx_dev->period_bytes;
+
+ /* azx_dev->fifo_size can't get FIFO size of in stream.
+ * Get from base address + offset.
+ */
+ fifo_size = readw(chip->remap_addr + VIA_IN_STREAM0_FIFO_SIZE_OFFSET);
+
+ if (azx_dev->insufficient) {
+ /* Link position never gather than FIFO size */
+ if (link_pos <= fifo_size)
+ return 0;
+
+ azx_dev->insufficient = 0;
+ }
+
+ if (link_pos <= fifo_size)
+ mini_pos = azx_dev->bufsize + link_pos - fifo_size;
+ else
+ mini_pos = link_pos - fifo_size;
+
+ /* Find nearest previous boudary */
+ mod_mini_pos = mini_pos % azx_dev->period_bytes;
+ mod_link_pos = link_pos % azx_dev->period_bytes;
+ if (mod_link_pos >= fifo_size)
+ bound_pos = link_pos - mod_link_pos;
+ else if (mod_dma_pos >= mod_mini_pos)
+ bound_pos = mini_pos - mod_mini_pos;
+ else {
+ bound_pos = mini_pos - mod_mini_pos + azx_dev->period_bytes;
+ if (bound_pos >= azx_dev->bufsize)
+ bound_pos = 0;
+ }
+
+ /* Calculate real DMA position we want */
+ return bound_pos + mod_dma_pos;
+}
+
#ifdef CONFIG_PM
static DEFINE_MUTEX(card_list_lock);
static LIST_HEAD(card_list);
static void azx_add_card_list(struct azx *chip)
{
+ struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
mutex_lock(&card_list_lock);
- list_add(&chip->list, &card_list);
+ list_add(&hda->list, &card_list);
mutex_unlock(&card_list_lock);
}
static void azx_del_card_list(struct azx *chip)
{
+ struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
mutex_lock(&card_list_lock);
- list_del_init(&chip->list);
+ list_del_init(&hda->list);
mutex_unlock(&card_list_lock);
}
/* trigger power-save check at writing parameter */
static int param_set_xint(const char *val, const struct kernel_param *kp)
{
+ struct hda_intel *hda;
struct azx *chip;
struct hda_codec *c;
int prev = power_save;
@@ -571,7 +749,8 @@ static int param_set_xint(const char *val, const struct kernel_param *kp)
return ret;
mutex_lock(&card_list_lock);
- list_for_each_entry(chip, &card_list, list) {
+ list_for_each_entry(hda, &card_list, list) {
+ chip = &hda->chip;
if (!chip->bus || chip->disabled)
continue;
list_for_each_entry(c, &chip->bus->codec_list, list)
@@ -593,10 +772,16 @@ static int azx_suspend(struct device *dev)
{
struct pci_dev *pci = to_pci_dev(dev);
struct snd_card *card = dev_get_drvdata(dev);
- struct azx *chip = card->private_data;
+ struct azx *chip;
+ struct hda_intel *hda;
struct azx_pcm *p;
- if (chip->disabled || chip->init_failed)
+ if (!card)
+ return 0;
+
+ chip = card->private_data;
+ hda = container_of(chip, struct hda_intel, chip);
+ if (chip->disabled || hda->init_failed)
return 0;
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -626,9 +811,15 @@ static int azx_resume(struct device *dev)
{
struct pci_dev *pci = to_pci_dev(dev);
struct snd_card *card = dev_get_drvdata(dev);
- struct azx *chip = card->private_data;
+ struct azx *chip;
+ struct hda_intel *hda;
- if (chip->disabled || chip->init_failed)
+ if (!card)
+ return 0;
+
+ chip = card->private_data;
+ hda = container_of(chip, struct hda_intel, chip);
+ if (chip->disabled || hda->init_failed)
return 0;
if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) {
@@ -663,9 +854,15 @@ static int azx_resume(struct device *dev)
static int azx_runtime_suspend(struct device *dev)
{
struct snd_card *card = dev_get_drvdata(dev);
- struct azx *chip = card->private_data;
+ struct azx *chip;
+ struct hda_intel *hda;
+
+ if (!card)
+ return 0;
- if (chip->disabled || chip->init_failed)
+ chip = card->private_data;
+ hda = container_of(chip, struct hda_intel, chip);
+ if (chip->disabled || hda->init_failed)
return 0;
if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME))
@@ -687,12 +884,18 @@ static int azx_runtime_suspend(struct device *dev)
static int azx_runtime_resume(struct device *dev)
{
struct snd_card *card = dev_get_drvdata(dev);
- struct azx *chip = card->private_data;
+ struct azx *chip;
+ struct hda_intel *hda;
struct hda_bus *bus;
struct hda_codec *codec;
int status;
- if (chip->disabled || chip->init_failed)
+ if (!card)
+ return 0;
+
+ chip = card->private_data;
+ hda = container_of(chip, struct hda_intel, chip);
+ if (chip->disabled || hda->init_failed)
return 0;
if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME))
@@ -727,9 +930,15 @@ static int azx_runtime_resume(struct device *dev)
static int azx_runtime_idle(struct device *dev)
{
struct snd_card *card = dev_get_drvdata(dev);
- struct azx *chip = card->private_data;
+ struct azx *chip;
+ struct hda_intel *hda;
+
+ if (!card)
+ return 0;
- if (chip->disabled || chip->init_failed)
+ chip = card->private_data;
+ hda = container_of(chip, struct hda_intel, chip);
+ if (chip->disabled || hda->init_failed)
return 0;
if (!power_save_controller ||
@@ -753,29 +962,6 @@ static const struct dev_pm_ops azx_pm = {
#endif /* CONFIG_PM */
-/*
- * reboot notifier for hang-up problem at power-down
- */
-static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf)
-{
- struct azx *chip = container_of(nb, struct azx, reboot_notifier);
- snd_hda_bus_reboot_notify(chip->bus);
- azx_stop_chip(chip);
- return NOTIFY_OK;
-}
-
-static void azx_notifier_register(struct azx *chip)
-{
- chip->reboot_notifier.notifier_call = azx_halt;
- register_reboot_notifier(&chip->reboot_notifier);
-}
-
-static void azx_notifier_unregister(struct azx *chip)
-{
- if (chip->reboot_notifier.notifier_call)
- unregister_reboot_notifier(&chip->reboot_notifier);
-}
-
static int azx_probe_continue(struct azx *chip);
#ifdef SUPPORT_VGA_SWITCHEROO
@@ -786,10 +972,11 @@ static void azx_vs_set_state(struct pci_dev *pci,
{
struct snd_card *card = pci_get_drvdata(pci);
struct azx *chip = card->private_data;
+ struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
bool disabled;
- wait_for_completion(&chip->probe_wait);
- if (chip->init_failed)
+ wait_for_completion(&hda->probe_wait);
+ if (hda->init_failed)
return;
disabled = (state == VGA_SWITCHEROO_OFF);
@@ -803,7 +990,7 @@ static void azx_vs_set_state(struct pci_dev *pci,
"Start delayed initialization\n");
if (azx_probe_continue(chip) < 0) {
dev_err(chip->card->dev, "initialization error\n");
- chip->init_failed = true;
+ hda->init_failed = true;
}
}
} else {
@@ -833,9 +1020,10 @@ static bool azx_vs_can_switch(struct pci_dev *pci)
{
struct snd_card *card = pci_get_drvdata(pci);
struct azx *chip = card->private_data;
+ struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
- wait_for_completion(&chip->probe_wait);
- if (chip->init_failed)
+ wait_for_completion(&hda->probe_wait);
+ if (hda->init_failed)
return false;
if (chip->disabled || !chip->bus)
return true;
@@ -847,11 +1035,12 @@ static bool azx_vs_can_switch(struct pci_dev *pci)
static void init_vga_switcheroo(struct azx *chip)
{
+ struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
struct pci_dev *p = get_bound_vga(chip->pci);
if (p) {
dev_info(chip->card->dev,
"Handle VGA-switcheroo audio client\n");
- chip->use_vga_switcheroo = 1;
+ hda->use_vga_switcheroo = 1;
pci_dev_put(p);
}
}
@@ -863,9 +1052,10 @@ static const struct vga_switcheroo_client_ops azx_vs_ops = {
static int register_vga_switcheroo(struct azx *chip)
{
+ struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
int err;
- if (!chip->use_vga_switcheroo)
+ if (!hda->use_vga_switcheroo)
return 0;
/* FIXME: currently only handling DIS controller
* is there any machine with two switchable HDMI audio controllers?
@@ -875,11 +1065,11 @@ static int register_vga_switcheroo(struct azx *chip)
chip->bus != NULL);
if (err < 0)
return err;
- chip->vga_switcheroo_registered = 1;
+ hda->vga_switcheroo_registered = 1;
/* register as an optimus hdmi audio power domain */
vga_switcheroo_init_domain_pm_optimus_hdmi_audio(chip->card->dev,
- &chip->hdmi_pm_domain);
+ &hda->hdmi_pm_domain);
return 0;
}
#else
@@ -895,7 +1085,6 @@ static int azx_free(struct azx *chip)
{
struct pci_dev *pci = chip->pci;
struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
-
int i;
if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME)
@@ -906,13 +1095,13 @@ static int azx_free(struct azx *chip)
azx_notifier_unregister(chip);
- chip->init_failed = 1; /* to be sure */
- complete_all(&chip->probe_wait);
+ hda->init_failed = 1; /* to be sure */
+ complete_all(&hda->probe_wait);
- if (use_vga_switcheroo(chip)) {
+ if (use_vga_switcheroo(hda)) {
if (chip->disabled && chip->bus)
snd_hda_unlock_devices(chip->bus);
- if (chip->vga_switcheroo_registered)
+ if (hda->vga_switcheroo_registered)
vga_switcheroo_unregister_client(chip->pci);
}
@@ -1048,6 +1237,30 @@ static int check_position_fix(struct azx *chip, int fix)
return POS_FIX_AUTO;
}
+static void assign_position_fix(struct azx *chip, int fix)
+{
+ static azx_get_pos_callback_t callbacks[] = {
+ [POS_FIX_AUTO] = NULL,
+ [POS_FIX_LPIB] = azx_get_pos_lpib,
+ [POS_FIX_POSBUF] = azx_get_pos_posbuf,
+ [POS_FIX_VIACOMBO] = azx_via_get_position,
+ [POS_FIX_COMBO] = azx_get_pos_lpib,
+ };
+
+ chip->get_position[0] = chip->get_position[1] = callbacks[fix];
+
+ /* combo mode uses LPIB only for playback */
+ if (fix == POS_FIX_COMBO)
+ chip->get_position[1] = NULL;
+
+ if (fix == POS_FIX_POSBUF &&
+ (chip->driver_caps & AZX_DCAPS_COUNT_LPIB_DELAY)) {
+ chip->get_delay[0] = chip->get_delay[1] =
+ azx_get_delay_from_lpib;
+ }
+
+}
+
/*
* black-lists for probe_mask
*/
@@ -1173,7 +1386,8 @@ static void azx_check_snoop_available(struct azx *chip)
static void azx_probe_work(struct work_struct *work)
{
- azx_probe_continue(container_of(work, struct azx, probe_work));
+ struct hda_intel *hda = container_of(work, struct hda_intel, probe_work);
+ azx_probe_continue(&hda->chip);
}
/*
@@ -1216,19 +1430,13 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci,
check_msi(chip);
chip->dev_index = dev;
chip->jackpoll_ms = jackpoll_ms;
- INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work);
INIT_LIST_HEAD(&chip->pcm_list);
- INIT_LIST_HEAD(&chip->list);
+ INIT_WORK(&hda->irq_pending_work, azx_irq_pending_work);
+ INIT_LIST_HEAD(&hda->list);
init_vga_switcheroo(chip);
- init_completion(&chip->probe_wait);
-
- chip->position_fix[0] = chip->position_fix[1] =
- check_position_fix(chip, position_fix[dev]);
- /* combo mode uses LPIB for playback */
- if (chip->position_fix[0] == POS_FIX_COMBO) {
- chip->position_fix[0] = POS_FIX_LPIB;
- chip->position_fix[1] = POS_FIX_AUTO;
- }
+ init_completion(&hda->probe_wait);
+
+ assign_position_fix(chip, check_position_fix(chip, position_fix[dev]));
check_probe_mask(chip, dev);
@@ -1257,7 +1465,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci,
}
/* continue probing in work context as may trigger request module */
- INIT_WORK(&chip->probe_work, azx_probe_work);
+ INIT_WORK(&hda->probe_work, azx_probe_work);
*rchip = chip;
@@ -1315,7 +1523,7 @@ static int azx_first_init(struct azx *chip)
NULL);
if (p_smbus) {
if (p_smbus->revision < 0x30)
- gcap &= ~ICH6_GCAP_64OK;
+ gcap &= ~AZX_GCAP_64OK;
pci_dev_put(p_smbus);
}
}
@@ -1323,7 +1531,7 @@ static int azx_first_init(struct azx *chip)
/* disable 64bit DMA address on some devices */
if (chip->driver_caps & AZX_DCAPS_NO_64BIT) {
dev_dbg(card->dev, "Disabling 64bit DMA\n");
- gcap &= ~ICH6_GCAP_64OK;
+ gcap &= ~AZX_GCAP_64OK;
}
/* disable buffer size rounding to 128-byte multiples if supported */
@@ -1339,7 +1547,7 @@ static int azx_first_init(struct azx *chip)
}
/* allow 64bit DMA address if supported by H/W */
- if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
+ if ((gcap & AZX_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64));
else {
pci_set_dma_mask(pci, DMA_BIT_MASK(32));
@@ -1583,6 +1791,7 @@ static int azx_probe(struct pci_dev *pci,
{
static int dev;
struct snd_card *card;
+ struct hda_intel *hda;
struct azx *chip;
bool schedule_probe;
int err;
@@ -1606,6 +1815,7 @@ static int azx_probe(struct pci_dev *pci,
if (err < 0)
goto out_free;
card->private_data = chip;
+ hda = container_of(chip, struct hda_intel, chip);
pci_set_drvdata(pci, card);
@@ -1642,11 +1852,11 @@ static int azx_probe(struct pci_dev *pci,
#endif
if (schedule_probe)
- schedule_work(&chip->probe_work);
+ schedule_work(&hda->probe_work);
dev++;
if (chip->disabled)
- complete_all(&chip->probe_wait);
+ complete_all(&hda->probe_wait);
return 0;
out_free:
@@ -1662,6 +1872,7 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = {
static int azx_probe_continue(struct azx *chip)
{
+ struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
struct pci_dev *pci = chip->pci;
int dev = chip->dev_index;
int err;
@@ -1735,13 +1946,13 @@ static int azx_probe_continue(struct azx *chip)
power_down_all_codecs(chip);
azx_notifier_register(chip);
azx_add_card_list(chip);
- if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) || chip->use_vga_switcheroo)
+ if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) || hda->use_vga_switcheroo)
pm_runtime_put_noidle(&pci->dev);
out_free:
if (err < 0)
- chip->init_failed = 1;
- complete_all(&chip->probe_wait);
+ hda->init_failed = 1;
+ complete_all(&hda->probe_wait);
return err;
}
@@ -1806,6 +2017,9 @@ static const struct pci_device_id azx_ids[] = {
/* BayTrail */
{ PCI_DEVICE(0x8086, 0x0f04),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
+ /* Braswell */
+ { PCI_DEVICE(0x8086, 0x2284),
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
/* ICH */
{ PCI_DEVICE(0x8086, 0x2668),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 4e2d4863daa1..364bb413e02a 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -268,7 +268,8 @@ int snd_hda_input_mux_put(struct hda_codec *codec,
const struct hda_input_mux *imux,
struct snd_ctl_elem_value *ucontrol, hda_nid_t nid,
unsigned int *cur_val);
-int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label,
+int snd_hda_add_imux_item(struct hda_codec *codec,
+ struct hda_input_mux *imux, const char *label,
int index, int *type_index_ret);
/*
@@ -437,6 +438,8 @@ struct snd_hda_pin_quirk {
#endif
+#define HDA_FIXUP_ID_NOT_SET -1
+#define HDA_FIXUP_ID_NO_FIXUP -2
/* fixup types */
enum {
@@ -773,9 +776,9 @@ struct hdmi_eld {
int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid);
int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid,
unsigned char *buf, int *eld_size);
-int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e,
+int snd_hdmi_parse_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e,
const unsigned char *buf, int size);
-void snd_hdmi_show_eld(struct parsed_hdmi_eld *e);
+void snd_hdmi_show_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e);
void snd_hdmi_eld_update_pcm_info(struct parsed_hdmi_eld *e,
struct hda_pcm_stream *hinfo);
diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h
index e9d1a5762a55..949cd437eeb2 100644
--- a/sound/pci/hda/hda_priv.h
+++ b/sound/pci/hda/hda_priv.h
@@ -22,107 +22,87 @@
/*
* registers
*/
-#define ICH6_REG_GCAP 0x00
-#define ICH6_GCAP_64OK (1 << 0) /* 64bit address support */
-#define ICH6_GCAP_NSDO (3 << 1) /* # of serial data out signals */
-#define ICH6_GCAP_BSS (31 << 3) /* # of bidirectional streams */
-#define ICH6_GCAP_ISS (15 << 8) /* # of input streams */
-#define ICH6_GCAP_OSS (15 << 12) /* # of output streams */
-#define ICH6_REG_VMIN 0x02
-#define ICH6_REG_VMAJ 0x03
-#define ICH6_REG_OUTPAY 0x04
-#define ICH6_REG_INPAY 0x06
-#define ICH6_REG_GCTL 0x08
-#define ICH6_GCTL_RESET (1 << 0) /* controller reset */
-#define ICH6_GCTL_FCNTRL (1 << 1) /* flush control */
-#define ICH6_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */
-#define ICH6_REG_WAKEEN 0x0c
-#define ICH6_REG_STATESTS 0x0e
-#define ICH6_REG_GSTS 0x10
-#define ICH6_GSTS_FSTS (1 << 1) /* flush status */
-#define ICH6_REG_INTCTL 0x20
-#define ICH6_REG_INTSTS 0x24
-#define ICH6_REG_WALLCLK 0x30 /* 24Mhz source */
-#define ICH6_REG_OLD_SSYNC 0x34 /* SSYNC for old ICH */
-#define ICH6_REG_SSYNC 0x38
-#define ICH6_REG_CORBLBASE 0x40
-#define ICH6_REG_CORBUBASE 0x44
-#define ICH6_REG_CORBWP 0x48
-#define ICH6_REG_CORBRP 0x4a
-#define ICH6_CORBRP_RST (1 << 15) /* read pointer reset */
-#define ICH6_REG_CORBCTL 0x4c
-#define ICH6_CORBCTL_RUN (1 << 1) /* enable DMA */
-#define ICH6_CORBCTL_CMEIE (1 << 0) /* enable memory error irq */
-#define ICH6_REG_CORBSTS 0x4d
-#define ICH6_CORBSTS_CMEI (1 << 0) /* memory error indication */
-#define ICH6_REG_CORBSIZE 0x4e
-
-#define ICH6_REG_RIRBLBASE 0x50
-#define ICH6_REG_RIRBUBASE 0x54
-#define ICH6_REG_RIRBWP 0x58
-#define ICH6_RIRBWP_RST (1 << 15) /* write pointer reset */
-#define ICH6_REG_RINTCNT 0x5a
-#define ICH6_REG_RIRBCTL 0x5c
-#define ICH6_RBCTL_IRQ_EN (1 << 0) /* enable IRQ */
-#define ICH6_RBCTL_DMA_EN (1 << 1) /* enable DMA */
-#define ICH6_RBCTL_OVERRUN_EN (1 << 2) /* enable overrun irq */
-#define ICH6_REG_RIRBSTS 0x5d
-#define ICH6_RBSTS_IRQ (1 << 0) /* response irq */
-#define ICH6_RBSTS_OVERRUN (1 << 2) /* overrun irq */
-#define ICH6_REG_RIRBSIZE 0x5e
-
-#define ICH6_REG_IC 0x60
-#define ICH6_REG_IR 0x64
-#define ICH6_REG_IRS 0x68
-#define ICH6_IRS_VALID (1<<1)
-#define ICH6_IRS_BUSY (1<<0)
-
-#define ICH6_REG_DPLBASE 0x70
-#define ICH6_REG_DPUBASE 0x74
-#define ICH6_DPLBASE_ENABLE 0x1 /* Enable position buffer */
+#define AZX_REG_GCAP 0x00
+#define AZX_GCAP_64OK (1 << 0) /* 64bit address support */
+#define AZX_GCAP_NSDO (3 << 1) /* # of serial data out signals */
+#define AZX_GCAP_BSS (31 << 3) /* # of bidirectional streams */
+#define AZX_GCAP_ISS (15 << 8) /* # of input streams */
+#define AZX_GCAP_OSS (15 << 12) /* # of output streams */
+#define AZX_REG_VMIN 0x02
+#define AZX_REG_VMAJ 0x03
+#define AZX_REG_OUTPAY 0x04
+#define AZX_REG_INPAY 0x06
+#define AZX_REG_GCTL 0x08
+#define AZX_GCTL_RESET (1 << 0) /* controller reset */
+#define AZX_GCTL_FCNTRL (1 << 1) /* flush control */
+#define AZX_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */
+#define AZX_REG_WAKEEN 0x0c
+#define AZX_REG_STATESTS 0x0e
+#define AZX_REG_GSTS 0x10
+#define AZX_GSTS_FSTS (1 << 1) /* flush status */
+#define AZX_REG_INTCTL 0x20
+#define AZX_REG_INTSTS 0x24
+#define AZX_REG_WALLCLK 0x30 /* 24Mhz source */
+#define AZX_REG_OLD_SSYNC 0x34 /* SSYNC for old ICH */
+#define AZX_REG_SSYNC 0x38
+#define AZX_REG_CORBLBASE 0x40
+#define AZX_REG_CORBUBASE 0x44
+#define AZX_REG_CORBWP 0x48
+#define AZX_REG_CORBRP 0x4a
+#define AZX_CORBRP_RST (1 << 15) /* read pointer reset */
+#define AZX_REG_CORBCTL 0x4c
+#define AZX_CORBCTL_RUN (1 << 1) /* enable DMA */
+#define AZX_CORBCTL_CMEIE (1 << 0) /* enable memory error irq */
+#define AZX_REG_CORBSTS 0x4d
+#define AZX_CORBSTS_CMEI (1 << 0) /* memory error indication */
+#define AZX_REG_CORBSIZE 0x4e
+
+#define AZX_REG_RIRBLBASE 0x50
+#define AZX_REG_RIRBUBASE 0x54
+#define AZX_REG_RIRBWP 0x58
+#define AZX_RIRBWP_RST (1 << 15) /* write pointer reset */
+#define AZX_REG_RINTCNT 0x5a
+#define AZX_REG_RIRBCTL 0x5c
+#define AZX_RBCTL_IRQ_EN (1 << 0) /* enable IRQ */
+#define AZX_RBCTL_DMA_EN (1 << 1) /* enable DMA */
+#define AZX_RBCTL_OVERRUN_EN (1 << 2) /* enable overrun irq */
+#define AZX_REG_RIRBSTS 0x5d
+#define AZX_RBSTS_IRQ (1 << 0) /* response irq */
+#define AZX_RBSTS_OVERRUN (1 << 2) /* overrun irq */
+#define AZX_REG_RIRBSIZE 0x5e
+
+#define AZX_REG_IC 0x60
+#define AZX_REG_IR 0x64
+#define AZX_REG_IRS 0x68
+#define AZX_IRS_VALID (1<<1)
+#define AZX_IRS_BUSY (1<<0)
+
+#define AZX_REG_DPLBASE 0x70
+#define AZX_REG_DPUBASE 0x74
+#define AZX_DPLBASE_ENABLE 0x1 /* Enable position buffer */
/* SD offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */
enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
/* stream register offsets from stream base */
-#define ICH6_REG_SD_CTL 0x00
-#define ICH6_REG_SD_STS 0x03
-#define ICH6_REG_SD_LPIB 0x04
-#define ICH6_REG_SD_CBL 0x08
-#define ICH6_REG_SD_LVI 0x0c
-#define ICH6_REG_SD_FIFOW 0x0e
-#define ICH6_REG_SD_FIFOSIZE 0x10
-#define ICH6_REG_SD_FORMAT 0x12
-#define ICH6_REG_SD_BDLPL 0x18
-#define ICH6_REG_SD_BDLPU 0x1c
+#define AZX_REG_SD_CTL 0x00
+#define AZX_REG_SD_STS 0x03
+#define AZX_REG_SD_LPIB 0x04
+#define AZX_REG_SD_CBL 0x08
+#define AZX_REG_SD_LVI 0x0c
+#define AZX_REG_SD_FIFOW 0x0e
+#define AZX_REG_SD_FIFOSIZE 0x10
+#define AZX_REG_SD_FORMAT 0x12
+#define AZX_REG_SD_BDLPL 0x18
+#define AZX_REG_SD_BDLPU 0x1c
/* PCI space */
-#define ICH6_PCIREG_TCSEL 0x44
+#define AZX_PCIREG_TCSEL 0x44
/*
* other constants
*/
-/* max number of SDs */
-/* ICH, ATI and VIA have 4 playback and 4 capture */
-#define ICH6_NUM_CAPTURE 4
-#define ICH6_NUM_PLAYBACK 4
-
-/* ULI has 6 playback and 5 capture */
-#define ULI_NUM_CAPTURE 5
-#define ULI_NUM_PLAYBACK 6
-
-/* ATI HDMI may have up to 8 playbacks and 0 capture */
-#define ATIHDMI_NUM_CAPTURE 0
-#define ATIHDMI_NUM_PLAYBACK 8
-
-/* TERA has 4 playback and 3 capture */
-#define TERA_NUM_CAPTURE 3
-#define TERA_NUM_PLAYBACK 4
-
-/* this number is statically defined for simplicity */
-#define MAX_AZX_DEV 16
-
/* max number of fragments - we may use more if allocating more pages for BDL */
#define BDL_SIZE 4096
#define AZX_MAX_BDL_ENTRIES (BDL_SIZE / 16)
@@ -160,13 +140,13 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
#define SD_STS_FIFO_READY 0x20 /* FIFO ready */
/* INTCTL and INTSTS */
-#define ICH6_INT_ALL_STREAM 0xff /* all stream interrupts */
-#define ICH6_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */
-#define ICH6_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */
+#define AZX_INT_ALL_STREAM 0xff /* all stream interrupts */
+#define AZX_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */
+#define AZX_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */
/* below are so far hardcoded - should read registers in future */
-#define ICH6_MAX_CORB_ENTRIES 256
-#define ICH6_MAX_RIRB_ENTRIES 256
+#define AZX_MAX_CORB_ENTRIES 256
+#define AZX_MAX_RIRB_ENTRIES 256
/* driver quirks (capabilities) */
/* bits 0-7 are used for indicating driver type */
@@ -192,35 +172,6 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
#define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */
#define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */
-/* position fix mode */
-enum {
- POS_FIX_AUTO,
- POS_FIX_LPIB,
- POS_FIX_POSBUF,
- POS_FIX_VIACOMBO,
- POS_FIX_COMBO,
-};
-
-/* Defines for ATI HD Audio support in SB450 south bridge */
-#define ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR 0x42
-#define ATI_SB450_HDAUDIO_ENABLE_SNOOP 0x02
-
-/* Defines for Nvidia HDA support */
-#define NVIDIA_HDA_TRANSREG_ADDR 0x4e
-#define NVIDIA_HDA_ENABLE_COHBITS 0x0f
-#define NVIDIA_HDA_ISTRM_COH 0x4d
-#define NVIDIA_HDA_OSTRM_COH 0x4c
-#define NVIDIA_HDA_ENABLE_COHBIT 0x01
-
-/* Defines for Intel SCH HDA snoop control */
-#define INTEL_SCH_HDA_DEVC 0x78
-#define INTEL_SCH_HDA_DEVC_NOSNOOP (0x1<<11)
-
-/* Define IN stream 0 FIFO size offset in VIA controller */
-#define VIA_IN_STREAM0_FIFO_SIZE_OFFSET 0x90
-/* Define VIA HD Audio Device ID*/
-#define VIA_HDAC_DEVICE_ID 0x3288
-
/* HD Audio class code */
#define PCI_CLASS_MULTIMEDIA_HD_AUDIO 0x0403
@@ -325,6 +276,9 @@ struct azx_pcm {
struct list_head list;
};
+typedef unsigned int (*azx_get_pos_callback_t)(struct azx *, struct azx_dev *);
+typedef int (*azx_get_delay_callback_t)(struct azx *, struct azx_dev *, unsigned int pos);
+
struct azx {
struct snd_card *card;
struct pci_dev *pci;
@@ -343,6 +297,10 @@ struct azx {
/* Register interaction. */
const struct hda_controller_ops *ops;
+ /* position adjustment callbacks */
+ azx_get_pos_callback_t get_position[2];
+ azx_get_delay_callback_t get_delay[2];
+
/* pci resources */
unsigned long addr;
void __iomem *remap_addr;
@@ -351,7 +309,6 @@ struct azx {
/* locks */
spinlock_t reg_lock;
struct mutex open_mutex; /* Prevents concurrent open/close operations */
- struct completion probe_wait;
/* streams (x num_streams) */
struct azx_dev *azx_dev;
@@ -378,7 +335,6 @@ struct azx {
#endif
/* flags */
- int position_fix[2]; /* for both playback/capture streams */
const int *bdl_pos_adj;
int poll_count;
unsigned int running:1;
@@ -386,46 +342,23 @@ struct azx {
unsigned int single_cmd:1;
unsigned int polling_mode:1;
unsigned int msi:1;
- unsigned int irq_pending_warned:1;
unsigned int probing:1; /* codec probing phase */
unsigned int snoop:1;
unsigned int align_buffer_size:1;
unsigned int region_requested:1;
-
- /* VGA-switcheroo setup */
- unsigned int use_vga_switcheroo:1;
- unsigned int vga_switcheroo_registered:1;
- unsigned int init_failed:1; /* delayed init failed */
unsigned int disabled:1; /* disabled by VGA-switcher */
/* for debugging */
unsigned int last_cmd[AZX_MAX_CODECS];
- /* for pending irqs */
- struct work_struct irq_pending_work;
-
- struct work_struct probe_work;
-
/* reboot notifier (for mysterious hangup problem at power-down) */
struct notifier_block reboot_notifier;
- /* card list (for power_save trigger) */
- struct list_head list;
-
#ifdef CONFIG_SND_HDA_DSP_LOADER
struct azx_dev saved_azx_dev;
#endif
-
- /* secondary power domain for hdmi audio under vga device */
- struct dev_pm_domain hdmi_pm_domain;
};
-#ifdef CONFIG_SND_VERBOSE_PRINTK
-#define SFX /* nop */
-#else
-#define SFX "hda-intel "
-#endif
-
#ifdef CONFIG_X86
#define azx_snoop(chip) ((chip)->snoop)
#else
@@ -437,29 +370,29 @@ struct azx {
*/
#define azx_writel(chip, reg, value) \
- ((chip)->ops->reg_writel(value, (chip)->remap_addr + ICH6_REG_##reg))
+ ((chip)->ops->reg_writel(value, (chip)->remap_addr + AZX_REG_##reg))
#define azx_readl(chip, reg) \
- ((chip)->ops->reg_readl((chip)->remap_addr + ICH6_REG_##reg))
+ ((chip)->ops->reg_readl((chip)->remap_addr + AZX_REG_##reg))
#define azx_writew(chip, reg, value) \
- ((chip)->ops->reg_writew(value, (chip)->remap_addr + ICH6_REG_##reg))
+ ((chip)->ops->reg_writew(value, (chip)->remap_addr + AZX_REG_##reg))
#define azx_readw(chip, reg) \
- ((chip)->ops->reg_readw((chip)->remap_addr + ICH6_REG_##reg))
+ ((chip)->ops->reg_readw((chip)->remap_addr + AZX_REG_##reg))
#define azx_writeb(chip, reg, value) \
- ((chip)->ops->reg_writeb(value, (chip)->remap_addr + ICH6_REG_##reg))
+ ((chip)->ops->reg_writeb(value, (chip)->remap_addr + AZX_REG_##reg))
#define azx_readb(chip, reg) \
- ((chip)->ops->reg_readb((chip)->remap_addr + ICH6_REG_##reg))
+ ((chip)->ops->reg_readb((chip)->remap_addr + AZX_REG_##reg))
#define azx_sd_writel(chip, dev, reg, value) \
- ((chip)->ops->reg_writel(value, (dev)->sd_addr + ICH6_REG_##reg))
+ ((chip)->ops->reg_writel(value, (dev)->sd_addr + AZX_REG_##reg))
#define azx_sd_readl(chip, dev, reg) \
- ((chip)->ops->reg_readl((dev)->sd_addr + ICH6_REG_##reg))
+ ((chip)->ops->reg_readl((dev)->sd_addr + AZX_REG_##reg))
#define azx_sd_writew(chip, dev, reg, value) \
- ((chip)->ops->reg_writew(value, (dev)->sd_addr + ICH6_REG_##reg))
+ ((chip)->ops->reg_writew(value, (dev)->sd_addr + AZX_REG_##reg))
#define azx_sd_readw(chip, dev, reg) \
- ((chip)->ops->reg_readw((dev)->sd_addr + ICH6_REG_##reg))
+ ((chip)->ops->reg_readw((dev)->sd_addr + AZX_REG_##reg))
#define azx_sd_writeb(chip, dev, reg, value) \
- ((chip)->ops->reg_writeb(value, (dev)->sd_addr + ICH6_REG_##reg))
+ ((chip)->ops->reg_writeb(value, (dev)->sd_addr + AZX_REG_##reg))
#define azx_sd_readb(chip, dev, reg) \
- ((chip)->ops->reg_readb((dev)->sd_addr + ICH6_REG_##reg))
+ ((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg))
#endif /* __SOUND_HDA_PRIV_H */
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 358414da6418..227990bc02e3 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -29,7 +29,6 @@
#include <linux/moduleparam.h>
#include <linux/mutex.h>
#include <linux/of_device.h>
-#include <linux/reboot.h>
#include <linux/slab.h>
#include <linux/time.h>
@@ -272,13 +271,9 @@ static int hda_tegra_resume(struct device *dev)
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip = card->private_data;
struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip);
- int status;
hda_tegra_enable_clocks(hda);
- /* Read STATESTS before controller reset */
- status = azx_readw(chip, STATESTS);
-
hda_tegra_init(hda);
azx_init_chip(chip, 1);
@@ -295,30 +290,6 @@ static const struct dev_pm_ops hda_tegra_pm = {
};
/*
- * reboot notifier for hang-up problem at power-down
- */
-static int hda_tegra_halt(struct notifier_block *nb, unsigned long event,
- void *buf)
-{
- struct azx *chip = container_of(nb, struct azx, reboot_notifier);
- snd_hda_bus_reboot_notify(chip->bus);
- azx_stop_chip(chip);
- return NOTIFY_OK;
-}
-
-static void hda_tegra_notifier_register(struct azx *chip)
-{
- chip->reboot_notifier.notifier_call = hda_tegra_halt;
- register_reboot_notifier(&chip->reboot_notifier);
-}
-
-static void hda_tegra_notifier_unregister(struct azx *chip)
-{
- if (chip->reboot_notifier.notifier_call)
- unregister_reboot_notifier(&chip->reboot_notifier);
-}
-
-/*
* destructor
*/
static int hda_tegra_dev_free(struct snd_device *device)
@@ -326,7 +297,7 @@ static int hda_tegra_dev_free(struct snd_device *device)
int i;
struct azx *chip = device->device_data;
- hda_tegra_notifier_unregister(chip);
+ azx_notifier_unregister(chip);
if (chip->initialized) {
for (i = 0; i < chip->num_streams; i++)
@@ -478,10 +449,7 @@ static int hda_tegra_create(struct snd_card *card,
chip->driver_type = driver_caps & 0xff;
chip->dev_index = 0;
INIT_LIST_HEAD(&chip->pcm_list);
- INIT_LIST_HEAD(&chip->list);
- chip->position_fix[0] = POS_FIX_AUTO;
- chip->position_fix[1] = POS_FIX_AUTO;
chip->codec_probe_mask = -1;
chip->single_cmd = false;
@@ -559,7 +527,7 @@ static int hda_tegra_probe(struct platform_device *pdev)
chip->running = 1;
power_down_all_codecs(chip);
- hda_tegra_notifier_register(chip);
+ azx_notifier_register(chip);
return 0;
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 092f2bd030bd..4f3aba78f720 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -2046,14 +2046,14 @@ enum dma_state {
DMA_STATE_RUN = 1
};
-static int dma_convert_to_hda_format(
+static int dma_convert_to_hda_format(struct hda_codec *codec,
unsigned int sample_rate,
unsigned short channels,
unsigned short *hda_format)
{
unsigned int format_val;
- format_val = snd_hda_calc_stream_format(
+ format_val = snd_hda_calc_stream_format(codec,
sample_rate,
channels,
SNDRV_PCM_FORMAT_S32_LE,
@@ -2452,7 +2452,7 @@ static int dspxfr_image(struct hda_codec *codec,
}
dma_engine->codec = codec;
- dma_convert_to_hda_format(sample_rate, channels, &hda_format);
+ dma_convert_to_hda_format(codec, sample_rate, channels, &hda_format);
dma_engine->m_converter_format = hda_format;
dma_engine->buf_size = (ovly ? DSP_DMA_WRITE_BUFLEN_OVLY :
DSP_DMA_WRITE_BUFLEN_INIT) * 2;
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 387f0b551889..3db724eaa53c 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -657,8 +657,10 @@ static void cs4208_fixup_mac(struct hda_codec *codec,
{
if (action != HDA_FIXUP_ACT_PRE_PROBE)
return;
+
+ codec->fixup_id = HDA_FIXUP_ID_NOT_SET;
snd_hda_pick_fixup(codec, NULL, cs4208_mac_fixup_tbl, cs4208_fixups);
- if (codec->fixup_id < 0 || codec->fixup_id == CS4208_MAC_AUTO)
+ if (codec->fixup_id == HDA_FIXUP_ID_NOT_SET)
codec->fixup_id = CS4208_GPIO0; /* default fixup */
snd_hda_apply_fixup(codec, action);
}
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 061ea5965dd5..ed3d133ffbb6 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -31,550 +31,11 @@
#include "hda_jack.h"
#include "hda_generic.h"
-#undef ENABLE_CMI_STATIC_QUIRKS
-
-#ifdef ENABLE_CMI_STATIC_QUIRKS
-#define NUM_PINS 11
-
-
-/* board config type */
-enum {
- CMI_MINIMAL, /* back 3-jack */
- CMI_MIN_FP, /* back 3-jack + front-panel 2-jack */
- CMI_FULL, /* back 6-jack + front-panel 2-jack */
- CMI_FULL_DIG, /* back 6-jack + front-panel 2-jack + digital I/O */
- CMI_ALLOUT, /* back 5-jack + front-panel 2-jack + digital out */
- CMI_AUTO, /* let driver guess it */
- CMI_MODELS
-};
-#endif /* ENABLE_CMI_STATIC_QUIRKS */
-
struct cmi_spec {
struct hda_gen_spec gen;
-
-#ifdef ENABLE_CMI_STATIC_QUIRKS
- /* below are only for static models */
-
- int board_config;
- unsigned int no_line_in: 1; /* no line-in (5-jack) */
- unsigned int front_panel: 1; /* has front-panel 2-jack */
-
- /* playback */
- struct hda_multi_out multiout;
- hda_nid_t dac_nids[AUTO_CFG_MAX_OUTS]; /* NID for each DAC */
- int num_dacs;
-
- /* capture */
- const hda_nid_t *adc_nids;
- hda_nid_t dig_in_nid;
-
- /* capture source */
- const struct hda_input_mux *input_mux;
- unsigned int cur_mux[2];
-
- /* channel mode */
- int num_channel_modes;
- const struct hda_channel_mode *channel_modes;
-
- struct hda_pcm pcm_rec[2]; /* PCM information */
-
- /* pin default configuration */
- hda_nid_t pin_nid[NUM_PINS];
- unsigned int def_conf[NUM_PINS];
- unsigned int pin_def_confs;
-
- /* multichannel pins */
- struct hda_verb multi_init[9]; /* 2 verbs for each pin + terminator */
-#endif /* ENABLE_CMI_STATIC_QUIRKS */
-};
-
-#ifdef ENABLE_CMI_STATIC_QUIRKS
-/*
- * input MUX
- */
-static int cmi_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct cmi_spec *spec = codec->spec;
- return snd_hda_input_mux_info(spec->input_mux, uinfo);
-}
-
-static int cmi_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct cmi_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 cmi_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct cmi_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->adc_nids[adc_idx], &spec->cur_mux[adc_idx]);
-}
-
-/*
- * shared line-in, mic for surrounds
- */
-
-/* 3-stack / 2 channel */
-static const struct hda_verb cmi9880_ch2_init[] = {
- /* set line-in PIN for input */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- /* set mic PIN for input, also enable vref */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- /* route front PCM (DAC1) to HP */
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
- {}
-};
-
-/* 3-stack / 6 channel */
-static const struct hda_verb cmi9880_ch6_init[] = {
- /* set line-in PIN for output */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- /* set mic PIN for output */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- /* route front PCM (DAC1) to HP */
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
- {}
-};
-
-/* 3-stack+front / 8 channel */
-static const struct hda_verb cmi9880_ch8_init[] = {
- /* set line-in PIN for output */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- /* set mic PIN for output */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- /* route rear-surround PCM (DAC4) to HP */
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x03 },
- {}
-};
-
-static const struct hda_channel_mode cmi9880_channel_modes[3] = {
- { 2, cmi9880_ch2_init },
- { 6, cmi9880_ch6_init },
- { 8, cmi9880_ch8_init },
-};
-
-static int cmi_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct cmi_spec *spec = codec->spec;
- return snd_hda_ch_mode_info(codec, uinfo, spec->channel_modes,
- spec->num_channel_modes);
-}
-
-static int cmi_ch_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct cmi_spec *spec = codec->spec;
- return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_modes,
- spec->num_channel_modes, spec->multiout.max_channels);
-}
-
-static int cmi_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct cmi_spec *spec = codec->spec;
- return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_modes,
- spec->num_channel_modes, &spec->multiout.max_channels);
-}
-
-/*
- */
-static const struct snd_kcontrol_new cmi9880_basic_mixer[] = {
- /* CMI9880 has no playback volumes! */
- HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), /* front */
- HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Side Playback Switch", 0x06, 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 = cmi_mux_enum_info,
- .get = cmi_mux_enum_get,
- .put = cmi_mux_enum_put,
- },
- HDA_CODEC_VOLUME("Capture Volume", 0x08, 0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x08, 0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Beep Playback Volume", 0x23, 0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Beep Playback Switch", 0x23, 0, HDA_OUTPUT),
- { } /* end */
};
/*
- * shared I/O pins
- */
-static const struct snd_kcontrol_new cmi9880_ch_mode_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = cmi_ch_mode_info,
- .get = cmi_ch_mode_get,
- .put = cmi_ch_mode_put,
- },
- { } /* end */
-};
-
-/* AUD-in selections:
- * 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x1f 0x20
- */
-static const struct hda_input_mux cmi9880_basic_mux = {
- .num_items = 4,
- .items = {
- { "Front Mic", 0x5 },
- { "Rear Mic", 0x2 },
- { "Line", 0x1 },
- { "CD", 0x7 },
- }
-};
-
-static const struct hda_input_mux cmi9880_no_line_mux = {
- .num_items = 3,
- .items = {
- { "Front Mic", 0x5 },
- { "Rear Mic", 0x2 },
- { "CD", 0x7 },
- }
-};
-
-/* front, rear, clfe, rear_surr */
-static const hda_nid_t cmi9880_dac_nids[4] = {
- 0x03, 0x04, 0x05, 0x06
-};
-/* ADC0, ADC1 */
-static const hda_nid_t cmi9880_adc_nids[2] = {
- 0x08, 0x09
-};
-
-#define CMI_DIG_OUT_NID 0x07
-#define CMI_DIG_IN_NID 0x0a
-
-/*
- */
-static const struct hda_verb cmi9880_basic_init[] = {
- /* port-D for line out (rear panel) */
- { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- /* port-E for HP out (front panel) */
- { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- /* route front PCM to HP */
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
- /* port-A for surround (rear panel) */
- { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- /* port-G for CLFE (rear panel) */
- { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- { 0x1f, AC_VERB_SET_CONNECT_SEL, 0x02 },
- /* port-H for side (rear panel) */
- { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- { 0x20, AC_VERB_SET_CONNECT_SEL, 0x01 },
- /* port-C for line-in (rear panel) */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- /* port-B for mic-in (rear panel) with vref */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- /* port-F for mic-in (front panel) with vref */
- { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- /* CD-in */
- { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- /* route front mic to ADC1/2 */
- { 0x08, AC_VERB_SET_CONNECT_SEL, 0x05 },
- { 0x09, AC_VERB_SET_CONNECT_SEL, 0x05 },
- {} /* terminator */
-};
-
-static const struct hda_verb cmi9880_allout_init[] = {
- /* port-D for line out (rear panel) */
- { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- /* port-E for HP out (front panel) */
- { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- /* route front PCM to HP */
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
- /* port-A for side (rear panel) */
- { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- /* port-G for CLFE (rear panel) */
- { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- { 0x1f, AC_VERB_SET_CONNECT_SEL, 0x02 },
- /* port-H for side (rear panel) */
- { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- { 0x20, AC_VERB_SET_CONNECT_SEL, 0x01 },
- /* port-C for surround (rear panel) */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- /* port-B for mic-in (rear panel) with vref */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- /* port-F for mic-in (front panel) with vref */
- { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- /* CD-in */
- { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- /* route front mic to ADC1/2 */
- { 0x08, AC_VERB_SET_CONNECT_SEL, 0x05 },
- { 0x09, AC_VERB_SET_CONNECT_SEL, 0x05 },
- {} /* terminator */
-};
-
-/*
- */
-static int cmi9880_build_controls(struct hda_codec *codec)
-{
- struct cmi_spec *spec = codec->spec;
- struct snd_kcontrol *kctl;
- int i, err;
-
- err = snd_hda_add_new_ctls(codec, cmi9880_basic_mixer);
- if (err < 0)
- return err;
- if (spec->channel_modes) {
- err = snd_hda_add_new_ctls(codec, cmi9880_ch_mode_mixer);
- 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;
- }
-
- /* assign Capture Source enums to NID */
- kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
- for (i = 0; kctl && i < kctl->count; i++) {
- err = snd_hda_add_nid(codec, kctl, i, spec->adc_nids[i]);
- if (err < 0)
- return err;
- }
- return 0;
-}
-
-static int cmi9880_init(struct hda_codec *codec)
-{
- struct cmi_spec *spec = codec->spec;
- if (spec->board_config == CMI_ALLOUT)
- snd_hda_sequence_write(codec, cmi9880_allout_init);
- else
- snd_hda_sequence_write(codec, cmi9880_basic_init);
- if (spec->board_config == CMI_AUTO)
- snd_hda_sequence_write(codec, spec->multi_init);
- return 0;
-}
-
-/*
- * Analog playback callbacks
- */
-static int cmi9880_playback_pcm_open(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct cmi_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
- hinfo);
-}
-
-static int cmi9880_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 cmi_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
- format, substream);
-}
-
-static int cmi9880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct cmi_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
-}
-
-/*
- * Digital out
- */
-static int cmi9880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct cmi_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_open(codec, &spec->multiout);
-}
-
-static int cmi9880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct cmi_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_close(codec, &spec->multiout);
-}
-
-static int cmi9880_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 cmi_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
- format, substream);
-}
-
-/*
- * Analog capture
- */
-static int cmi9880_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 cmi_spec *spec = codec->spec;
-
- snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
- stream_tag, 0, format);
- return 0;
-}
-
-static int cmi9880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct cmi_spec *spec = codec->spec;
-
- snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
- return 0;
-}
-
-
-/*
- */
-static const struct hda_pcm_stream cmi9880_pcm_analog_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 8,
- .nid = 0x03, /* NID to query formats and rates */
- .ops = {
- .open = cmi9880_playback_pcm_open,
- .prepare = cmi9880_playback_pcm_prepare,
- .cleanup = cmi9880_playback_pcm_cleanup
- },
-};
-
-static const struct hda_pcm_stream cmi9880_pcm_analog_capture = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0x08, /* NID to query formats and rates */
- .ops = {
- .prepare = cmi9880_capture_pcm_prepare,
- .cleanup = cmi9880_capture_pcm_cleanup
- },
-};
-
-static const struct hda_pcm_stream cmi9880_pcm_digital_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- /* NID is set in cmi9880_build_pcms */
- .ops = {
- .open = cmi9880_dig_playback_pcm_open,
- .close = cmi9880_dig_playback_pcm_close,
- .prepare = cmi9880_dig_playback_pcm_prepare
- },
-};
-
-static const struct hda_pcm_stream cmi9880_pcm_digital_capture = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- /* NID is set in cmi9880_build_pcms */
-};
-
-static int cmi9880_build_pcms(struct hda_codec *codec)
-{
- struct cmi_spec *spec = codec->spec;
- struct hda_pcm *info = spec->pcm_rec;
-
- codec->num_pcms = 1;
- codec->pcm_info = info;
-
- info->name = "CMI9880";
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cmi9880_pcm_analog_playback;
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = cmi9880_pcm_analog_capture;
-
- if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
- codec->num_pcms++;
- info++;
- info->name = "CMI9880 Digital";
- info->pcm_type = HDA_PCM_TYPE_SPDIF;
- if (spec->multiout.dig_out_nid) {
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cmi9880_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] = cmi9880_pcm_digital_capture;
- info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
- }
- }
-
- return 0;
-}
-
-static void cmi9880_free(struct hda_codec *codec)
-{
- kfree(codec->spec);
-}
-
-/*
- */
-
-static const char * const cmi9880_models[CMI_MODELS] = {
- [CMI_MINIMAL] = "minimal",
- [CMI_MIN_FP] = "min_fp",
- [CMI_FULL] = "full",
- [CMI_FULL_DIG] = "full_dig",
- [CMI_ALLOUT] = "allout",
- [CMI_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk cmi9880_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", CMI_FULL_DIG),
- SND_PCI_QUIRK(0x1854, 0x002b, "LG LS75", CMI_MINIMAL),
- SND_PCI_QUIRK(0x1854, 0x0032, "LG", CMI_FULL_DIG),
- {} /* terminator */
-};
-
-static const struct hda_codec_ops cmi9880_patch_ops = {
- .build_controls = cmi9880_build_controls,
- .build_pcms = cmi9880_build_pcms,
- .init = cmi9880_init,
- .free = cmi9880_free,
-};
-#endif /* ENABLE_CMI_STATIC_QUIRKS */
-
-/*
* stuff for auto-parser
*/
static const struct hda_codec_ops cmi_auto_patch_ops = {
@@ -585,12 +46,18 @@ static const struct hda_codec_ops cmi_auto_patch_ops = {
.unsol_event = snd_hda_jack_unsol_event,
};
-static int cmi_parse_auto_config(struct hda_codec *codec)
+static int patch_cmi9880(struct hda_codec *codec)
{
- struct cmi_spec *spec = codec->spec;
- struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+ struct cmi_spec *spec;
+ struct auto_pin_cfg *cfg;
int err;
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+ cfg = &spec->gen.autocfg;
snd_hda_gen_spec_init(&spec->gen);
err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0);
@@ -608,79 +75,6 @@ static int cmi_parse_auto_config(struct hda_codec *codec)
return err;
}
-
-static int patch_cmi9880(struct hda_codec *codec)
-{
- struct cmi_spec *spec;
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-#ifdef ENABLE_CMI_STATIC_QUIRKS
- spec->board_config = snd_hda_check_board_config(codec, CMI_MODELS,
- cmi9880_models,
- cmi9880_cfg_tbl);
- if (spec->board_config < 0) {
- codec_dbg(codec, "%s: BIOS auto-probing.\n",
- codec->chip_name);
- spec->board_config = CMI_AUTO; /* try everything */
- }
-
- if (spec->board_config == CMI_AUTO)
- return cmi_parse_auto_config(codec);
-
- /* copy default DAC NIDs */
- memcpy(spec->dac_nids, cmi9880_dac_nids, sizeof(spec->dac_nids));
- spec->num_dacs = 4;
-
- switch (spec->board_config) {
- case CMI_MINIMAL:
- case CMI_MIN_FP:
- spec->channel_modes = cmi9880_channel_modes;
- if (spec->board_config == CMI_MINIMAL)
- spec->num_channel_modes = 2;
- else {
- spec->front_panel = 1;
- spec->num_channel_modes = 3;
- }
- spec->multiout.max_channels = cmi9880_channel_modes[0].channels;
- spec->input_mux = &cmi9880_basic_mux;
- break;
- case CMI_FULL:
- case CMI_FULL_DIG:
- spec->front_panel = 1;
- spec->multiout.max_channels = 8;
- spec->input_mux = &cmi9880_basic_mux;
- if (spec->board_config == CMI_FULL_DIG) {
- spec->multiout.dig_out_nid = CMI_DIG_OUT_NID;
- spec->dig_in_nid = CMI_DIG_IN_NID;
- }
- break;
- case CMI_ALLOUT:
- default:
- spec->front_panel = 1;
- spec->multiout.max_channels = 8;
- spec->no_line_in = 1;
- spec->input_mux = &cmi9880_no_line_mux;
- spec->multiout.dig_out_nid = CMI_DIG_OUT_NID;
- break;
- }
-
- spec->multiout.num_dacs = spec->num_dacs;
- spec->multiout.dac_nids = spec->dac_nids;
-
- spec->adc_nids = cmi9880_adc_nids;
-
- codec->patch_ops = cmi9880_patch_ops;
-
- return 0;
-#else
- return cmi_parse_auto_config(codec);
-#endif
-}
-
/*
* patch entries
*/
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 1dc7e974f3b1..7627a69ca6d7 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -34,27 +34,6 @@
#include "hda_jack.h"
#include "hda_generic.h"
-#undef ENABLE_CXT_STATIC_QUIRKS
-
-#define CXT_PIN_DIR_IN 0x00
-#define CXT_PIN_DIR_OUT 0x01
-#define CXT_PIN_DIR_INOUT 0x02
-#define CXT_PIN_DIR_IN_NOMICBIAS 0x03
-#define CXT_PIN_DIR_INOUT_NOMICBIAS 0x04
-
-#define CONEXANT_HP_EVENT 0x37
-#define CONEXANT_MIC_EVENT 0x38
-#define CONEXANT_LINE_EVENT 0x39
-
-/* Conexant 5051 specific */
-
-#define CXT5051_SPDIF_OUT 0x12
-#define CXT5051_PORTB_EVENT 0x38
-#define CXT5051_PORTC_EVENT 0x39
-
-#define AUTO_MIC_PORTB (1 << 1)
-#define AUTO_MIC_PORTC (1 << 2)
-
struct conexant_spec {
struct hda_gen_spec gen;
@@ -72,64 +51,6 @@ struct conexant_spec {
bool dc_enable;
unsigned int dc_input_bias; /* offset into olpc_xo_dc_bias */
struct nid_path *dc_mode_path;
-
-#ifdef ENABLE_CXT_STATIC_QUIRKS
- const struct snd_kcontrol_new *mixers[5];
- int num_mixers;
- hda_nid_t vmaster_nid;
-
- const struct hda_verb *init_verbs[5]; /* 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 hp_present;
- unsigned int line_present;
- unsigned int auto_mic;
-
- /* capture */
- unsigned int num_adc_nids;
- const hda_nid_t *adc_nids;
- hda_nid_t dig_in_nid; /* digital-in NID; optional */
-
- unsigned int cur_adc_idx;
- hda_nid_t cur_adc;
- unsigned int cur_adc_stream_tag;
- unsigned int cur_adc_format;
-
- const struct hda_pcm_stream *capture_stream;
-
- /* 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[2]; /* used in build_pcms() */
-
- unsigned int spdif_route;
-
- unsigned int port_d_mode;
- unsigned int dell_automute:1;
- unsigned int dell_vostro:1;
- unsigned int ideapad:1;
- unsigned int thinkpad:1;
- unsigned int hp_laptop:1;
- unsigned int asus:1;
-
- unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */
-#endif /* ENABLE_CXT_STATIC_QUIRKS */
};
@@ -173,2533 +94,6 @@ static int add_beep_ctls(struct hda_codec *codec)
#define add_beep_ctls(codec) 0
#endif
-
-#ifdef ENABLE_CXT_STATIC_QUIRKS
-static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct conexant_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
- hinfo);
-}
-
-static int conexant_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 conexant_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
- stream_tag,
- format, substream);
-}
-
-static int conexant_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct conexant_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
-}
-
-/*
- * Digital out
- */
-static int conexant_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct conexant_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_open(codec, &spec->multiout);
-}
-
-static int conexant_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct conexant_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_close(codec, &spec->multiout);
-}
-
-static int conexant_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 conexant_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
- stream_tag,
- format, substream);
-}
-
-/*
- * Analog capture
- */
-static int conexant_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 conexant_spec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
- stream_tag, 0, format);
- return 0;
-}
-
-static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct conexant_spec *spec = codec->spec;
- snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
- return 0;
-}
-
-
-
-static const struct hda_pcm_stream conexant_pcm_analog_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0, /* fill later */
- .ops = {
- .open = conexant_playback_pcm_open,
- .prepare = conexant_playback_pcm_prepare,
- .cleanup = conexant_playback_pcm_cleanup
- },
-};
-
-static const struct hda_pcm_stream conexant_pcm_analog_capture = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0, /* fill later */
- .ops = {
- .prepare = conexant_capture_pcm_prepare,
- .cleanup = conexant_capture_pcm_cleanup
- },
-};
-
-
-static const struct hda_pcm_stream conexant_pcm_digital_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0, /* fill later */
- .ops = {
- .open = conexant_dig_playback_pcm_open,
- .close = conexant_dig_playback_pcm_close,
- .prepare = conexant_dig_playback_pcm_prepare
- },
-};
-
-static const struct hda_pcm_stream conexant_pcm_digital_capture = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- /* NID is set in alc_build_pcms */
-};
-
-static int cx5051_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 conexant_spec *spec = codec->spec;
- spec->cur_adc = spec->adc_nids[spec->cur_adc_idx];
- spec->cur_adc_stream_tag = stream_tag;
- spec->cur_adc_format = format;
- snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format);
- return 0;
-}
-
-static int cx5051_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct conexant_spec *spec = codec->spec;
- snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
- spec->cur_adc = 0;
- return 0;
-}
-
-static const struct hda_pcm_stream cx5051_pcm_analog_capture = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0, /* fill later */
- .ops = {
- .prepare = cx5051_capture_pcm_prepare,
- .cleanup = cx5051_capture_pcm_cleanup
- },
-};
-
-static int conexant_build_pcms(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- struct hda_pcm *info = spec->pcm_rec;
-
- codec->num_pcms = 1;
- codec->pcm_info = info;
-
- info->name = "CONEXANT Analog";
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = conexant_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];
- if (spec->capture_stream)
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = *spec->capture_stream;
- else {
- if (codec->vendor_id == 0x14f15051)
- info->stream[SNDRV_PCM_STREAM_CAPTURE] =
- cx5051_pcm_analog_capture;
- else {
- info->stream[SNDRV_PCM_STREAM_CAPTURE] =
- conexant_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++;
- info->name = "Conexant Digital";
- info->pcm_type = HDA_PCM_TYPE_SPDIF;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
- conexant_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] =
- conexant_pcm_digital_capture;
- info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
- spec->dig_in_nid;
- }
- }
-
- return 0;
-}
-
-static int conexant_mux_enum_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_spec *spec = codec->spec;
-
- return snd_hda_input_mux_info(spec->input_mux, uinfo);
-}
-
-static int conexant_mux_enum_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_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 conexant_mux_enum_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_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]);
-}
-
-static void conexant_set_power(struct hda_codec *codec, hda_nid_t fg,
- unsigned int power_state)
-{
- if (power_state == AC_PWRST_D3)
- msleep(100);
- snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
- power_state);
- /* partial workaround for "azx_get_response timeout" */
- if (power_state == AC_PWRST_D0)
- msleep(10);
- snd_hda_codec_set_power_to_all(codec, fg, power_state);
-}
-
-static int conexant_init(struct hda_codec *codec)
-{
- struct conexant_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 void conexant_free(struct hda_codec *codec)
-{
- kfree(codec->spec);
-}
-
-static const struct snd_kcontrol_new cxt_capture_mixers[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = conexant_mux_enum_info,
- .get = conexant_mux_enum_get,
- .put = conexant_mux_enum_put
- },
- {}
-};
-
-static const char * const slave_pfxs[] = {
- "Headphone", "Speaker", "Bass Speaker", "Front", "Surround", "CLFE",
- NULL
-};
-
-static int conexant_build_controls(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- 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;
- }
-
- /* if we have no master control, let's create it */
- if (spec->vmaster_nid &&
- !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, slave_pfxs,
- "Playback Volume");
- if (err < 0)
- return err;
- }
- if (spec->vmaster_nid &&
- !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
- err = snd_hda_add_vmaster(codec, "Master Playback Switch",
- NULL, slave_pfxs,
- "Playback Switch");
- if (err < 0)
- return err;
- }
-
- if (spec->input_mux) {
- err = snd_hda_add_new_ctls(codec, cxt_capture_mixers);
- if (err < 0)
- return err;
- }
-
- err = add_beep_ctls(codec);
- if (err < 0)
- return err;
-
- return 0;
-}
-
-static const struct hda_codec_ops conexant_patch_ops = {
- .build_controls = conexant_build_controls,
- .build_pcms = conexant_build_pcms,
- .init = conexant_init,
- .free = conexant_free,
- .set_power_state = conexant_set_power,
-};
-
-static int patch_conexant_auto(struct hda_codec *codec);
-/*
- * EAPD control
- * the private value = nid | (invert << 8)
- */
-
-#define cxt_eapd_info snd_ctl_boolean_mono_info
-
-static int cxt_eapd_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_spec *spec = codec->spec;
- int invert = (kcontrol->private_value >> 8) & 1;
- if (invert)
- ucontrol->value.integer.value[0] = !spec->cur_eapd;
- else
- ucontrol->value.integer.value[0] = spec->cur_eapd;
- return 0;
-
-}
-
-static int cxt_eapd_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_spec *spec = codec->spec;
- int invert = (kcontrol->private_value >> 8) & 1;
- hda_nid_t nid = kcontrol->private_value & 0xff;
- unsigned int eapd;
-
- eapd = !!ucontrol->value.integer.value[0];
- if (invert)
- 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;
-}
-
-/* controls for test mode */
-#ifdef CONFIG_SND_DEBUG
-
-#define CXT_EAPD_SWITCH(xname, nid, mask) \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .info = cxt_eapd_info, \
- .get = cxt_eapd_get, \
- .put = cxt_eapd_put, \
- .private_value = nid | (mask<<16) }
-
-
-
-static int conexant_ch_mode_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_spec *spec = codec->spec;
- return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode,
- spec->num_channel_mode);
-}
-
-static int conexant_ch_mode_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_spec *spec = codec->spec;
- return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
- spec->num_channel_mode,
- spec->multiout.max_channels);
-}
-
-static int conexant_ch_mode_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_spec *spec = codec->spec;
- int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
- spec->num_channel_mode,
- &spec->multiout.max_channels);
- return err;
-}
-
-#define CXT_PIN_MODE(xname, nid, dir) \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .info = conexant_ch_mode_info, \
- .get = conexant_ch_mode_get, \
- .put = conexant_ch_mode_put, \
- .private_value = nid | (dir<<16) }
-
-#endif /* CONFIG_SND_DEBUG */
-
-/* Conexant 5045 specific */
-
-static const hda_nid_t cxt5045_dac_nids[1] = { 0x19 };
-static const hda_nid_t cxt5045_adc_nids[1] = { 0x1a };
-static const hda_nid_t cxt5045_capsrc_nids[1] = { 0x1a };
-#define CXT5045_SPDIF_OUT 0x18
-
-static const struct hda_channel_mode cxt5045_modes[1] = {
- { 2, NULL },
-};
-
-static const struct hda_input_mux cxt5045_capture_source = {
- .num_items = 2,
- .items = {
- { "Internal Mic", 0x1 },
- { "Mic", 0x2 },
- }
-};
-
-static const struct hda_input_mux cxt5045_capture_source_benq = {
- .num_items = 4,
- .items = {
- { "Internal Mic", 0x1 },
- { "Mic", 0x2 },
- { "Line", 0x3 },
- { "Mixer", 0x0 },
- }
-};
-
-/* turn on/off EAPD (+ mute HP) as a master switch */
-static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_spec *spec = codec->spec;
- unsigned int bits;
-
- if (!cxt_eapd_put(kcontrol, ucontrol))
- return 0;
-
- /* toggle internal speakers mute depending of presence of
- * the headphone jack
- */
- bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE;
- snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
-
- bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE;
- snd_hda_codec_amp_stereo(codec, 0x11, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
- return 1;
-}
-
-/* bind volumes of both NID 0x10 and 0x11 */
-static const struct hda_bind_ctls cxt5045_hp_bind_master_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x10, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-/* toggle input of built-in and mic jack appropriately */
-static void cxt5045_hp_automic(struct hda_codec *codec)
-{
- static const struct hda_verb mic_jack_on[] = {
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- {}
- };
- static const struct hda_verb mic_jack_off[] = {
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- {}
- };
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x12);
- if (present)
- snd_hda_sequence_write(codec, mic_jack_on);
- else
- snd_hda_sequence_write(codec, mic_jack_off);
-}
-
-
-/* mute internal speaker if HP is plugged */
-static void cxt5045_hp_automute(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- unsigned int bits;
-
- spec->hp_present = snd_hda_jack_detect(codec, 0x11);
-
- bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
-}
-
-/* unsolicited event for HP jack sensing */
-static void cxt5045_hp_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- res >>= 26;
- switch (res) {
- case CONEXANT_HP_EVENT:
- cxt5045_hp_automute(codec);
- break;
- case CONEXANT_MIC_EVENT:
- cxt5045_hp_automic(codec);
- break;
-
- }
-}
-
-static const struct snd_kcontrol_new cxt5045_mixers[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x17, 0x2, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x17, 0x2, HDA_INPUT),
- HDA_BIND_VOL("Master Playback Volume", &cxt5045_hp_bind_master_vol),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = cxt_eapd_info,
- .get = cxt_eapd_get,
- .put = cxt5045_hp_master_sw_put,
- .private_value = 0x10,
- },
-
- {}
-};
-
-static const struct snd_kcontrol_new cxt5045_benq_mixers[] = {
- HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT),
-
- {}
-};
-
-static const struct hda_verb cxt5045_init_verbs[] = {
- /* Line in, Mic */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
- /* HP, Amp */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x10, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Record selector: Internal mic */
- {0x1a, AC_VERB_SET_CONNECT_SEL,0x1},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE,
- AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17},
- /* SPDIF route: PCM */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- { 0x13, AC_VERB_SET_CONNECT_SEL, 0x0 },
- /* EAPD */
- {0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2 }, /* default on */
- { } /* end */
-};
-
-static const struct hda_verb cxt5045_benq_init_verbs[] = {
- /* Internal Mic, Mic */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
- /* Line In,HP, Amp */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x10, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Record selector: Internal mic */
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE,
- AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17},
- /* SPDIF route: PCM */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* EAPD */
- {0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
- { } /* end */
-};
-
-static const struct hda_verb cxt5045_hp_sense_init_verbs[] = {
- /* pin sensing on HP jack */
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
- { } /* end */
-};
-
-static const struct hda_verb cxt5045_mic_sense_init_verbs[] = {
- /* pin sensing on HP jack */
- {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
- { } /* end */
-};
-
-#ifdef CONFIG_SND_DEBUG
-/* Test configuration for debugging, modelled after the ALC260 test
- * configuration.
- */
-static const struct hda_input_mux cxt5045_test_capture_source = {
- .num_items = 5,
- .items = {
- { "MIXER", 0x0 },
- { "MIC1 pin", 0x1 },
- { "LINE1 pin", 0x2 },
- { "HP-OUT pin", 0x3 },
- { "CD pin", 0x4 },
- },
-};
-
-static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
-
- /* Output controls */
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x10, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("HP-OUT Playback Volume", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("HP-OUT Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("LINE1 Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-
- /* Modes for retasking pin widgets */
- CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT),
- CXT_PIN_MODE("LINE1 pin mode", 0x12, CXT_PIN_DIR_INOUT),
-
- /* EAPD Switch Control */
- CXT_EAPD_SWITCH("External Amplifier", 0x10, 0x0),
-
- /* Loopback mixer controls */
-
- HDA_CODEC_VOLUME("PCM Volume", 0x17, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Switch", 0x17, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("MIC1 pin Volume", 0x17, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("MIC1 pin Switch", 0x17, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("LINE1 pin Volume", 0x17, 0x2, HDA_INPUT),
- HDA_CODEC_MUTE("LINE1 pin Switch", 0x17, 0x2, HDA_INPUT),
- HDA_CODEC_VOLUME("HP-OUT pin Volume", 0x17, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("HP-OUT pin Switch", 0x17, 0x3, HDA_INPUT),
- HDA_CODEC_VOLUME("CD pin Volume", 0x17, 0x4, HDA_INPUT),
- HDA_CODEC_MUTE("CD pin Switch", 0x17, 0x4, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Input Source",
- .info = conexant_mux_enum_info,
- .get = conexant_mux_enum_get,
- .put = conexant_mux_enum_put,
- },
- /* Audio input controls */
- HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_verb cxt5045_test_init_verbs[] = {
- /* Set connections */
- { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
- { 0x11, AC_VERB_SET_CONNECT_SEL, 0x0 },
- { 0x12, AC_VERB_SET_CONNECT_SEL, 0x0 },
- /* Enable retasking pins as output, initially without power amp */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- /* Disable digital (SPDIF) pins initially, but users can enable
- * them via a mixer switch. In the case of SPDIF-out, this initverb
- * payload also sets the generation to 0, output to be in "consumer"
- * PCM format, copyright asserted, no pre-emphasis and no validity
- * control.
- */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0},
-
- /* Unmute retasking pin widget output buffers since the default
- * state appears to be output. As the pin mode is changed by the
- * user the pin mode control will take care of enabling the pin's
- * input/output buffers as needed.
- */
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Mute capture amp left and right */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-
- /* Set ADC connection select to match default mixer setting (mic1
- * pin)
- */
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x01},
-
- /* Mute all inputs to mixer widget (even unconnected ones) */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Mic1 pin */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* Line pin */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* HP pin */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
-
- { }
-};
-#endif
-
-
-/* initialize jack-sensing, too */
-static int cxt5045_init(struct hda_codec *codec)
-{
- conexant_init(codec);
- cxt5045_hp_automute(codec);
- return 0;
-}
-
-
-enum {
- CXT5045_LAPTOP_HPSENSE,
- CXT5045_LAPTOP_MICSENSE,
- CXT5045_LAPTOP_HPMICSENSE,
- CXT5045_BENQ,
-#ifdef CONFIG_SND_DEBUG
- CXT5045_TEST,
-#endif
- CXT5045_AUTO,
- CXT5045_MODELS
-};
-
-static const char * const cxt5045_models[CXT5045_MODELS] = {
- [CXT5045_LAPTOP_HPSENSE] = "laptop-hpsense",
- [CXT5045_LAPTOP_MICSENSE] = "laptop-micsense",
- [CXT5045_LAPTOP_HPMICSENSE] = "laptop-hpmicsense",
- [CXT5045_BENQ] = "benq",
-#ifdef CONFIG_SND_DEBUG
- [CXT5045_TEST] = "test",
-#endif
- [CXT5045_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk cxt5045_cfg_tbl[] = {
- SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ),
- SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE),
- SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP_HPMICSENSE),
- SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505",
- CXT5045_LAPTOP_HPMICSENSE),
- SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE),
- SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE),
- SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE),
- SND_PCI_QUIRK_MASK(0x1631, 0xff00, 0xc100, "Packard Bell",
- CXT5045_LAPTOP_HPMICSENSE),
- SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP_HPSENSE),
- {}
-};
-
-static int patch_cxt5045(struct hda_codec *codec)
-{
- struct conexant_spec *spec;
- int board_config;
-
- board_config = snd_hda_check_board_config(codec, CXT5045_MODELS,
- cxt5045_models,
- cxt5045_cfg_tbl);
- if (board_config < 0)
- board_config = CXT5045_AUTO; /* model=auto as default */
- if (board_config == CXT5045_AUTO)
- return patch_conexant_auto(codec);
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (!spec)
- return -ENOMEM;
- codec->spec = spec;
- codec->single_adc_amp = 1;
-
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids);
- spec->multiout.dac_nids = cxt5045_dac_nids;
- spec->multiout.dig_out_nid = CXT5045_SPDIF_OUT;
- spec->num_adc_nids = 1;
- spec->adc_nids = cxt5045_adc_nids;
- spec->capsrc_nids = cxt5045_capsrc_nids;
- spec->input_mux = &cxt5045_capture_source;
- spec->num_mixers = 1;
- spec->mixers[0] = cxt5045_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = cxt5045_init_verbs;
- spec->spdif_route = 0;
- spec->num_channel_mode = ARRAY_SIZE(cxt5045_modes);
- spec->channel_mode = cxt5045_modes;
-
- set_beep_amp(spec, 0x16, 0, 1);
-
- codec->patch_ops = conexant_patch_ops;
-
- switch (board_config) {
- case CXT5045_LAPTOP_HPSENSE:
- codec->patch_ops.unsol_event = cxt5045_hp_unsol_event;
- spec->input_mux = &cxt5045_capture_source;
- spec->num_init_verbs = 2;
- spec->init_verbs[1] = cxt5045_hp_sense_init_verbs;
- spec->mixers[0] = cxt5045_mixers;
- codec->patch_ops.init = cxt5045_init;
- break;
- case CXT5045_LAPTOP_MICSENSE:
- codec->patch_ops.unsol_event = cxt5045_hp_unsol_event;
- spec->input_mux = &cxt5045_capture_source;
- spec->num_init_verbs = 2;
- spec->init_verbs[1] = cxt5045_mic_sense_init_verbs;
- spec->mixers[0] = cxt5045_mixers;
- codec->patch_ops.init = cxt5045_init;
- break;
- default:
- case CXT5045_LAPTOP_HPMICSENSE:
- codec->patch_ops.unsol_event = cxt5045_hp_unsol_event;
- spec->input_mux = &cxt5045_capture_source;
- spec->num_init_verbs = 3;
- spec->init_verbs[1] = cxt5045_hp_sense_init_verbs;
- spec->init_verbs[2] = cxt5045_mic_sense_init_verbs;
- spec->mixers[0] = cxt5045_mixers;
- codec->patch_ops.init = cxt5045_init;
- break;
- case CXT5045_BENQ:
- codec->patch_ops.unsol_event = cxt5045_hp_unsol_event;
- spec->input_mux = &cxt5045_capture_source_benq;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = cxt5045_benq_init_verbs;
- spec->mixers[0] = cxt5045_mixers;
- spec->mixers[1] = cxt5045_benq_mixers;
- spec->num_mixers = 2;
- codec->patch_ops.init = cxt5045_init;
- break;
-#ifdef CONFIG_SND_DEBUG
- case CXT5045_TEST:
- spec->input_mux = &cxt5045_test_capture_source;
- spec->mixers[0] = cxt5045_test_mixer;
- spec->init_verbs[0] = cxt5045_test_init_verbs;
- break;
-
-#endif
- }
-
- switch (codec->subsystem_id >> 16) {
- case 0x103c:
- case 0x1631:
- case 0x1734:
- case 0x17aa:
- /* HP, Packard Bell, Fujitsu-Siemens & Lenovo laptops have
- * really bad sound over 0dB on NID 0x17. Fix max PCM level to
- * 0 dB (originally it has 0x2b steps with 0dB offset 0x14)
- */
- snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT,
- (0x14 << AC_AMPCAP_OFFSET_SHIFT) |
- (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) |
- (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
- (1 << AC_AMPCAP_MUTE_SHIFT));
- break;
- }
-
- if (spec->beep_amp)
- snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
-
- return 0;
-}
-
-
-/* Conexant 5047 specific */
-#define CXT5047_SPDIF_OUT 0x11
-
-static const hda_nid_t cxt5047_dac_nids[1] = { 0x10 }; /* 0x1c */
-static const hda_nid_t cxt5047_adc_nids[1] = { 0x12 };
-static const hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a };
-
-static const struct hda_channel_mode cxt5047_modes[1] = {
- { 2, NULL },
-};
-
-static const struct hda_input_mux cxt5047_toshiba_capture_source = {
- .num_items = 2,
- .items = {
- { "ExtMic", 0x2 },
- { "Line-In", 0x1 },
- }
-};
-
-/* turn on/off EAPD (+ mute HP) as a master switch */
-static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_spec *spec = codec->spec;
- unsigned int bits;
-
- if (!cxt_eapd_put(kcontrol, ucontrol))
- return 0;
-
- /* toggle internal speakers mute depending of presence of
- * the headphone jack
- */
- bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE;
- /* NOTE: Conexat codec needs the index for *OUTPUT* amp of
- * pin widgets unlike other codecs. In this case, we need to
- * set index 0x01 for the volume from the mixer amp 0x19.
- */
- snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01,
- HDA_AMP_MUTE, bits);
- bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE;
- snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
- return 1;
-}
-
-/* mute internal speaker if HP is plugged */
-static void cxt5047_hp_automute(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- unsigned int bits;
-
- spec->hp_present = snd_hda_jack_detect(codec, 0x13);
-
- bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0;
- /* See the note in cxt5047_hp_master_sw_put */
- snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01,
- HDA_AMP_MUTE, bits);
-}
-
-/* toggle input of built-in and mic jack appropriately */
-static void cxt5047_hp_automic(struct hda_codec *codec)
-{
- static const struct hda_verb mic_jack_on[] = {
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {}
- };
- static const struct hda_verb mic_jack_off[] = {
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {}
- };
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x15);
- 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 cxt5047_hp_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case CONEXANT_HP_EVENT:
- cxt5047_hp_automute(codec);
- break;
- case CONEXANT_MIC_EVENT:
- cxt5047_hp_automic(codec);
- break;
- }
-}
-
-static const struct snd_kcontrol_new cxt5047_base_mixers[] = {
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x19, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x19, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x1a, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT),
- HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = cxt_eapd_info,
- .get = cxt_eapd_get,
- .put = cxt5047_hp_master_sw_put,
- .private_value = 0x13,
- },
-
- {}
-};
-
-static const struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = {
- /* See the note in cxt5047_hp_master_sw_put */
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT),
- {}
-};
-
-static const struct snd_kcontrol_new cxt5047_hp_only_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x13, 0x00, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct hda_verb cxt5047_init_verbs[] = {
- /* Line in, Mic, Built-in Mic */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 },
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 },
- /* HP, Speaker */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, /* mixer(0x19) */
- {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mixer(0x19) */
- /* Record selector: Mic */
- {0x12, AC_VERB_SET_CONNECT_SEL,0x03},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE,
- AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17},
- {0x1A, AC_VERB_SET_CONNECT_SEL,0x02},
- {0x1A, AC_VERB_SET_AMP_GAIN_MUTE,
- AC_AMP_SET_OUTPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x00},
- {0x1A, AC_VERB_SET_AMP_GAIN_MUTE,
- AC_AMP_SET_OUTPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x03},
- /* SPDIF route: PCM */
- { 0x18, AC_VERB_SET_CONNECT_SEL, 0x0 },
- /* Enable unsolicited events */
- {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
- { } /* end */
-};
-
-/* configuration for Toshiba Laptops */
-static const struct hda_verb cxt5047_toshiba_init_verbs[] = {
- {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */
- {}
-};
-
-/* Test configuration for debugging, modelled after the ALC260 test
- * configuration.
- */
-#ifdef CONFIG_SND_DEBUG
-static const struct hda_input_mux cxt5047_test_capture_source = {
- .num_items = 4,
- .items = {
- { "LINE1 pin", 0x0 },
- { "MIC1 pin", 0x1 },
- { "MIC2 pin", 0x2 },
- { "CD pin", 0x3 },
- },
-};
-
-static const struct snd_kcontrol_new cxt5047_test_mixer[] = {
-
- /* Output only controls */
- HDA_CODEC_VOLUME("OutAmp-1 Volume", 0x10, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("OutAmp-1 Switch", 0x10,0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("OutAmp-2 Volume", 0x1c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("OutAmp-2 Switch", 0x1c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("HeadPhone Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("HeadPhone Playback Switch", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line1-Out Playback Volume", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Line1-Out Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line2-Out Playback Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Line2-Out Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-
- /* Modes for retasking pin widgets */
- CXT_PIN_MODE("LINE1 pin mode", 0x14, CXT_PIN_DIR_INOUT),
- CXT_PIN_MODE("MIC1 pin mode", 0x15, CXT_PIN_DIR_INOUT),
-
- /* EAPD Switch Control */
- CXT_EAPD_SWITCH("External Amplifier", 0x13, 0x0),
-
- /* Loopback mixer controls */
- HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x12, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("MIC1 Playback Switch", 0x12, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x12, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("MIC2 Playback Switch", 0x12, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("LINE Playback Volume", 0x12, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("LINE Playback Switch", 0x12, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x12, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x12, 0x04, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Capture-1 Volume", 0x19, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture-1 Switch", 0x19, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture-2 Volume", 0x19, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Capture-2 Switch", 0x19, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture-3 Volume", 0x19, 0x2, HDA_INPUT),
- HDA_CODEC_MUTE("Capture-3 Switch", 0x19, 0x2, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture-4 Volume", 0x19, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("Capture-4 Switch", 0x19, 0x3, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Input Source",
- .info = conexant_mux_enum_info,
- .get = conexant_mux_enum_get,
- .put = conexant_mux_enum_put,
- },
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x1a, 0x0, HDA_OUTPUT),
-
- { } /* end */
-};
-
-static const struct hda_verb cxt5047_test_init_verbs[] = {
- /* Enable retasking pins as output, initially without power amp */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- /* Disable digital (SPDIF) pins initially, but users can enable
- * them via a mixer switch. In the case of SPDIF-out, this initverb
- * payload also sets the generation to 0, output to be in "consumer"
- * PCM format, copyright asserted, no pre-emphasis and no validity
- * control.
- */
- {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0},
-
- /* Ensure mic1, mic2, line1 pin widgets take input from the
- * OUT1 sum bus when acting as an output.
- */
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* Start with output sum widgets muted and their output gains at min */
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
- /* Unmute retasking pin widget output buffers since the default
- * state appears to be output. As the pin mode is changed by the
- * user the pin mode control will take care of enabling the pin's
- * input/output buffers as needed.
- */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Mute capture amp left and right */
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-
- /* Set ADC connection select to match default mixer setting (mic1
- * pin)
- */
- {0x12, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Mute all inputs to mixer widget (even unconnected ones) */
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
-
- { }
-};
-#endif
-
-
-/* initialize jack-sensing, too */
-static int cxt5047_hp_init(struct hda_codec *codec)
-{
- conexant_init(codec);
- cxt5047_hp_automute(codec);
- return 0;
-}
-
-
-enum {
- CXT5047_LAPTOP, /* Laptops w/o EAPD support */
- CXT5047_LAPTOP_HP, /* Some HP laptops */
- CXT5047_LAPTOP_EAPD, /* Laptops with EAPD support */
-#ifdef CONFIG_SND_DEBUG
- CXT5047_TEST,
-#endif
- CXT5047_AUTO,
- CXT5047_MODELS
-};
-
-static const char * const cxt5047_models[CXT5047_MODELS] = {
- [CXT5047_LAPTOP] = "laptop",
- [CXT5047_LAPTOP_HP] = "laptop-hp",
- [CXT5047_LAPTOP_EAPD] = "laptop-eapd",
-#ifdef CONFIG_SND_DEBUG
- [CXT5047_TEST] = "test",
-#endif
- [CXT5047_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk cxt5047_cfg_tbl[] = {
- SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP),
- SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series",
- CXT5047_LAPTOP),
- SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD),
- {}
-};
-
-static int patch_cxt5047(struct hda_codec *codec)
-{
- struct conexant_spec *spec;
- int board_config;
-
- board_config = snd_hda_check_board_config(codec, CXT5047_MODELS,
- cxt5047_models,
- cxt5047_cfg_tbl);
- if (board_config < 0)
- board_config = CXT5047_AUTO; /* model=auto as default */
- if (board_config == CXT5047_AUTO)
- return patch_conexant_auto(codec);
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (!spec)
- return -ENOMEM;
- codec->spec = spec;
- codec->pin_amp_workaround = 1;
-
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = ARRAY_SIZE(cxt5047_dac_nids);
- spec->multiout.dac_nids = cxt5047_dac_nids;
- spec->multiout.dig_out_nid = CXT5047_SPDIF_OUT;
- spec->num_adc_nids = 1;
- spec->adc_nids = cxt5047_adc_nids;
- spec->capsrc_nids = cxt5047_capsrc_nids;
- spec->num_mixers = 1;
- spec->mixers[0] = cxt5047_base_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = cxt5047_init_verbs;
- spec->spdif_route = 0;
- spec->num_channel_mode = ARRAY_SIZE(cxt5047_modes),
- spec->channel_mode = cxt5047_modes,
-
- codec->patch_ops = conexant_patch_ops;
-
- switch (board_config) {
- case CXT5047_LAPTOP:
- spec->num_mixers = 2;
- spec->mixers[1] = cxt5047_hp_spk_mixers;
- codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
- break;
- case CXT5047_LAPTOP_HP:
- spec->num_mixers = 2;
- spec->mixers[1] = cxt5047_hp_only_mixers;
- codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
- codec->patch_ops.init = cxt5047_hp_init;
- break;
- case CXT5047_LAPTOP_EAPD:
- spec->input_mux = &cxt5047_toshiba_capture_source;
- spec->num_mixers = 2;
- spec->mixers[1] = cxt5047_hp_spk_mixers;
- spec->num_init_verbs = 2;
- spec->init_verbs[1] = cxt5047_toshiba_init_verbs;
- codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
- break;
-#ifdef CONFIG_SND_DEBUG
- case CXT5047_TEST:
- spec->input_mux = &cxt5047_test_capture_source;
- spec->mixers[0] = cxt5047_test_mixer;
- spec->init_verbs[0] = cxt5047_test_init_verbs;
- codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
-#endif
- }
- spec->vmaster_nid = 0x13;
-
- switch (codec->subsystem_id >> 16) {
- case 0x103c:
- /* HP laptops have really bad sound over 0 dB on NID 0x10.
- * Fix max PCM level to 0 dB (originally it has 0x1e steps
- * with 0 dB offset 0x17)
- */
- snd_hda_override_amp_caps(codec, 0x10, 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;
- }
-
- return 0;
-}
-
-/* Conexant 5051 specific */
-static const hda_nid_t cxt5051_dac_nids[1] = { 0x10 };
-static const hda_nid_t cxt5051_adc_nids[2] = { 0x14, 0x15 };
-
-static const struct hda_channel_mode cxt5051_modes[1] = {
- { 2, NULL },
-};
-
-static void cxt5051_update_speaker(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- unsigned int pinctl;
- /* headphone pin */
- pinctl = (spec->hp_present && spec->cur_eapd) ? PIN_HP : 0;
- snd_hda_set_pin_ctl(codec, 0x16, pinctl);
- /* speaker pin */
- pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
- snd_hda_set_pin_ctl(codec, 0x1a, pinctl);
- /* on ideapad there is an additional speaker (subwoofer) to mute */
- if (spec->ideapad)
- snd_hda_set_pin_ctl(codec, 0x1b, pinctl);
-}
-
-/* turn on/off EAPD (+ mute HP) as a master switch */
-static int cxt5051_hp_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-
- if (!cxt_eapd_put(kcontrol, ucontrol))
- return 0;
- cxt5051_update_speaker(codec);
- return 1;
-}
-
-/* toggle input of built-in and mic jack appropriately */
-static void cxt5051_portb_automic(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- unsigned int present;
-
- if (!(spec->auto_mic & AUTO_MIC_PORTB))
- return;
- present = snd_hda_jack_detect(codec, 0x17);
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_CONNECT_SEL,
- present ? 0x01 : 0x00);
-}
-
-/* switch the current ADC according to the jack state */
-static void cxt5051_portc_automic(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- unsigned int present;
- hda_nid_t new_adc;
-
- if (!(spec->auto_mic & AUTO_MIC_PORTC))
- return;
- present = snd_hda_jack_detect(codec, 0x18);
- if (present)
- spec->cur_adc_idx = 1;
- else
- spec->cur_adc_idx = 0;
- new_adc = spec->adc_nids[spec->cur_adc_idx];
- if (spec->cur_adc && spec->cur_adc != new_adc) {
- /* stream is running, let's swap the current ADC */
- __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
- spec->cur_adc = new_adc;
- snd_hda_codec_setup_stream(codec, new_adc,
- spec->cur_adc_stream_tag, 0,
- spec->cur_adc_format);
- }
-}
-
-/* mute internal speaker if HP is plugged */
-static void cxt5051_hp_automute(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
-
- spec->hp_present = snd_hda_jack_detect(codec, 0x16);
- cxt5051_update_speaker(codec);
-}
-
-/* unsolicited event for HP jack sensing */
-static void cxt5051_hp_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case CONEXANT_HP_EVENT:
- cxt5051_hp_automute(codec);
- break;
- case CXT5051_PORTB_EVENT:
- cxt5051_portb_automic(codec);
- break;
- case CXT5051_PORTC_EVENT:
- cxt5051_portc_automic(codec);
- break;
- }
-}
-
-static const struct snd_kcontrol_new cxt5051_playback_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = cxt_eapd_info,
- .get = cxt_eapd_get,
- .put = cxt5051_hp_master_sw_put,
- .private_value = 0x1a,
- },
- {}
-};
-
-static const struct snd_kcontrol_new cxt5051_capture_mixers[] = {
- HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Dock Mic Volume", 0x15, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Dock Mic Switch", 0x15, 0x00, HDA_INPUT),
- {}
-};
-
-static const struct snd_kcontrol_new cxt5051_hp_mixers[] = {
- HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Volume", 0x15, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Switch", 0x15, 0x00, HDA_INPUT),
- {}
-};
-
-static const struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x14, 0x00, HDA_INPUT),
- {}
-};
-
-static const struct snd_kcontrol_new cxt5051_f700_mixers[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x14, 0x01, HDA_INPUT),
- {}
-};
-
-static const struct snd_kcontrol_new cxt5051_toshiba_mixers[] = {
- HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT),
- {}
-};
-
-static const struct hda_verb cxt5051_init_verbs[] = {
- /* Line in, Mic */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
- /* SPK */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* HP, Amp */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* DAC1 */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Record selector: Internal mic */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44},
- /* SPDIF route: PCM */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* EAPD */
- {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
- {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
- { } /* end */
-};
-
-static const struct hda_verb cxt5051_hp_dv6736_init_verbs[] = {
- /* Line in, Mic */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0},
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0},
- /* SPK */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* HP, Amp */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* DAC1 */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Record selector: Internal mic */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x1},
- /* SPDIF route: PCM */
- {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* EAPD */
- {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
- {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
- { } /* end */
-};
-
-static const struct hda_verb cxt5051_f700_init_verbs[] = {
- /* Line in, Mic */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0},
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0},
- /* SPK */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* HP, Amp */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* DAC1 */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Record selector: Internal mic */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x1},
- /* SPDIF route: PCM */
- {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* EAPD */
- {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
- {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
- { } /* end */
-};
-
-static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid,
- unsigned int event)
-{
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | event);
-}
-
-static const struct hda_verb cxt5051_ideapad_init_verbs[] = {
- /* Subwoofer */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- { } /* end */
-};
-
-/* initialize jack-sensing, too */
-static int cxt5051_init(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
-
- conexant_init(codec);
-
- if (spec->auto_mic & AUTO_MIC_PORTB)
- cxt5051_init_mic_port(codec, 0x17, CXT5051_PORTB_EVENT);
- if (spec->auto_mic & AUTO_MIC_PORTC)
- cxt5051_init_mic_port(codec, 0x18, CXT5051_PORTC_EVENT);
-
- if (codec->patch_ops.unsol_event) {
- cxt5051_hp_automute(codec);
- cxt5051_portb_automic(codec);
- cxt5051_portc_automic(codec);
- }
- return 0;
-}
-
-
-enum {
- CXT5051_LAPTOP, /* Laptops w/ EAPD support */
- CXT5051_HP, /* no docking */
- CXT5051_HP_DV6736, /* HP without mic switch */
- CXT5051_F700, /* HP Compaq Presario F700 */
- CXT5051_TOSHIBA, /* Toshiba M300 & co */
- CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */
- CXT5051_AUTO, /* auto-parser */
- CXT5051_MODELS
-};
-
-static const char *const cxt5051_models[CXT5051_MODELS] = {
- [CXT5051_LAPTOP] = "laptop",
- [CXT5051_HP] = "hp",
- [CXT5051_HP_DV6736] = "hp-dv6736",
- [CXT5051_F700] = "hp-700",
- [CXT5051_TOSHIBA] = "toshiba",
- [CXT5051_IDEAPAD] = "ideapad",
- [CXT5051_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk cxt5051_cfg_tbl[] = {
- SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736),
- SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP),
- SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700),
- SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba M30x", CXT5051_TOSHIBA),
- SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
- CXT5051_LAPTOP),
- SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),
- SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD),
- {}
-};
-
-static int patch_cxt5051(struct hda_codec *codec)
-{
- struct conexant_spec *spec;
- int board_config;
-
- board_config = snd_hda_check_board_config(codec, CXT5051_MODELS,
- cxt5051_models,
- cxt5051_cfg_tbl);
- if (board_config < 0)
- board_config = CXT5051_AUTO; /* model=auto as default */
- if (board_config == CXT5051_AUTO)
- return patch_conexant_auto(codec);
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (!spec)
- return -ENOMEM;
- codec->spec = spec;
- codec->pin_amp_workaround = 1;
-
- codec->patch_ops = conexant_patch_ops;
- codec->patch_ops.init = cxt5051_init;
-
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = ARRAY_SIZE(cxt5051_dac_nids);
- spec->multiout.dac_nids = cxt5051_dac_nids;
- spec->multiout.dig_out_nid = CXT5051_SPDIF_OUT;
- spec->num_adc_nids = 1; /* not 2; via auto-mic switch */
- spec->adc_nids = cxt5051_adc_nids;
- spec->num_mixers = 2;
- spec->mixers[0] = cxt5051_capture_mixers;
- spec->mixers[1] = cxt5051_playback_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = cxt5051_init_verbs;
- spec->spdif_route = 0;
- spec->num_channel_mode = ARRAY_SIZE(cxt5051_modes);
- spec->channel_mode = cxt5051_modes;
- spec->cur_adc = 0;
- spec->cur_adc_idx = 0;
-
- set_beep_amp(spec, 0x13, 0, HDA_OUTPUT);
-
- codec->patch_ops.unsol_event = cxt5051_hp_unsol_event;
-
- spec->auto_mic = AUTO_MIC_PORTB | AUTO_MIC_PORTC;
- switch (board_config) {
- case CXT5051_HP:
- spec->mixers[0] = cxt5051_hp_mixers;
- break;
- case CXT5051_HP_DV6736:
- spec->init_verbs[0] = cxt5051_hp_dv6736_init_verbs;
- spec->mixers[0] = cxt5051_hp_dv6736_mixers;
- spec->auto_mic = 0;
- break;
- case CXT5051_F700:
- spec->init_verbs[0] = cxt5051_f700_init_verbs;
- spec->mixers[0] = cxt5051_f700_mixers;
- spec->auto_mic = 0;
- break;
- case CXT5051_TOSHIBA:
- spec->mixers[0] = cxt5051_toshiba_mixers;
- spec->auto_mic = AUTO_MIC_PORTB;
- break;
- case CXT5051_IDEAPAD:
- spec->init_verbs[spec->num_init_verbs++] =
- cxt5051_ideapad_init_verbs;
- spec->ideapad = 1;
- break;
- }
-
- if (spec->beep_amp)
- snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
-
- return 0;
-}
-
-/* Conexant 5066 specific */
-
-static const hda_nid_t cxt5066_dac_nids[1] = { 0x10 };
-static const hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
-static const hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
-static const hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 };
-
-static const struct hda_channel_mode cxt5066_modes[1] = {
- { 2, NULL },
-};
-
-#define HP_PRESENT_PORT_A (1 << 0)
-#define HP_PRESENT_PORT_D (1 << 1)
-#define hp_port_a_present(spec) ((spec)->hp_present & HP_PRESENT_PORT_A)
-#define hp_port_d_present(spec) ((spec)->hp_present & HP_PRESENT_PORT_D)
-
-static void cxt5066_update_speaker(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- unsigned int pinctl;
-
- codec_dbg(codec,
- "CXT5066: update speaker, hp_present=%d, cur_eapd=%d\n",
- spec->hp_present, spec->cur_eapd);
-
- /* Port A (HP) */
- pinctl = (hp_port_a_present(spec) && spec->cur_eapd) ? PIN_HP : 0;
- snd_hda_set_pin_ctl(codec, 0x19, pinctl);
-
- /* Port D (HP/LO) */
- pinctl = spec->cur_eapd ? spec->port_d_mode : 0;
- if (spec->dell_automute || spec->thinkpad) {
- /* Mute if Port A is connected */
- if (hp_port_a_present(spec))
- pinctl = 0;
- } else {
- /* Thinkpad/Dell doesn't give pin-D status */
- if (!hp_port_d_present(spec))
- pinctl = 0;
- }
- snd_hda_set_pin_ctl(codec, 0x1c, pinctl);
-
- /* CLASS_D AMP */
- pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
- snd_hda_set_pin_ctl(codec, 0x1f, pinctl);
-}
-
-/* turn on/off EAPD (+ mute HP) as a master switch */
-static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-
- if (!cxt_eapd_put(kcontrol, ucontrol))
- return 0;
-
- cxt5066_update_speaker(codec);
- return 1;
-}
-
-/* toggle input of built-in digital mic and mic jack appropriately */
-static void cxt5066_vostro_automic(struct hda_codec *codec)
-{
- unsigned int present;
-
- struct hda_verb ext_mic_present[] = {
- /* enable external mic, port B */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-
- /* switch to external mic input */
- {0x17, AC_VERB_SET_CONNECT_SEL, 0},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* disable internal digital mic */
- {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {}
- };
- static const struct hda_verb ext_mic_absent[] = {
- /* enable internal mic, port C */
- {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- /* switch to internal mic input */
- {0x14, AC_VERB_SET_CONNECT_SEL, 2},
-
- /* disable external mic, port B */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {}
- };
-
- present = snd_hda_jack_detect(codec, 0x1a);
- if (present) {
- codec_dbg(codec, "CXT5066: external microphone detected\n");
- snd_hda_sequence_write(codec, ext_mic_present);
- } else {
- codec_dbg(codec, "CXT5066: external microphone absent\n");
- snd_hda_sequence_write(codec, ext_mic_absent);
- }
-}
-
-/* toggle input of built-in digital mic and mic jack appropriately */
-static void cxt5066_ideapad_automic(struct hda_codec *codec)
-{
- unsigned int present;
-
- struct hda_verb ext_mic_present[] = {
- {0x14, AC_VERB_SET_CONNECT_SEL, 0},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {}
- };
- static const struct hda_verb ext_mic_absent[] = {
- {0x14, AC_VERB_SET_CONNECT_SEL, 2},
- {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {}
- };
-
- present = snd_hda_jack_detect(codec, 0x1b);
- if (present) {
- codec_dbg(codec, "CXT5066: external microphone detected\n");
- snd_hda_sequence_write(codec, ext_mic_present);
- } else {
- codec_dbg(codec, "CXT5066: external microphone absent\n");
- snd_hda_sequence_write(codec, ext_mic_absent);
- }
-}
-
-
-/* toggle input of built-in digital mic and mic jack appropriately */
-static void cxt5066_asus_automic(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x1b);
- codec_dbg(codec, "CXT5066: external microphone present=%d\n", present);
- snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL,
- present ? 1 : 0);
-}
-
-
-/* toggle input of built-in digital mic and mic jack appropriately */
-static void cxt5066_hp_laptop_automic(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x1b);
- codec_dbg(codec, "CXT5066: external microphone present=%d\n", present);
- snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL,
- present ? 1 : 3);
-}
-
-
-/* toggle input of built-in digital mic and mic jack appropriately
- order is: external mic -> dock mic -> interal mic */
-static void cxt5066_thinkpad_automic(struct hda_codec *codec)
-{
- unsigned int ext_present, dock_present;
-
- static const struct hda_verb ext_mic_present[] = {
- {0x14, AC_VERB_SET_CONNECT_SEL, 0},
- {0x17, AC_VERB_SET_CONNECT_SEL, 1},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {}
- };
- static const struct hda_verb dock_mic_present[] = {
- {0x14, AC_VERB_SET_CONNECT_SEL, 0},
- {0x17, AC_VERB_SET_CONNECT_SEL, 0},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {}
- };
- static const struct hda_verb ext_mic_absent[] = {
- {0x14, AC_VERB_SET_CONNECT_SEL, 2},
- {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {}
- };
-
- ext_present = snd_hda_jack_detect(codec, 0x1b);
- dock_present = snd_hda_jack_detect(codec, 0x1a);
- if (ext_present) {
- codec_dbg(codec, "CXT5066: external microphone detected\n");
- snd_hda_sequence_write(codec, ext_mic_present);
- } else if (dock_present) {
- codec_dbg(codec, "CXT5066: dock microphone detected\n");
- snd_hda_sequence_write(codec, dock_mic_present);
- } else {
- codec_dbg(codec, "CXT5066: external microphone absent\n");
- snd_hda_sequence_write(codec, ext_mic_absent);
- }
-}
-
-/* mute internal speaker if HP is plugged */
-static void cxt5066_hp_automute(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- unsigned int portA, portD;
-
- /* Port A */
- portA = snd_hda_jack_detect(codec, 0x19);
-
- /* Port D */
- portD = snd_hda_jack_detect(codec, 0x1c);
-
- spec->hp_present = portA ? HP_PRESENT_PORT_A : 0;
- spec->hp_present |= portD ? HP_PRESENT_PORT_D : 0;
- codec_dbg(codec, "CXT5066: hp automute portA=%x portD=%x present=%d\n",
- portA, portD, spec->hp_present);
- cxt5066_update_speaker(codec);
-}
-
-/* Dispatch the right mic autoswitch function */
-static void cxt5066_automic(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
-
- if (spec->dell_vostro)
- cxt5066_vostro_automic(codec);
- else if (spec->ideapad)
- cxt5066_ideapad_automic(codec);
- else if (spec->thinkpad)
- cxt5066_thinkpad_automic(codec);
- else if (spec->hp_laptop)
- cxt5066_hp_laptop_automic(codec);
- else if (spec->asus)
- cxt5066_asus_automic(codec);
-}
-
-/* unsolicited event for jack sensing */
-static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- codec_dbg(codec, "CXT5066: unsol event %x (%x)\n", res, res >> 26);
- switch (res >> 26) {
- case CONEXANT_HP_EVENT:
- cxt5066_hp_automute(codec);
- break;
- case CONEXANT_MIC_EVENT:
- cxt5066_automic(codec);
- break;
- }
-}
-
-
-static const struct hda_input_mux cxt5066_analog_mic_boost = {
- .num_items = 5,
- .items = {
- { "0dB", 0 },
- { "10dB", 1 },
- { "20dB", 2 },
- { "30dB", 3 },
- { "40dB", 4 },
- },
-};
-
-static void cxt5066_set_mic_boost(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- snd_hda_codec_write_cache(codec, 0x17, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT |
- cxt5066_analog_mic_boost.items[spec->mic_boost].index);
- if (spec->ideapad || spec->thinkpad) {
- /* adjust the internal mic as well...it is not through 0x17 */
- snd_hda_codec_write_cache(codec, 0x23, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_INPUT |
- cxt5066_analog_mic_boost.
- items[spec->mic_boost].index);
- }
-}
-
-static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- return snd_hda_input_mux_info(&cxt5066_analog_mic_boost, uinfo);
-}
-
-static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_spec *spec = codec->spec;
- ucontrol->value.enumerated.item[0] = spec->mic_boost;
- return 0;
-}
-
-static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_spec *spec = codec->spec;
- const struct hda_input_mux *imux = &cxt5066_analog_mic_boost;
- unsigned int idx;
- idx = ucontrol->value.enumerated.item[0];
- if (idx >= imux->num_items)
- idx = imux->num_items - 1;
-
- spec->mic_boost = idx;
- cxt5066_set_mic_boost(codec);
- return 1;
-}
-
-static void conexant_check_dig_outs(struct hda_codec *codec,
- const hda_nid_t *dig_pins,
- int num_pins)
-{
- struct conexant_spec *spec = codec->spec;
- hda_nid_t *nid_loc = &spec->multiout.dig_out_nid;
- int i;
-
- for (i = 0; i < num_pins; i++, dig_pins++) {
- unsigned int cfg = snd_hda_codec_get_pincfg(codec, *dig_pins);
- if (get_defcfg_connect(cfg) == AC_JACK_PORT_NONE)
- continue;
- if (snd_hda_get_connections(codec, *dig_pins, nid_loc, 1) != 1)
- continue;
- }
-}
-
-static const struct hda_input_mux cxt5066_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic B", 0 },
- { "Mic C", 1 },
- { "Mic E", 2 },
- { "Mic F", 3 },
- },
-};
-
-static const struct hda_bind_ctls cxt5066_bind_capture_vol_others = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT),
- HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT),
- 0
- },
-};
-
-static const struct hda_bind_ctls cxt5066_bind_capture_sw_others = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT),
- HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT),
- 0
- },
-};
-
-static const struct snd_kcontrol_new cxt5066_mixer_master[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
- {}
-};
-
-static const struct snd_kcontrol_new cxt5066_mixers[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = cxt_eapd_info,
- .get = cxt_eapd_get,
- .put = cxt5066_hp_master_sw_put,
- .private_value = 0x1d,
- },
-
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Analog Mic Boost Capture Enum",
- .info = cxt5066_mic_boost_mux_enum_info,
- .get = cxt5066_mic_boost_mux_enum_get,
- .put = cxt5066_mic_boost_mux_enum_put,
- },
-
- HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others),
- HDA_BIND_SW("Capture Switch", &cxt5066_bind_capture_sw_others),
- {}
-};
-
-static const struct snd_kcontrol_new cxt5066_vostro_mixers[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Internal Mic Boost Capture Enum",
- .info = cxt5066_mic_boost_mux_enum_info,
- .get = cxt5066_mic_boost_mux_enum_get,
- .put = cxt5066_mic_boost_mux_enum_put,
- .private_value = 0x23 | 0x100,
- },
- {}
-};
-
-static const struct hda_verb cxt5066_init_verbs[] = {
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */
- {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */
-
- /* Speakers */
- {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
-
- /* HP, Amp */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
-
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
-
- /* DAC1 */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
- /* no digital microphone support yet */
- {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* Audio input selector */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3},
-
- /* SPDIF route: PCM */
- {0x20, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x22, AC_VERB_SET_CONNECT_SEL, 0x0},
-
- {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- /* EAPD */
- {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
-
- /* not handling these yet */
- {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
- {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
- {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
- {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
- {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
- {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
- {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
- { } /* end */
-};
-
-static const struct hda_verb cxt5066_init_verbs_vostro[] = {
- /* Port A: headphones */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
-
- /* Port B: external microphone */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* Port C: unused */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* Port D: unused */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* Port E: unused, but has primary EAPD */
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
-
- /* Port F: unused */
- {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* Port G: internal speakers */
- {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
-
- /* DAC1 */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* DAC2: unused */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-
- /* Digital microphone port */
- {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- /* Audio input selectors */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-
- /* Disable SPDIF */
- {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* enable unsolicited events for Port A and B */
- {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
- {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
- { } /* end */
-};
-
-static const struct hda_verb cxt5066_init_verbs_ideapad[] = {
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */
- {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */
-
- /* Speakers */
- {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
-
- /* HP, Amp */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
-
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
-
- /* DAC1 */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x14, AC_VERB_SET_CONNECT_SEL, 2}, /* default to internal mic */
-
- /* Audio input selector */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x2},
- {0x17, AC_VERB_SET_CONNECT_SEL, 1}, /* route ext mic */
-
- /* SPDIF route: PCM */
- {0x20, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x22, AC_VERB_SET_CONNECT_SEL, 0x0},
-
- {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- /* internal microphone */
- {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable internal mic */
-
- /* EAPD */
- {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
-
- {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
- { } /* end */
-};
-
-static const struct hda_verb cxt5066_init_verbs_thinkpad[] = {
- {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */
-
- /* Port G: internal speakers */
- {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
-
- /* Port A: HP, Amp */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
-
- /* Port B: Mic Dock */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* Port C: Mic */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* Port D: HP Dock, Amp */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
-
- /* DAC1 */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x14, AC_VERB_SET_CONNECT_SEL, 2}, /* default to internal mic */
-
- /* Audio input selector */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x2},
- {0x17, AC_VERB_SET_CONNECT_SEL, 1}, /* route ext mic */
-
- /* SPDIF route: PCM */
- {0x20, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x22, AC_VERB_SET_CONNECT_SEL, 0x0},
-
- {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- /* internal microphone */
- {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable internal mic */
-
- /* EAPD */
- {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
-
- /* enable unsolicited events for Port A, B, C and D */
- {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
- {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
- {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
- { } /* end */
-};
-
-static const struct hda_verb cxt5066_init_verbs_portd_lo[] = {
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- { } /* end */
-};
-
-
-static const struct hda_verb cxt5066_init_verbs_hp_laptop[] = {
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
- { } /* end */
-};
-
-/* initialize jack-sensing, too */
-static int cxt5066_init(struct hda_codec *codec)
-{
- codec_dbg(codec, "CXT5066: init\n");
- conexant_init(codec);
- if (codec->patch_ops.unsol_event) {
- cxt5066_hp_automute(codec);
- cxt5066_automic(codec);
- }
- cxt5066_set_mic_boost(codec);
- return 0;
-}
-
-enum {
- CXT5066_LAPTOP, /* Laptops w/ EAPD support */
- CXT5066_DELL_LAPTOP, /* Dell Laptop */
- CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */
- CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */
- CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */
- CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */
- CXT5066_HP_LAPTOP, /* HP Laptop */
- CXT5066_AUTO, /* BIOS auto-parser */
- CXT5066_MODELS
-};
-
-static const char * const cxt5066_models[CXT5066_MODELS] = {
- [CXT5066_LAPTOP] = "laptop",
- [CXT5066_DELL_LAPTOP] = "dell-laptop",
- [CXT5066_DELL_VOSTRO] = "dell-vostro",
- [CXT5066_IDEAPAD] = "ideapad",
- [CXT5066_THINKPAD] = "thinkpad",
- [CXT5066_ASUS] = "asus",
- [CXT5066_HP_LAPTOP] = "hp-laptop",
- [CXT5066_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
- SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD),
- SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO),
- SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD),
- SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO),
- SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
- SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD),
- SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
- SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS),
- SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
- SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS),
- SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD),
- SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
- CXT5066_LAPTOP),
- SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
- SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
- SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
- SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS),
- SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
- {}
-};
-
-static int patch_cxt5066(struct hda_codec *codec)
-{
- struct conexant_spec *spec;
- int board_config;
-
- board_config = snd_hda_check_board_config(codec, CXT5066_MODELS,
- cxt5066_models, cxt5066_cfg_tbl);
- if (board_config < 0)
- board_config = CXT5066_AUTO; /* model=auto as default */
- if (board_config == CXT5066_AUTO)
- return patch_conexant_auto(codec);
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (!spec)
- return -ENOMEM;
- codec->spec = spec;
-
- codec->patch_ops = conexant_patch_ops;
- codec->patch_ops.init = conexant_init;
-
- spec->dell_automute = 0;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids);
- spec->multiout.dac_nids = cxt5066_dac_nids;
- conexant_check_dig_outs(codec, cxt5066_digout_pin_nids,
- ARRAY_SIZE(cxt5066_digout_pin_nids));
- spec->num_adc_nids = 1;
- spec->adc_nids = cxt5066_adc_nids;
- spec->capsrc_nids = cxt5066_capsrc_nids;
- spec->input_mux = &cxt5066_capture_source;
-
- spec->port_d_mode = PIN_HP;
-
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = cxt5066_init_verbs;
- spec->num_channel_mode = ARRAY_SIZE(cxt5066_modes);
- spec->channel_mode = cxt5066_modes;
- spec->cur_adc = 0;
- spec->cur_adc_idx = 0;
-
- set_beep_amp(spec, 0x13, 0, HDA_OUTPUT);
-
- switch (board_config) {
- default:
- case CXT5066_LAPTOP:
- spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
- spec->mixers[spec->num_mixers++] = cxt5066_mixers;
- break;
- case CXT5066_DELL_LAPTOP:
- spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
- spec->mixers[spec->num_mixers++] = cxt5066_mixers;
-
- spec->port_d_mode = PIN_OUT;
- spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_portd_lo;
- spec->num_init_verbs++;
- spec->dell_automute = 1;
- break;
- case CXT5066_ASUS:
- case CXT5066_HP_LAPTOP:
- codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_unsol_event;
- spec->init_verbs[spec->num_init_verbs] =
- cxt5066_init_verbs_hp_laptop;
- spec->num_init_verbs++;
- spec->hp_laptop = board_config == CXT5066_HP_LAPTOP;
- spec->asus = board_config == CXT5066_ASUS;
- spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
- spec->mixers[spec->num_mixers++] = cxt5066_mixers;
- /* no S/PDIF out */
- if (board_config == CXT5066_HP_LAPTOP)
- spec->multiout.dig_out_nid = 0;
- /* input source automatically selected */
- spec->input_mux = NULL;
- spec->port_d_mode = 0;
- spec->mic_boost = 3; /* default 30dB gain */
- break;
-
- case CXT5066_DELL_VOSTRO:
- codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_unsol_event;
- spec->init_verbs[0] = cxt5066_init_verbs_vostro;
- spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
- spec->mixers[spec->num_mixers++] = cxt5066_mixers;
- spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers;
- spec->port_d_mode = 0;
- spec->dell_vostro = 1;
- spec->mic_boost = 3; /* default 30dB gain */
-
- /* no S/PDIF out */
- spec->multiout.dig_out_nid = 0;
-
- /* input source automatically selected */
- spec->input_mux = NULL;
- break;
- case CXT5066_IDEAPAD:
- codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_unsol_event;
- spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
- spec->mixers[spec->num_mixers++] = cxt5066_mixers;
- spec->init_verbs[0] = cxt5066_init_verbs_ideapad;
- spec->port_d_mode = 0;
- spec->ideapad = 1;
- spec->mic_boost = 2; /* default 20dB gain */
-
- /* no S/PDIF out */
- spec->multiout.dig_out_nid = 0;
-
- /* input source automatically selected */
- spec->input_mux = NULL;
- break;
- case CXT5066_THINKPAD:
- codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_unsol_event;
- spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
- spec->mixers[spec->num_mixers++] = cxt5066_mixers;
- spec->init_verbs[0] = cxt5066_init_verbs_thinkpad;
- spec->thinkpad = 1;
- spec->port_d_mode = PIN_OUT;
- spec->mic_boost = 2; /* default 20dB gain */
-
- /* no S/PDIF out */
- spec->multiout.dig_out_nid = 0;
-
- /* input source automatically selected */
- spec->input_mux = NULL;
- break;
- }
-
- if (spec->beep_amp)
- snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
-
- return 0;
-}
-
-#endif /* ENABLE_CXT_STATIC_QUIRKS */
-
-
/*
* Automatic parser for CX20641 & co
*/
@@ -3487,35 +881,28 @@ static int patch_conexant_auto(struct hda_codec *codec)
return err;
}
-#ifndef ENABLE_CXT_STATIC_QUIRKS
-#define patch_cxt5045 patch_conexant_auto
-#define patch_cxt5047 patch_conexant_auto
-#define patch_cxt5051 patch_conexant_auto
-#define patch_cxt5066 patch_conexant_auto
-#endif
-
/*
*/
static const struct hda_codec_preset snd_hda_preset_conexant[] = {
{ .id = 0x14f15045, .name = "CX20549 (Venice)",
- .patch = patch_cxt5045 },
+ .patch = patch_conexant_auto },
{ .id = 0x14f15047, .name = "CX20551 (Waikiki)",
- .patch = patch_cxt5047 },
+ .patch = patch_conexant_auto },
{ .id = 0x14f15051, .name = "CX20561 (Hermosa)",
- .patch = patch_cxt5051 },
+ .patch = patch_conexant_auto },
{ .id = 0x14f15066, .name = "CX20582 (Pebble)",
- .patch = patch_cxt5066 },
+ .patch = patch_conexant_auto },
{ .id = 0x14f15067, .name = "CX20583 (Pebble HSF)",
- .patch = patch_cxt5066 },
+ .patch = patch_conexant_auto },
{ .id = 0x14f15068, .name = "CX20584",
- .patch = patch_cxt5066 },
+ .patch = patch_conexant_auto },
{ .id = 0x14f15069, .name = "CX20585",
- .patch = patch_cxt5066 },
+ .patch = patch_conexant_auto },
{ .id = 0x14f1506c, .name = "CX20588",
- .patch = patch_cxt5066 },
+ .patch = patch_conexant_auto },
{ .id = 0x14f1506e, .name = "CX20590",
- .patch = patch_cxt5066 },
+ .patch = patch_conexant_auto },
{ .id = 0x14f15097, .name = "CX20631",
.patch = patch_conexant_auto },
{ .id = 0x14f15098, .name = "CX20632",
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index ba4ca52072ff..36badba2dcec 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -648,7 +648,8 @@ static int get_channel_allocation_order(int ca)
*
* TODO: it could select the wrong CA from multiple candidates.
*/
-static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels)
+static int hdmi_channel_allocation(struct hda_codec *codec,
+ struct hdmi_eld *eld, int channels)
{
int i;
int ca = 0;
@@ -694,7 +695,7 @@ static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels)
}
snd_print_channel_allocation(eld->info.spk_alloc, buf, sizeof(buf));
- snd_printdd("HDMI: select CA 0x%x for %d-channel allocation: %s\n",
+ codec_dbg(codec, "HDMI: select CA 0x%x for %d-channel allocation: %s\n",
ca, channels, buf);
return ca;
@@ -1131,7 +1132,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
if (!non_pcm && per_pin->chmap_set)
ca = hdmi_manual_channel_allocation(channels, per_pin->chmap);
else
- ca = hdmi_channel_allocation(eld, channels);
+ ca = hdmi_channel_allocation(codec, eld, channels);
if (ca < 0)
ca = 0;
@@ -1557,13 +1558,13 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
eld->eld_valid = false;
else {
memset(&eld->info, 0, sizeof(struct parsed_hdmi_eld));
- if (snd_hdmi_parse_eld(&eld->info, eld->eld_buffer,
+ if (snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer,
eld->eld_size) < 0)
eld->eld_valid = false;
}
if (eld->eld_valid) {
- snd_hdmi_show_eld(&eld->info);
+ snd_hdmi_show_eld(codec, &eld->info);
update_eld = true;
}
else if (repoll) {
@@ -3355,6 +3356,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
{ .id = 0x80862808, .name = "Broadwell HDMI", .patch = patch_generic_hdmi },
{ .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi },
{ .id = 0x80862882, .name = "Valleyview2 HDMI", .patch = patch_generic_hdmi },
+{ .id = 0x80862883, .name = "Braswell HDMI", .patch = patch_generic_hdmi },
{ .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi },
{} /* terminator */
};
@@ -3414,6 +3416,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862807");
MODULE_ALIAS("snd-hda-codec-id:80862808");
MODULE_ALIAS("snd-hda-codec-id:80862880");
MODULE_ALIAS("snd-hda-codec-id:80862882");
+MODULE_ALIAS("snd-hda-codec-id:80862883");
MODULE_ALIAS("snd-hda-codec-id:808629fb");
MODULE_LICENSE("GPL");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b60824e90408..654c8f16d150 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -101,6 +101,7 @@ struct alc_spec {
/* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */
int mute_led_polarity;
hda_nid_t mute_led_nid;
+ hda_nid_t cap_mute_led_nid;
unsigned int gpio_led; /* used for alc269_fixup_hp_gpio_led() */
@@ -3402,7 +3403,8 @@ static unsigned int led_power_filter(struct hda_codec *codec,
{
struct alc_spec *spec = codec->spec;
- if (power_state != AC_PWRST_D3 || nid != spec->mute_led_nid)
+ if (power_state != AC_PWRST_D3 || nid == 0 ||
+ (nid != spec->mute_led_nid && nid != spec->cap_mute_led_nid))
return power_state;
/* Set pin ctl again, it might have just been set to 0 */
@@ -3520,6 +3522,68 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec,
}
}
+/* turn on/off mic-mute LED per capture hook */
+static void alc269_fixup_hp_cap_mic_mute_hook(struct hda_codec *codec,
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int pinval, enable, disable;
+
+ pinval = snd_hda_codec_get_pin_target(codec, spec->cap_mute_led_nid);
+ pinval &= ~AC_PINCTL_VREFEN;
+ enable = pinval | AC_PINCTL_VREF_80;
+ disable = pinval | AC_PINCTL_VREF_HIZ;
+
+ if (!ucontrol)
+ return;
+
+ if (ucontrol->value.integer.value[0] ||
+ ucontrol->value.integer.value[1])
+ pinval = disable;
+ else
+ pinval = enable;
+
+ if (spec->cap_mute_led_nid)
+ snd_hda_set_pin_ctl_cache(codec, spec->cap_mute_led_nid, pinval);
+}
+
+static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ static const struct hda_verb gpio_init[] = {
+ { 0x01, AC_VERB_SET_GPIO_MASK, 0x08 },
+ { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x08 },
+ {}
+ };
+
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ spec->gen.vmaster_mute.hook = alc269_fixup_hp_gpio_mute_hook;
+ spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook;
+ spec->gpio_led = 0;
+ spec->cap_mute_led_nid = 0x18;
+ snd_hda_add_verbs(codec, gpio_init);
+ codec->power_filter = led_power_filter;
+ }
+}
+
+static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook;
+ spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook;
+ spec->mute_led_polarity = 0;
+ spec->mute_led_nid = 0x1a;
+ spec->cap_mute_led_nid = 0x18;
+ spec->gen.vmaster_mute_enum = 1;
+ codec->power_filter = led_power_filter;
+ }
+}
+
static void alc_headset_mode_unplugged(struct hda_codec *codec)
{
int val;
@@ -4231,6 +4295,9 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec,
/* for hda_fixup_thinkpad_acpi() */
#include "thinkpad_helper.c"
+/* for dell wmi mic mute led */
+#include "dell_wmi_helper.c"
+
enum {
ALC269_FIXUP_SONY_VAIO,
ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -4255,6 +4322,8 @@ enum {
ALC269_FIXUP_HP_MUTE_LED_MIC1,
ALC269_FIXUP_HP_MUTE_LED_MIC2,
ALC269_FIXUP_HP_GPIO_LED,
+ ALC269_FIXUP_HP_GPIO_MIC1_LED,
+ ALC269_FIXUP_HP_LINE1_MIC1_LED,
ALC269_FIXUP_INV_DMIC,
ALC269_FIXUP_LENOVO_DOCK,
ALC269_FIXUP_NO_SHUTUP,
@@ -4292,6 +4361,8 @@ enum {
ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC,
ALC293_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC292_FIXUP_TPT440_DOCK,
+ ALC283_FIXUP_BXBT2807_MIC,
+ ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -4447,6 +4518,14 @@ static const struct hda_fixup alc269_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc269_fixup_hp_gpio_led,
},
+ [ALC269_FIXUP_HP_GPIO_MIC1_LED] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc269_fixup_hp_gpio_mic1_led,
+ },
+ [ALC269_FIXUP_HP_LINE1_MIC1_LED] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc269_fixup_hp_line1_mic1_led,
+ },
[ALC269_FIXUP_INV_DMIC] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_inv_dmic_0x12,
@@ -4718,6 +4797,20 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST
},
+ [ALC283_FIXUP_BXBT2807_MIC] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x04a110f0 },
+ { },
+ },
+ },
+ [ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_dell_wmi,
+ .chained_before = true,
+ .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
+ },
+
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -4727,7 +4820,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
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(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
@@ -4761,10 +4853,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED),
SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK),
SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK),
+ SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED),
SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK),
SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
@@ -4782,6 +4876,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED),
/* ALC282 */
+ SND_PCI_QUIRK(0x103c, 0x21f8, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x21f9, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x220d, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x220e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x220f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
@@ -4790,6 +4886,20 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x2212, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2213, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2214, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x2234, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2235, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2236, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2237, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2238, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2239, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2246, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2247, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2248, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2249, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x224a, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x224b, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x224c, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x224d, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x2266, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2267, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2268, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
@@ -4814,13 +4924,43 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x22ce, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x22cf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x22d0, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x22da, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x22db, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x22dc, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x22fb, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x8004, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
/* ALC290 */
+ SND_PCI_QUIRK(0x103c, 0x221b, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x221c, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x221d, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2220, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2222, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2223, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2224, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2225, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2246, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2247, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2248, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2249, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2253, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2254, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2255, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2256, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2257, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2258, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2259, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x2260, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2261, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2262, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2263, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2264, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2265, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2277, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2278, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x227d, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x227e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x227f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
@@ -4843,7 +4983,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
- SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED),
SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
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),
@@ -4864,9 +5003,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
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(0x104d, 0x9099, "Sony VAIO S13", ALC275_FIXUP_SONY_DISABLE_AAMIX),
- SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK),
SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC),
+ SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_BXBT2807_MIC),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
@@ -4891,7 +5030,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
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_VENDOR(0x17aa, "Thinkpad", ALC269_FIXUP_THINKPAD_ACPI),
SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
#if 0
@@ -4945,6 +5083,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
{}
};
+static const struct snd_pci_quirk alc269_fixup_vendor_tbl[] = {
+ SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
+ SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED),
+ SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
+ SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", ALC269_FIXUP_THINKPAD_ACPI),
+ {}
+};
+
static const struct hda_model_fixup alc269_fixup_models[] = {
{.id = ALC269_FIXUP_AMIC, .name = "laptop-amic"},
{.id = ALC269_FIXUP_DMIC, .name = "laptop-dmic"},
@@ -5040,6 +5186,17 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x1d, 0x40700001},
{0x1e, 0x411111f0},
{0x21, 0x02211040}),
+ SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP 15 Touchsmart", ALC269_FIXUP_HP_MUTE_LED_MIC1,
+ {0x12, 0x99a30130},
+ {0x14, 0x90170110},
+ {0x17, 0x40000000},
+ {0x18, 0x411111f0},
+ {0x19, 0x03a11020},
+ {0x1a, 0x411111f0},
+ {0x1b, 0x411111f0},
+ {0x1d, 0x40f41905},
+ {0x1e, 0x411111f0},
+ {0x21, 0x0321101f}),
SND_HDA_PIN_QUIRK(0x10ec0283, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
{0x12, 0x90a60130},
{0x14, 0x90170110},
@@ -5162,6 +5319,8 @@ static int patch_alc269(struct hda_codec *codec)
snd_hda_pick_fixup(codec, alc269_fixup_models,
alc269_fixup_tbl, alc269_fixups);
snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups);
+ snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl,
+ alc269_fixups);
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
alc_auto_parse_customize_define(codec);
@@ -5858,6 +6017,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05fe, "Dell XPS 15", ALC668_FIXUP_DELL_XPS13),
SND_PCI_QUIRK(0x1028, 0x060a, "Dell XPS 13", ALC668_FIXUP_DELL_XPS13),
SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 3744ea4e843d..ea823e1100da 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -84,6 +84,7 @@ enum {
STAC_DELL_EQ,
STAC_ALIENWARE_M17X,
STAC_92HD89XX_HP_FRONT_JACK,
+ STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK,
STAC_92HD73XX_MODELS
};
@@ -103,6 +104,7 @@ enum {
STAC_92HD83XXX_HP,
STAC_HP_ENVY_BASS,
STAC_HP_BNB13_EQ,
+ STAC_HP_ENVY_TS_BASS,
STAC_92HD83XXX_MODELS
};
@@ -1017,7 +1019,7 @@ static int stac_create_spdif_mux_ctls(struct hda_codec *codec)
for (i = 0; i < num_cons; i++) {
if (snd_BUG_ON(!labels[i]))
return -EINVAL;
- snd_hda_add_imux_item(&spec->spdif_mux, labels[i], i, NULL);
+ snd_hda_add_imux_item(codec, &spec->spdif_mux, labels[i], i, NULL);
}
kctl = snd_hda_gen_add_kctl(&spec->gen, NULL, &stac_smux_mixer);
@@ -1809,6 +1811,11 @@ static const struct hda_pintbl stac92hd89xx_hp_front_jack_pin_configs[] = {
{}
};
+static const struct hda_pintbl stac92hd89xx_hp_z1_g2_right_mic_jack_pin_configs[] = {
+ { 0x0e, 0x400000f0 },
+ {}
+};
+
static void stac92hd73xx_fixup_ref(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
@@ -1931,6 +1938,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = {
[STAC_92HD89XX_HP_FRONT_JACK] = {
.type = HDA_FIXUP_PINS,
.v.pins = stac92hd89xx_hp_front_jack_pin_configs,
+ },
+ [STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = stac92hd89xx_hp_z1_g2_right_mic_jack_pin_configs,
}
};
@@ -1991,6 +2002,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = {
"Alienware M17x", STAC_ALIENWARE_M17X),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490,
"Alienware M17x R3", STAC_DELL_EQ),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1927,
+ "HP Z1 G2", STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17,
"unknown HP", STAC_92HD89XX_HP_FRONT_JACK),
{} /* terminator */
@@ -2668,6 +2681,13 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = {
.chained = true,
.chain_id = STAC_92HD83XXX_HP_MIC_LED,
},
+ [STAC_HP_ENVY_TS_BASS] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x10, 0x92170111 },
+ {}
+ },
+ },
};
static const struct hda_model_fixup stac92hd83xxx_models[] = {
@@ -2684,6 +2704,7 @@ static const struct hda_model_fixup stac92hd83xxx_models[] = {
{ .id = STAC_92HD83XXX_HEADSET_JACK, .name = "headset-jack" },
{ .id = STAC_HP_ENVY_BASS, .name = "hp-envy-bass" },
{ .id = STAC_HP_BNB13_EQ, .name = "hp-bnb13-eq" },
+ { .id = STAC_HP_ENVY_TS_BASS, .name = "hp-envy-ts-bass" },
{}
};
@@ -2739,6 +2760,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = {
"HP bNB13", STAC_HP_BNB13_EQ),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x190A,
"HP bNB13", STAC_HP_BNB13_EQ),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x190e,
+ "HP ENVY TS", STAC_HP_ENVY_TS_BASS),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1940,
"HP bNB13", STAC_HP_BNB13_EQ),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1941,
@@ -3438,9 +3461,11 @@ static void stac922x_fixup_intel_mac_auto(struct hda_codec *codec,
{
if (action != HDA_FIXUP_ACT_PRE_PROBE)
return;
+
+ codec->fixup_id = HDA_FIXUP_ID_NOT_SET;
snd_hda_pick_fixup(codec, NULL, stac922x_intel_mac_fixup_tbl,
stac922x_fixups);
- if (codec->fixup_id != STAC_INTEL_MAC_AUTO)
+ if (codec->fixup_id != HDA_FIXUP_ID_NOT_SET)
snd_hda_apply_fixup(codec, action);
}
diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h
index b209fc30b334..58f8f2ae758d 100644
--- a/sound/pci/ice1712/ice1712.h
+++ b/sound/pci/ice1712/ice1712.h
@@ -41,14 +41,17 @@
#define ICEREG(ice, x) ((ice)->port + ICE1712_REG_##x)
#define ICE1712_REG_CONTROL 0x00 /* byte */
-#define ICE1712_RESET 0x80 /* reset whole chip */
-#define ICE1712_SERR_LEVEL 0x04 /* SERR# level otherwise edge */
+#define ICE1712_RESET 0x80 /* soft reset whole chip */
+#define ICE1712_SERR_ASSERT_DS_DMA 0x40 /* disabled SERR# assertion for the DS DMA Ch-C irq otherwise enabled */
+#define ICE1712_DOS_VOL 0x10 /* DOS WT/FM volume control */
+#define ICE1712_SERR_LEVEL 0x08 /* SERR# level otherwise edge */
+#define ICE1712_SERR_ASSERT_SB 0x02 /* disabled SERR# assertion for SB irq otherwise enabled */
#define ICE1712_NATIVE 0x01 /* native mode otherwise SB */
#define ICE1712_REG_IRQMASK 0x01 /* byte */
-#define ICE1712_IRQ_MPU1 0x80
-#define ICE1712_IRQ_TIMER 0x40
-#define ICE1712_IRQ_MPU2 0x20
-#define ICE1712_IRQ_PROPCM 0x10
+#define ICE1712_IRQ_MPU1 0x80 /* MIDI irq mask */
+#define ICE1712_IRQ_TIMER 0x40 /* Timer mask */
+#define ICE1712_IRQ_MPU2 0x20 /* Secondary MIDI irq mask */
+#define ICE1712_IRQ_PROPCM 0x10 /* professional multi-track */
#define ICE1712_IRQ_FM 0x08 /* FM/MIDI - legacy */
#define ICE1712_IRQ_PBKDS 0x04 /* playback DS channels */
#define ICE1712_IRQ_CONCAP 0x02 /* consumer capture */
diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c
index 71f4bdcc4055..84f67450924e 100644
--- a/sound/pci/mixart/mixart_core.c
+++ b/sound/pci/mixart/mixart_core.c
@@ -151,13 +151,11 @@ static int send_msg( struct mixart_mgr *mgr,
{
u32 headptr, tailptr;
u32 msg_frame_address;
- int err, i;
+ int i;
if (snd_BUG_ON(msg->size % 4))
return -EINVAL;
- err = 0;
-
/* get message frame address */
tailptr = readl_be(MIXART_MEM(mgr, MSG_INBOUND_FREE_TAIL));
headptr = readl_be(MIXART_MEM(mgr, MSG_INBOUND_FREE_HEAD));
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 64b9fda5f04a..dbbbacfd535e 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -53,6 +53,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = {
{ OXYGEN_PCI_SUBID(0x1043, 0x835e) },
{ OXYGEN_PCI_SUBID(0x1043, 0x838e) },
{ OXYGEN_PCI_SUBID(0x1043, 0x8522) },
+ { OXYGEN_PCI_SUBID(0x1043, 0x85f4) },
{ OXYGEN_PCI_SUBID_BROKEN_EEPROM },
{ }
};
diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c
index c8c7f2c9b355..e02605931669 100644
--- a/sound/pci/oxygen/xonar_pcm179x.c
+++ b/sound/pci/oxygen/xonar_pcm179x.c
@@ -100,8 +100,8 @@
*/
/*
- * Xonar Essence ST (Deluxe)/STX
- * -----------------------------
+ * Xonar Essence ST (Deluxe)/STX (II)
+ * ----------------------------------
*
* CMI8788:
*
@@ -1138,6 +1138,14 @@ int get_xonar_pcm179x_model(struct oxygen *chip,
chip->model.resume = xonar_stx_resume;
chip->model.set_dac_params = set_pcm1796_params;
break;
+ case 0x85f4:
+ chip->model = model_xonar_st;
+ /* TODO: daughterboard support */
+ chip->model.shortname = "Xonar STX II";
+ chip->model.init = xonar_stx_init;
+ chip->model.resume = xonar_stx_resume;
+ chip->model.set_dac_params = set_pcm1796_params;
+ break;
default:
return -EINVAL;
}
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index b4a8278241b1..f0315c3f7de4 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -941,7 +941,7 @@ setmixer(struct cmdif *cif, short num, unsigned short rval, unsigned short lval)
union cmdret rptr = CMDRET_ZERO;
int i = 0;
- snd_printdd("sent mixer %d: 0x%d 0x%d\n", num, rval, lval);
+ snd_printdd("sent mixer %d: 0x%x 0x%x\n", num, rval, lval);
do {
SEND_SDGV(cif, num, num, rval, lval);
SEND_RDGV(cif, num, num, &rptr);
@@ -1080,7 +1080,7 @@ getmixer(struct cmdif *cif, short num, unsigned short *rval,
return -EIO;
*rval = rptr.retwords[0];
*lval = rptr.retwords[1];
- snd_printdd("got mixer %d: 0x%d 0x%d\n", num, *rval, *lval);
+ snd_printdd("got mixer %d: 0x%x 0x%x\n", num, *rval, *lval);
return 0;
}
diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index 1272c18a2544..da875dced2ef 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -3880,14 +3880,12 @@ void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voi
{
unsigned long flags;
void (*private_free)(struct snd_trident_voice *);
- void *private_data;
if (voice == NULL || !voice->use)
return;
snd_trident_clear_voices(trident, voice->number, voice->number);
spin_lock_irqsave(&trident->voice_alloc, flags);
private_free = voice->private_free;
- private_data = voice->private_data;
voice->private_free = NULL;
voice->private_data = NULL;
if (voice->pcm)
diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c
index 3102a579660b..04c474658e3c 100644
--- a/sound/pci/trident/trident_memory.c
+++ b/sound/pci/trident/trident_memory.c
@@ -139,12 +139,11 @@ static inline void *offset_ptr(struct snd_trident *trident, int offset)
static struct snd_util_memblk *
search_empty(struct snd_util_memhdr *hdr, int size)
{
- struct snd_util_memblk *blk, *prev;
+ struct snd_util_memblk *blk;
int page, psize;
struct list_head *p;
psize = get_aligned_page(size + ALIGN_PAGE_SIZE -1);
- prev = NULL;
page = 0;
list_for_each(p, &hdr->block) {
blk = list_entry(p, struct snd_util_memblk, list);
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 0060b31cc3f3..0e9623368ab0 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -47,6 +47,7 @@ source "sound/soc/kirkwood/Kconfig"
source "sound/soc/intel/Kconfig"
source "sound/soc/mxs/Kconfig"
source "sound/soc/pxa/Kconfig"
+source "sound/soc/rockchip/Kconfig"
source "sound/soc/samsung/Kconfig"
source "sound/soc/s6000/Kconfig"
source "sound/soc/sh/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 5f1df02984f8..534714a1ca44 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -24,6 +24,7 @@ obj-$(CONFIG_SND_SOC) += nuc900/
obj-$(CONFIG_SND_SOC) += omap/
obj-$(CONFIG_SND_SOC) += kirkwood/
obj-$(CONFIG_SND_SOC) += pxa/
+obj-$(CONFIG_SND_SOC) += rockchip/
obj-$(CONFIG_SND_SOC) += samsung/
obj-$(CONFIG_SND_SOC) += s6000/
obj-$(CONFIG_SND_SOC) += sh/
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index de433cfd044c..f403f399808a 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -347,6 +347,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
u32 tfmr, rfmr, tcmr, rcmr;
int start_event;
int ret;
+ int fslen, fslen_ext;
/*
* Currently, there is only one set of dma params for
@@ -388,18 +389,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
}
/*
- * The SSC only supports up to 16-bit samples in I2S format, due
- * to the size of the Frame Mode Register FSLEN field.
- */
- if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S
- && bits > 16) {
- printk(KERN_WARNING
- "atmel_ssc_dai: sample size %d "
- "is too large for I2S\n", bits);
- return -EINVAL;
- }
-
- /*
* Compute SSC register settings.
*/
switch (ssc_p->daifmt
@@ -413,6 +402,17 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
* from the MCK divider, and the BCLK signal
* is output on the SSC TK line.
*/
+
+ if (bits > 16 && !ssc->pdata->has_fslen_ext) {
+ dev_err(dai->dev,
+ "sample size %d is too large for SSC device\n",
+ bits);
+ return -EINVAL;
+ }
+
+ fslen_ext = (bits - 1) / 16;
+ fslen = (bits - 1) % 16;
+
rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period)
| SSC_BF(RCMR_STTDLY, START_DELAY)
| SSC_BF(RCMR_START, SSC_START_FALLING_RF)
@@ -420,9 +420,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
| SSC_BF(RCMR_CKS, SSC_CKS_DIV);
- rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+ rfmr = SSC_BF(RFMR_FSLEN_EXT, fslen_ext)
+ | SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
| SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE)
- | SSC_BF(RFMR_FSLEN, (bits - 1))
+ | SSC_BF(RFMR_FSLEN, fslen)
| SSC_BF(RFMR_DATNB, (channels - 1))
| SSC_BIT(RFMR_MSBF)
| SSC_BF(RFMR_LOOP, 0)
@@ -435,10 +436,11 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
| SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS)
| SSC_BF(TCMR_CKS, SSC_CKS_DIV);
- tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+ tfmr = SSC_BF(TFMR_FSLEN_EXT, fslen_ext)
+ | SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
| SSC_BF(TFMR_FSDEN, 0)
| SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE)
- | SSC_BF(TFMR_FSLEN, (bits - 1))
+ | SSC_BF(TFMR_FSLEN, fslen)
| SSC_BF(TFMR_DATNB, (channels - 1))
| SSC_BIT(TFMR_MSBF)
| SSC_BF(TFMR_DATDEF, 0)
diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c
index b4e36901a40b..4052268ce462 100644
--- a/sound/soc/atmel/atmel_wm8904.c
+++ b/sound/soc/atmel/atmel_wm8904.c
@@ -18,10 +18,6 @@
#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),
@@ -61,26 +57,6 @@ 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",
@@ -94,7 +70,6 @@ static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = {
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,
@@ -153,7 +128,6 @@ 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;
int id, ret;
card->dev = &pdev->dev;
@@ -170,30 +144,6 @@ static int atmel_asoc_wm8904_probe(struct platform_device *pdev)
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");
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index a3881c4381c9..bcf591373a7a 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -290,19 +290,19 @@ static int bf5xx_pcm_silence(struct snd_pcm_substream *substream,
unsigned int sample_size = runtime->sample_bits / 8;
void *buf = runtime->dma_area;
struct bf5xx_i2s_pcm_data *dma_data;
- unsigned int offset, size;
+ unsigned int offset, samples;
dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
if (dma_data->tdm_mode) {
offset = pos * 8 * sample_size;
- size = count * 8 * sample_size;
+ samples = count * 8;
} else {
offset = frames_to_bytes(runtime, pos);
- size = frames_to_bytes(runtime, count);
+ samples = count * runtime->channels;
}
- snd_pcm_format_set_silence(runtime->format, buf + offset, size);
+ snd_pcm_format_set_silence(runtime->format, buf + offset, samples);
return 0;
}
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 3c4b10ff48c1..922006dd0583 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -945,11 +945,11 @@ static int pm860x_pcm_hw_params(struct snd_pcm_substream *substream,
unsigned char inf = 0, mask = 0;
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
inf &= ~PCM_INF2_18WL;
break;
- case SNDRV_PCM_FORMAT_S18_3LE:
+ case 18:
inf |= PCM_INF2_18WL;
break;
default:
@@ -1044,11 +1044,11 @@ static int pm860x_i2s_hw_params(struct snd_pcm_substream *substream,
unsigned char inf;
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
inf = 0;
break;
- case SNDRV_PCM_FORMAT_S18_3LE:
+ case 18:
inf = PCM_INF2_18WL;
break;
default:
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 0b9571c858f8..8838838e25ed 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -47,6 +47,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_CS42L52 if I2C && INPUT
select SND_SOC_CS42L56 if I2C && INPUT
select SND_SOC_CS42L73 if I2C
+ select SND_SOC_CS4265 if I2C
select SND_SOC_CS4270 if I2C
select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
select SND_SOC_CS42XX8_I2C if I2C
@@ -74,10 +75,12 @@ config SND_SOC_ALL_CODECS
select SND_SOC_PCM3008
select SND_SOC_PCM512x_I2C if I2C
select SND_SOC_PCM512x_SPI if SPI_MASTER
+ select SND_SOC_RT286 if I2C
select SND_SOC_RT5631 if I2C
select SND_SOC_RT5640 if I2C
select SND_SOC_RT5645 if I2C
select SND_SOC_RT5651 if I2C
+ select SND_SOC_RT5670 if I2C
select SND_SOC_RT5677 if I2C
select SND_SOC_SGTL5000 if I2C
select SND_SOC_SI476X if MFD_SI476X_CORE
@@ -91,6 +94,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_STA350 if I2C
select SND_SOC_STA529 if I2C
select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
+ select SND_SOC_TAS2552 if I2C
select SND_SOC_TAS5086 if I2C
select SND_SOC_TLV320AIC23_I2C if I2C
select SND_SOC_TLV320AIC23_SPI if SPI_MASTER
@@ -338,6 +342,11 @@ config SND_SOC_CS42L73
tristate "Cirrus Logic CS42L73 CODEC"
depends on I2C
+config SND_SOC_CS4265
+ tristate "Cirrus Logic CS4265 CODEC"
+ depends on I2C
+ select REGMAP_I2C
+
# Cirrus Logic CS4270 Codec
config SND_SOC_CS4270
tristate "Cirrus Logic CS4270 CODEC"
@@ -445,9 +454,16 @@ config SND_SOC_RL6231
default y if SND_SOC_RT5640=y
default y if SND_SOC_RT5645=y
default y if SND_SOC_RT5651=y
+ default y if SND_SOC_RT5670=y
+ default y if SND_SOC_RT5677=y
default m if SND_SOC_RT5640=m
default m if SND_SOC_RT5645=m
default m if SND_SOC_RT5651=m
+ default m if SND_SOC_RT5670=m
+ default m if SND_SOC_RT5677=m
+
+config SND_SOC_RT286
+ tristate
config SND_SOC_RT5631
tristate
@@ -461,6 +477,9 @@ config SND_SOC_RT5645
config SND_SOC_RT5651
tristate
+config SND_SOC_RT5670
+ tristate
+
config SND_SOC_RT5677
tristate
@@ -521,6 +540,10 @@ config SND_SOC_STA529
config SND_SOC_STAC9766
tristate
+config SND_SOC_TAS2552
+ tristate "Texas Instruments TAS2552 Mono Audio amplifier"
+ depends on I2C
+
config SND_SOC_TAS5086
tristate "Texas Instruments TAS5086 speaker amplifier"
depends on I2C
@@ -541,7 +564,9 @@ config SND_SOC_TLV320AIC26
depends on SPI
config SND_SOC_TLV320AIC31XX
- tristate
+ tristate "Texas Instruments TLV320AIC31xx CODECs"
+ depends on I2C
+ select REGMAP_I2C
config SND_SOC_TLV320AIC32X4
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 1bd6e1cf6f82..20afe0f0c5be 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -37,6 +37,7 @@ snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o
snd-soc-cs42l52-objs := cs42l52.o
snd-soc-cs42l56-objs := cs42l56.o
snd-soc-cs42l73-objs := cs42l73.o
+snd-soc-cs4265-objs := cs4265.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-cs4271-objs := cs4271.o
snd-soc-cs42xx8-objs := cs42xx8.o
@@ -68,10 +69,12 @@ snd-soc-pcm512x-objs := pcm512x.o
snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
snd-soc-pcm512x-spi-objs := pcm512x-spi.o
snd-soc-rl6231-objs := rl6231.o
+snd-soc-rt286-objs := rt286.o
snd-soc-rt5631-objs := rt5631.o
snd-soc-rt5640-objs := rt5640.o
snd-soc-rt5645-objs := rt5645.o
snd-soc-rt5651-objs := rt5651.o
+snd-soc-rt5670-objs := rt5670.o
snd-soc-rt5677-objs := rt5677.o
snd-soc-sgtl5000-objs := sgtl5000.o
snd-soc-alc5623-objs := alc5623.o
@@ -162,6 +165,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o
# Amp
snd-soc-max9877-objs := max9877.o
snd-soc-tpa6130a2-objs := tpa6130a2.o
+snd-soc-tas2552-objs := tas2552.o
obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o
obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o
@@ -204,6 +208,7 @@ obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o
obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o
obj-$(CONFIG_SND_SOC_CS42L56) += snd-soc-cs42l56.o
obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o
+obj-$(CONFIG_SND_SOC_CS4265) += snd-soc-cs4265.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o
obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o
@@ -235,10 +240,12 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o
obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o
obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o
obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o
+obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o
obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o
obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o
obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o
obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o
+obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o
obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o
obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o
obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o
@@ -255,6 +262,7 @@ obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o
obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o
obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o
obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
+obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o
obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 8d9ba4ba4bfe..e889e1b84192 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -89,8 +89,8 @@ static int ac97_soc_probe(struct snd_soc_codec *codec)
int ret;
/* add codec as bus device for standard ac97 */
- ret = snd_ac97_bus(codec->card->snd_card, 0, soc_ac97_ops, NULL,
- &ac97_bus);
+ ret = snd_ac97_bus(codec->component.card->snd_card, 0, soc_ac97_ops,
+ NULL, &ac97_bus);
if (ret < 0)
return ret;
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index d71c59cf7bdd..370b742117ef 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -230,8 +230,10 @@ static int adau1701_reg_read(void *context, unsigned int reg,
*value = 0;
- for (i = 0; i < size; i++)
- *value |= recv_buf[i] << (i * 8);
+ for (i = 0; i < size; i++) {
+ *value <<= 8;
+ *value |= recv_buf[i];
+ }
return 0;
}
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 2961fae9670a..0b659704e60c 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -359,14 +359,14 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream,
if (adau->dai_fmt != SND_SOC_DAIFMT_RIGHT_J)
return 0;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
val = ADAU17X1_SERIAL_PORT1_DELAY16;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
val = ADAU17X1_SERIAL_PORT1_DELAY8;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
val = ADAU17X1_SERIAL_PORT1_DELAY0;
break;
default:
diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c
index fd55da7cb9d4..70ab35744aba 100644
--- a/sound/soc/codecs/adau1977.c
+++ b/sound/soc/codecs/adau1977.c
@@ -968,7 +968,7 @@ int adau1977_probe(struct device *dev, struct regmap *regmap,
if (adau1977->dvdd_reg)
power_off_mask = ~0;
else
- power_off_mask = ~ADAU1977_BLOCK_POWER_SAI_LDO_EN;
+ power_off_mask = (unsigned int)~ADAU1977_BLOCK_POWER_SAI_LDO_EN;
ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI,
power_off_mask, 0x00);
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 3ba4c0f11418..041712592e29 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -547,7 +547,7 @@ static const struct ak4642_drvdata ak4648_drvdata = {
.extended_frequencies = 1,
};
-static struct of_device_id ak4642_of_match[];
+static const struct of_device_id ak4642_of_match[];
static int ak4642_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -593,7 +593,7 @@ static int ak4642_i2c_remove(struct i2c_client *client)
return 0;
}
-static struct of_device_id ak4642_of_match[] = {
+static const struct of_device_id ak4642_of_match[] = {
{ .compatible = "asahi-kasei,ak4642", .data = &ak4642_drvdata},
{ .compatible = "asahi-kasei,ak4643", .data = &ak4643_drvdata},
{ .compatible = "asahi-kasei,ak4648", .data = &ak4648_drvdata},
diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c
index 72e953b2cb41..8107a1cac876 100644
--- a/sound/soc/codecs/ak5386.c
+++ b/sound/soc/codecs/ak5386.c
@@ -14,12 +14,18 @@
#include <linux/of.h>
#include <linux/of_gpio.h>
#include <linux/of_device.h>
+#include <linux/regulator/consumer.h>
#include <sound/soc.h>
#include <sound/pcm.h>
#include <sound/initval.h>
+static const char * const supply_names[] = {
+ "va", "vd"
+};
+
struct ak5386_priv {
int reset_gpio;
+ struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
};
static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = {
@@ -32,7 +38,42 @@ static const struct snd_soc_dapm_route ak5386_dapm_routes[] = {
{ "Capture", NULL, "AINR" },
};
+static int ak5386_soc_probe(struct snd_soc_codec *codec)
+{
+ struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec);
+ return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+
+static int ak5386_soc_remove(struct snd_soc_codec *codec)
+{
+ struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec);
+ regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int ak5386_soc_suspend(struct snd_soc_codec *codec)
+{
+ struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec);
+ regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+ return 0;
+}
+
+static int ak5386_soc_resume(struct snd_soc_codec *codec)
+{
+ struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec);
+ return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+#else
+#define ak5386_soc_suspend NULL
+#define ak5386_soc_resume NULL
+#endif /* CONFIG_PM */
+
static struct snd_soc_codec_driver soc_codec_ak5386 = {
+ .probe = ak5386_soc_probe,
+ .remove = ak5386_soc_remove,
+ .suspend = ak5386_soc_suspend,
+ .resume = ak5386_soc_resume,
.dapm_widgets = ak5386_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ak5386_dapm_widgets),
.dapm_routes = ak5386_dapm_routes,
@@ -122,6 +163,7 @@ static int ak5386_probe(struct platform_device *pdev)
{
struct device *dev = &pdev->dev;
struct ak5386_priv *priv;
+ int ret, i;
priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
@@ -130,6 +172,14 @@ static int ak5386_probe(struct platform_device *pdev)
priv->reset_gpio = -EINVAL;
dev_set_drvdata(dev, priv);
+ for (i = 0; i < ARRAY_SIZE(supply_names); i++)
+ priv->supplies[i].supply = supply_names[i];
+
+ ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies),
+ priv->supplies);
+ if (ret < 0)
+ return ret;
+
if (of_match_device(of_match_ptr(ak5386_dt_ids), dev))
priv->reset_gpio = of_get_named_gpio(dev->of_node,
"reset-gpio", 0);
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 29e198f57d4c..2f2e91ac690f 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -243,6 +243,31 @@ int arizona_init_spk(struct snd_soc_codec *codec)
}
EXPORT_SYMBOL_GPL(arizona_init_spk);
+static const struct snd_soc_dapm_route arizona_mono_routes[] = {
+ { "OUT1R", NULL, "OUT1L" },
+ { "OUT2R", NULL, "OUT2L" },
+ { "OUT3R", NULL, "OUT3L" },
+ { "OUT4R", NULL, "OUT4L" },
+ { "OUT5R", NULL, "OUT5L" },
+ { "OUT6R", NULL, "OUT6L" },
+};
+
+int arizona_init_mono(struct snd_soc_codec *codec)
+{
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+ int i;
+
+ for (i = 0; i < ARIZONA_MAX_OUTPUT; ++i) {
+ if (arizona->pdata.out_mono[i])
+ snd_soc_dapm_add_routes(&codec->dapm,
+ &arizona_mono_routes[i], 1);
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_mono);
+
int arizona_init_gpio(struct snd_soc_codec *codec)
{
struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
@@ -1127,6 +1152,31 @@ static int arizona_startup(struct snd_pcm_substream *substream,
constraint);
}
+static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec,
+ unsigned int rate)
+{
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+ struct reg_default dac_comp[] = {
+ { 0x80, 0x3 },
+ { ARIZONA_DAC_COMP_1, 0 },
+ { ARIZONA_DAC_COMP_2, 0 },
+ { 0x80, 0x0 },
+ };
+
+ mutex_lock(&codec->mutex);
+
+ dac_comp[1].def = arizona->dac_comp_coeff;
+ if (rate >= 176400)
+ dac_comp[2].def = arizona->dac_comp_enabled;
+
+ mutex_unlock(&codec->mutex);
+
+ regmap_multi_reg_write(arizona->regmap,
+ dac_comp,
+ ARRAY_SIZE(dac_comp));
+}
+
static int arizona_hw_params_rate(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -1153,6 +1203,15 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream,
switch (dai_priv->clk) {
case ARIZONA_CLK_SYSCLK:
+ switch (priv->arizona->type) {
+ case WM5102:
+ arizona_wm5102_set_dac_comp(codec,
+ params_rate(params));
+ break;
+ default:
+ break;
+ }
+
snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1,
ARIZONA_SAMPLE_RATE_1_MASK, sr_val);
if (base)
@@ -1175,6 +1234,27 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream,
return 0;
}
+static bool arizona_aif_cfg_changed(struct snd_soc_codec *codec,
+ int base, int bclk, int lrclk, int frame)
+{
+ int val;
+
+ val = snd_soc_read(codec, base + ARIZONA_AIF_BCLK_CTRL);
+ if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK))
+ return true;
+
+ val = snd_soc_read(codec, base + ARIZONA_AIF_TX_BCLK_RATE);
+ if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK))
+ return true;
+
+ val = snd_soc_read(codec, base + ARIZONA_AIF_FRAME_CTRL_1);
+ if (frame != (val & (ARIZONA_AIF1TX_WL_MASK |
+ ARIZONA_AIF1TX_SLOT_LEN_MASK)))
+ return true;
+
+ return false;
+}
+
static int arizona_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -1185,26 +1265,40 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
int base = dai->driver->base;
const int *rates;
int i, ret, val;
+ int channels = params_channels(params);
int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1];
+ int tdm_width = arizona->tdm_width[dai->id - 1];
+ int tdm_slots = arizona->tdm_slots[dai->id - 1];
int bclk, lrclk, wl, frame, bclk_target;
+ bool reconfig;
+ unsigned int aif_tx_state, aif_rx_state;
if (params_rate(params) % 8000)
rates = &arizona_44k1_bclk_rates[0];
else
rates = &arizona_48k_bclk_rates[0];
- bclk_target = snd_soc_params_to_bclk(params);
- if (chan_limit && chan_limit < params_channels(params)) {
+ if (tdm_slots) {
+ arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n",
+ tdm_slots, tdm_width);
+ bclk_target = tdm_slots * tdm_width * params_rate(params);
+ channels = tdm_slots;
+ } else {
+ bclk_target = snd_soc_params_to_bclk(params);
+ }
+
+ if (chan_limit && chan_limit < channels) {
arizona_aif_dbg(dai, "Limiting to %d channels\n", chan_limit);
- bclk_target /= params_channels(params);
+ bclk_target /= channels;
bclk_target *= chan_limit;
}
- /* Force stereo for I2S mode */
+ /* Force multiple of 2 channels for I2S mode */
val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT);
- if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) {
+ if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) {
arizona_aif_dbg(dai, "Forcing stereo mode\n");
- bclk_target *= 2;
+ bclk_target /= channels;
+ bclk_target *= channels + 1;
}
for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) {
@@ -1228,28 +1322,56 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
wl = snd_pcm_format_width(params_format(params));
frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl;
+ reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame);
+
+ if (reconfig) {
+ /* Save AIF TX/RX state */
+ aif_tx_state = snd_soc_read(codec,
+ base + ARIZONA_AIF_TX_ENABLES);
+ aif_rx_state = snd_soc_read(codec,
+ base + ARIZONA_AIF_RX_ENABLES);
+ /* Disable AIF TX/RX before reconfiguring it */
+ regmap_update_bits_async(arizona->regmap,
+ base + ARIZONA_AIF_TX_ENABLES, 0xff, 0x0);
+ regmap_update_bits(arizona->regmap,
+ base + ARIZONA_AIF_RX_ENABLES, 0xff, 0x0);
+ }
+
ret = arizona_hw_params_rate(substream, params, dai);
if (ret != 0)
- return ret;
+ goto restore_aif;
- regmap_update_bits_async(arizona->regmap,
- base + ARIZONA_AIF_BCLK_CTRL,
- ARIZONA_AIF1_BCLK_FREQ_MASK, bclk);
- regmap_update_bits_async(arizona->regmap,
- base + ARIZONA_AIF_TX_BCLK_RATE,
- ARIZONA_AIF1TX_BCPF_MASK, lrclk);
- regmap_update_bits_async(arizona->regmap,
- base + ARIZONA_AIF_RX_BCLK_RATE,
- ARIZONA_AIF1RX_BCPF_MASK, lrclk);
- regmap_update_bits_async(arizona->regmap,
- base + ARIZONA_AIF_FRAME_CTRL_1,
- ARIZONA_AIF1TX_WL_MASK |
- ARIZONA_AIF1TX_SLOT_LEN_MASK, frame);
- regmap_update_bits(arizona->regmap, base + ARIZONA_AIF_FRAME_CTRL_2,
- ARIZONA_AIF1RX_WL_MASK |
- ARIZONA_AIF1RX_SLOT_LEN_MASK, frame);
+ if (reconfig) {
+ regmap_update_bits_async(arizona->regmap,
+ base + ARIZONA_AIF_BCLK_CTRL,
+ ARIZONA_AIF1_BCLK_FREQ_MASK, bclk);
+ regmap_update_bits_async(arizona->regmap,
+ base + ARIZONA_AIF_TX_BCLK_RATE,
+ ARIZONA_AIF1TX_BCPF_MASK, lrclk);
+ regmap_update_bits_async(arizona->regmap,
+ base + ARIZONA_AIF_RX_BCLK_RATE,
+ ARIZONA_AIF1RX_BCPF_MASK, lrclk);
+ regmap_update_bits_async(arizona->regmap,
+ base + ARIZONA_AIF_FRAME_CTRL_1,
+ ARIZONA_AIF1TX_WL_MASK |
+ ARIZONA_AIF1TX_SLOT_LEN_MASK, frame);
+ regmap_update_bits(arizona->regmap,
+ base + ARIZONA_AIF_FRAME_CTRL_2,
+ ARIZONA_AIF1RX_WL_MASK |
+ ARIZONA_AIF1RX_SLOT_LEN_MASK, frame);
+ }
- return 0;
+restore_aif:
+ if (reconfig) {
+ /* Restore AIF TX/RX state */
+ regmap_update_bits_async(arizona->regmap,
+ base + ARIZONA_AIF_TX_ENABLES,
+ 0xff, aif_tx_state);
+ regmap_update_bits(arizona->regmap,
+ base + ARIZONA_AIF_RX_ENABLES,
+ 0xff, aif_rx_state);
+ }
+ return ret;
}
static const char *arizona_dai_clk_str(int clk_id)
@@ -1324,9 +1446,63 @@ static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate)
ARIZONA_AIF1_TRI, reg);
}
+static void arizona_set_channels_to_mask(struct snd_soc_dai *dai,
+ unsigned int base,
+ int channels, unsigned int mask)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+ int slot, i;
+
+ for (i = 0; i < channels; ++i) {
+ slot = ffs(mask) - 1;
+ if (slot < 0)
+ return;
+
+ regmap_write(arizona->regmap, base + i, slot);
+
+ mask &= ~(1 << slot);
+ }
+
+ if (mask)
+ arizona_aif_warn(dai, "Too many channels in TDM mask\n");
+}
+
+static int arizona_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+ unsigned int rx_mask, int slots, int slot_width)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+ int base = dai->driver->base;
+ int rx_max_chan = dai->driver->playback.channels_max;
+ int tx_max_chan = dai->driver->capture.channels_max;
+
+ /* Only support TDM for the physical AIFs */
+ if (dai->id > ARIZONA_MAX_AIF)
+ return -ENOTSUPP;
+
+ if (slots == 0) {
+ tx_mask = (1 << tx_max_chan) - 1;
+ rx_mask = (1 << rx_max_chan) - 1;
+ }
+
+ arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_3,
+ tx_max_chan, tx_mask);
+ arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_11,
+ rx_max_chan, rx_mask);
+
+ arizona->tdm_width[dai->id - 1] = slot_width;
+ arizona->tdm_slots[dai->id - 1] = slots;
+
+ return 0;
+}
+
const struct snd_soc_dai_ops arizona_dai_ops = {
.startup = arizona_startup,
.set_fmt = arizona_set_fmt,
+ .set_tdm_slot = arizona_set_tdm_slot,
.hw_params = arizona_hw_params,
.set_sysclk = arizona_dai_set_sysclk,
.set_tristate = arizona_set_tristate,
@@ -1400,6 +1576,12 @@ static int arizona_validate_fll(struct arizona_fll *fll,
{
unsigned int Fvco_min;
+ if (fll->fout && Fout != fll->fout) {
+ arizona_fll_err(fll,
+ "Can't change output on active FLL\n");
+ return -EINVAL;
+ }
+
if (Fref / ARIZONA_FLL_MAX_REFDIV > ARIZONA_FLL_MAX_FREF) {
arizona_fll_err(fll,
"Can't scale %dMHz in to <=13.5MHz\n",
@@ -1478,6 +1660,10 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
while (div <= ARIZONA_FLL_MAX_REFDIV) {
for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO;
ratio++) {
+ if ((ARIZONA_FLL_VCO_CORNER / 2) /
+ (fll->vco_mult * ratio) < Fref)
+ break;
+
if (target % (ratio * Fref)) {
cfg->refdiv = refdiv;
cfg->fratio = ratio - 1;
@@ -1485,11 +1671,7 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
}
}
- for (ratio = init_ratio - 1; ratio >= 0; ratio--) {
- if (ARIZONA_FLL_VCO_CORNER / (fll->vco_mult * ratio) <
- Fref)
- break;
-
+ for (ratio = init_ratio - 1; ratio > 0; ratio--) {
if (target % (ratio * Fref)) {
cfg->refdiv = refdiv;
cfg->fratio = ratio - 1;
@@ -1616,7 +1798,7 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base,
ARIZONA_FLL1_CTRL_UPD | cfg->n);
}
-static bool arizona_is_enabled_fll(struct arizona_fll *fll)
+static int arizona_is_enabled_fll(struct arizona_fll *fll)
{
struct arizona *arizona = fll->arizona;
unsigned int reg;
@@ -1632,13 +1814,26 @@ static bool arizona_is_enabled_fll(struct arizona_fll *fll)
return reg & ARIZONA_FLL1_ENA;
}
-static void arizona_enable_fll(struct arizona_fll *fll)
+static int arizona_enable_fll(struct arizona_fll *fll)
{
struct arizona *arizona = fll->arizona;
int ret;
bool use_sync = false;
+ int already_enabled = arizona_is_enabled_fll(fll);
struct arizona_fll_cfg cfg;
+ if (already_enabled < 0)
+ return already_enabled;
+
+ if (already_enabled) {
+ /* Facilitate smooth refclk across the transition */
+ regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x7,
+ ARIZONA_FLL1_GAIN_MASK, 0);
+ regmap_update_bits_async(fll->arizona->regmap, fll->base + 1,
+ ARIZONA_FLL1_FREERUN,
+ ARIZONA_FLL1_FREERUN);
+ }
+
/*
* If we have both REFCLK and SYNCCLK then enable both,
* otherwise apply the SYNCCLK settings to REFCLK.
@@ -1666,7 +1861,7 @@ static void arizona_enable_fll(struct arizona_fll *fll)
ARIZONA_FLL1_SYNC_ENA, 0);
} else {
arizona_fll_err(fll, "No clocks provided\n");
- return;
+ return -EINVAL;
}
/*
@@ -1681,25 +1876,29 @@ static void arizona_enable_fll(struct arizona_fll *fll)
ARIZONA_FLL1_SYNC_BW,
ARIZONA_FLL1_SYNC_BW);
- if (!arizona_is_enabled_fll(fll))
+ if (!already_enabled)
pm_runtime_get(arizona->dev);
/* Clear any pending completions */
try_wait_for_completion(&fll->ok);
regmap_update_bits_async(arizona->regmap, fll->base + 1,
- ARIZONA_FLL1_FREERUN, 0);
- regmap_update_bits_async(arizona->regmap, fll->base + 1,
ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA);
if (use_sync)
regmap_update_bits_async(arizona->regmap, fll->base + 0x11,
ARIZONA_FLL1_SYNC_ENA,
ARIZONA_FLL1_SYNC_ENA);
+ if (already_enabled)
+ regmap_update_bits_async(arizona->regmap, fll->base + 1,
+ ARIZONA_FLL1_FREERUN, 0);
+
ret = wait_for_completion_timeout(&fll->ok,
msecs_to_jiffies(250));
if (ret == 0)
arizona_fll_warn(fll, "Timed out waiting for lock\n");
+
+ return 0;
}
static void arizona_disable_fll(struct arizona_fll *fll)
@@ -1713,6 +1912,8 @@ static void arizona_disable_fll(struct arizona_fll *fll)
ARIZONA_FLL1_ENA, 0, &change);
regmap_update_bits(arizona->regmap, fll->base + 0x11,
ARIZONA_FLL1_SYNC_ENA, 0);
+ regmap_update_bits_async(arizona->regmap, fll->base + 1,
+ ARIZONA_FLL1_FREERUN, 0);
if (change)
pm_runtime_put_autosuspend(arizona->dev);
@@ -1721,7 +1922,7 @@ static void arizona_disable_fll(struct arizona_fll *fll)
int arizona_set_fll_refclk(struct arizona_fll *fll, int source,
unsigned int Fref, unsigned int Fout)
{
- int ret;
+ int ret = 0;
if (fll->ref_src == source && fll->ref_freq == Fref)
return 0;
@@ -1736,17 +1937,17 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source,
fll->ref_freq = Fref;
if (fll->fout && Fref > 0) {
- arizona_enable_fll(fll);
+ ret = arizona_enable_fll(fll);
}
- return 0;
+ return ret;
}
EXPORT_SYMBOL_GPL(arizona_set_fll_refclk);
int arizona_set_fll(struct arizona_fll *fll, int source,
unsigned int Fref, unsigned int Fout)
{
- int ret;
+ int ret = 0;
if (fll->sync_src == source &&
fll->sync_freq == Fref && fll->fout == Fout)
@@ -1768,13 +1969,12 @@ int arizona_set_fll(struct arizona_fll *fll, int source,
fll->sync_freq = Fref;
fll->fout = Fout;
- if (Fout) {
- arizona_enable_fll(fll);
- } else {
+ if (Fout)
+ ret = arizona_enable_fll(fll);
+ else
arizona_disable_fll(fll);
- }
- return 0;
+ return ret;
}
EXPORT_SYMBOL_GPL(arizona_set_fll);
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 05ae17f5bca3..942cfb197b6d 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -249,6 +249,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source,
extern int arizona_init_spk(struct snd_soc_codec *codec);
extern int arizona_init_gpio(struct snd_soc_codec *codec);
+extern int arizona_init_mono(struct snd_soc_codec *codec);
extern int arizona_init_dai(struct arizona_priv *priv, int dai);
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
new file mode 100644
index 000000000000..a20b30ca52c0
--- /dev/null
+++ b/sound/soc/codecs/cs4265.c
@@ -0,0 +1,682 @@
+/*
+ * cs4265.c -- CS4265 ALSA SoC audio driver
+ *
+ * Copyright 2014 Cirrus Logic, Inc.
+ *
+ * Author: Paul Handrigan <paul.handrigan@cirrus.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/kernel.h>
+#include <linux/gpio/consumer.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/i2c.h>
+#include <linux/input.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include "cs4265.h"
+
+struct cs4265_private {
+ struct device *dev;
+ struct regmap *regmap;
+ struct gpio_desc *reset_gpio;
+ u8 format;
+ u32 sysclk;
+};
+
+static const struct reg_default cs4265_reg_defaults[] = {
+ { CS4265_PWRCTL, 0x0F },
+ { CS4265_DAC_CTL, 0x08 },
+ { CS4265_ADC_CTL, 0x00 },
+ { CS4265_MCLK_FREQ, 0x00 },
+ { CS4265_SIG_SEL, 0x40 },
+ { CS4265_CHB_PGA_CTL, 0x00 },
+ { CS4265_CHA_PGA_CTL, 0x00 },
+ { CS4265_ADC_CTL2, 0x19 },
+ { CS4265_DAC_CHA_VOL, 0x00 },
+ { CS4265_DAC_CHB_VOL, 0x00 },
+ { CS4265_DAC_CTL2, 0xC0 },
+ { CS4265_SPDIF_CTL1, 0x00 },
+ { CS4265_SPDIF_CTL2, 0x00 },
+ { CS4265_INT_MASK, 0x00 },
+ { CS4265_STATUS_MODE_MSB, 0x00 },
+ { CS4265_STATUS_MODE_LSB, 0x00 },
+};
+
+static bool cs4265_readable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CS4265_PWRCTL:
+ case CS4265_DAC_CTL:
+ case CS4265_ADC_CTL:
+ case CS4265_MCLK_FREQ:
+ case CS4265_SIG_SEL:
+ case CS4265_CHB_PGA_CTL:
+ case CS4265_CHA_PGA_CTL:
+ case CS4265_ADC_CTL2:
+ case CS4265_DAC_CHA_VOL:
+ case CS4265_DAC_CHB_VOL:
+ case CS4265_DAC_CTL2:
+ case CS4265_SPDIF_CTL1:
+ case CS4265_SPDIF_CTL2:
+ case CS4265_INT_MASK:
+ case CS4265_STATUS_MODE_MSB:
+ case CS4265_STATUS_MODE_LSB:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool cs4265_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CS4265_INT_STATUS:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static DECLARE_TLV_DB_SCALE(pga_tlv, -1200, 50, 0);
+
+static DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 0);
+
+static const char * const digital_input_mux_text[] = {
+ "SDIN1", "SDIN2"
+};
+
+static SOC_ENUM_SINGLE_DECL(digital_input_mux_enum, CS4265_SIG_SEL, 7,
+ digital_input_mux_text);
+
+static const struct snd_kcontrol_new digital_input_mux =
+ SOC_DAPM_ENUM("Digital Input Mux", digital_input_mux_enum);
+
+static const char * const mic_linein_text[] = {
+ "MIC", "LINEIN"
+};
+
+static SOC_ENUM_SINGLE_DECL(mic_linein_enum, CS4265_ADC_CTL2, 0,
+ mic_linein_text);
+
+static const char * const cam_mode_text[] = {
+ "One Byte", "Two Byte"
+};
+
+static SOC_ENUM_SINGLE_DECL(cam_mode_enum, CS4265_SPDIF_CTL1, 5,
+ cam_mode_text);
+
+static const char * const cam_mono_stereo_text[] = {
+ "Stereo", "Mono"
+};
+
+static SOC_ENUM_SINGLE_DECL(spdif_mono_stereo_enum, CS4265_SPDIF_CTL2, 2,
+ cam_mono_stereo_text);
+
+static const char * const mono_select_text[] = {
+ "Channel A", "Channel B"
+};
+
+static SOC_ENUM_SINGLE_DECL(spdif_mono_select_enum, CS4265_SPDIF_CTL2, 0,
+ mono_select_text);
+
+static const struct snd_kcontrol_new mic_linein_mux =
+ SOC_DAPM_ENUM("ADC Input Capture Mux", mic_linein_enum);
+
+static const struct snd_kcontrol_new loopback_ctl =
+ SOC_DAPM_SINGLE("Switch", CS4265_SIG_SEL, 1, 1, 0);
+
+static const struct snd_kcontrol_new spdif_switch =
+ SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 0, 0);
+
+static const struct snd_kcontrol_new dac_switch =
+ SOC_DAPM_SINGLE("Switch", CS4265_PWRCTL, 1, 1, 0);
+
+static const struct snd_kcontrol_new cs4265_snd_controls[] = {
+
+ SOC_DOUBLE_R_SX_TLV("PGA Volume", CS4265_CHA_PGA_CTL,
+ CS4265_CHB_PGA_CTL, 0, 0x28, 0x30, pga_tlv),
+ SOC_DOUBLE_R_TLV("DAC Volume", CS4265_DAC_CHA_VOL,
+ CS4265_DAC_CHB_VOL, 0, 0xFF, 1, dac_tlv),
+ SOC_SINGLE("De-emp 44.1kHz Switch", CS4265_DAC_CTL, 1,
+ 1, 0),
+ SOC_SINGLE("DAC INV Switch", CS4265_DAC_CTL2, 5,
+ 1, 0),
+ SOC_SINGLE("DAC Zero Cross Switch", CS4265_DAC_CTL2, 6,
+ 1, 0),
+ SOC_SINGLE("DAC Soft Ramp Switch", CS4265_DAC_CTL2, 7,
+ 1, 0),
+ SOC_SINGLE("ADC HPF Switch", CS4265_ADC_CTL, 1,
+ 1, 0),
+ SOC_SINGLE("ADC Zero Cross Switch", CS4265_ADC_CTL2, 3,
+ 1, 1),
+ SOC_SINGLE("ADC Soft Ramp Switch", CS4265_ADC_CTL2, 7,
+ 1, 0),
+ SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1,
+ 6, 1, 0),
+ SOC_ENUM("C Data Access", cam_mode_enum),
+ SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2,
+ 3, 1, 0),
+ SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum),
+ SOC_SINGLE("MMTLR Data Switch", 0,
+ 1, 1, 0),
+ SOC_ENUM("Mono Channel Select", spdif_mono_select_enum),
+ SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24),
+};
+
+static const struct snd_soc_dapm_widget cs4265_dapm_widgets[] = {
+
+ SND_SOC_DAPM_INPUT("LINEINL"),
+ SND_SOC_DAPM_INPUT("LINEINR"),
+ SND_SOC_DAPM_INPUT("MICL"),
+ SND_SOC_DAPM_INPUT("MICR"),
+
+ SND_SOC_DAPM_AIF_OUT("DOUT", NULL, 0,
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SPDIFOUT", NULL, 0,
+ SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_MUX("ADC Mux", SND_SOC_NOPM, 0, 0, &mic_linein_mux),
+
+ SND_SOC_DAPM_ADC("ADC", NULL, CS4265_PWRCTL, 2, 1),
+ SND_SOC_DAPM_PGA("Pre-amp MIC", CS4265_PWRCTL, 3,
+ 1, NULL, 0),
+
+ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM,
+ 0, 0, &digital_input_mux),
+
+ SND_SOC_DAPM_MIXER("SDIN1 Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SDIN2 Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SPDIF Transmitter", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_SWITCH("Loopback", SND_SOC_NOPM, 0, 0,
+ &loopback_ctl),
+ SND_SOC_DAPM_SWITCH("SPDIF", SND_SOC_NOPM, 0, 0,
+ &spdif_switch),
+ SND_SOC_DAPM_SWITCH("DAC", CS4265_PWRCTL, 1, 1,
+ &dac_switch),
+
+ SND_SOC_DAPM_AIF_IN("DIN1", NULL, 0,
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("DIN2", NULL, 0,
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("TXIN", NULL, 0,
+ CS4265_SPDIF_CTL2, 5, 1),
+
+ SND_SOC_DAPM_OUTPUT("LINEOUTL"),
+ SND_SOC_DAPM_OUTPUT("LINEOUTR"),
+
+};
+
+static const struct snd_soc_dapm_route cs4265_audio_map[] = {
+
+ {"DIN1", NULL, "DAI1 Playback"},
+ {"DIN2", NULL, "DAI2 Playback"},
+ {"SDIN1 Input Mixer", NULL, "DIN1"},
+ {"SDIN2 Input Mixer", NULL, "DIN2"},
+ {"Input Mux", "SDIN1", "SDIN1 Input Mixer"},
+ {"Input Mux", "SDIN2", "SDIN2 Input Mixer"},
+ {"DAC", "Switch", "Input Mux"},
+ {"SPDIF", "Switch", "Input Mux"},
+ {"LINEOUTL", NULL, "DAC"},
+ {"LINEOUTR", NULL, "DAC"},
+ {"SPDIFOUT", NULL, "SPDIF"},
+
+ {"ADC Mux", "LINEIN", "LINEINL"},
+ {"ADC Mux", "LINEIN", "LINEINR"},
+ {"ADC Mux", "MIC", "MICL"},
+ {"ADC Mux", "MIC", "MICR"},
+ {"ADC", NULL, "ADC Mux"},
+ {"DOUT", NULL, "ADC"},
+ {"DAI1 Capture", NULL, "DOUT"},
+ {"DAI2 Capture", NULL, "DOUT"},
+
+ /* Loopback */
+ {"Loopback", "Switch", "ADC"},
+ {"DAC", NULL, "Loopback"},
+};
+
+struct cs4265_clk_para {
+ u32 mclk;
+ u32 rate;
+ u8 fm_mode; /* values 1, 2, or 4 */
+ u8 mclkdiv;
+};
+
+static const struct cs4265_clk_para clk_map_table[] = {
+ /*32k*/
+ {8192000, 32000, 0, 0},
+ {12288000, 32000, 0, 1},
+ {16384000, 32000, 0, 2},
+ {24576000, 32000, 0, 3},
+ {32768000, 32000, 0, 4},
+
+ /*44.1k*/
+ {11289600, 44100, 0, 0},
+ {16934400, 44100, 0, 1},
+ {22579200, 44100, 0, 2},
+ {33868000, 44100, 0, 3},
+ {45158400, 44100, 0, 4},
+
+ /*48k*/
+ {12288000, 48000, 0, 0},
+ {18432000, 48000, 0, 1},
+ {24576000, 48000, 0, 2},
+ {36864000, 48000, 0, 3},
+ {49152000, 48000, 0, 4},
+
+ /*64k*/
+ {8192000, 64000, 1, 0},
+ {1228800, 64000, 1, 1},
+ {1693440, 64000, 1, 2},
+ {2457600, 64000, 1, 3},
+ {3276800, 64000, 1, 4},
+
+ /* 88.2k */
+ {11289600, 88200, 1, 0},
+ {16934400, 88200, 1, 1},
+ {22579200, 88200, 1, 2},
+ {33868000, 88200, 1, 3},
+ {45158400, 88200, 1, 4},
+
+ /* 96k */
+ {12288000, 96000, 1, 0},
+ {18432000, 96000, 1, 1},
+ {24576000, 96000, 1, 2},
+ {36864000, 96000, 1, 3},
+ {49152000, 96000, 1, 4},
+
+ /* 128k */
+ {8192000, 128000, 2, 0},
+ {12288000, 128000, 2, 1},
+ {16934400, 128000, 2, 2},
+ {24576000, 128000, 2, 3},
+ {32768000, 128000, 2, 4},
+
+ /* 176.4k */
+ {11289600, 176400, 2, 0},
+ {16934400, 176400, 2, 1},
+ {22579200, 176400, 2, 2},
+ {33868000, 176400, 2, 3},
+ {49152000, 176400, 2, 4},
+
+ /* 192k */
+ {12288000, 192000, 2, 0},
+ {18432000, 192000, 2, 1},
+ {24576000, 192000, 2, 2},
+ {36864000, 192000, 2, 3},
+ {49152000, 192000, 2, 4},
+};
+
+static int cs4265_get_clk_index(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
+ if (clk_map_table[i].rate == rate &&
+ clk_map_table[i].mclk == mclk)
+ return i;
+ }
+ return -EINVAL;
+}
+
+static int cs4265_set_sysclk(struct snd_soc_dai *codec_dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec);
+ int i;
+
+ if (clk_id != 0) {
+ dev_err(codec->dev, "Invalid clk_id %d\n", clk_id);
+ return -EINVAL;
+ }
+ for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
+ if (clk_map_table[i].mclk == freq) {
+ cs4265->sysclk = freq;
+ return 0;
+ }
+ }
+ cs4265->sysclk = 0;
+ dev_err(codec->dev, "Invalid freq parameter %d\n", freq);
+ return -EINVAL;
+}
+
+static int cs4265_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec);
+ u8 iface = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ snd_soc_update_bits(codec, CS4265_ADC_CTL,
+ CS4265_ADC_MASTER,
+ CS4265_ADC_MASTER);
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ snd_soc_update_bits(codec, CS4265_ADC_CTL,
+ CS4265_ADC_MASTER,
+ 0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= SND_SOC_DAIFMT_I2S;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ iface |= SND_SOC_DAIFMT_RIGHT_J;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= SND_SOC_DAIFMT_LEFT_J;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ cs4265->format = iface;
+ return 0;
+}
+
+static int cs4265_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ if (mute) {
+ snd_soc_update_bits(codec, CS4265_DAC_CTL,
+ CS4265_DAC_CTL_MUTE,
+ CS4265_DAC_CTL_MUTE);
+ snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
+ CS4265_SPDIF_CTL2_MUTE,
+ CS4265_SPDIF_CTL2_MUTE);
+ } else {
+ snd_soc_update_bits(codec, CS4265_DAC_CTL,
+ CS4265_DAC_CTL_MUTE,
+ 0);
+ snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
+ CS4265_SPDIF_CTL2_MUTE,
+ 0);
+ }
+ return 0;
+}
+
+static int cs4265_pcm_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 cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec);
+ int index;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
+ ((cs4265->format & SND_SOC_DAIFMT_FORMAT_MASK)
+ == SND_SOC_DAIFMT_RIGHT_J))
+ return -EINVAL;
+
+ index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params));
+ if (index >= 0) {
+ snd_soc_update_bits(codec, CS4265_ADC_CTL,
+ CS4265_ADC_FM, clk_map_table[index].fm_mode);
+ snd_soc_update_bits(codec, CS4265_MCLK_FREQ,
+ CS4265_MCLK_FREQ_MASK,
+ clk_map_table[index].mclkdiv);
+
+ } else {
+ dev_err(codec->dev, "can't get correct mclk\n");
+ return -EINVAL;
+ }
+
+ switch (cs4265->format & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ snd_soc_update_bits(codec, CS4265_DAC_CTL,
+ CS4265_DAC_CTL_DIF, (1 << 4));
+ snd_soc_update_bits(codec, CS4265_ADC_CTL,
+ CS4265_ADC_DIF, (1 << 4));
+ snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
+ CS4265_SPDIF_CTL2_DIF, (1 << 6));
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ if (params_width(params) == 16) {
+ snd_soc_update_bits(codec, CS4265_DAC_CTL,
+ CS4265_DAC_CTL_DIF, (1 << 5));
+ snd_soc_update_bits(codec, CS4265_ADC_CTL,
+ CS4265_SPDIF_CTL2_DIF, (1 << 7));
+ } else {
+ snd_soc_update_bits(codec, CS4265_DAC_CTL,
+ CS4265_DAC_CTL_DIF, (3 << 5));
+ snd_soc_update_bits(codec, CS4265_ADC_CTL,
+ CS4265_SPDIF_CTL2_DIF, (1 << 7));
+ }
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ snd_soc_update_bits(codec, CS4265_DAC_CTL,
+ CS4265_DAC_CTL_DIF, 0);
+ snd_soc_update_bits(codec, CS4265_ADC_CTL,
+ CS4265_ADC_DIF, 0);
+ snd_soc_update_bits(codec, CS4265_ADC_CTL,
+ CS4265_SPDIF_CTL2_DIF, (1 << 6));
+
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int cs4265_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ snd_soc_update_bits(codec, CS4265_PWRCTL,
+ CS4265_PWRCTL_PDN, 0);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ snd_soc_update_bits(codec, CS4265_PWRCTL,
+ CS4265_PWRCTL_PDN,
+ CS4265_PWRCTL_PDN);
+ break;
+ case SND_SOC_BIAS_OFF:
+ snd_soc_update_bits(codec, CS4265_PWRCTL,
+ CS4265_PWRCTL_PDN,
+ CS4265_PWRCTL_PDN);
+ break;
+ }
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+#define CS4265_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
+#define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE)
+
+static const struct snd_soc_dai_ops cs4265_ops = {
+ .hw_params = cs4265_pcm_hw_params,
+ .digital_mute = cs4265_digital_mute,
+ .set_fmt = cs4265_set_fmt,
+ .set_sysclk = cs4265_set_sysclk,
+};
+
+static struct snd_soc_dai_driver cs4265_dai[] = {
+ {
+ .name = "cs4265-dai1",
+ .playback = {
+ .stream_name = "DAI1 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = CS4265_RATES,
+ .formats = CS4265_FORMATS,
+ },
+ .capture = {
+ .stream_name = "DAI1 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = CS4265_RATES,
+ .formats = CS4265_FORMATS,
+ },
+ .ops = &cs4265_ops,
+ },
+ {
+ .name = "cs4265-dai2",
+ .playback = {
+ .stream_name = "DAI2 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = CS4265_RATES,
+ .formats = CS4265_FORMATS,
+ },
+ .capture = {
+ .stream_name = "DAI2 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = CS4265_RATES,
+ .formats = CS4265_FORMATS,
+ },
+ .ops = &cs4265_ops,
+ },
+};
+
+static const struct snd_soc_codec_driver soc_codec_cs4265 = {
+ .set_bias_level = cs4265_set_bias_level,
+
+ .dapm_widgets = cs4265_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cs4265_dapm_widgets),
+ .dapm_routes = cs4265_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(cs4265_audio_map),
+
+ .controls = cs4265_snd_controls,
+ .num_controls = ARRAY_SIZE(cs4265_snd_controls),
+};
+
+static const struct regmap_config cs4265_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = CS4265_MAX_REGISTER,
+ .reg_defaults = cs4265_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(cs4265_reg_defaults),
+ .readable_reg = cs4265_readable_register,
+ .volatile_reg = cs4265_volatile_register,
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static int cs4265_i2c_probe(struct i2c_client *i2c_client,
+ const struct i2c_device_id *id)
+{
+ struct cs4265_private *cs4265;
+ int ret = 0;
+ unsigned int devid = 0;
+ unsigned int reg;
+
+ cs4265 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4265_private),
+ GFP_KERNEL);
+ if (cs4265 == NULL)
+ return -ENOMEM;
+ cs4265->dev = &i2c_client->dev;
+
+ cs4265->regmap = devm_regmap_init_i2c(i2c_client, &cs4265_regmap);
+ if (IS_ERR(cs4265->regmap)) {
+ ret = PTR_ERR(cs4265->regmap);
+ dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
+ return ret;
+ }
+
+ cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev,
+ "reset-gpios");
+ if (IS_ERR(cs4265->reset_gpio)) {
+ ret = PTR_ERR(cs4265->reset_gpio);
+ if (ret != -ENOENT && ret != -ENOSYS)
+ return ret;
+
+ cs4265->reset_gpio = NULL;
+ } else {
+ ret = gpiod_direction_output(cs4265->reset_gpio, 0);
+ if (ret)
+ return ret;
+ mdelay(1);
+ gpiod_set_value_cansleep(cs4265->reset_gpio, 1);
+
+ }
+
+ i2c_set_clientdata(i2c_client, cs4265);
+
+ ret = regmap_read(cs4265->regmap, CS4265_CHIP_ID, &reg);
+ devid = reg & CS4265_CHIP_ID_MASK;
+ if (devid != CS4265_CHIP_ID_VAL) {
+ ret = -ENODEV;
+ dev_err(&i2c_client->dev,
+ "CS4265 Device ID (%X). Expected %X\n",
+ devid, CS4265_CHIP_ID);
+ return ret;
+ }
+ dev_info(&i2c_client->dev,
+ "CS4265 Version %x\n",
+ reg & CS4265_REV_ID_MASK);
+
+ regmap_write(cs4265->regmap, CS4265_PWRCTL, 0x0F);
+
+ ret = snd_soc_register_codec(&i2c_client->dev,
+ &soc_codec_cs4265, cs4265_dai,
+ ARRAY_SIZE(cs4265_dai));
+ return ret;
+}
+
+static int cs4265_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static const struct of_device_id cs4265_of_match[] = {
+ { .compatible = "cirrus,cs4265", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, cs4265_of_match);
+
+static const struct i2c_device_id cs4265_id[] = {
+ { "cs4265", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, cs4265_id);
+
+static struct i2c_driver cs4265_i2c_driver = {
+ .driver = {
+ .name = "cs4265",
+ .owner = THIS_MODULE,
+ .of_match_table = cs4265_of_match,
+ },
+ .id_table = cs4265_id,
+ .probe = cs4265_i2c_probe,
+ .remove = cs4265_i2c_remove,
+};
+
+module_i2c_driver(cs4265_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC CS4265 driver");
+MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <paul.handrigan@cirrus.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs4265.h b/sound/soc/codecs/cs4265.h
new file mode 100644
index 000000000000..0a80a8dcec67
--- /dev/null
+++ b/sound/soc/codecs/cs4265.h
@@ -0,0 +1,64 @@
+/*
+ * cs4265.h -- CS4265 ALSA SoC audio driver
+ *
+ * Copyright 2014 Cirrus Logic, Inc.
+ *
+ * Author: Paul Handrigan <paul.handrigan@cirrus.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 __CS4265_H__
+#define __CS4265_H__
+
+#define CS4265_CHIP_ID 0x1
+#define CS4265_CHIP_ID_VAL 0xD0
+#define CS4265_CHIP_ID_MASK 0xF0
+#define CS4265_REV_ID_MASK 0x0F
+
+#define CS4265_PWRCTL 0x02
+#define CS4265_PWRCTL_PDN 1
+
+#define CS4265_DAC_CTL 0x3
+#define CS4265_DAC_CTL_MUTE (1 << 2)
+#define CS4265_DAC_CTL_DIF (3 << 4)
+
+#define CS4265_ADC_CTL 0x4
+#define CS4265_ADC_MASTER 1
+#define CS4265_ADC_DIF (1 << 4)
+#define CS4265_ADC_FM (3 << 6)
+
+#define CS4265_MCLK_FREQ 0x5
+#define CS4265_MCLK_FREQ_MASK (7 << 4)
+
+#define CS4265_SIG_SEL 0x6
+#define CS4265_SIG_SEL_LOOP (1 << 1)
+
+#define CS4265_CHB_PGA_CTL 0x7
+#define CS4265_CHA_PGA_CTL 0x8
+
+#define CS4265_ADC_CTL2 0x9
+
+#define CS4265_DAC_CHA_VOL 0xA
+#define CS4265_DAC_CHB_VOL 0xB
+
+#define CS4265_DAC_CTL2 0xC
+
+#define CS4265_INT_STATUS 0xD
+#define CS4265_INT_MASK 0xE
+#define CS4265_STATUS_MODE_MSB 0xF
+#define CS4265_STATUS_MODE_LSB 0x10
+
+#define CS4265_SPDIF_CTL1 0x11
+
+#define CS4265_SPDIF_CTL2 0x12
+#define CS4265_SPDIF_CTL2_MUTE (1 << 4)
+#define CS4265_SPDIF_CTL2_DIF (3 << 6)
+
+#define CS4265_C_DATA_BUFF 0x13
+#define CS4265_MAX_REGISTER 0x2A
+
+#endif
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 9947a9583679..e6d4ff9fd992 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -664,10 +664,8 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
cs4270 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4270_private),
GFP_KERNEL);
- if (!cs4270) {
- dev_err(&i2c_client->dev, "could not allocate codec\n");
+ if (!cs4270)
return -ENOMEM;
- }
/* get the power supply regulators */
for (i = 0; i < ARRAY_SIZE(supply_names); i++)
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 071fc77f2f06..969167d8b71e 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -399,15 +399,15 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv),
SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL,
- CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv),
+ CS42L52_HPB_VOL, 0, 0x34, 0xC0, hpd_tlv),
SOC_ENUM("Headphone Analog Gain", hp_gain_enum),
SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL,
- CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv),
+ CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv),
SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
- CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv),
+ CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv),
SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0),
@@ -417,10 +417,10 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
SOC_ENUM("MIC Bias Level", mic_bias_level_enum),
SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL,
- CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv),
+ CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv),
SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL,
- 6, 0x7f, 0x19, ipd_tlv),
+ 0, 0x19, 0x7F, ipd_tlv),
SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0),
@@ -428,11 +428,11 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
CS42L52_ADCB_MIXER_VOL, 7, 1, 1),
SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL,
- CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv),
+ CS42L52_PGAB_CTL, 0, 0x28, 0x24, pga_tlv),
SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
- 0, 0x7f, 0x19, mix_tlv),
+ 0, 0x19, 0x7f, mix_tlv),
SOC_DOUBLE_R("PCM Mixer Switch",
CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index fdc4bd27b0df..c766a5a9ce80 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -318,24 +318,32 @@ static const struct soc_enum adca_swap_enum =
ARRAY_SIZE(left_swap_text),
left_swap_text,
swap_values);
+static const struct snd_kcontrol_new adca_swap_mux =
+ SOC_DAPM_ENUM("Route", adca_swap_enum);
static const struct soc_enum pcma_swap_enum =
SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 4, 3,
ARRAY_SIZE(left_swap_text),
left_swap_text,
swap_values);
+static const struct snd_kcontrol_new pcma_swap_mux =
+ SOC_DAPM_ENUM("Route", pcma_swap_enum);
static const struct soc_enum adcb_swap_enum =
SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 2, 3,
ARRAY_SIZE(right_swap_text),
right_swap_text,
swap_values);
+static const struct snd_kcontrol_new adcb_swap_mux =
+ SOC_DAPM_ENUM("Route", adcb_swap_enum);
static const struct soc_enum pcmb_swap_enum =
SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 6, 3,
ARRAY_SIZE(right_swap_text),
right_swap_text,
swap_values);
+static const struct snd_kcontrol_new pcmb_swap_mux =
+ SOC_DAPM_ENUM("Route", pcmb_swap_enum);
static const struct snd_kcontrol_new hpa_switch =
SOC_DAPM_SINGLE("Switch", CS42L56_PWRCTL_2, 6, 1, 1);
@@ -421,15 +429,15 @@ static const struct soc_enum ng_delay_enum =
static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L56_MASTER_A_VOLUME,
- CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xfd, adv_tlv),
+ CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xE4, adv_tlv),
SOC_DOUBLE("Master Mute Switch", CS42L56_DSP_MUTE_CTL, 0, 1, 1, 1),
SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L56_ADCA_MIX_VOLUME,
- CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv),
+ CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv),
SOC_DOUBLE("ADC Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 6, 7, 1, 1),
SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L56_PCMA_MIX_VOLUME,
- CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv),
+ CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv),
SOC_DOUBLE("PCM Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 4, 5, 1, 1),
SOC_SINGLE_TLV("Analog Advisory Volume",
@@ -438,16 +446,16 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
CS42L56_DIGINPUT_ADV_VOLUME, 0, 0x00, 1, adv_tlv),
SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L56_PGAA_MUX_VOLUME,
- CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0xfd, pga_tlv),
+ CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0x24, pga_tlv),
SOC_DOUBLE_R_TLV("ADC Volume", CS42L56_ADCA_ATTENUATOR,
CS42L56_ADCB_ATTENUATOR, 0, 0x00, 1, adc_tlv),
SOC_DOUBLE("ADC Mute Switch", CS42L56_MISC_ADC_CTL, 2, 3, 1, 1),
SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1),
SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME,
- CS42L56_HPA_VOLUME, 0, 0x44, 0x55, hl_tlv),
+ CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv),
SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME,
- CS42L56_LOA_VOLUME, 0, 0x44, 0x55, hl_tlv),
+ CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv),
SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL,
0, 0x00, 1, tone_tlv),
@@ -467,11 +475,6 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
SOC_SINGLE("ADCA Invert", CS42L56_MISC_ADC_CTL, 2, 1, 1),
SOC_SINGLE("ADCB Invert", CS42L56_MISC_ADC_CTL, 3, 1, 1),
- SOC_ENUM("PCMA Swap", pcma_swap_enum),
- SOC_ENUM("PCMB Swap", pcmb_swap_enum),
- SOC_ENUM("ADCA Swap", adca_swap_enum),
- SOC_ENUM("ADCB Swap", adcb_swap_enum),
-
SOC_DOUBLE("HPF Switch", CS42L56_HPF_CTL, 5, 7, 1, 1),
SOC_DOUBLE("HPF Freeze Switch", CS42L56_HPF_CTL, 4, 6, 1, 1),
SOC_ENUM("HPFA Corner Freq", hpfa_freq_enum),
@@ -570,6 +573,16 @@ static const struct snd_soc_dapm_widget cs42l56_dapm_widgets[] = {
SND_SOC_DAPM_ADC("ADCA", NULL, CS42L56_PWRCTL_1, 1, 1),
SND_SOC_DAPM_ADC("ADCB", NULL, CS42L56_PWRCTL_1, 2, 1),
+ SND_SOC_DAPM_MUX("ADCA Swap Mux", SND_SOC_NOPM, 0, 0,
+ &adca_swap_mux),
+ SND_SOC_DAPM_MUX("ADCB Swap Mux", SND_SOC_NOPM, 0, 0,
+ &adcb_swap_mux),
+
+ SND_SOC_DAPM_MUX("PCMA Swap Mux", SND_SOC_NOPM, 0, 0,
+ &pcma_swap_mux),
+ SND_SOC_DAPM_MUX("PCMB Swap Mux", SND_SOC_NOPM, 0, 0,
+ &pcmb_swap_mux),
+
SND_SOC_DAPM_DAC("DACA", NULL, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC("DACB", NULL, SND_SOC_NOPM, 0, 0),
@@ -607,8 +620,19 @@ static const struct snd_soc_dapm_route cs42l56_audio_map[] = {
{"Digital Output Mux", NULL, "ADCA"},
{"Digital Output Mux", NULL, "ADCB"},
- {"ADCB", NULL, "ADCB Mux"},
- {"ADCA", NULL, "ADCA Mux"},
+ {"ADCB", NULL, "ADCB Swap Mux"},
+ {"ADCA", NULL, "ADCA Swap Mux"},
+
+ {"ADCA Swap Mux", NULL, "ADCA"},
+ {"ADCB Swap Mux", NULL, "ADCB"},
+
+ {"DACA", "Left", "ADCA Swap Mux"},
+ {"DACA", "LR 2", "ADCA Swap Mux"},
+ {"DACA", "Right", "ADCA Swap Mux"},
+
+ {"DACB", "Left", "ADCB Swap Mux"},
+ {"DACB", "LR 2", "ADCB Swap Mux"},
+ {"DACB", "Right", "ADCB Swap Mux"},
{"ADCA Mux", NULL, "AIN3A"},
{"ADCA Mux", NULL, "AIN2A"},
@@ -633,30 +657,32 @@ static const struct snd_soc_dapm_route cs42l56_audio_map[] = {
{"PGAB Input Mux", NULL, "AIN2B"},
{"PGAB Input Mux", NULL, "AIN3B"},
- {"LOB", NULL, "Lineout Right"},
- {"LOA", NULL, "Lineout Left"},
-
- {"Lineout Right", "Switch", "LINEOUTB Input Mux"},
- {"Lineout Left", "Switch", "LINEOUTA Input Mux"},
+ {"LOB", "Switch", "LINEOUTB Input Mux"},
+ {"LOA", "Switch", "LINEOUTA Input Mux"},
{"LINEOUTA Input Mux", "PGAA", "PGAA"},
{"LINEOUTB Input Mux", "PGAB", "PGAB"},
{"LINEOUTA Input Mux", "DACA", "DACA"},
{"LINEOUTB Input Mux", "DACB", "DACB"},
- {"HPA", NULL, "Headphone Left"},
- {"HPB", NULL, "Headphone Right"},
-
- {"Headphone Right", "Switch", "HPB Input Mux"},
- {"Headphone Left", "Switch", "HPA Input Mux"},
+ {"HPA", "Switch", "HPB Input Mux"},
+ {"HPB", "Switch", "HPA Input Mux"},
{"HPA Input Mux", "PGAA", "PGAA"},
{"HPB Input Mux", "PGAB", "PGAB"},
{"HPA Input Mux", "DACA", "DACA"},
{"HPB Input Mux", "DACB", "DACB"},
- {"DACB", NULL, "HiFi Playback"},
- {"DACA", NULL, "HiFi Playback"},
+ {"DACA", NULL, "PCMA Swap Mux"},
+ {"DACB", NULL, "PCMB Swap Mux"},
+
+ {"PCMB Swap Mux", "Left", "HiFi Playback"},
+ {"PCMB Swap Mux", "LR 2", "HiFi Playback"},
+ {"PCMB Swap Mux", "Right", "HiFi Playback"},
+
+ {"PCMA Swap Mux", "Left", "HiFi Playback"},
+ {"PCMA Swap Mux", "LR 2", "HiFi Playback"},
+ {"PCMA Swap Mux", "Right", "HiFi Playback"},
};
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index ae3717992d56..0e7b9eb2ba61 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -401,7 +401,7 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv),
SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL,
- CS42L73_MICBPREPGABVOL, 5, 0x34,
+ CS42L73_MICBPREPGABVOL, 0, 0x34,
0x24, micpga_tlv),
SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL,
@@ -1408,10 +1408,8 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client,
cs42l73 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l73_private),
GFP_KERNEL);
- if (!cs42l73) {
- dev_err(&i2c_client->dev, "could not allocate codec\n");
+ if (!cs42l73)
return -ENOMEM;
- }
cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap);
if (IS_ERR(cs42l73->regmap)) {
diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c
index a25bc6061a30..02b1520ae0bc 100644
--- a/sound/soc/codecs/cs42xx8.c
+++ b/sound/soc/codecs/cs42xx8.c
@@ -219,6 +219,9 @@ static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai,
case SND_SOC_DAIFMT_RIGHT_J:
val = CS42XX8_INTF_DAC_DIF_RIGHTJ | CS42XX8_INTF_ADC_DIF_RIGHTJ;
break;
+ case SND_SOC_DAIFMT_DSP_A:
+ val = CS42XX8_INTF_DAC_DIF_TDM | CS42XX8_INTF_ADC_DIF_TDM;
+ break;
default:
dev_err(codec->dev, "unsupported dai format\n");
return -EINVAL;
@@ -422,7 +425,7 @@ const struct cs42xx8_driver_data cs42888_data = {
};
EXPORT_SYMBOL_GPL(cs42888_data);
-const struct of_device_id cs42xx8_of_match[] = {
+static const struct of_device_id cs42xx8_of_match[] = {
{ .compatible = "cirrus,cs42448", .data = &cs42448_data, },
{ .compatible = "cirrus,cs42888", .data = &cs42888_data, },
{ /* sentinel */ }
diff --git a/sound/soc/codecs/cs42xx8.h b/sound/soc/codecs/cs42xx8.h
index da0b94aee419..b2c10e537ef6 100644
--- a/sound/soc/codecs/cs42xx8.h
+++ b/sound/soc/codecs/cs42xx8.h
@@ -128,8 +128,8 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap);
#define CS42XX8_INTF_DAC_DIF_RIGHTJ (2 << CS42XX8_INTF_DAC_DIF_SHIFT)
#define CS42XX8_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_DAC_DIF_SHIFT)
#define CS42XX8_INTF_DAC_DIF_ONELINE_20 (4 << CS42XX8_INTF_DAC_DIF_SHIFT)
-#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (6 << CS42XX8_INTF_DAC_DIF_SHIFT)
-#define CS42XX8_INTF_DAC_DIF_TDM (7 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (5 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_TDM (6 << CS42XX8_INTF_DAC_DIF_SHIFT)
#define CS42XX8_INTF_ADC_DIF_SHIFT 0
#define CS42XX8_INTF_ADC_DIF_WIDTH 3
#define CS42XX8_INTF_ADC_DIF_MASK (((1 << CS42XX8_INTF_ADC_DIF_WIDTH) - 1) << CS42XX8_INTF_ADC_DIF_SHIFT)
@@ -138,8 +138,8 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap);
#define CS42XX8_INTF_ADC_DIF_RIGHTJ (2 << CS42XX8_INTF_ADC_DIF_SHIFT)
#define CS42XX8_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_ADC_DIF_SHIFT)
#define CS42XX8_INTF_ADC_DIF_ONELINE_20 (4 << CS42XX8_INTF_ADC_DIF_SHIFT)
-#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (6 << CS42XX8_INTF_ADC_DIF_SHIFT)
-#define CS42XX8_INTF_ADC_DIF_TDM (7 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (5 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_TDM (6 << CS42XX8_INTF_ADC_DIF_SHIFT)
/* ADC Control & DAC De-Emphasis (Address 05h) */
#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT 7
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index d5fd00a64748..8f95b0300f1a 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -253,7 +253,7 @@ static void v253_close(struct tty_struct *tty)
/* Prevent the codec driver from further accessing the modem */
codec->hw_write = NULL;
cx20442->control_data = NULL;
- codec->card->pop_time = 0;
+ codec->component.card->pop_time = 0;
}
/* Line discipline .hangup() */
@@ -281,7 +281,7 @@ static void v253_receive(struct tty_struct *tty,
/* Set up codec driver access to modem controls */
cx20442->control_data = tty;
codec->hw_write = (hw_write_t)tty->ops->write;
- codec->card->pop_time = 1;
+ codec->component.card->pop_time = 1;
}
}
@@ -372,7 +372,7 @@ static int cx20442_codec_probe(struct snd_soc_codec *codec)
snd_soc_codec_set_drvdata(codec, cx20442);
codec->hw_write = NULL;
- codec->card->pop_time = 0;
+ codec->component.card->pop_time = 0;
return 0;
}
@@ -383,8 +383,8 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec)
struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec);
if (cx20442->control_data) {
- struct tty_struct *tty = cx20442->control_data;
- tty_hangup(tty);
+ struct tty_struct *tty = cx20442->control_data;
+ tty_hangup(tty);
}
if (!IS_ERR(cx20442->por)) {
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 9134982807b5..2cd3e5427441 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1299,12 +1299,12 @@ static int max98088_dai2_hw_params(struct snd_pcm_substream *substream,
rate = params_rate(params);
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
snd_soc_update_bits(codec, M98088_REG_1C_DAI2_FORMAT,
M98088_DAI_WS, 0);
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
snd_soc_update_bits(codec, M98088_REG_1C_DAI2_FORMAT,
M98088_DAI_WS, M98088_DAI_WS);
break;
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index f5fccc7a8e89..4a063fa88526 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -26,10 +26,6 @@
#include <sound/max98090.h>
#include "max98090.h"
-#define DEBUG
-#define EXTMIC_METHOD
-#define EXTMIC_METHOD_TEST
-
/* Allows for sparsely populated register maps */
static struct reg_default max98090_reg[] = {
{ 0x00, 0x00 }, /* 00 Software Reset */
@@ -820,7 +816,6 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w,
else
val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT;
-
if (val >= 1) {
if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) {
max98090->pa1en = val - 1; /* Update for volatile */
@@ -1140,7 +1135,6 @@ static const struct snd_kcontrol_new max98090_mixhprsel_mux =
SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum);
static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = {
-
SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2"),
SND_SOC_DAPM_INPUT("DMICL"),
@@ -1304,7 +1298,6 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = {
};
static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = {
-
SND_SOC_DAPM_INPUT("DMIC3"),
SND_SOC_DAPM_INPUT("DMIC4"),
@@ -1315,7 +1308,6 @@ static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = {
};
static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
-
{"MIC1 Input", NULL, "MIC1"},
{"MIC2 Input", NULL, "MIC2"},
@@ -1493,17 +1485,14 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
{"SPKR", NULL, "SPK Right Out"},
{"RCVL", NULL, "RCV Left Out"},
{"RCVR", NULL, "RCV Right Out"},
-
};
static const struct snd_soc_dapm_route max98091_dapm_routes[] = {
-
/* DMIC inputs */
{"DMIC3", NULL, "DMIC3_ENA"},
{"DMIC4", NULL, "DMIC4_ENA"},
{"DMIC3", NULL, "AHPF"},
{"DMIC4", NULL, "AHPF"},
-
};
static int max98090_add_widgets(struct snd_soc_codec *codec)
@@ -1531,7 +1520,6 @@ static int max98090_add_widgets(struct snd_soc_codec *codec)
snd_soc_dapm_add_routes(dapm, max98091_dapm_routes,
ARRAY_SIZE(max98091_dapm_routes));
-
}
return 0;
@@ -2212,22 +2200,11 @@ static struct snd_soc_dai_driver max98090_dai[] = {
}
};
-static void max98090_handle_pdata(struct snd_soc_codec *codec)
-{
- struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
- struct max98090_pdata *pdata = max98090->pdata;
-
- if (!pdata) {
- dev_err(codec->dev, "No platform data\n");
- return;
- }
-
-}
-
static int max98090_probe(struct snd_soc_codec *codec)
{
struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
struct max98090_cdata *cdata;
+ enum max98090_type devtype;
int ret = 0;
dev_dbg(codec->dev, "max98090_probe\n");
@@ -2263,16 +2240,21 @@ static int max98090_probe(struct snd_soc_codec *codec)
}
if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) {
- max98090->devtype = MAX98090;
+ devtype = MAX98090;
dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret);
} else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) {
- max98090->devtype = MAX98091;
+ devtype = MAX98091;
dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret);
} else {
- max98090->devtype = MAX98090;
+ devtype = MAX98090;
dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret);
}
+ if (max98090->devtype != devtype) {
+ dev_warn(codec->dev, "Mismatch in DT specified CODEC type.\n");
+ max98090->devtype = devtype;
+ }
+
max98090->jack_state = M98090_JACK_STATE_NO_HEADSET;
INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work);
@@ -2284,7 +2266,7 @@ static int max98090_probe(struct snd_soc_codec *codec)
/* Register for interrupts */
dev_dbg(codec->dev, "irq = %d\n", max98090->irq);
- ret = request_threaded_irq(max98090->irq, NULL,
+ ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL,
max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
"max98090_interrupt", codec);
if (ret < 0) {
@@ -2317,8 +2299,6 @@ static int max98090_probe(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE,
M98090_MBVSEL_MASK, M98090_MBVSEL_2V8);
- max98090_handle_pdata(codec);
-
max98090_add_widgets(codec);
err_access:
@@ -2428,7 +2408,7 @@ static int max98090_runtime_suspend(struct device *dev)
}
#endif
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int max98090_resume(struct device *dev)
{
struct max98090_priv *max98090 = dev_get_drvdata(dev);
@@ -2460,12 +2440,14 @@ static const struct dev_pm_ops max98090_pm = {
static const struct i2c_device_id max98090_i2c_id[] = {
{ "max98090", MAX98090 },
+ { "max98091", MAX98091 },
{ }
};
MODULE_DEVICE_TABLE(i2c, max98090_i2c_id);
static const struct of_device_id max98090_of_match[] = {
{ .compatible = "maxim,max98090", },
+ { .compatible = "maxim,max98091", },
{ }
};
MODULE_DEVICE_TABLE(of, max98090_of_match);
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 89ec00424880..0ee6797d5083 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -1280,12 +1280,12 @@ static int max98095_dai2_hw_params(struct snd_pcm_substream *substream,
rate = params_rate(params);
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
snd_soc_update_bits(codec, M98095_034_DAI2_FORMAT,
M98095_DAI_WS, 0);
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
snd_soc_update_bits(codec, M98095_034_DAI2_FORMAT,
M98095_DAI_WS, M98095_DAI_WS);
break;
@@ -1341,12 +1341,12 @@ static int max98095_dai3_hw_params(struct snd_pcm_substream *substream,
rate = params_rate(params);
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
snd_soc_update_bits(codec, M98095_03E_DAI3_FORMAT,
M98095_DAI_WS, 0);
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
snd_soc_update_bits(codec, M98095_03E_DAI3_FORMAT,
M98095_DAI_WS, M98095_DAI_WS);
break;
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 9965277b595a..388f90a597fa 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -766,11 +766,11 @@ static int __init mc13783_codec_probe(struct platform_device *pdev)
ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port);
if (ret)
- return ret;
+ goto out;
ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port);
if (ret)
- return ret;
+ goto out;
}
dev_set_drvdata(&pdev->dev, priv);
@@ -783,6 +783,8 @@ static int __init mc13783_codec_probe(struct platform_device *pdev)
ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async));
+out:
+ of_node_put(np);
return ret;
}
diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c
index 3a80ba4452df..57b0c94a710b 100644
--- a/sound/soc/codecs/pcm1792a.c
+++ b/sound/soc/codecs/pcm1792a.c
@@ -36,6 +36,7 @@
#define PCM1792A_DAC_VOL_LEFT 0x10
#define PCM1792A_DAC_VOL_RIGHT 0x11
#define PCM1792A_FMT_CONTROL 0x12
+#define PCM1792A_MODE_CONTROL 0x13
#define PCM1792A_SOFT_MUTE PCM1792A_FMT_CONTROL
#define PCM1792A_FMT_MASK 0x70
@@ -164,6 +165,8 @@ 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),
+ SOC_SINGLE("DAC Invert Output Switch", PCM1792A_MODE_CONTROL, 7, 1, 0),
+ SOC_SINGLE("DAC Rolloff Filter Switch", PCM1792A_MODE_CONTROL, 1, 1, 0),
};
static const struct snd_soc_dapm_widget pcm1792a_dapm_widgets[] = {
diff --git a/sound/soc/codecs/pcm1792a.h b/sound/soc/codecs/pcm1792a.h
index 7a83d1fc102a..51d5470fee16 100644
--- a/sound/soc/codecs/pcm1792a.h
+++ b/sound/soc/codecs/pcm1792a.h
@@ -18,7 +18,8 @@
#define __PCM1792A_H__
#define PCM1792A_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_8000_48000 | \
- SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000)
+ SNDRV_PCM_RATE_88200 | 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)
diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c
index 7b82fbe0d14c..56650d6c2f53 100644
--- a/sound/soc/codecs/rl6231.c
+++ b/sound/soc/codecs/rl6231.c
@@ -11,25 +11,6 @@
*/
#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/init.h>
-#include <linux/delay.h>
-#include <linux/pm.h>
-#include <linux/gpio.h>
-#include <linux/i2c.h>
-#include <linux/regmap.h>
-#include <linux/of.h>
-#include <linux/of_gpio.h>
-#include <linux/platform_device.h>
-#include <linux/spi/spi.h>
-#include <linux/acpi.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/soc-dapm.h>
-#include <sound/initval.h>
-#include <sound/tlv.h>
#include "rl6231.h"
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
new file mode 100644
index 000000000000..e4f6102efc1a
--- /dev/null
+++ b/sound/soc/codecs/rt286.c
@@ -0,0 +1,1222 @@
+/*
+ * rt286.c -- RT286 ALSA SoC audio codec driver
+ *
+ * Copyright 2013 Realtek Semiconductor Corp.
+ * Author: Bard Liao <bardliao@realtek.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/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <linux/acpi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/jack.h>
+#include <linux/workqueue.h>
+#include <sound/rt286.h>
+#include <sound/hda_verbs.h>
+
+#include "rt286.h"
+
+#define RT286_VENDOR_ID 0x10ec0286
+
+struct rt286_priv {
+ struct regmap *regmap;
+ struct rt286_platform_data pdata;
+ struct i2c_client *i2c;
+ struct snd_soc_jack *jack;
+ struct delayed_work jack_detect_work;
+ int sys_clk;
+ struct reg_default *index_cache;
+};
+
+static struct reg_default rt286_index_def[] = {
+ { 0x01, 0xaaaa },
+ { 0x02, 0x8aaa },
+ { 0x03, 0x0002 },
+ { 0x04, 0xaf01 },
+ { 0x08, 0x000d },
+ { 0x09, 0xd810 },
+ { 0x0a, 0x0060 },
+ { 0x0b, 0x0000 },
+ { 0x0d, 0x2800 },
+ { 0x0f, 0x0000 },
+ { 0x19, 0x0a17 },
+ { 0x20, 0x0020 },
+ { 0x33, 0x0208 },
+ { 0x49, 0x0004 },
+ { 0x4f, 0x50e9 },
+ { 0x50, 0x2c00 },
+ { 0x63, 0x2902 },
+ { 0x67, 0x1111 },
+ { 0x68, 0x1016 },
+ { 0x69, 0x273f },
+};
+#define INDEX_CACHE_SIZE ARRAY_SIZE(rt286_index_def)
+
+static const struct reg_default rt286_reg[] = {
+ { 0x00170500, 0x00000400 },
+ { 0x00220000, 0x00000031 },
+ { 0x00239000, 0x0000007f },
+ { 0x0023a000, 0x0000007f },
+ { 0x00270500, 0x00000400 },
+ { 0x00370500, 0x00000400 },
+ { 0x00870500, 0x00000400 },
+ { 0x00920000, 0x00000031 },
+ { 0x00935000, 0x000000c3 },
+ { 0x00936000, 0x000000c3 },
+ { 0x00970500, 0x00000400 },
+ { 0x00b37000, 0x00000097 },
+ { 0x00b37200, 0x00000097 },
+ { 0x00b37300, 0x00000097 },
+ { 0x00c37000, 0x00000000 },
+ { 0x00c37100, 0x00000080 },
+ { 0x01270500, 0x00000400 },
+ { 0x01370500, 0x00000400 },
+ { 0x01371f00, 0x411111f0 },
+ { 0x01439000, 0x00000080 },
+ { 0x0143a000, 0x00000080 },
+ { 0x01470700, 0x00000000 },
+ { 0x01470500, 0x00000400 },
+ { 0x01470c00, 0x00000000 },
+ { 0x01470100, 0x00000000 },
+ { 0x01837000, 0x00000000 },
+ { 0x01870500, 0x00000400 },
+ { 0x02050000, 0x00000000 },
+ { 0x02139000, 0x00000080 },
+ { 0x0213a000, 0x00000080 },
+ { 0x02170100, 0x00000000 },
+ { 0x02170500, 0x00000400 },
+ { 0x02170700, 0x00000000 },
+ { 0x02270100, 0x00000000 },
+ { 0x02370100, 0x00000000 },
+ { 0x02040000, 0x00004002 },
+ { 0x01870700, 0x00000020 },
+ { 0x00830000, 0x000000c3 },
+ { 0x00930000, 0x000000c3 },
+ { 0x01270700, 0x00000000 },
+};
+
+static bool rt286_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case 0 ... 0xff:
+ case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID):
+ case RT286_GET_HP_SENSE:
+ case RT286_GET_MIC1_SENSE:
+ case RT286_PROC_COEF:
+ return true;
+ default:
+ return false;
+ }
+
+
+}
+
+static bool rt286_readable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case 0 ... 0xff:
+ case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID):
+ case RT286_GET_HP_SENSE:
+ case RT286_GET_MIC1_SENSE:
+ case RT286_SET_AUDIO_POWER:
+ case RT286_SET_HPO_POWER:
+ case RT286_SET_SPK_POWER:
+ case RT286_SET_DMIC1_POWER:
+ case RT286_SPK_MUX:
+ case RT286_HPO_MUX:
+ case RT286_ADC0_MUX:
+ case RT286_ADC1_MUX:
+ case RT286_SET_MIC1:
+ case RT286_SET_PIN_HPO:
+ case RT286_SET_PIN_SPK:
+ case RT286_SET_PIN_DMIC1:
+ case RT286_SPK_EAPD:
+ case RT286_SET_AMP_GAIN_HPO:
+ case RT286_SET_DMIC2_DEFAULT:
+ case RT286_DACL_GAIN:
+ case RT286_DACR_GAIN:
+ case RT286_ADCL_GAIN:
+ case RT286_ADCR_GAIN:
+ case RT286_MIC_GAIN:
+ case RT286_SPOL_GAIN:
+ case RT286_SPOR_GAIN:
+ case RT286_HPOL_GAIN:
+ case RT286_HPOR_GAIN:
+ case RT286_F_DAC_SWITCH:
+ case RT286_F_RECMIX_SWITCH:
+ case RT286_REC_MIC_SWITCH:
+ case RT286_REC_I2S_SWITCH:
+ case RT286_REC_LINE_SWITCH:
+ case RT286_REC_BEEP_SWITCH:
+ case RT286_DAC_FORMAT:
+ case RT286_ADC_FORMAT:
+ case RT286_COEF_INDEX:
+ case RT286_PROC_COEF:
+ case RT286_SET_AMP_GAIN_ADC_IN1:
+ case RT286_SET_AMP_GAIN_ADC_IN2:
+ case RT286_SET_POWER(RT286_DAC_OUT1):
+ case RT286_SET_POWER(RT286_DAC_OUT2):
+ case RT286_SET_POWER(RT286_ADC_IN1):
+ case RT286_SET_POWER(RT286_ADC_IN2):
+ case RT286_SET_POWER(RT286_DMIC2):
+ case RT286_SET_POWER(RT286_MIC1):
+ return true;
+ default:
+ return false;
+ }
+}
+
+static int rt286_hw_write(void *context, unsigned int reg, unsigned int value)
+{
+ struct i2c_client *client = context;
+ struct rt286_priv *rt286 = i2c_get_clientdata(client);
+ u8 data[4];
+ int ret, i;
+
+ /*handle index registers*/
+ if (reg <= 0xff) {
+ rt286_hw_write(client, RT286_COEF_INDEX, reg);
+ reg = RT286_PROC_COEF;
+ for (i = 0; i < INDEX_CACHE_SIZE; i++) {
+ if (reg == rt286->index_cache[i].reg) {
+ rt286->index_cache[i].def = value;
+ break;
+ }
+
+ }
+ }
+
+ data[0] = (reg >> 24) & 0xff;
+ data[1] = (reg >> 16) & 0xff;
+ /*
+ * 4 bit VID: reg should be 0
+ * 12 bit VID: value should be 0
+ * So we use an OR operator to handle it rather than use if condition.
+ */
+ data[2] = ((reg >> 8) & 0xff) | ((value >> 8) & 0xff);
+ data[3] = value & 0xff;
+
+ ret = i2c_master_send(client, data, 4);
+
+ if (ret == 4)
+ return 0;
+ else
+ pr_err("ret=%d\n", ret);
+ if (ret < 0)
+ return ret;
+ else
+ return -EIO;
+}
+
+static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value)
+{
+ struct i2c_client *client = context;
+ struct i2c_msg xfer[2];
+ int ret;
+ __be32 be_reg;
+ unsigned int index, vid, buf = 0x0;
+
+ /*handle index registers*/
+ if (reg <= 0xff) {
+ rt286_hw_write(client, RT286_COEF_INDEX, reg);
+ reg = RT286_PROC_COEF;
+ }
+
+ reg = reg | 0x80000;
+ vid = (reg >> 8) & 0xfff;
+
+ if (AC_VERB_GET_AMP_GAIN_MUTE == (vid & 0xf00)) {
+ index = (reg >> 8) & 0xf;
+ reg = (reg & ~0xf0f) | index;
+ }
+ be_reg = cpu_to_be32(reg);
+
+ /* Write register */
+ xfer[0].addr = client->addr;
+ xfer[0].flags = 0;
+ xfer[0].len = 4;
+ xfer[0].buf = (u8 *)&be_reg;
+
+ /* Read data */
+ xfer[1].addr = client->addr;
+ xfer[1].flags = I2C_M_RD;
+ xfer[1].len = 4;
+ xfer[1].buf = (u8 *)&buf;
+
+ ret = i2c_transfer(client->adapter, xfer, 2);
+ if (ret < 0)
+ return ret;
+ else if (ret != 2)
+ return -EIO;
+
+ *value = be32_to_cpu(buf);
+
+ return 0;
+}
+
+static void rt286_index_sync(struct snd_soc_codec *codec)
+{
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+ int i;
+
+ for (i = 0; i < INDEX_CACHE_SIZE; i++) {
+ snd_soc_write(codec, rt286->index_cache[i].reg,
+ rt286->index_cache[i].def);
+ }
+}
+
+static int rt286_support_power_controls[] = {
+ RT286_DAC_OUT1,
+ RT286_DAC_OUT2,
+ RT286_ADC_IN1,
+ RT286_ADC_IN2,
+ RT286_MIC1,
+ RT286_DMIC1,
+ RT286_DMIC2,
+ RT286_SPK_OUT,
+ RT286_HP_OUT,
+};
+#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls)
+
+static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
+{
+ unsigned int val, buf;
+ int i;
+
+ *hp = false;
+ *mic = false;
+
+ if (rt286->pdata.cbj_en) {
+ regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf);
+ *hp = buf & 0x80000000;
+ if (*hp) {
+ /* power on HV,VERF */
+ regmap_update_bits(rt286->regmap,
+ RT286_POWER_CTRL1, 0x1001, 0x0);
+ /* power LDO1 */
+ regmap_update_bits(rt286->regmap,
+ RT286_POWER_CTRL2, 0x4, 0x4);
+ regmap_write(rt286->regmap, RT286_SET_MIC1, 0x24);
+ regmap_read(rt286->regmap, RT286_CBJ_CTRL2, &val);
+
+ msleep(200);
+ i = 40;
+ while (((val & 0x0800) == 0) && (i > 0)) {
+ regmap_read(rt286->regmap,
+ RT286_CBJ_CTRL2, &val);
+ i--;
+ msleep(20);
+ }
+
+ if (0x0400 == (val & 0x0700)) {
+ *mic = false;
+
+ regmap_write(rt286->regmap,
+ RT286_SET_MIC1, 0x20);
+ /* power off HV,VERF */
+ regmap_update_bits(rt286->regmap,
+ RT286_POWER_CTRL1, 0x1001, 0x1001);
+ regmap_update_bits(rt286->regmap,
+ RT286_A_BIAS_CTRL3, 0xc000, 0x0000);
+ regmap_update_bits(rt286->regmap,
+ RT286_CBJ_CTRL1, 0x0030, 0x0000);
+ regmap_update_bits(rt286->regmap,
+ RT286_A_BIAS_CTRL2, 0xc000, 0x0000);
+ } else if ((0x0200 == (val & 0x0700)) ||
+ (0x0100 == (val & 0x0700))) {
+ *mic = true;
+ regmap_update_bits(rt286->regmap,
+ RT286_A_BIAS_CTRL3, 0xc000, 0x8000);
+ regmap_update_bits(rt286->regmap,
+ RT286_CBJ_CTRL1, 0x0030, 0x0020);
+ regmap_update_bits(rt286->regmap,
+ RT286_A_BIAS_CTRL2, 0xc000, 0x8000);
+ } else {
+ *mic = false;
+ }
+
+ regmap_update_bits(rt286->regmap,
+ RT286_MISC_CTRL1,
+ 0x0060, 0x0000);
+ } else {
+ regmap_update_bits(rt286->regmap,
+ RT286_MISC_CTRL1,
+ 0x0060, 0x0020);
+ regmap_update_bits(rt286->regmap,
+ RT286_A_BIAS_CTRL3,
+ 0xc000, 0x8000);
+ regmap_update_bits(rt286->regmap,
+ RT286_CBJ_CTRL1,
+ 0x0030, 0x0020);
+ regmap_update_bits(rt286->regmap,
+ RT286_A_BIAS_CTRL2,
+ 0xc000, 0x8000);
+
+ *mic = false;
+ }
+ } else {
+ regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf);
+ *hp = buf & 0x80000000;
+ regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf);
+ *mic = buf & 0x80000000;
+ }
+
+ return 0;
+}
+
+static void rt286_jack_detect_work(struct work_struct *work)
+{
+ struct rt286_priv *rt286 =
+ container_of(work, struct rt286_priv, jack_detect_work.work);
+ int status = 0;
+ bool hp = false;
+ bool mic = false;
+
+ rt286_jack_detect(rt286, &hp, &mic);
+
+ if (hp == true)
+ status |= SND_JACK_HEADPHONE;
+
+ if (mic == true)
+ status |= SND_JACK_MICROPHONE;
+
+ snd_soc_jack_report(rt286->jack, status,
+ SND_JACK_MICROPHONE | SND_JACK_HEADPHONE);
+}
+
+int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
+{
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+ rt286->jack = jack;
+
+ /* Send an initial empty report */
+ snd_soc_jack_report(rt286->jack, 0,
+ SND_JACK_MICROPHONE | SND_JACK_HEADPHONE);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(rt286_mic_detect);
+
+static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0);
+static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0);
+
+static const struct snd_kcontrol_new rt286_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN,
+ RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv),
+ SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN,
+ RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv),
+ SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN,
+ 0, 0x3, 0, mic_vol_tlv),
+ SOC_DOUBLE_R("Speaker Playback Switch", RT286_SPOL_GAIN,
+ RT286_SPOR_GAIN, RT286_MUTE_SFT, 1, 1),
+};
+
+/* Digital Mixer */
+static const struct snd_kcontrol_new rt286_front_mix[] = {
+ SOC_DAPM_SINGLE("DAC Switch", RT286_F_DAC_SWITCH,
+ RT286_MUTE_SFT, 1, 1),
+ SOC_DAPM_SINGLE("RECMIX Switch", RT286_F_RECMIX_SWITCH,
+ RT286_MUTE_SFT, 1, 1),
+};
+
+/* Analog Input Mixer */
+static const struct snd_kcontrol_new rt286_rec_mix[] = {
+ SOC_DAPM_SINGLE("Mic1 Switch", RT286_REC_MIC_SWITCH,
+ RT286_MUTE_SFT, 1, 1),
+ SOC_DAPM_SINGLE("I2S Switch", RT286_REC_I2S_SWITCH,
+ RT286_MUTE_SFT, 1, 1),
+ SOC_DAPM_SINGLE("Line1 Switch", RT286_REC_LINE_SWITCH,
+ RT286_MUTE_SFT, 1, 1),
+ SOC_DAPM_SINGLE("Beep Switch", RT286_REC_BEEP_SWITCH,
+ RT286_MUTE_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new spo_enable_control =
+ SOC_DAPM_SINGLE("Switch", RT286_SET_PIN_SPK,
+ RT286_SET_PIN_SFT, 1, 0);
+
+static const struct snd_kcontrol_new hpol_enable_control =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOL_GAIN,
+ RT286_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new hpor_enable_control =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOR_GAIN,
+ RT286_MUTE_SFT, 1, 1);
+
+/* ADC0 source */
+static const char * const rt286_adc_src[] = {
+ "Mic", "RECMIX", "Dmic"
+};
+
+static const int rt286_adc_values[] = {
+ 0, 4, 5,
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(
+ rt286_adc0_enum, RT286_ADC0_MUX, RT286_ADC_SEL_SFT,
+ RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values);
+
+static const struct snd_kcontrol_new rt286_adc0_mux =
+ SOC_DAPM_ENUM("ADC 0 source", rt286_adc0_enum);
+
+static SOC_VALUE_ENUM_SINGLE_DECL(
+ rt286_adc1_enum, RT286_ADC1_MUX, RT286_ADC_SEL_SFT,
+ RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values);
+
+static const struct snd_kcontrol_new rt286_adc1_mux =
+ SOC_DAPM_ENUM("ADC 1 source", rt286_adc1_enum);
+
+static const char * const rt286_dac_src[] = {
+ "Front", "Surround"
+};
+/* HP-OUT source */
+static SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, RT286_HPO_MUX,
+ 0, rt286_dac_src);
+
+static const struct snd_kcontrol_new rt286_hpo_mux =
+SOC_DAPM_ENUM("HPO source", rt286_hpo_enum);
+
+/* SPK-OUT source */
+static SOC_ENUM_SINGLE_DECL(rt286_spo_enum, RT286_SPK_MUX,
+ 0, rt286_dac_src);
+
+static const struct snd_kcontrol_new rt286_spo_mux =
+SOC_DAPM_ENUM("SPO source", rt286_spo_enum);
+
+static int rt286_spk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_write(codec,
+ RT286_SPK_EAPD, RT286_SET_EAPD_HIGH);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_write(codec,
+ RT286_SPK_EAPD, RT286_SET_EAPD_LOW);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0x20);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0);
+ break;
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int rt286_adc_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ unsigned int nid;
+
+ nid = (w->reg >> 20) & 0xff;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec,
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0),
+ 0x7080, 0x7000);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_update_bits(codec,
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0),
+ 0x7080, 0x7080);
+ break;
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = {
+ /* Input Lines */
+ SND_SOC_DAPM_INPUT("DMIC1 Pin"),
+ SND_SOC_DAPM_INPUT("DMIC2 Pin"),
+ SND_SOC_DAPM_INPUT("MIC1"),
+ SND_SOC_DAPM_INPUT("LINE1"),
+ SND_SOC_DAPM_INPUT("Beep"),
+
+ /* DMIC */
+ SND_SOC_DAPM_PGA_E("DMIC1", RT286_SET_POWER(RT286_DMIC1), 0, 1,
+ NULL, 0, rt286_set_dmic1_event,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PGA("DMIC2", RT286_SET_POWER(RT286_DMIC2), 0, 1,
+ NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM,
+ 0, 0, NULL, 0),
+
+ /* REC Mixer */
+ SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0,
+ rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)),
+
+ /* ADCs */
+ SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0),
+
+ /* ADC Mux */
+ SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1,
+ &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD |
+ SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1,
+ &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD |
+ SND_SOC_DAPM_POST_PMU),
+
+ /* Audio Interface */
+ SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
+
+ /* Output Side */
+ /* DACs */
+ SND_SOC_DAPM_DAC("DAC 0", NULL, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("DAC 1", NULL, SND_SOC_NOPM, 0, 0),
+
+ /* Output Mux */
+ SND_SOC_DAPM_MUX("SPK Mux", SND_SOC_NOPM, 0, 0, &rt286_spo_mux),
+ SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt286_hpo_mux),
+
+ SND_SOC_DAPM_SUPPLY("HP Power", RT286_SET_PIN_HPO,
+ RT286_SET_PIN_SFT, 0, NULL, 0),
+
+ /* Output Mixer */
+ SND_SOC_DAPM_MIXER("Front", RT286_SET_POWER(RT286_DAC_OUT1), 0, 1,
+ rt286_front_mix, ARRAY_SIZE(rt286_front_mix)),
+ SND_SOC_DAPM_PGA("Surround", RT286_SET_POWER(RT286_DAC_OUT2), 0, 1,
+ NULL, 0),
+
+ /* Output Pga */
+ SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0,
+ &spo_enable_control, rt286_spk_event,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0,
+ &hpol_enable_control),
+ SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0,
+ &hpor_enable_control),
+
+ /* Output Lines */
+ SND_SOC_DAPM_OUTPUT("SPOL"),
+ SND_SOC_DAPM_OUTPUT("SPOR"),
+ SND_SOC_DAPM_OUTPUT("HPO Pin"),
+ SND_SOC_DAPM_OUTPUT("SPDIF"),
+};
+
+static const struct snd_soc_dapm_route rt286_dapm_routes[] = {
+ {"DMIC1", NULL, "DMIC1 Pin"},
+ {"DMIC2", NULL, "DMIC2 Pin"},
+ {"DMIC1", NULL, "DMIC Receiver"},
+ {"DMIC2", NULL, "DMIC Receiver"},
+
+ {"RECMIX", "Beep Switch", "Beep"},
+ {"RECMIX", "Line1 Switch", "LINE1"},
+ {"RECMIX", "Mic1 Switch", "MIC1"},
+
+ {"ADC 0 Mux", "Dmic", "DMIC1"},
+ {"ADC 0 Mux", "RECMIX", "RECMIX"},
+ {"ADC 0 Mux", "Mic", "MIC1"},
+ {"ADC 1 Mux", "Dmic", "DMIC2"},
+ {"ADC 1 Mux", "RECMIX", "RECMIX"},
+ {"ADC 1 Mux", "Mic", "MIC1"},
+
+ {"ADC 0", NULL, "ADC 0 Mux"},
+ {"ADC 1", NULL, "ADC 1 Mux"},
+
+ {"AIF1TX", NULL, "ADC 0"},
+ {"AIF2TX", NULL, "ADC 1"},
+
+ {"DAC 0", NULL, "AIF1RX"},
+ {"DAC 1", NULL, "AIF2RX"},
+
+ {"Front", "DAC Switch", "DAC 0"},
+ {"Front", "RECMIX Switch", "RECMIX"},
+
+ {"Surround", NULL, "DAC 1"},
+
+ {"SPK Mux", "Front", "Front"},
+ {"SPK Mux", "Surround", "Surround"},
+
+ {"HPO Mux", "Front", "Front"},
+ {"HPO Mux", "Surround", "Surround"},
+
+ {"SPO", "Switch", "SPK Mux"},
+ {"HPO L", "Switch", "HPO Mux"},
+ {"HPO R", "Switch", "HPO Mux"},
+ {"HPO L", NULL, "HP Power"},
+ {"HPO R", NULL, "HP Power"},
+
+ {"SPOL", NULL, "SPO"},
+ {"SPOR", NULL, "SPO"},
+ {"HPO Pin", NULL, "HPO L"},
+ {"HPO Pin", NULL, "HPO R"},
+};
+
+static int rt286_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 rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+ unsigned int val = 0;
+ int d_len_code;
+
+ switch (params_rate(params)) {
+ /* bit 14 0:48K 1:44.1K */
+ case 44100:
+ val |= 0x4000;
+ break;
+ case 48000:
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported sample rate %d\n",
+ params_rate(params));
+ return -EINVAL;
+ }
+ switch (rt286->sys_clk) {
+ case 12288000:
+ case 24576000:
+ if (params_rate(params) != 48000) {
+ dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n",
+ params_rate(params), rt286->sys_clk);
+ return -EINVAL;
+ }
+ break;
+ case 11289600:
+ case 22579200:
+ if (params_rate(params) != 44100) {
+ dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n",
+ params_rate(params), rt286->sys_clk);
+ return -EINVAL;
+ }
+ break;
+ }
+
+ if (params_channels(params) <= 16) {
+ /* bit 3:0 Number of Channel */
+ val |= (params_channels(params) - 1);
+ } else {
+ dev_err(codec->dev, "Unsupported channels %d\n",
+ params_channels(params));
+ return -EINVAL;
+ }
+
+ d_len_code = 0;
+ switch (params_width(params)) {
+ /* bit 6:4 Bits per Sample */
+ case 16:
+ d_len_code = 0;
+ val |= (0x1 << 4);
+ break;
+ case 32:
+ d_len_code = 2;
+ val |= (0x4 << 4);
+ break;
+ case 20:
+ d_len_code = 1;
+ val |= (0x2 << 4);
+ break;
+ case 24:
+ d_len_code = 2;
+ val |= (0x3 << 4);
+ break;
+ case 8:
+ d_len_code = 3;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x0018, d_len_code << 3);
+ dev_dbg(codec->dev, "format val = 0x%x\n", val);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val);
+ else
+ snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val);
+
+ return 0;
+}
+
+static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x800, 0x800);
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x800, 0x0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x300, 0x0);
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x300, 0x1 << 8);
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x300, 0x2 << 8);
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x300, 0x3 << 8);
+ break;
+ default:
+ return -EINVAL;
+ }
+ /* bit 15 Stream Type 0:PCM 1:Non-PCM */
+ snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x8000, 0);
+ snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x8000, 0);
+
+ return 0;
+}
+
+static int rt286_set_dai_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+ dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq);
+
+ if (RT286_SCLK_S_MCLK == clk_id) {
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL2, 0x0100, 0x0);
+ snd_soc_update_bits(codec,
+ RT286_PLL_CTRL1, 0x20, 0x20);
+ } else {
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL2, 0x0100, 0x0100);
+ snd_soc_update_bits(codec,
+ RT286_PLL_CTRL, 0x4, 0x4);
+ snd_soc_update_bits(codec,
+ RT286_PLL_CTRL1, 0x20, 0x0);
+ }
+
+ switch (freq) {
+ case 19200000:
+ if (RT286_SCLK_S_MCLK == clk_id) {
+ dev_err(codec->dev, "Should not use MCLK\n");
+ return -EINVAL;
+ }
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL2, 0x40, 0x40);
+ break;
+ case 24000000:
+ if (RT286_SCLK_S_MCLK == clk_id) {
+ dev_err(codec->dev, "Should not use MCLK\n");
+ return -EINVAL;
+ }
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL2, 0x40, 0x0);
+ break;
+ case 12288000:
+ case 11289600:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL2, 0x8, 0x0);
+ snd_soc_update_bits(codec,
+ RT286_CLK_DIV, 0xfc1e, 0x0004);
+ break;
+ case 24576000:
+ case 22579200:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL2, 0x8, 0x8);
+ snd_soc_update_bits(codec,
+ RT286_CLK_DIV, 0xfc1e, 0x5406);
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported system clock\n");
+ return -EINVAL;
+ }
+
+ rt286->sys_clk = freq;
+
+ return 0;
+}
+
+static int rt286_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio);
+ if (50 == ratio)
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x1000, 0x1000);
+ else
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x1000, 0x0);
+
+
+ return 0;
+}
+
+static int rt286_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) {
+ snd_soc_write(codec,
+ RT286_SET_AUDIO_POWER, AC_PWRST_D0);
+ snd_soc_update_bits(codec,
+ RT286_DC_GAIN, 0x200, 0x200);
+ }
+ break;
+
+ case SND_SOC_BIAS_ON:
+ mdelay(10);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ snd_soc_write(codec,
+ RT286_SET_AUDIO_POWER, AC_PWRST_D3);
+ snd_soc_update_bits(codec,
+ RT286_DC_GAIN, 0x200, 0x0);
+ break;
+
+ default:
+ break;
+ }
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+static irqreturn_t rt286_irq(int irq, void *data)
+{
+ struct rt286_priv *rt286 = data;
+ bool hp = false;
+ bool mic = false;
+ int status = 0;
+
+ rt286_jack_detect(rt286, &hp, &mic);
+
+ /* Clear IRQ */
+ regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x1, 0x1);
+
+ if (hp == true)
+ status |= SND_JACK_HEADPHONE;
+
+ if (mic == true)
+ status |= SND_JACK_MICROPHONE;
+
+ snd_soc_jack_report(rt286->jack, status,
+ SND_JACK_MICROPHONE | SND_JACK_HEADPHONE);
+
+ pm_wakeup_event(&rt286->i2c->dev, 300);
+
+ return IRQ_HANDLED;
+}
+
+static int rt286_probe(struct snd_soc_codec *codec)
+{
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+ codec->dapm.bias_level = SND_SOC_BIAS_OFF;
+
+ if (rt286->i2c->irq) {
+ regmap_update_bits(rt286->regmap,
+ RT286_IRQ_CTRL, 0x2, 0x2);
+
+ INIT_DELAYED_WORK(&rt286->jack_detect_work,
+ rt286_jack_detect_work);
+ schedule_delayed_work(&rt286->jack_detect_work,
+ msecs_to_jiffies(1250));
+ }
+
+ return 0;
+}
+
+static int rt286_remove(struct snd_soc_codec *codec)
+{
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+ cancel_delayed_work_sync(&rt286->jack_detect_work);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int rt286_suspend(struct snd_soc_codec *codec)
+{
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+ regcache_cache_only(rt286->regmap, true);
+ regcache_mark_dirty(rt286->regmap);
+
+ return 0;
+}
+
+static int rt286_resume(struct snd_soc_codec *codec)
+{
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+ regcache_cache_only(rt286->regmap, false);
+ rt286_index_sync(codec);
+ regcache_sync(rt286->regmap);
+
+ return 0;
+}
+#else
+#define rt286_suspend NULL
+#define rt286_resume NULL
+#endif
+
+#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
+
+static const struct snd_soc_dai_ops rt286_aif_dai_ops = {
+ .hw_params = rt286_hw_params,
+ .set_fmt = rt286_set_dai_fmt,
+ .set_sysclk = rt286_set_dai_sysclk,
+ .set_bclk_ratio = rt286_set_bclk_ratio,
+};
+
+static struct snd_soc_dai_driver rt286_dai[] = {
+ {
+ .name = "rt286-aif1",
+ .id = RT286_AIF1,
+ .playback = {
+ .stream_name = "AIF1 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT286_STEREO_RATES,
+ .formats = RT286_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF1 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT286_STEREO_RATES,
+ .formats = RT286_FORMATS,
+ },
+ .ops = &rt286_aif_dai_ops,
+ .symmetric_rates = 1,
+ },
+ {
+ .name = "rt286-aif2",
+ .id = RT286_AIF2,
+ .playback = {
+ .stream_name = "AIF2 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT286_STEREO_RATES,
+ .formats = RT286_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF2 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT286_STEREO_RATES,
+ .formats = RT286_FORMATS,
+ },
+ .ops = &rt286_aif_dai_ops,
+ .symmetric_rates = 1,
+ },
+
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_rt286 = {
+ .probe = rt286_probe,
+ .remove = rt286_remove,
+ .suspend = rt286_suspend,
+ .resume = rt286_resume,
+ .set_bias_level = rt286_set_bias_level,
+ .idle_bias_off = true,
+ .controls = rt286_snd_controls,
+ .num_controls = ARRAY_SIZE(rt286_snd_controls),
+ .dapm_widgets = rt286_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets),
+ .dapm_routes = rt286_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes),
+};
+
+static const struct regmap_config rt286_regmap = {
+ .reg_bits = 32,
+ .val_bits = 32,
+ .max_register = 0x02370100,
+ .volatile_reg = rt286_volatile_register,
+ .readable_reg = rt286_readable_register,
+ .reg_write = rt286_hw_write,
+ .reg_read = rt286_hw_read,
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults = rt286_reg,
+ .num_reg_defaults = ARRAY_SIZE(rt286_reg),
+};
+
+static const struct i2c_device_id rt286_i2c_id[] = {
+ {"rt286", 0},
+ {}
+};
+MODULE_DEVICE_TABLE(i2c, rt286_i2c_id);
+
+static const struct acpi_device_id rt286_acpi_match[] = {
+ { "INT343A", 0 },
+ {},
+};
+MODULE_DEVICE_TABLE(acpi, rt286_acpi_match);
+
+static int rt286_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev);
+ struct rt286_priv *rt286;
+ int i, ret;
+
+ rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286),
+ GFP_KERNEL);
+ if (NULL == rt286)
+ return -ENOMEM;
+
+ rt286->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt286_regmap);
+ if (IS_ERR(rt286->regmap)) {
+ ret = PTR_ERR(rt286->regmap);
+ dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+ ret);
+ return ret;
+ }
+
+ regmap_read(rt286->regmap,
+ RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret);
+ if (ret != RT286_VENDOR_ID) {
+ dev_err(&i2c->dev,
+ "Device with ID register %x is not rt286\n", ret);
+ return -ENODEV;
+ }
+
+ rt286->index_cache = rt286_index_def;
+ rt286->i2c = i2c;
+ i2c_set_clientdata(i2c, rt286);
+
+ if (pdata)
+ rt286->pdata = *pdata;
+
+ regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3);
+
+ for (i = 0; i < RT286_POWER_REG_LEN; i++)
+ regmap_write(rt286->regmap,
+ RT286_SET_POWER(rt286_support_power_controls[i]),
+ AC_PWRST_D1);
+
+ if (!rt286->pdata.cbj_en) {
+ regmap_write(rt286->regmap, RT286_CBJ_CTRL2, 0x0000);
+ regmap_write(rt286->regmap, RT286_MIC1_DET_CTRL, 0x0816);
+ regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000);
+ regmap_update_bits(rt286->regmap,
+ RT286_CBJ_CTRL1, 0xf000, 0xb000);
+ } else {
+ regmap_update_bits(rt286->regmap,
+ RT286_CBJ_CTRL1, 0xf000, 0x5000);
+ }
+
+ mdelay(10);
+
+ if (!rt286->pdata.gpio2_en)
+ regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0x4000);
+ else
+ regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0);
+
+ mdelay(10);
+
+ /*Power down LDO2*/
+ regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0);
+
+ /*Set depop parameter*/
+ regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a);
+ regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737);
+ regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f);
+
+ if (rt286->i2c->irq) {
+ ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq,
+ IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286);
+ if (ret != 0) {
+ dev_err(&i2c->dev,
+ "Failed to reguest IRQ: %d\n", ret);
+ return ret;
+ }
+ }
+
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286,
+ rt286_dai, ARRAY_SIZE(rt286_dai));
+
+ return ret;
+}
+
+static int rt286_i2c_remove(struct i2c_client *i2c)
+{
+ struct rt286_priv *rt286 = i2c_get_clientdata(i2c);
+
+ if (i2c->irq)
+ free_irq(i2c->irq, rt286);
+ snd_soc_unregister_codec(&i2c->dev);
+
+ return 0;
+}
+
+
+static struct i2c_driver rt286_i2c_driver = {
+ .driver = {
+ .name = "rt286",
+ .owner = THIS_MODULE,
+ .acpi_match_table = ACPI_PTR(rt286_acpi_match),
+ },
+ .probe = rt286_i2c_probe,
+ .remove = rt286_i2c_remove,
+ .id_table = rt286_i2c_id,
+};
+
+module_i2c_driver(rt286_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC RT286 driver");
+MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h
new file mode 100644
index 000000000000..b539b7320a79
--- /dev/null
+++ b/sound/soc/codecs/rt286.h
@@ -0,0 +1,198 @@
+/*
+ * rt286.h -- RT286 ALSA SoC audio driver
+ *
+ * Copyright 2011 Realtek Microelectronics
+ * Author: Johnny Hsu <johnnyhsu@realtek.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 __RT286_H__
+#define __RT286_H__
+
+#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D)
+
+#define RT286_AUDIO_FUNCTION_GROUP 0x01
+#define RT286_DAC_OUT1 0x02
+#define RT286_DAC_OUT2 0x03
+#define RT286_ADC_IN1 0x09
+#define RT286_ADC_IN2 0x08
+#define RT286_MIXER_IN 0x0b
+#define RT286_MIXER_OUT1 0x0c
+#define RT286_MIXER_OUT2 0x0d
+#define RT286_DMIC1 0x12
+#define RT286_DMIC2 0x13
+#define RT286_SPK_OUT 0x14
+#define RT286_MIC1 0x18
+#define RT286_LINE1 0x1a
+#define RT286_BEEP 0x1d
+#define RT286_SPDIF 0x1e
+#define RT286_VENDOR_REGISTERS 0x20
+#define RT286_HP_OUT 0x21
+#define RT286_MIXER_IN1 0x22
+#define RT286_MIXER_IN2 0x23
+
+#define RT286_SET_PIN_SFT 6
+#define RT286_SET_PIN_ENABLE 0x40
+#define RT286_SET_PIN_DISABLE 0
+#define RT286_SET_EAPD_HIGH 0x2
+#define RT286_SET_EAPD_LOW 0
+
+#define RT286_MUTE_SFT 7
+
+/* Verb commands */
+#define RT286_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM)
+#define RT286_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0)
+#define RT286_SET_AUDIO_POWER RT286_SET_POWER(RT286_AUDIO_FUNCTION_GROUP)
+#define RT286_SET_HPO_POWER RT286_SET_POWER(RT286_HP_OUT)
+#define RT286_SET_SPK_POWER RT286_SET_POWER(RT286_SPK_OUT)
+#define RT286_SET_DMIC1_POWER RT286_SET_POWER(RT286_DMIC1)
+#define RT286_SPK_MUX\
+ VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_SPK_OUT, 0)
+#define RT286_HPO_MUX\
+ VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_HP_OUT, 0)
+#define RT286_ADC0_MUX\
+ VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN1, 0)
+#define RT286_ADC1_MUX\
+ VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN2, 0)
+#define RT286_SET_MIC1\
+ VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0)
+#define RT286_SET_PIN_HPO\
+ VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0)
+#define RT286_SET_PIN_SPK\
+ VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0)
+#define RT286_SET_PIN_DMIC1\
+ VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0)
+#define RT286_SPK_EAPD\
+ VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0)
+#define RT286_SET_AMP_GAIN_HPO\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0)
+#define RT286_SET_AMP_GAIN_ADC_IN1\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0)
+#define RT286_SET_AMP_GAIN_ADC_IN2\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN2, 0)
+#define RT286_GET_HP_SENSE\
+ VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0)
+#define RT286_GET_MIC1_SENSE\
+ VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0)
+#define RT286_SET_DMIC2_DEFAULT\
+ VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0)
+#define RT286_DACL_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0xa000)
+#define RT286_DACR_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0x9000)
+#define RT286_ADCL_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x6000)
+#define RT286_ADCR_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x5000)
+#define RT286_MIC_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIC1, 0x7000)
+#define RT286_SPOL_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0xa000)
+#define RT286_SPOR_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0x9000)
+#define RT286_HPOL_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000)
+#define RT286_HPOR_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000)
+#define RT286_F_DAC_SWITCH\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7000)
+#define RT286_F_RECMIX_SWITCH\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7100)
+#define RT286_REC_MIC_SWITCH\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7000)
+#define RT286_REC_I2S_SWITCH\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7100)
+#define RT286_REC_LINE_SWITCH\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7200)
+#define RT286_REC_BEEP_SWITCH\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7300)
+#define RT286_DAC_FORMAT\
+ VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_DAC_OUT1, 0)
+#define RT286_ADC_FORMAT\
+ VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_ADC_IN1, 0)
+#define RT286_COEF_INDEX\
+ VERB_CMD(AC_VERB_SET_COEF_INDEX, RT286_VENDOR_REGISTERS, 0)
+#define RT286_PROC_COEF\
+ VERB_CMD(AC_VERB_SET_PROC_COEF, RT286_VENDOR_REGISTERS, 0)
+
+/* Index registers */
+#define RT286_A_BIAS_CTRL1 0x01
+#define RT286_A_BIAS_CTRL2 0x02
+#define RT286_POWER_CTRL1 0x03
+#define RT286_A_BIAS_CTRL3 0x04
+#define RT286_POWER_CTRL2 0x08
+#define RT286_I2S_CTRL1 0x09
+#define RT286_I2S_CTRL2 0x0a
+#define RT286_CLK_DIV 0x0b
+#define RT286_DC_GAIN 0x0d
+#define RT286_POWER_CTRL3 0x0f
+#define RT286_MIC1_DET_CTRL 0x19
+#define RT286_MISC_CTRL1 0x20
+#define RT286_IRQ_CTRL 0x33
+#define RT286_PLL_CTRL1 0x49
+#define RT286_CBJ_CTRL1 0x4f
+#define RT286_CBJ_CTRL2 0x50
+#define RT286_PLL_CTRL 0x63
+#define RT286_DEPOP_CTRL1 0x66
+#define RT286_DEPOP_CTRL2 0x67
+#define RT286_DEPOP_CTRL3 0x68
+#define RT286_DEPOP_CTRL4 0x69
+
+/* SPDIF (0x06) */
+#define RT286_SPDIF_SEL_SFT 0
+#define RT286_SPDIF_SEL_PCM0 0
+#define RT286_SPDIF_SEL_PCM1 1
+#define RT286_SPDIF_SEL_SPOUT 2
+#define RT286_SPDIF_SEL_PP 3
+
+/* RECMIX (0x0b) */
+#define RT286_M_REC_BEEP_SFT 0
+#define RT286_M_REC_LINE1_SFT 1
+#define RT286_M_REC_MIC1_SFT 2
+#define RT286_M_REC_I2S_SFT 3
+
+/* Front (0x0c) */
+#define RT286_M_FRONT_DAC_SFT 0
+#define RT286_M_FRONT_REC_SFT 1
+
+/* SPK-OUT (0x14) */
+#define RT286_M_SPK_MUX_SFT 14
+#define RT286_SPK_SEL_MASK 0x1
+#define RT286_SPK_SEL_SFT 0
+#define RT286_SPK_SEL_F 0
+#define RT286_SPK_SEL_S 1
+
+/* HP-OUT (0x21) */
+#define RT286_M_HP_MUX_SFT 14
+#define RT286_HP_SEL_MASK 0x1
+#define RT286_HP_SEL_SFT 0
+#define RT286_HP_SEL_F 0
+#define RT286_HP_SEL_S 1
+
+/* ADC (0x22) (0x23) */
+#define RT286_ADC_SEL_MASK 0x7
+#define RT286_ADC_SEL_SFT 0
+#define RT286_ADC_SEL_SURR 0
+#define RT286_ADC_SEL_FRONT 1
+#define RT286_ADC_SEL_DMIC 2
+#define RT286_ADC_SEL_BEEP 4
+#define RT286_ADC_SEL_LINE1 5
+#define RT286_ADC_SEL_I2S 6
+#define RT286_ADC_SEL_MIC1 7
+
+#define RT286_SCLK_S_MCLK 0
+#define RT286_SCLK_S_PLL 1
+
+enum {
+ RT286_AIF1,
+ RT286_AIF2,
+ RT286_AIFS,
+};
+
+int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack);
+
+#endif /* __RT286_H__ */
+
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 30e234708579..1ba27db660a6 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -1370,16 +1370,16 @@ static int rt5631_hifi_pcm_params(struct snd_pcm_substream *substream,
return coeff;
}
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface |= RT5631_SDP_I2S_DL_20;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface |= RT5631_SDP_I2S_DL_24;
break;
- case SNDRV_PCM_FORMAT_S8:
+ case 8:
iface |= RT5631_SDP_I2S_DL_8;
break;
default:
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index de80e89b5fd8..6bc6efdec550 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -2215,14 +2215,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c,
rt5640->hp_mute = 1;
- ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640,
- rt5640_dai, ARRAY_SIZE(rt5640_dai));
- if (ret < 0)
- goto err;
-
- return 0;
-err:
- return ret;
+ return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640,
+ rt5640_dai, ARRAY_SIZE(rt5640_dai));
}
static int rt5640_i2c_remove(struct i2c_client *i2c)
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 02147be2b302..a7762d0a623e 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -2345,14 +2345,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
}
- ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645,
- rt5645_dai, ARRAY_SIZE(rt5645_dai));
- if (ret < 0)
- goto err;
-
- return 0;
-err:
- return ret;
+ return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645,
+ rt5645_dai, ARRAY_SIZE(rt5645_dai));
}
static int rt5645_i2c_remove(struct i2c_client *i2c)
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index ea4b1c652a26..bb0a3ab5416c 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -1366,16 +1366,16 @@ static int rt5651_hw_params(struct snd_pcm_substream *substream,
dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n",
bclk_ms, pre_div, dai->id);
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
val_len |= RT5651_I2S_DL_20;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
val_len |= RT5651_I2S_DL_24;
break;
- case SNDRV_PCM_FORMAT_S8:
+ case 8:
val_len |= RT5651_I2S_DL_8;
break;
default:
diff --git a/sound/soc/codecs/rt5670-dsp.h b/sound/soc/codecs/rt5670-dsp.h
new file mode 100644
index 000000000000..a34d0cdb8198
--- /dev/null
+++ b/sound/soc/codecs/rt5670-dsp.h
@@ -0,0 +1,54 @@
+/*
+ * rt5670-dsp.h -- RT5670 ALSA SoC DSP driver
+ *
+ * Copyright 2014 Realtek Microelectronics
+ * Author: Bard Liao <bardliao@realtek.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 __RT5670_DSP_H__
+#define __RT5670_DSP_H__
+
+#define RT5670_DSP_CTRL1 0xe0
+#define RT5670_DSP_CTRL2 0xe1
+#define RT5670_DSP_CTRL3 0xe2
+#define RT5670_DSP_CTRL4 0xe3
+#define RT5670_DSP_CTRL5 0xe4
+
+/* DSP Control 1 (0xe0) */
+#define RT5670_DSP_CMD_MASK (0xff << 8)
+#define RT5670_DSP_CMD_PE (0x0d << 8) /* Patch Entry */
+#define RT5670_DSP_CMD_MW (0x3b << 8) /* Memory Write */
+#define RT5670_DSP_CMD_MR (0x37 << 8) /* Memory Read */
+#define RT5670_DSP_CMD_RR (0x60 << 8) /* Register Read */
+#define RT5670_DSP_CMD_RW (0x68 << 8) /* Register Write */
+#define RT5670_DSP_REG_DATHI (0x26 << 8) /* High Data Addr */
+#define RT5670_DSP_REG_DATLO (0x25 << 8) /* Low Data Addr */
+#define RT5670_DSP_CLK_MASK (0x3 << 6)
+#define RT5670_DSP_CLK_SFT 6
+#define RT5670_DSP_CLK_768K (0x0 << 6)
+#define RT5670_DSP_CLK_384K (0x1 << 6)
+#define RT5670_DSP_CLK_192K (0x2 << 6)
+#define RT5670_DSP_CLK_96K (0x3 << 6)
+#define RT5670_DSP_BUSY_MASK (0x1 << 5)
+#define RT5670_DSP_RW_MASK (0x1 << 4)
+#define RT5670_DSP_DL_MASK (0x3 << 2)
+#define RT5670_DSP_DL_0 (0x0 << 2)
+#define RT5670_DSP_DL_1 (0x1 << 2)
+#define RT5670_DSP_DL_2 (0x2 << 2)
+#define RT5670_DSP_DL_3 (0x3 << 2)
+#define RT5670_DSP_I2C_AL_16 (0x1 << 1)
+#define RT5670_DSP_CMD_EN (0x1)
+
+struct rt5670_dsp_param {
+ u16 cmd_fmt;
+ u16 addr;
+ u16 data;
+ u8 cmd;
+};
+
+#endif /* __RT5670_DSP_H__ */
+
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
new file mode 100644
index 000000000000..ba9d9b4d4857
--- /dev/null
+++ b/sound/soc/codecs/rt5670.c
@@ -0,0 +1,2657 @@
+/*
+ * rt5670.c -- RT5670 ALSA SoC audio codec driver
+ *
+ * Copyright 2014 Realtek Semiconductor Corp.
+ * Author: Bard Liao <bardliao@realtek.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/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/jack.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/rt5670.h>
+
+#include "rl6231.h"
+#include "rt5670.h"
+#include "rt5670-dsp.h"
+
+#define RT5670_DEVICE_ID 0x6271
+
+#define RT5670_PR_RANGE_BASE (0xff + 1)
+#define RT5670_PR_SPACING 0x100
+
+#define RT5670_PR_BASE (RT5670_PR_RANGE_BASE + (0 * RT5670_PR_SPACING))
+
+static const struct regmap_range_cfg rt5670_ranges[] = {
+ { .name = "PR", .range_min = RT5670_PR_BASE,
+ .range_max = RT5670_PR_BASE + 0xf8,
+ .selector_reg = RT5670_PRIV_INDEX,
+ .selector_mask = 0xff,
+ .selector_shift = 0x0,
+ .window_start = RT5670_PRIV_DATA,
+ .window_len = 0x1, },
+};
+
+static struct reg_default init_list[] = {
+ { RT5670_PR_BASE + 0x14, 0x9a8a },
+ { RT5670_PR_BASE + 0x38, 0x3ba1 },
+ { RT5670_PR_BASE + 0x3d, 0x3640 },
+};
+#define RT5670_INIT_REG_LEN ARRAY_SIZE(init_list)
+
+static const struct reg_default rt5670_reg[] = {
+ { 0x00, 0x0000 },
+ { 0x02, 0x8888 },
+ { 0x03, 0x8888 },
+ { 0x0a, 0x0001 },
+ { 0x0b, 0x0827 },
+ { 0x0c, 0x0000 },
+ { 0x0d, 0x0008 },
+ { 0x0e, 0x0000 },
+ { 0x0f, 0x0808 },
+ { 0x19, 0xafaf },
+ { 0x1a, 0xafaf },
+ { 0x1b, 0x0011 },
+ { 0x1c, 0x2f2f },
+ { 0x1d, 0x2f2f },
+ { 0x1e, 0x0000 },
+ { 0x1f, 0x2f2f },
+ { 0x20, 0x0000 },
+ { 0x26, 0x7860 },
+ { 0x27, 0x7860 },
+ { 0x28, 0x7871 },
+ { 0x29, 0x8080 },
+ { 0x2a, 0x5656 },
+ { 0x2b, 0x5454 },
+ { 0x2c, 0xaaa0 },
+ { 0x2d, 0x0000 },
+ { 0x2e, 0x2f2f },
+ { 0x2f, 0x1002 },
+ { 0x30, 0x0000 },
+ { 0x31, 0x5f00 },
+ { 0x32, 0x0000 },
+ { 0x33, 0x0000 },
+ { 0x34, 0x0000 },
+ { 0x35, 0x0000 },
+ { 0x36, 0x0000 },
+ { 0x37, 0x0000 },
+ { 0x38, 0x0000 },
+ { 0x3b, 0x0000 },
+ { 0x3c, 0x007f },
+ { 0x3d, 0x0000 },
+ { 0x3e, 0x007f },
+ { 0x45, 0xe00f },
+ { 0x4c, 0x5380 },
+ { 0x4f, 0x0073 },
+ { 0x52, 0x00d3 },
+ { 0x53, 0xf0f0 },
+ { 0x61, 0x0000 },
+ { 0x62, 0x0001 },
+ { 0x63, 0x00c3 },
+ { 0x64, 0x0000 },
+ { 0x65, 0x0000 },
+ { 0x66, 0x0000 },
+ { 0x6f, 0x8000 },
+ { 0x70, 0x8000 },
+ { 0x71, 0x8000 },
+ { 0x72, 0x8000 },
+ { 0x73, 0x1110 },
+ { 0x74, 0x0e00 },
+ { 0x75, 0x1505 },
+ { 0x76, 0x0015 },
+ { 0x77, 0x0c00 },
+ { 0x78, 0x4000 },
+ { 0x79, 0x0123 },
+ { 0x7f, 0x1100 },
+ { 0x80, 0x0000 },
+ { 0x81, 0x0000 },
+ { 0x82, 0x0000 },
+ { 0x83, 0x0000 },
+ { 0x84, 0x0000 },
+ { 0x85, 0x0000 },
+ { 0x86, 0x0008 },
+ { 0x87, 0x0000 },
+ { 0x88, 0x0000 },
+ { 0x89, 0x0000 },
+ { 0x8a, 0x0000 },
+ { 0x8b, 0x0000 },
+ { 0x8c, 0x0007 },
+ { 0x8d, 0x0000 },
+ { 0x8e, 0x0004 },
+ { 0x8f, 0x1100 },
+ { 0x90, 0x0646 },
+ { 0x91, 0x0c06 },
+ { 0x93, 0x0000 },
+ { 0x94, 0x0000 },
+ { 0x95, 0x0000 },
+ { 0x97, 0x0000 },
+ { 0x98, 0x0000 },
+ { 0x99, 0x0000 },
+ { 0x9a, 0x2184 },
+ { 0x9b, 0x010a },
+ { 0x9c, 0x0aea },
+ { 0x9d, 0x000c },
+ { 0x9e, 0x0400 },
+ { 0xae, 0x7000 },
+ { 0xaf, 0x0000 },
+ { 0xb0, 0x6000 },
+ { 0xb1, 0x0000 },
+ { 0xb2, 0x0000 },
+ { 0xb3, 0x001f },
+ { 0xb4, 0x2206 },
+ { 0xb5, 0x1f00 },
+ { 0xb6, 0x0000 },
+ { 0xb7, 0x0000 },
+ { 0xbb, 0x0000 },
+ { 0xbc, 0x0000 },
+ { 0xbd, 0x0000 },
+ { 0xbe, 0x0000 },
+ { 0xbf, 0x0000 },
+ { 0xc0, 0x0000 },
+ { 0xc1, 0x0000 },
+ { 0xc2, 0x0000 },
+ { 0xcd, 0x0000 },
+ { 0xce, 0x0000 },
+ { 0xcf, 0x1813 },
+ { 0xd0, 0x0690 },
+ { 0xd1, 0x1c17 },
+ { 0xd3, 0xb320 },
+ { 0xd4, 0x0000 },
+ { 0xd6, 0x0400 },
+ { 0xd9, 0x0809 },
+ { 0xda, 0x0000 },
+ { 0xdb, 0x0001 },
+ { 0xdc, 0x0049 },
+ { 0xdd, 0x0009 },
+ { 0xe6, 0x8000 },
+ { 0xe7, 0x0000 },
+ { 0xec, 0xb300 },
+ { 0xed, 0x0000 },
+ { 0xee, 0xb300 },
+ { 0xef, 0x0000 },
+ { 0xf8, 0x0000 },
+ { 0xf9, 0x0000 },
+ { 0xfa, 0x8010 },
+ { 0xfb, 0x0033 },
+ { 0xfc, 0x0080 },
+};
+
+static bool rt5670_volatile_register(struct device *dev, unsigned int reg)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(rt5670_ranges); i++) {
+ if ((reg >= rt5670_ranges[i].window_start &&
+ reg <= rt5670_ranges[i].window_start +
+ rt5670_ranges[i].window_len) ||
+ (reg >= rt5670_ranges[i].range_min &&
+ reg <= rt5670_ranges[i].range_max)) {
+ return true;
+ }
+ }
+
+ switch (reg) {
+ case RT5670_RESET:
+ case RT5670_PDM_DATA_CTRL1:
+ case RT5670_PDM1_DATA_CTRL4:
+ case RT5670_PDM2_DATA_CTRL4:
+ case RT5670_PRIV_DATA:
+ case RT5670_ASRC_5:
+ case RT5670_CJ_CTRL1:
+ case RT5670_CJ_CTRL2:
+ case RT5670_CJ_CTRL3:
+ case RT5670_A_JD_CTRL1:
+ case RT5670_A_JD_CTRL2:
+ case RT5670_VAD_CTRL5:
+ case RT5670_ADC_EQ_CTRL1:
+ case RT5670_EQ_CTRL1:
+ case RT5670_ALC_CTRL_1:
+ case RT5670_IRQ_CTRL1:
+ case RT5670_IRQ_CTRL2:
+ case RT5670_INT_IRQ_ST:
+ case RT5670_IL_CMD:
+ case RT5670_DSP_CTRL1:
+ case RT5670_DSP_CTRL2:
+ case RT5670_DSP_CTRL3:
+ case RT5670_DSP_CTRL4:
+ case RT5670_DSP_CTRL5:
+ case RT5670_VENDOR_ID:
+ case RT5670_VENDOR_ID1:
+ case RT5670_VENDOR_ID2:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool rt5670_readable_register(struct device *dev, unsigned int reg)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(rt5670_ranges); i++) {
+ if ((reg >= rt5670_ranges[i].window_start &&
+ reg <= rt5670_ranges[i].window_start +
+ rt5670_ranges[i].window_len) ||
+ (reg >= rt5670_ranges[i].range_min &&
+ reg <= rt5670_ranges[i].range_max)) {
+ return true;
+ }
+ }
+
+ switch (reg) {
+ case RT5670_RESET:
+ case RT5670_HP_VOL:
+ case RT5670_LOUT1:
+ case RT5670_CJ_CTRL1:
+ case RT5670_CJ_CTRL2:
+ case RT5670_CJ_CTRL3:
+ case RT5670_IN2:
+ case RT5670_INL1_INR1_VOL:
+ case RT5670_DAC1_DIG_VOL:
+ case RT5670_DAC2_DIG_VOL:
+ case RT5670_DAC_CTRL:
+ case RT5670_STO1_ADC_DIG_VOL:
+ case RT5670_MONO_ADC_DIG_VOL:
+ case RT5670_STO2_ADC_DIG_VOL:
+ case RT5670_ADC_BST_VOL1:
+ case RT5670_ADC_BST_VOL2:
+ case RT5670_STO2_ADC_MIXER:
+ case RT5670_STO1_ADC_MIXER:
+ case RT5670_MONO_ADC_MIXER:
+ case RT5670_AD_DA_MIXER:
+ case RT5670_STO_DAC_MIXER:
+ case RT5670_DD_MIXER:
+ case RT5670_DIG_MIXER:
+ case RT5670_DSP_PATH1:
+ case RT5670_DSP_PATH2:
+ case RT5670_DIG_INF1_DATA:
+ case RT5670_DIG_INF2_DATA:
+ case RT5670_PDM_OUT_CTRL:
+ case RT5670_PDM_DATA_CTRL1:
+ case RT5670_PDM1_DATA_CTRL2:
+ case RT5670_PDM1_DATA_CTRL3:
+ case RT5670_PDM1_DATA_CTRL4:
+ case RT5670_PDM2_DATA_CTRL2:
+ case RT5670_PDM2_DATA_CTRL3:
+ case RT5670_PDM2_DATA_CTRL4:
+ case RT5670_REC_L1_MIXER:
+ case RT5670_REC_L2_MIXER:
+ case RT5670_REC_R1_MIXER:
+ case RT5670_REC_R2_MIXER:
+ case RT5670_HPO_MIXER:
+ case RT5670_MONO_MIXER:
+ case RT5670_OUT_L1_MIXER:
+ case RT5670_OUT_R1_MIXER:
+ case RT5670_LOUT_MIXER:
+ case RT5670_PWR_DIG1:
+ case RT5670_PWR_DIG2:
+ case RT5670_PWR_ANLG1:
+ case RT5670_PWR_ANLG2:
+ case RT5670_PWR_MIXER:
+ case RT5670_PWR_VOL:
+ case RT5670_PRIV_INDEX:
+ case RT5670_PRIV_DATA:
+ case RT5670_I2S4_SDP:
+ case RT5670_I2S1_SDP:
+ case RT5670_I2S2_SDP:
+ case RT5670_I2S3_SDP:
+ case RT5670_ADDA_CLK1:
+ case RT5670_ADDA_CLK2:
+ case RT5670_DMIC_CTRL1:
+ case RT5670_DMIC_CTRL2:
+ case RT5670_TDM_CTRL_1:
+ case RT5670_TDM_CTRL_2:
+ case RT5670_TDM_CTRL_3:
+ case RT5670_DSP_CLK:
+ case RT5670_GLB_CLK:
+ case RT5670_PLL_CTRL1:
+ case RT5670_PLL_CTRL2:
+ case RT5670_ASRC_1:
+ case RT5670_ASRC_2:
+ case RT5670_ASRC_3:
+ case RT5670_ASRC_4:
+ case RT5670_ASRC_5:
+ case RT5670_ASRC_7:
+ case RT5670_ASRC_8:
+ case RT5670_ASRC_9:
+ case RT5670_ASRC_10:
+ case RT5670_ASRC_11:
+ case RT5670_ASRC_12:
+ case RT5670_ASRC_13:
+ case RT5670_ASRC_14:
+ case RT5670_DEPOP_M1:
+ case RT5670_DEPOP_M2:
+ case RT5670_DEPOP_M3:
+ case RT5670_CHARGE_PUMP:
+ case RT5670_MICBIAS:
+ case RT5670_A_JD_CTRL1:
+ case RT5670_A_JD_CTRL2:
+ case RT5670_VAD_CTRL1:
+ case RT5670_VAD_CTRL2:
+ case RT5670_VAD_CTRL3:
+ case RT5670_VAD_CTRL4:
+ case RT5670_VAD_CTRL5:
+ case RT5670_ADC_EQ_CTRL1:
+ case RT5670_ADC_EQ_CTRL2:
+ case RT5670_EQ_CTRL1:
+ case RT5670_EQ_CTRL2:
+ case RT5670_ALC_DRC_CTRL1:
+ case RT5670_ALC_DRC_CTRL2:
+ case RT5670_ALC_CTRL_1:
+ case RT5670_ALC_CTRL_2:
+ case RT5670_ALC_CTRL_3:
+ case RT5670_JD_CTRL:
+ case RT5670_IRQ_CTRL1:
+ case RT5670_IRQ_CTRL2:
+ case RT5670_INT_IRQ_ST:
+ case RT5670_GPIO_CTRL1:
+ case RT5670_GPIO_CTRL2:
+ case RT5670_GPIO_CTRL3:
+ case RT5670_SCRABBLE_FUN:
+ case RT5670_SCRABBLE_CTRL:
+ case RT5670_BASE_BACK:
+ case RT5670_MP3_PLUS1:
+ case RT5670_MP3_PLUS2:
+ case RT5670_ADJ_HPF1:
+ case RT5670_ADJ_HPF2:
+ case RT5670_HP_CALIB_AMP_DET:
+ case RT5670_SV_ZCD1:
+ case RT5670_SV_ZCD2:
+ case RT5670_IL_CMD:
+ case RT5670_IL_CMD2:
+ case RT5670_IL_CMD3:
+ case RT5670_DRC_HL_CTRL1:
+ case RT5670_DRC_HL_CTRL2:
+ case RT5670_ADC_MONO_HP_CTRL1:
+ case RT5670_ADC_MONO_HP_CTRL2:
+ case RT5670_ADC_STO2_HP_CTRL1:
+ case RT5670_ADC_STO2_HP_CTRL2:
+ case RT5670_JD_CTRL3:
+ case RT5670_JD_CTRL4:
+ case RT5670_DIG_MISC:
+ case RT5670_DSP_CTRL1:
+ case RT5670_DSP_CTRL2:
+ case RT5670_DSP_CTRL3:
+ case RT5670_DSP_CTRL4:
+ case RT5670_DSP_CTRL5:
+ case RT5670_GEN_CTRL2:
+ case RT5670_GEN_CTRL3:
+ case RT5670_VENDOR_ID:
+ case RT5670_VENDOR_ID1:
+ case RT5670_VENDOR_ID2:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
+static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
+
+/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
+static unsigned int bst_tlv[] = {
+ TLV_DB_RANGE_HEAD(7),
+ 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+ 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
+ 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0),
+ 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0),
+ 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0),
+ 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0),
+ 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0),
+};
+
+/* Interface data select */
+static const char * const rt5670_data_select[] = {
+ "Normal", "Swap", "left copy to right", "right copy to left"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA,
+ RT5670_IF2_DAC_SEL_SFT, rt5670_data_select);
+
+static SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA,
+ RT5670_IF2_ADC_SEL_SFT, rt5670_data_select);
+
+static const struct snd_kcontrol_new rt5670_snd_controls[] = {
+ /* Headphone Output Volume */
+ SOC_DOUBLE("HP Playback Switch", RT5670_HP_VOL,
+ RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1),
+ SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL,
+ RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
+ 39, 0, out_vol_tlv),
+ /* OUTPUT Control */
+ SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1,
+ RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1),
+ SOC_DOUBLE_TLV("OUT Playback Volume", RT5670_LOUT1,
+ RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 39, 1, out_vol_tlv),
+ /* DAC Digital Volume */
+ SOC_DOUBLE("DAC2 Playback Switch", RT5670_DAC_CTRL,
+ RT5670_M_DAC_L2_VOL_SFT, RT5670_M_DAC_R2_VOL_SFT, 1, 1),
+ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5670_DAC1_DIG_VOL,
+ RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
+ 175, 0, dac_vol_tlv),
+ SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5670_DAC2_DIG_VOL,
+ RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
+ 175, 0, dac_vol_tlv),
+ /* IN1/IN2 Control */
+ SOC_SINGLE_TLV("IN1 Boost Volume", RT5670_CJ_CTRL1,
+ RT5670_BST_SFT1, 8, 0, bst_tlv),
+ SOC_SINGLE_TLV("IN2 Boost Volume", RT5670_IN2,
+ RT5670_BST_SFT1, 8, 0, bst_tlv),
+ /* INL/INR Volume Control */
+ SOC_DOUBLE_TLV("IN Capture Volume", RT5670_INL1_INR1_VOL,
+ RT5670_INL_VOL_SFT, RT5670_INR_VOL_SFT,
+ 31, 1, in_vol_tlv),
+ /* ADC Digital Volume Control */
+ SOC_DOUBLE("ADC Capture Switch", RT5670_STO1_ADC_DIG_VOL,
+ RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1),
+ SOC_DOUBLE_TLV("ADC Capture Volume", RT5670_STO1_ADC_DIG_VOL,
+ RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
+ 127, 0, adc_vol_tlv),
+
+ SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5670_MONO_ADC_DIG_VOL,
+ RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
+ 127, 0, adc_vol_tlv),
+
+ /* ADC Boost Volume Control */
+ SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5670_ADC_BST_VOL1,
+ RT5670_STO1_ADC_L_BST_SFT, RT5670_STO1_ADC_R_BST_SFT,
+ 3, 0, adc_bst_tlv),
+
+ SOC_DOUBLE_TLV("STO2 ADC Boost Gain Volume", RT5670_ADC_BST_VOL1,
+ RT5670_STO2_ADC_L_BST_SFT, RT5670_STO2_ADC_R_BST_SFT,
+ 3, 0, adc_bst_tlv),
+
+ SOC_ENUM("ADC IF2 Data Switch", rt5670_if2_adc_enum),
+ SOC_ENUM("DAC IF2 Data Switch", rt5670_if2_dac_enum),
+};
+
+/**
+ * set_dmic_clk - Set parameter of dmic.
+ *
+ * @w: DAPM widget.
+ * @kcontrol: The kcontrol of this widget.
+ * @event: Event id.
+ *
+ * Choose dmic clock between 1MHz and 3MHz.
+ * It is better for clock to approximate 3MHz.
+ */
+static int set_dmic_clk(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
+ int idx = -EINVAL;
+
+ idx = rl6231_calc_dmic_clk(rt5670->sysclk);
+
+ if (idx < 0)
+ dev_err(codec->dev, "Failed to set DMIC clock\n");
+ else
+ snd_soc_update_bits(codec, RT5670_DMIC_CTRL1,
+ RT5670_DMIC_CLK_MASK, idx << RT5670_DMIC_CLK_SFT);
+ return idx;
+}
+
+static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
+{
+ unsigned int val;
+
+ val = snd_soc_read(source->codec, RT5670_GLB_CLK);
+ val &= RT5670_SCLK_SRC_MASK;
+ if (val == RT5670_SCLK_SRC_PLL1)
+ return 1;
+ else
+ return 0;
+}
+
+static int is_using_asrc(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
+{
+ unsigned int reg, shift, val;
+
+ switch (source->shift) {
+ case 0:
+ reg = RT5670_ASRC_3;
+ shift = 0;
+ break;
+ case 1:
+ reg = RT5670_ASRC_3;
+ shift = 4;
+ break;
+ case 2:
+ reg = RT5670_ASRC_5;
+ shift = 12;
+ break;
+ case 3:
+ reg = RT5670_ASRC_2;
+ shift = 0;
+ break;
+ case 8:
+ reg = RT5670_ASRC_2;
+ shift = 4;
+ break;
+ case 9:
+ reg = RT5670_ASRC_2;
+ shift = 8;
+ break;
+ case 10:
+ reg = RT5670_ASRC_2;
+ shift = 12;
+ break;
+ default:
+ return 0;
+ }
+
+ val = (snd_soc_read(source->codec, reg) >> shift) & 0xf;
+ switch (val) {
+ case 1:
+ case 2:
+ case 3:
+ case 4:
+ return 1;
+ default:
+ return 0;
+ }
+
+}
+
+/* Digital Mixer */
+static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = {
+ SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER,
+ RT5670_M_ADC_L1_SFT, 1, 1),
+ SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO1_ADC_MIXER,
+ RT5670_M_ADC_L2_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_sto1_adc_r_mix[] = {
+ SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER,
+ RT5670_M_ADC_R1_SFT, 1, 1),
+ SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO1_ADC_MIXER,
+ RT5670_M_ADC_R2_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_sto2_adc_l_mix[] = {
+ SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO2_ADC_MIXER,
+ RT5670_M_ADC_L1_SFT, 1, 1),
+ SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO2_ADC_MIXER,
+ RT5670_M_ADC_L2_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_sto2_adc_r_mix[] = {
+ SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO2_ADC_MIXER,
+ RT5670_M_ADC_R1_SFT, 1, 1),
+ SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO2_ADC_MIXER,
+ RT5670_M_ADC_R2_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_mono_adc_l_mix[] = {
+ SOC_DAPM_SINGLE("ADC1 Switch", RT5670_MONO_ADC_MIXER,
+ RT5670_M_MONO_ADC_L1_SFT, 1, 1),
+ SOC_DAPM_SINGLE("ADC2 Switch", RT5670_MONO_ADC_MIXER,
+ RT5670_M_MONO_ADC_L2_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_mono_adc_r_mix[] = {
+ SOC_DAPM_SINGLE("ADC1 Switch", RT5670_MONO_ADC_MIXER,
+ RT5670_M_MONO_ADC_R1_SFT, 1, 1),
+ SOC_DAPM_SINGLE("ADC2 Switch", RT5670_MONO_ADC_MIXER,
+ RT5670_M_MONO_ADC_R2_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_dac_l_mix[] = {
+ SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER,
+ RT5670_M_ADCMIX_L_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER,
+ RT5670_M_DAC1_L_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_dac_r_mix[] = {
+ SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER,
+ RT5670_M_ADCMIX_R_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER,
+ RT5670_M_DAC1_R_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_sto_dac_l_mix[] = {
+ SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_STO_DAC_MIXER,
+ RT5670_M_DAC_L1_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_STO_DAC_MIXER,
+ RT5670_M_DAC_L2_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_STO_DAC_MIXER,
+ RT5670_M_DAC_R1_STO_L_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_sto_dac_r_mix[] = {
+ SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_STO_DAC_MIXER,
+ RT5670_M_DAC_R1_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_STO_DAC_MIXER,
+ RT5670_M_DAC_R2_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_STO_DAC_MIXER,
+ RT5670_M_DAC_L1_STO_R_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_mono_dac_l_mix[] = {
+ SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_DD_MIXER,
+ RT5670_M_DAC_L1_MONO_L_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DD_MIXER,
+ RT5670_M_DAC_L2_MONO_L_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DD_MIXER,
+ RT5670_M_DAC_R2_MONO_L_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_mono_dac_r_mix[] = {
+ SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_DD_MIXER,
+ RT5670_M_DAC_R1_MONO_R_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DD_MIXER,
+ RT5670_M_DAC_R2_MONO_R_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DD_MIXER,
+ RT5670_M_DAC_L2_MONO_R_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_dig_l_mix[] = {
+ SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5670_DIG_MIXER,
+ RT5670_M_STO_L_DAC_L_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DIG_MIXER,
+ RT5670_M_DAC_L2_DAC_L_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DIG_MIXER,
+ RT5670_M_DAC_R2_DAC_L_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_dig_r_mix[] = {
+ SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5670_DIG_MIXER,
+ RT5670_M_STO_R_DAC_R_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DIG_MIXER,
+ RT5670_M_DAC_R2_DAC_R_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DIG_MIXER,
+ RT5670_M_DAC_L2_DAC_R_SFT, 1, 1),
+};
+
+/* Analog Input Mixer */
+static const struct snd_kcontrol_new rt5670_rec_l_mix[] = {
+ SOC_DAPM_SINGLE("INL Switch", RT5670_REC_L2_MIXER,
+ RT5670_M_IN_L_RM_L_SFT, 1, 1),
+ SOC_DAPM_SINGLE("BST2 Switch", RT5670_REC_L2_MIXER,
+ RT5670_M_BST2_RM_L_SFT, 1, 1),
+ SOC_DAPM_SINGLE("BST1 Switch", RT5670_REC_L2_MIXER,
+ RT5670_M_BST1_RM_L_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_rec_r_mix[] = {
+ SOC_DAPM_SINGLE("INR Switch", RT5670_REC_R2_MIXER,
+ RT5670_M_IN_R_RM_R_SFT, 1, 1),
+ SOC_DAPM_SINGLE("BST2 Switch", RT5670_REC_R2_MIXER,
+ RT5670_M_BST2_RM_R_SFT, 1, 1),
+ SOC_DAPM_SINGLE("BST1 Switch", RT5670_REC_R2_MIXER,
+ RT5670_M_BST1_RM_R_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_out_l_mix[] = {
+ SOC_DAPM_SINGLE("BST1 Switch", RT5670_OUT_L1_MIXER,
+ RT5670_M_BST1_OM_L_SFT, 1, 1),
+ SOC_DAPM_SINGLE("INL Switch", RT5670_OUT_L1_MIXER,
+ RT5670_M_IN_L_OM_L_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_OUT_L1_MIXER,
+ RT5670_M_DAC_L2_OM_L_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_OUT_L1_MIXER,
+ RT5670_M_DAC_L1_OM_L_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_out_r_mix[] = {
+ SOC_DAPM_SINGLE("BST2 Switch", RT5670_OUT_R1_MIXER,
+ RT5670_M_BST2_OM_R_SFT, 1, 1),
+ SOC_DAPM_SINGLE("INR Switch", RT5670_OUT_R1_MIXER,
+ RT5670_M_IN_R_OM_R_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_OUT_R1_MIXER,
+ RT5670_M_DAC_R2_OM_R_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_OUT_R1_MIXER,
+ RT5670_M_DAC_R1_OM_R_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_hpo_mix[] = {
+ SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER,
+ RT5670_M_DAC1_HM_SFT, 1, 1),
+ SOC_DAPM_SINGLE("HPVOL Switch", RT5670_HPO_MIXER,
+ RT5670_M_HPVOL_HM_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_hpvoll_mix[] = {
+ SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER,
+ RT5670_M_DACL1_HML_SFT, 1, 1),
+ SOC_DAPM_SINGLE("INL Switch", RT5670_HPO_MIXER,
+ RT5670_M_INL1_HML_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_hpvolr_mix[] = {
+ SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER,
+ RT5670_M_DACR1_HMR_SFT, 1, 1),
+ SOC_DAPM_SINGLE("INR Switch", RT5670_HPO_MIXER,
+ RT5670_M_INR1_HMR_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_lout_mix[] = {
+ SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_LOUT_MIXER,
+ RT5670_M_DAC_L1_LM_SFT, 1, 1),
+ SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_LOUT_MIXER,
+ RT5670_M_DAC_R1_LM_SFT, 1, 1),
+ SOC_DAPM_SINGLE("OUTMIX L Switch", RT5670_LOUT_MIXER,
+ RT5670_M_OV_L_LM_SFT, 1, 1),
+ SOC_DAPM_SINGLE("OUTMIX R Switch", RT5670_LOUT_MIXER,
+ RT5670_M_OV_R_LM_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_hpl_mix[] = {
+ SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_HPO_MIXER,
+ RT5670_M_DACL1_HML_SFT, 1, 1),
+ SOC_DAPM_SINGLE("INL1 Switch", RT5670_HPO_MIXER,
+ RT5670_M_INL1_HML_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5670_hpr_mix[] = {
+ SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_HPO_MIXER,
+ RT5670_M_DACR1_HMR_SFT, 1, 1),
+ SOC_DAPM_SINGLE("INR1 Switch", RT5670_HPO_MIXER,
+ RT5670_M_INR1_HMR_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new lout_l_enable_control =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5670_LOUT1,
+ RT5670_L_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new lout_r_enable_control =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5670_LOUT1,
+ RT5670_R_MUTE_SFT, 1, 1);
+
+/* DAC1 L/R source */ /* MX-29 [9:8] [11:10] */
+static const char * const rt5670_dac1_src[] = {
+ "IF1 DAC", "IF2 DAC"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_dac1l_enum, RT5670_AD_DA_MIXER,
+ RT5670_DAC1_L_SEL_SFT, rt5670_dac1_src);
+
+static const struct snd_kcontrol_new rt5670_dac1l_mux =
+ SOC_DAPM_ENUM("DAC1 L source", rt5670_dac1l_enum);
+
+static SOC_ENUM_SINGLE_DECL(rt5670_dac1r_enum, RT5670_AD_DA_MIXER,
+ RT5670_DAC1_R_SEL_SFT, rt5670_dac1_src);
+
+static const struct snd_kcontrol_new rt5670_dac1r_mux =
+ SOC_DAPM_ENUM("DAC1 R source", rt5670_dac1r_enum);
+
+/*DAC2 L/R source*/ /* MX-1B [6:4] [2:0] */
+/* TODO Use SOC_VALUE_ENUM_SINGLE_DECL */
+static const char * const rt5670_dac12_src[] = {
+ "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC",
+ "Bass", "VAD_ADC", "IF4 DAC"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_dac2l_enum, RT5670_DAC_CTRL,
+ RT5670_DAC2_L_SEL_SFT, rt5670_dac12_src);
+
+static const struct snd_kcontrol_new rt5670_dac_l2_mux =
+ SOC_DAPM_ENUM("DAC2 L source", rt5670_dac2l_enum);
+
+static const char * const rt5670_dacr2_src[] = {
+ "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC", "TxDP ADC", "IF4 DAC"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_dac2r_enum, RT5670_DAC_CTRL,
+ RT5670_DAC2_R_SEL_SFT, rt5670_dacr2_src);
+
+static const struct snd_kcontrol_new rt5670_dac_r2_mux =
+ SOC_DAPM_ENUM("DAC2 R source", rt5670_dac2r_enum);
+
+/*RxDP source*/ /* MX-2D [15:13] */
+static const char * const rt5670_rxdp_src[] = {
+ "IF2 DAC", "IF1 DAC", "STO1 ADC Mixer", "STO2 ADC Mixer",
+ "Mono ADC Mixer L", "Mono ADC Mixer R", "DAC1"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_rxdp_enum, RT5670_DSP_PATH1,
+ RT5670_RXDP_SEL_SFT, rt5670_rxdp_src);
+
+static const struct snd_kcontrol_new rt5670_rxdp_mux =
+ SOC_DAPM_ENUM("DAC2 L source", rt5670_rxdp_enum);
+
+/* MX-2D [1] [0] */
+static const char * const rt5670_dsp_bypass_src[] = {
+ "DSP", "Bypass"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_dsp_ul_enum, RT5670_DSP_PATH1,
+ RT5670_DSP_UL_SFT, rt5670_dsp_bypass_src);
+
+static const struct snd_kcontrol_new rt5670_dsp_ul_mux =
+ SOC_DAPM_ENUM("DSP UL source", rt5670_dsp_ul_enum);
+
+static SOC_ENUM_SINGLE_DECL(rt5670_dsp_dl_enum, RT5670_DSP_PATH1,
+ RT5670_DSP_DL_SFT, rt5670_dsp_bypass_src);
+
+static const struct snd_kcontrol_new rt5670_dsp_dl_mux =
+ SOC_DAPM_ENUM("DSP DL source", rt5670_dsp_dl_enum);
+
+/* Stereo2 ADC source */
+/* MX-26 [15] */
+static const char * const rt5670_stereo2_adc_lr_src[] = {
+ "L", "LR"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc_lr_enum, RT5670_STO2_ADC_MIXER,
+ RT5670_STO2_ADC_SRC_SFT, rt5670_stereo2_adc_lr_src);
+
+static const struct snd_kcontrol_new rt5670_sto2_adc_lr_mux =
+ SOC_DAPM_ENUM("Stereo2 ADC LR source", rt5670_stereo2_adc_lr_enum);
+
+/* Stereo1 ADC source */
+/* MX-27 MX-26 [12] */
+static const char * const rt5670_stereo_adc1_src[] = {
+ "DAC MIX", "ADC"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc1_enum, RT5670_STO1_ADC_MIXER,
+ RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src);
+
+static const struct snd_kcontrol_new rt5670_sto_adc_l1_mux =
+ SOC_DAPM_ENUM("Stereo1 ADC L1 source", rt5670_stereo1_adc1_enum);
+
+static const struct snd_kcontrol_new rt5670_sto_adc_r1_mux =
+ SOC_DAPM_ENUM("Stereo1 ADC R1 source", rt5670_stereo1_adc1_enum);
+
+static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc1_enum, RT5670_STO2_ADC_MIXER,
+ RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src);
+
+static const struct snd_kcontrol_new rt5670_sto2_adc_l1_mux =
+ SOC_DAPM_ENUM("Stereo2 ADC L1 source", rt5670_stereo2_adc1_enum);
+
+static const struct snd_kcontrol_new rt5670_sto2_adc_r1_mux =
+ SOC_DAPM_ENUM("Stereo2 ADC R1 source", rt5670_stereo2_adc1_enum);
+
+/* MX-27 MX-26 [11] */
+static const char * const rt5670_stereo_adc2_src[] = {
+ "DAC MIX", "DMIC"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc2_enum, RT5670_STO1_ADC_MIXER,
+ RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src);
+
+static const struct snd_kcontrol_new rt5670_sto_adc_l2_mux =
+ SOC_DAPM_ENUM("Stereo1 ADC L2 source", rt5670_stereo1_adc2_enum);
+
+static const struct snd_kcontrol_new rt5670_sto_adc_r2_mux =
+ SOC_DAPM_ENUM("Stereo1 ADC R2 source", rt5670_stereo1_adc2_enum);
+
+static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER,
+ RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src);
+
+static const struct snd_kcontrol_new rt5670_sto2_adc_l2_mux =
+ SOC_DAPM_ENUM("Stereo2 ADC L2 source", rt5670_stereo2_adc2_enum);
+
+static const struct snd_kcontrol_new rt5670_sto2_adc_r2_mux =
+ SOC_DAPM_ENUM("Stereo2 ADC R2 source", rt5670_stereo2_adc2_enum);
+
+/* MX-27 MX26 [10] */
+static const char * const rt5670_stereo_adc_src[] = {
+ "ADC1L ADC2R", "ADC3"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc_enum, RT5670_STO1_ADC_MIXER,
+ RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src);
+
+static const struct snd_kcontrol_new rt5670_sto_adc_mux =
+ SOC_DAPM_ENUM("Stereo1 ADC source", rt5670_stereo1_adc_enum);
+
+static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc_enum, RT5670_STO2_ADC_MIXER,
+ RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src);
+
+static const struct snd_kcontrol_new rt5670_sto2_adc_mux =
+ SOC_DAPM_ENUM("Stereo2 ADC source", rt5670_stereo2_adc_enum);
+
+/* MX-27 MX-26 [9:8] */
+static const char * const rt5670_stereo_dmic_src[] = {
+ "DMIC1", "DMIC2", "DMIC3"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_dmic_enum, RT5670_STO1_ADC_MIXER,
+ RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src);
+
+static const struct snd_kcontrol_new rt5670_sto1_dmic_mux =
+ SOC_DAPM_ENUM("Stereo1 DMIC source", rt5670_stereo1_dmic_enum);
+
+static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_dmic_enum, RT5670_STO2_ADC_MIXER,
+ RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src);
+
+static const struct snd_kcontrol_new rt5670_sto2_dmic_mux =
+ SOC_DAPM_ENUM("Stereo2 DMIC source", rt5670_stereo2_dmic_enum);
+
+/* MX-27 [0] */
+static const char * const rt5670_stereo_dmic3_src[] = {
+ "DMIC3", "PDM ADC"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_stereo_dmic3_enum, RT5670_STO1_ADC_MIXER,
+ RT5670_DMIC3_SRC_SFT, rt5670_stereo_dmic3_src);
+
+static const struct snd_kcontrol_new rt5670_sto_dmic3_mux =
+ SOC_DAPM_ENUM("Stereo DMIC3 source", rt5670_stereo_dmic3_enum);
+
+/* Mono ADC source */
+/* MX-28 [12] */
+static const char * const rt5670_mono_adc_l1_src[] = {
+ "Mono DAC MIXL", "ADC1"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_mono_adc_l1_enum, RT5670_MONO_ADC_MIXER,
+ RT5670_MONO_ADC_L1_SRC_SFT, rt5670_mono_adc_l1_src);
+
+static const struct snd_kcontrol_new rt5670_mono_adc_l1_mux =
+ SOC_DAPM_ENUM("Mono ADC1 left source", rt5670_mono_adc_l1_enum);
+/* MX-28 [11] */
+static const char * const rt5670_mono_adc_l2_src[] = {
+ "Mono DAC MIXL", "DMIC"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_mono_adc_l2_enum, RT5670_MONO_ADC_MIXER,
+ RT5670_MONO_ADC_L2_SRC_SFT, rt5670_mono_adc_l2_src);
+
+static const struct snd_kcontrol_new rt5670_mono_adc_l2_mux =
+ SOC_DAPM_ENUM("Mono ADC2 left source", rt5670_mono_adc_l2_enum);
+
+/* MX-28 [9:8] */
+static const char * const rt5670_mono_dmic_src[] = {
+ "DMIC1", "DMIC2", "DMIC3"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_mono_dmic_l_enum, RT5670_MONO_ADC_MIXER,
+ RT5670_MONO_DMIC_L_SRC_SFT, rt5670_mono_dmic_src);
+
+static const struct snd_kcontrol_new rt5670_mono_dmic_l_mux =
+ SOC_DAPM_ENUM("Mono DMIC left source", rt5670_mono_dmic_l_enum);
+/* MX-28 [1:0] */
+static SOC_ENUM_SINGLE_DECL(rt5670_mono_dmic_r_enum, RT5670_MONO_ADC_MIXER,
+ RT5670_MONO_DMIC_R_SRC_SFT, rt5670_mono_dmic_src);
+
+static const struct snd_kcontrol_new rt5670_mono_dmic_r_mux =
+ SOC_DAPM_ENUM("Mono DMIC Right source", rt5670_mono_dmic_r_enum);
+/* MX-28 [4] */
+static const char * const rt5670_mono_adc_r1_src[] = {
+ "Mono DAC MIXR", "ADC2"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_mono_adc_r1_enum, RT5670_MONO_ADC_MIXER,
+ RT5670_MONO_ADC_R1_SRC_SFT, rt5670_mono_adc_r1_src);
+
+static const struct snd_kcontrol_new rt5670_mono_adc_r1_mux =
+ SOC_DAPM_ENUM("Mono ADC1 right source", rt5670_mono_adc_r1_enum);
+/* MX-28 [3] */
+static const char * const rt5670_mono_adc_r2_src[] = {
+ "Mono DAC MIXR", "DMIC"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_mono_adc_r2_enum, RT5670_MONO_ADC_MIXER,
+ RT5670_MONO_ADC_R2_SRC_SFT, rt5670_mono_adc_r2_src);
+
+static const struct snd_kcontrol_new rt5670_mono_adc_r2_mux =
+ SOC_DAPM_ENUM("Mono ADC2 right source", rt5670_mono_adc_r2_enum);
+
+/* MX-2D [3:2] */
+static const char * const rt5670_txdp_slot_src[] = {
+ "Slot 0-1", "Slot 2-3", "Slot 4-5", "Slot 6-7"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_txdp_slot_enum, RT5670_DSP_PATH1,
+ RT5670_TXDP_SLOT_SEL_SFT, rt5670_txdp_slot_src);
+
+static const struct snd_kcontrol_new rt5670_txdp_slot_mux =
+ SOC_DAPM_ENUM("TxDP Slot source", rt5670_txdp_slot_enum);
+
+/* MX-2F [15] */
+static const char * const rt5670_if1_adc2_in_src[] = {
+ "IF_ADC2", "VAD_ADC"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_if1_adc2_in_enum, RT5670_DIG_INF1_DATA,
+ RT5670_IF1_ADC2_IN_SFT, rt5670_if1_adc2_in_src);
+
+static const struct snd_kcontrol_new rt5670_if1_adc2_in_mux =
+ SOC_DAPM_ENUM("IF1 ADC2 IN source", rt5670_if1_adc2_in_enum);
+
+/* MX-2F [14:12] */
+static const char * const rt5670_if2_adc_in_src[] = {
+ "IF_ADC1", "IF_ADC2", "IF_ADC3", "TxDC_DAC", "TxDP_ADC", "VAD_ADC"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_in_enum, RT5670_DIG_INF1_DATA,
+ RT5670_IF2_ADC_IN_SFT, rt5670_if2_adc_in_src);
+
+static const struct snd_kcontrol_new rt5670_if2_adc_in_mux =
+ SOC_DAPM_ENUM("IF2 ADC IN source", rt5670_if2_adc_in_enum);
+
+/* MX-30 [5:4] */
+static const char * const rt5670_if4_adc_in_src[] = {
+ "IF_ADC1", "IF_ADC2", "IF_ADC3"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_if4_adc_in_enum, RT5670_DIG_INF2_DATA,
+ RT5670_IF4_ADC_IN_SFT, rt5670_if4_adc_in_src);
+
+static const struct snd_kcontrol_new rt5670_if4_adc_in_mux =
+ SOC_DAPM_ENUM("IF4 ADC IN source", rt5670_if4_adc_in_enum);
+
+/* MX-31 [15] [13] [11] [9] */
+static const char * const rt5670_pdm_src[] = {
+ "Mono DAC", "Stereo DAC"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_pdm1_l_enum, RT5670_PDM_OUT_CTRL,
+ RT5670_PDM1_L_SFT, rt5670_pdm_src);
+
+static const struct snd_kcontrol_new rt5670_pdm1_l_mux =
+ SOC_DAPM_ENUM("PDM1 L source", rt5670_pdm1_l_enum);
+
+static SOC_ENUM_SINGLE_DECL(rt5670_pdm1_r_enum, RT5670_PDM_OUT_CTRL,
+ RT5670_PDM1_R_SFT, rt5670_pdm_src);
+
+static const struct snd_kcontrol_new rt5670_pdm1_r_mux =
+ SOC_DAPM_ENUM("PDM1 R source", rt5670_pdm1_r_enum);
+
+static SOC_ENUM_SINGLE_DECL(rt5670_pdm2_l_enum, RT5670_PDM_OUT_CTRL,
+ RT5670_PDM2_L_SFT, rt5670_pdm_src);
+
+static const struct snd_kcontrol_new rt5670_pdm2_l_mux =
+ SOC_DAPM_ENUM("PDM2 L source", rt5670_pdm2_l_enum);
+
+static SOC_ENUM_SINGLE_DECL(rt5670_pdm2_r_enum, RT5670_PDM_OUT_CTRL,
+ RT5670_PDM2_R_SFT, rt5670_pdm_src);
+
+static const struct snd_kcontrol_new rt5670_pdm2_r_mux =
+ SOC_DAPM_ENUM("PDM2 R source", rt5670_pdm2_r_enum);
+
+/* MX-FA [12] */
+static const char * const rt5670_if1_adc1_in1_src[] = {
+ "IF_ADC1", "IF1_ADC3"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_if1_adc1_in1_enum, RT5670_DIG_MISC,
+ RT5670_IF1_ADC1_IN1_SFT, rt5670_if1_adc1_in1_src);
+
+static const struct snd_kcontrol_new rt5670_if1_adc1_in1_mux =
+ SOC_DAPM_ENUM("IF1 ADC1 IN1 source", rt5670_if1_adc1_in1_enum);
+
+/* MX-FA [11] */
+static const char * const rt5670_if1_adc1_in2_src[] = {
+ "IF1_ADC1_IN1", "IF1_ADC4"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_if1_adc1_in2_enum, RT5670_DIG_MISC,
+ RT5670_IF1_ADC1_IN2_SFT, rt5670_if1_adc1_in2_src);
+
+static const struct snd_kcontrol_new rt5670_if1_adc1_in2_mux =
+ SOC_DAPM_ENUM("IF1 ADC1 IN2 source", rt5670_if1_adc1_in2_enum);
+
+/* MX-FA [10] */
+static const char * const rt5670_if1_adc2_in1_src[] = {
+ "IF1_ADC2_IN", "IF1_ADC4"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_if1_adc2_in1_enum, RT5670_DIG_MISC,
+ RT5670_IF1_ADC2_IN1_SFT, rt5670_if1_adc2_in1_src);
+
+static const struct snd_kcontrol_new rt5670_if1_adc2_in1_mux =
+ SOC_DAPM_ENUM("IF1 ADC2 IN1 source", rt5670_if1_adc2_in1_enum);
+
+/* MX-9D [9:8] */
+static const char * const rt5670_vad_adc_src[] = {
+ "Sto1 ADC L", "Mono ADC L", "Mono ADC R", "Sto2 ADC L"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5670_vad_adc_enum, RT5670_VAD_CTRL4,
+ RT5670_VAD_SEL_SFT, rt5670_vad_adc_src);
+
+static const struct snd_kcontrol_new rt5670_vad_adc_mux =
+ SOC_DAPM_ENUM("VAD ADC source", rt5670_vad_adc_enum);
+
+static int rt5670_hp_power_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ regmap_update_bits(rt5670->regmap, RT5670_CHARGE_PUMP,
+ RT5670_PM_HP_MASK, RT5670_PM_HP_HV);
+ regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2,
+ 0x0400, 0x0400);
+ /* headphone amp power on */
+ regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1,
+ RT5670_PWR_HA | RT5670_PWR_FV1 |
+ RT5670_PWR_FV2, RT5670_PWR_HA |
+ RT5670_PWR_FV1 | RT5670_PWR_FV2);
+ /* depop parameters */
+ regmap_write(rt5670->regmap, RT5670_DEPOP_M2, 0x3100);
+ regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8009);
+ regmap_write(rt5670->regmap, RT5670_PR_BASE +
+ RT5670_HP_DCC_INT1, 0x9f00);
+ mdelay(20);
+ regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x0004);
+ msleep(30);
+ break;
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int rt5670_hp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ /* headphone unmute sequence */
+ regmap_write(rt5670->regmap, RT5670_PR_BASE +
+ RT5670_MAMP_INT_REG2, 0xb400);
+ regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0772);
+ regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x805d);
+ regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x831d);
+ regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2,
+ 0x0300, 0x0300);
+ regmap_update_bits(rt5670->regmap, RT5670_HP_VOL,
+ RT5670_L_MUTE | RT5670_R_MUTE, 0);
+ msleep(80);
+ regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019);
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ /* headphone mute sequence */
+ regmap_write(rt5670->regmap, RT5670_PR_BASE +
+ RT5670_MAMP_INT_REG2, 0xb400);
+ regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0772);
+ regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x803d);
+ mdelay(10);
+ regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x831d);
+ mdelay(10);
+ regmap_update_bits(rt5670->regmap, RT5670_HP_VOL,
+ RT5670_L_MUTE | RT5670_R_MUTE,
+ RT5670_L_MUTE | RT5670_R_MUTE);
+ msleep(20);
+ regmap_update_bits(rt5670->regmap,
+ RT5670_GEN_CTRL2, 0x0300, 0x0);
+ regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019);
+ regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0707);
+ regmap_write(rt5670->regmap, RT5670_PR_BASE +
+ RT5670_MAMP_INT_REG2, 0xfc00);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int rt5670_bst1_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec, RT5670_PWR_ANLG2,
+ RT5670_PWR_BST1_P, RT5670_PWR_BST1_P);
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_update_bits(codec, RT5670_PWR_ANLG2,
+ RT5670_PWR_BST1_P, 0);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int rt5670_bst2_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec, RT5670_PWR_ANLG2,
+ RT5670_PWR_BST2_P, RT5670_PWR_BST2_P);
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_update_bits(codec, RT5670_PWR_ANLG2,
+ RT5670_PWR_BST2_P, 0);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = {
+ SND_SOC_DAPM_SUPPLY("PLL1", RT5670_PWR_ANLG2,
+ RT5670_PWR_PLL_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("I2S DSP", RT5670_PWR_DIG2,
+ RT5670_PWR_I2S_DSP_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5670_PWR_VOL,
+ RT5670_PWR_MIC_DET_BIT, 0, NULL, 0),
+
+ /* ASRC */
+ SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5670_ASRC_1,
+ 11, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("I2S2 ASRC", 1, RT5670_ASRC_1,
+ 12, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DAC STO ASRC", 1, RT5670_ASRC_1,
+ 10, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DAC MONO L ASRC", 1, RT5670_ASRC_1,
+ 9, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DAC MONO R ASRC", 1, RT5670_ASRC_1,
+ 8, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5670_ASRC_1,
+ 3, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5670_ASRC_1,
+ 2, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("ADC MONO L ASRC", 1, RT5670_ASRC_1,
+ 1, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("ADC MONO R ASRC", 1, RT5670_ASRC_1,
+ 0, 0, NULL, 0),
+
+ /* Input Side */
+ /* micbias */
+ SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5670_PWR_ANLG2,
+ RT5670_PWR_MB1_BIT, 0, NULL, 0),
+
+ /* Input Lines */
+ SND_SOC_DAPM_INPUT("DMIC L1"),
+ SND_SOC_DAPM_INPUT("DMIC R1"),
+ SND_SOC_DAPM_INPUT("DMIC L2"),
+ SND_SOC_DAPM_INPUT("DMIC R2"),
+ SND_SOC_DAPM_INPUT("DMIC L3"),
+ SND_SOC_DAPM_INPUT("DMIC R3"),
+
+ SND_SOC_DAPM_INPUT("IN1P"),
+ SND_SOC_DAPM_INPUT("IN1N"),
+ SND_SOC_DAPM_INPUT("IN2P"),
+ SND_SOC_DAPM_INPUT("IN2N"),
+
+ SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DMIC3", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0,
+ set_dmic_clk, SND_SOC_DAPM_PRE_PMU),
+ SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5670_DMIC_CTRL1,
+ RT5670_DMIC_1_EN_SFT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5670_DMIC_CTRL1,
+ RT5670_DMIC_2_EN_SFT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DMIC3 Power", RT5670_DMIC_CTRL1,
+ RT5670_DMIC_3_EN_SFT, 0, NULL, 0),
+ /* Boost */
+ SND_SOC_DAPM_PGA_E("BST1", RT5670_PWR_ANLG2, RT5670_PWR_BST1_BIT,
+ 0, NULL, 0, rt5670_bst1_event,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PGA_E("BST2", RT5670_PWR_ANLG2, RT5670_PWR_BST2_BIT,
+ 0, NULL, 0, rt5670_bst2_event,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ /* Input Volume */
+ SND_SOC_DAPM_PGA("INL VOL", RT5670_PWR_VOL,
+ RT5670_PWR_IN_L_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("INR VOL", RT5670_PWR_VOL,
+ RT5670_PWR_IN_R_BIT, 0, NULL, 0),
+
+ /* REC Mixer */
+ SND_SOC_DAPM_MIXER("RECMIXL", RT5670_PWR_MIXER, RT5670_PWR_RM_L_BIT, 0,
+ rt5670_rec_l_mix, ARRAY_SIZE(rt5670_rec_l_mix)),
+ SND_SOC_DAPM_MIXER("RECMIXR", RT5670_PWR_MIXER, RT5670_PWR_RM_R_BIT, 0,
+ rt5670_rec_r_mix, ARRAY_SIZE(rt5670_rec_r_mix)),
+ /* ADCs */
+ SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_ADC("ADC 2", NULL, SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_PGA("ADC 1_2", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("ADC 1 power", RT5670_PWR_DIG1,
+ RT5670_PWR_ADC_L_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC 2 power", RT5670_PWR_DIG1,
+ RT5670_PWR_ADC_R_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC clock", RT5670_PR_BASE +
+ RT5670_CHOP_DAC_ADC, 12, 0, NULL, 0),
+ /* ADC Mux */
+ SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_sto1_dmic_mux),
+ SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_sto_adc_l2_mux),
+ SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_sto_adc_r2_mux),
+ SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_sto_adc_l1_mux),
+ SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_sto_adc_r1_mux),
+ SND_SOC_DAPM_MUX("Stereo2 DMIC Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_sto2_dmic_mux),
+ SND_SOC_DAPM_MUX("Stereo2 ADC L2 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_sto2_adc_l2_mux),
+ SND_SOC_DAPM_MUX("Stereo2 ADC R2 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_sto2_adc_r2_mux),
+ SND_SOC_DAPM_MUX("Stereo2 ADC L1 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_sto2_adc_l1_mux),
+ SND_SOC_DAPM_MUX("Stereo2 ADC R1 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_sto2_adc_r1_mux),
+ SND_SOC_DAPM_MUX("Stereo2 ADC LR Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_sto2_adc_lr_mux),
+ SND_SOC_DAPM_MUX("Mono DMIC L Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_mono_dmic_l_mux),
+ SND_SOC_DAPM_MUX("Mono DMIC R Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_mono_dmic_r_mux),
+ SND_SOC_DAPM_MUX("Mono ADC L2 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_mono_adc_l2_mux),
+ SND_SOC_DAPM_MUX("Mono ADC L1 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_mono_adc_l1_mux),
+ SND_SOC_DAPM_MUX("Mono ADC R1 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_mono_adc_r1_mux),
+ SND_SOC_DAPM_MUX("Mono ADC R2 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_mono_adc_r2_mux),
+ /* ADC Mixer */
+ SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5670_PWR_DIG2,
+ RT5670_PWR_ADC_S1F_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC Stereo2 Filter", RT5670_PWR_DIG2,
+ RT5670_PWR_ADC_S2F_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", RT5670_STO1_ADC_DIG_VOL,
+ RT5670_L_MUTE_SFT, 1, rt5670_sto1_adc_l_mix,
+ ARRAY_SIZE(rt5670_sto1_adc_l_mix)),
+ SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", RT5670_STO1_ADC_DIG_VOL,
+ RT5670_R_MUTE_SFT, 1, rt5670_sto1_adc_r_mix,
+ ARRAY_SIZE(rt5670_sto1_adc_r_mix)),
+ SND_SOC_DAPM_MIXER("Sto2 ADC MIXL", SND_SOC_NOPM, 0, 0,
+ rt5670_sto2_adc_l_mix,
+ ARRAY_SIZE(rt5670_sto2_adc_l_mix)),
+ SND_SOC_DAPM_MIXER("Sto2 ADC MIXR", SND_SOC_NOPM, 0, 0,
+ rt5670_sto2_adc_r_mix,
+ ARRAY_SIZE(rt5670_sto2_adc_r_mix)),
+ SND_SOC_DAPM_SUPPLY("ADC Mono Left Filter", RT5670_PWR_DIG2,
+ RT5670_PWR_ADC_MF_L_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Mono ADC MIXL", RT5670_MONO_ADC_DIG_VOL,
+ RT5670_L_MUTE_SFT, 1, rt5670_mono_adc_l_mix,
+ ARRAY_SIZE(rt5670_mono_adc_l_mix)),
+ SND_SOC_DAPM_SUPPLY("ADC Mono Right Filter", RT5670_PWR_DIG2,
+ RT5670_PWR_ADC_MF_R_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Mono ADC MIXR", RT5670_MONO_ADC_DIG_VOL,
+ RT5670_R_MUTE_SFT, 1, rt5670_mono_adc_r_mix,
+ ARRAY_SIZE(rt5670_mono_adc_r_mix)),
+
+ /* ADC PGA */
+ SND_SOC_DAPM_PGA("Stereo1 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Stereo1 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Stereo2 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Stereo2 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Sto2 ADC LR MIX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Stereo2 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mono ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("VAD_ADC", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* DSP */
+ SND_SOC_DAPM_PGA("TxDP_ADC", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("TxDP_ADC_L", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("TxDP_ADC_R", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("TxDC_DAC", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_MUX("TDM Data Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_txdp_slot_mux),
+
+ SND_SOC_DAPM_MUX("DSP UL Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_dsp_ul_mux),
+ SND_SOC_DAPM_MUX("DSP DL Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_dsp_dl_mux),
+
+ SND_SOC_DAPM_MUX("RxDP Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_rxdp_mux),
+
+ /* IF2 Mux */
+ SND_SOC_DAPM_MUX("IF2 ADC Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_if2_adc_in_mux),
+
+ /* Digital Interface */
+ SND_SOC_DAPM_SUPPLY("I2S1", RT5670_PWR_DIG1,
+ RT5670_PWR_I2S1_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1 DAC2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1 DAC2 L", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1 DAC2 R", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("I2S2", RT5670_PWR_DIG1,
+ RT5670_PWR_I2S2_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF2 DAC", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF2 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF2 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF2 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF2 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Digital Interface Select */
+ SND_SOC_DAPM_MUX("IF1 ADC1 IN1 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_if1_adc1_in1_mux),
+ SND_SOC_DAPM_MUX("IF1 ADC1 IN2 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_if1_adc1_in2_mux),
+ SND_SOC_DAPM_MUX("IF1 ADC2 IN Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_if1_adc2_in_mux),
+ SND_SOC_DAPM_MUX("IF1 ADC2 IN1 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_if1_adc2_in1_mux),
+ SND_SOC_DAPM_MUX("VAD ADC Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_vad_adc_mux),
+
+ /* Audio Interface */
+ SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0,
+ RT5670_GPIO_CTRL1, RT5670_I2S2_PIN_SFT, 1),
+
+ /* Audio DSP */
+ SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Output Side */
+ /* DAC mixer before sound effect */
+ SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0,
+ rt5670_dac_l_mix, ARRAY_SIZE(rt5670_dac_l_mix)),
+ SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0,
+ rt5670_dac_r_mix, ARRAY_SIZE(rt5670_dac_r_mix)),
+ SND_SOC_DAPM_PGA("DAC MIX", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* DAC2 channel Mux */
+ SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_dac_l2_mux),
+ SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5670_dac_r2_mux),
+ SND_SOC_DAPM_PGA("DAC L2 Volume", RT5670_PWR_DIG1,
+ RT5670_PWR_DAC_L2_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DAC R2 Volume", RT5670_PWR_DIG1,
+ RT5670_PWR_DAC_R2_BIT, 0, NULL, 0),
+
+ SND_SOC_DAPM_MUX("DAC1 L Mux", SND_SOC_NOPM, 0, 0, &rt5670_dac1l_mux),
+ SND_SOC_DAPM_MUX("DAC1 R Mux", SND_SOC_NOPM, 0, 0, &rt5670_dac1r_mux),
+
+ /* DAC Mixer */
+ SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5670_PWR_DIG2,
+ RT5670_PWR_DAC_S1F_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DAC Mono Left Filter", RT5670_PWR_DIG2,
+ RT5670_PWR_DAC_MF_L_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DAC Mono Right Filter", RT5670_PWR_DIG2,
+ RT5670_PWR_DAC_MF_R_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0,
+ rt5670_sto_dac_l_mix,
+ ARRAY_SIZE(rt5670_sto_dac_l_mix)),
+ SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0,
+ rt5670_sto_dac_r_mix,
+ ARRAY_SIZE(rt5670_sto_dac_r_mix)),
+ SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0,
+ rt5670_mono_dac_l_mix,
+ ARRAY_SIZE(rt5670_mono_dac_l_mix)),
+ SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0,
+ rt5670_mono_dac_r_mix,
+ ARRAY_SIZE(rt5670_mono_dac_r_mix)),
+ SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0,
+ rt5670_dig_l_mix,
+ ARRAY_SIZE(rt5670_dig_l_mix)),
+ SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0,
+ rt5670_dig_r_mix,
+ ARRAY_SIZE(rt5670_dig_r_mix)),
+
+ /* DACs */
+ SND_SOC_DAPM_SUPPLY("DAC L1 Power", RT5670_PWR_DIG1,
+ RT5670_PWR_DAC_L1_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DAC R1 Power", RT5670_PWR_DIG1,
+ RT5670_PWR_DAC_R1_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_DAC("DAC L1", NULL, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("DAC R1", NULL, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("DAC L2", NULL, RT5670_PWR_DIG1,
+ RT5670_PWR_DAC_L2_BIT, 0),
+
+ SND_SOC_DAPM_DAC("DAC R2", NULL, RT5670_PWR_DIG1,
+ RT5670_PWR_DAC_R2_BIT, 0),
+ /* OUT Mixer */
+
+ SND_SOC_DAPM_MIXER("OUT MIXL", RT5670_PWR_MIXER, RT5670_PWR_OM_L_BIT,
+ 0, rt5670_out_l_mix, ARRAY_SIZE(rt5670_out_l_mix)),
+ SND_SOC_DAPM_MIXER("OUT MIXR", RT5670_PWR_MIXER, RT5670_PWR_OM_R_BIT,
+ 0, rt5670_out_r_mix, ARRAY_SIZE(rt5670_out_r_mix)),
+ /* Ouput Volume */
+ SND_SOC_DAPM_MIXER("HPOVOL MIXL", RT5670_PWR_VOL,
+ RT5670_PWR_HV_L_BIT, 0,
+ rt5670_hpvoll_mix, ARRAY_SIZE(rt5670_hpvoll_mix)),
+ SND_SOC_DAPM_MIXER("HPOVOL MIXR", RT5670_PWR_VOL,
+ RT5670_PWR_HV_R_BIT, 0,
+ rt5670_hpvolr_mix, ARRAY_SIZE(rt5670_hpvolr_mix)),
+ SND_SOC_DAPM_PGA("DAC 1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("DAC 2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("HPOVOL", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* HPO/LOUT/Mono Mixer */
+ SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0,
+ rt5670_hpo_mix, ARRAY_SIZE(rt5670_hpo_mix)),
+ SND_SOC_DAPM_MIXER("LOUT MIX", RT5670_PWR_ANLG1, RT5670_PWR_LM_BIT,
+ 0, rt5670_lout_mix, ARRAY_SIZE(rt5670_lout_mix)),
+ SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0,
+ rt5670_hp_power_event, SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_SUPPLY("HP L Amp", RT5670_PWR_ANLG1,
+ RT5670_PWR_HP_L_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("HP R Amp", RT5670_PWR_ANLG1,
+ RT5670_PWR_HP_R_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0,
+ rt5670_hp_event, SND_SOC_DAPM_PRE_PMD |
+ SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_SWITCH("LOUT L Playback", SND_SOC_NOPM, 0, 0,
+ &lout_l_enable_control),
+ SND_SOC_DAPM_SWITCH("LOUT R Playback", SND_SOC_NOPM, 0, 0,
+ &lout_r_enable_control),
+ SND_SOC_DAPM_PGA("LOUT Amp", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* PDM */
+ SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5670_PWR_DIG2,
+ RT5670_PWR_PDM1_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2,
+ RT5670_PWR_PDM2_BIT, 0, NULL, 0),
+
+ SND_SOC_DAPM_MUX("PDM1 L Mux", RT5670_PDM_OUT_CTRL,
+ RT5670_M_PDM1_L_SFT, 1, &rt5670_pdm1_l_mux),
+ SND_SOC_DAPM_MUX("PDM1 R Mux", RT5670_PDM_OUT_CTRL,
+ RT5670_M_PDM1_R_SFT, 1, &rt5670_pdm1_r_mux),
+ SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL,
+ RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux),
+ SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL,
+ RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux),
+
+ /* Output Lines */
+ SND_SOC_DAPM_OUTPUT("HPOL"),
+ SND_SOC_DAPM_OUTPUT("HPOR"),
+ SND_SOC_DAPM_OUTPUT("LOUTL"),
+ SND_SOC_DAPM_OUTPUT("LOUTR"),
+ SND_SOC_DAPM_OUTPUT("PDM1L"),
+ SND_SOC_DAPM_OUTPUT("PDM1R"),
+ SND_SOC_DAPM_OUTPUT("PDM2L"),
+ SND_SOC_DAPM_OUTPUT("PDM2R"),
+};
+
+static const struct snd_soc_dapm_route rt5670_dapm_routes[] = {
+ { "ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc },
+ { "ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc },
+ { "ADC Mono Left Filter", NULL, "ADC MONO L ASRC", is_using_asrc },
+ { "ADC Mono Right Filter", NULL, "ADC MONO R ASRC", is_using_asrc },
+ { "DAC Mono Left Filter", NULL, "DAC MONO L ASRC", is_using_asrc },
+ { "DAC Mono Right Filter", NULL, "DAC MONO R ASRC", is_using_asrc },
+ { "DAC Stereo1 Filter", NULL, "DAC STO ASRC", is_using_asrc },
+
+ { "I2S1", NULL, "I2S1 ASRC" },
+ { "I2S2", NULL, "I2S2 ASRC" },
+
+ { "DMIC1", NULL, "DMIC L1" },
+ { "DMIC1", NULL, "DMIC R1" },
+ { "DMIC2", NULL, "DMIC L2" },
+ { "DMIC2", NULL, "DMIC R2" },
+ { "DMIC3", NULL, "DMIC L3" },
+ { "DMIC3", NULL, "DMIC R3" },
+
+ { "BST1", NULL, "IN1P" },
+ { "BST1", NULL, "IN1N" },
+ { "BST1", NULL, "Mic Det Power" },
+ { "BST2", NULL, "IN2P" },
+ { "BST2", NULL, "IN2N" },
+
+ { "INL VOL", NULL, "IN2P" },
+ { "INR VOL", NULL, "IN2N" },
+
+ { "RECMIXL", "INL Switch", "INL VOL" },
+ { "RECMIXL", "BST2 Switch", "BST2" },
+ { "RECMIXL", "BST1 Switch", "BST1" },
+
+ { "RECMIXR", "INR Switch", "INR VOL" },
+ { "RECMIXR", "BST2 Switch", "BST2" },
+ { "RECMIXR", "BST1 Switch", "BST1" },
+
+ { "ADC 1", NULL, "RECMIXL" },
+ { "ADC 1", NULL, "ADC 1 power" },
+ { "ADC 1", NULL, "ADC clock" },
+ { "ADC 2", NULL, "RECMIXR" },
+ { "ADC 2", NULL, "ADC 2 power" },
+ { "ADC 2", NULL, "ADC clock" },
+
+ { "DMIC L1", NULL, "DMIC CLK" },
+ { "DMIC L1", NULL, "DMIC1 Power" },
+ { "DMIC R1", NULL, "DMIC CLK" },
+ { "DMIC R1", NULL, "DMIC1 Power" },
+ { "DMIC L2", NULL, "DMIC CLK" },
+ { "DMIC L2", NULL, "DMIC2 Power" },
+ { "DMIC R2", NULL, "DMIC CLK" },
+ { "DMIC R2", NULL, "DMIC2 Power" },
+ { "DMIC L3", NULL, "DMIC CLK" },
+ { "DMIC L3", NULL, "DMIC3 Power" },
+ { "DMIC R3", NULL, "DMIC CLK" },
+ { "DMIC R3", NULL, "DMIC3 Power" },
+
+ { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" },
+ { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" },
+ { "Stereo1 DMIC Mux", "DMIC3", "DMIC3" },
+
+ { "Stereo2 DMIC Mux", "DMIC1", "DMIC1" },
+ { "Stereo2 DMIC Mux", "DMIC2", "DMIC2" },
+ { "Stereo2 DMIC Mux", "DMIC3", "DMIC3" },
+
+ { "Mono DMIC L Mux", "DMIC1", "DMIC L1" },
+ { "Mono DMIC L Mux", "DMIC2", "DMIC L2" },
+ { "Mono DMIC L Mux", "DMIC3", "DMIC L3" },
+
+ { "Mono DMIC R Mux", "DMIC1", "DMIC R1" },
+ { "Mono DMIC R Mux", "DMIC2", "DMIC R2" },
+ { "Mono DMIC R Mux", "DMIC3", "DMIC R3" },
+
+ { "ADC 1_2", NULL, "ADC 1" },
+ { "ADC 1_2", NULL, "ADC 2" },
+
+ { "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" },
+ { "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" },
+ { "Stereo1 ADC L1 Mux", "ADC", "ADC 1_2" },
+ { "Stereo1 ADC L1 Mux", "DAC MIX", "DAC MIXL" },
+
+ { "Stereo1 ADC R1 Mux", "ADC", "ADC 1_2" },
+ { "Stereo1 ADC R1 Mux", "DAC MIX", "DAC MIXR" },
+ { "Stereo1 ADC R2 Mux", "DMIC", "Stereo1 DMIC Mux" },
+ { "Stereo1 ADC R2 Mux", "DAC MIX", "DAC MIXR" },
+
+ { "Mono ADC L2 Mux", "DMIC", "Mono DMIC L Mux" },
+ { "Mono ADC L2 Mux", "Mono DAC MIXL", "Mono DAC MIXL" },
+ { "Mono ADC L1 Mux", "Mono DAC MIXL", "Mono DAC MIXL" },
+ { "Mono ADC L1 Mux", "ADC1", "ADC 1" },
+
+ { "Mono ADC R1 Mux", "Mono DAC MIXR", "Mono DAC MIXR" },
+ { "Mono ADC R1 Mux", "ADC2", "ADC 2" },
+ { "Mono ADC R2 Mux", "DMIC", "Mono DMIC R Mux" },
+ { "Mono ADC R2 Mux", "Mono DAC MIXR", "Mono DAC MIXR" },
+
+ { "Sto1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux" },
+ { "Sto1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux" },
+ { "Sto1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux" },
+ { "Sto1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux" },
+
+ { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" },
+ { "Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter" },
+ { "ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll },
+
+ { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" },
+ { "Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter" },
+ { "ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll },
+
+ { "Mono ADC MIXL", "ADC1 Switch", "Mono ADC L1 Mux" },
+ { "Mono ADC MIXL", "ADC2 Switch", "Mono ADC L2 Mux" },
+ { "Mono ADC MIXL", NULL, "ADC Mono Left Filter" },
+ { "ADC Mono Left Filter", NULL, "PLL1", is_sys_clk_from_pll },
+
+ { "Mono ADC MIXR", "ADC1 Switch", "Mono ADC R1 Mux" },
+ { "Mono ADC MIXR", "ADC2 Switch", "Mono ADC R2 Mux" },
+ { "Mono ADC MIXR", NULL, "ADC Mono Right Filter" },
+ { "ADC Mono Right Filter", NULL, "PLL1", is_sys_clk_from_pll },
+
+ { "Stereo2 ADC L2 Mux", "DMIC", "Stereo2 DMIC Mux" },
+ { "Stereo2 ADC L2 Mux", "DAC MIX", "DAC MIXL" },
+ { "Stereo2 ADC L1 Mux", "ADC", "ADC 1_2" },
+ { "Stereo2 ADC L1 Mux", "DAC MIX", "DAC MIXL" },
+
+ { "Stereo2 ADC R1 Mux", "ADC", "ADC 1_2" },
+ { "Stereo2 ADC R1 Mux", "DAC MIX", "DAC MIXR" },
+ { "Stereo2 ADC R2 Mux", "DMIC", "Stereo2 DMIC Mux" },
+ { "Stereo2 ADC R2 Mux", "DAC MIX", "DAC MIXR" },
+
+ { "Sto2 ADC MIXL", "ADC1 Switch", "Stereo2 ADC L1 Mux" },
+ { "Sto2 ADC MIXL", "ADC2 Switch", "Stereo2 ADC L2 Mux" },
+ { "Sto2 ADC MIXR", "ADC1 Switch", "Stereo2 ADC R1 Mux" },
+ { "Sto2 ADC MIXR", "ADC2 Switch", "Stereo2 ADC R2 Mux" },
+
+ { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXL" },
+ { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXR" },
+
+ { "Stereo2 ADC LR Mux", "L", "Sto2 ADC MIXL" },
+ { "Stereo2 ADC LR Mux", "LR", "Sto2 ADC LR MIX" },
+
+ { "Stereo2 ADC MIXL", NULL, "Stereo2 ADC LR Mux" },
+ { "Stereo2 ADC MIXL", NULL, "ADC Stereo2 Filter" },
+ { "ADC Stereo2 Filter", NULL, "PLL1", is_sys_clk_from_pll },
+
+ { "Stereo2 ADC MIXR", NULL, "Sto2 ADC MIXR" },
+ { "Stereo2 ADC MIXR", NULL, "ADC Stereo2 Filter" },
+ { "ADC Stereo2 Filter", NULL, "PLL1", is_sys_clk_from_pll },
+
+ { "VAD ADC Mux", "Sto1 ADC L", "Stereo1 ADC MIXL" },
+ { "VAD ADC Mux", "Mono ADC L", "Mono ADC MIXL" },
+ { "VAD ADC Mux", "Mono ADC R", "Mono ADC MIXR" },
+ { "VAD ADC Mux", "Sto2 ADC L", "Sto2 ADC MIXL" },
+
+ { "VAD_ADC", NULL, "VAD ADC Mux" },
+
+ { "IF_ADC1", NULL, "Stereo1 ADC MIXL" },
+ { "IF_ADC1", NULL, "Stereo1 ADC MIXR" },
+ { "IF_ADC2", NULL, "Mono ADC MIXL" },
+ { "IF_ADC2", NULL, "Mono ADC MIXR" },
+ { "IF_ADC3", NULL, "Stereo2 ADC MIXL" },
+ { "IF_ADC3", NULL, "Stereo2 ADC MIXR" },
+
+ { "IF1 ADC1 IN1 Mux", "IF_ADC1", "IF_ADC1" },
+ { "IF1 ADC1 IN1 Mux", "IF1_ADC3", "IF1_ADC3" },
+
+ { "IF1 ADC1 IN2 Mux", "IF1_ADC1_IN1", "IF1 ADC1 IN1 Mux" },
+ { "IF1 ADC1 IN2 Mux", "IF1_ADC4", "IF1_ADC4" },
+
+ { "IF1 ADC2 IN Mux", "IF_ADC2", "IF_ADC2" },
+ { "IF1 ADC2 IN Mux", "VAD_ADC", "VAD_ADC" },
+
+ { "IF1 ADC2 IN1 Mux", "IF1_ADC2_IN", "IF1 ADC2 IN Mux" },
+ { "IF1 ADC2 IN1 Mux", "IF1_ADC4", "IF1_ADC4" },
+
+ { "IF1_ADC1" , NULL, "IF1 ADC1 IN2 Mux" },
+ { "IF1_ADC2" , NULL, "IF1 ADC2 IN1 Mux" },
+
+ { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL" },
+ { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR" },
+ { "Stereo2 ADC MIX", NULL, "Sto2 ADC MIXL" },
+ { "Stereo2 ADC MIX", NULL, "Sto2 ADC MIXR" },
+ { "Mono ADC MIX", NULL, "Mono ADC MIXL" },
+ { "Mono ADC MIX", NULL, "Mono ADC MIXR" },
+
+ { "RxDP Mux", "IF2 DAC", "IF2 DAC" },
+ { "RxDP Mux", "IF1 DAC", "IF1 DAC2" },
+ { "RxDP Mux", "STO1 ADC Mixer", "Stereo1 ADC MIX" },
+ { "RxDP Mux", "STO2 ADC Mixer", "Stereo2 ADC MIX" },
+ { "RxDP Mux", "Mono ADC Mixer L", "Mono ADC MIXL" },
+ { "RxDP Mux", "Mono ADC Mixer R", "Mono ADC MIXR" },
+ { "RxDP Mux", "DAC1", "DAC MIX" },
+
+ { "TDM Data Mux", "Slot 0-1", "Stereo1 ADC MIX" },
+ { "TDM Data Mux", "Slot 2-3", "Mono ADC MIX" },
+ { "TDM Data Mux", "Slot 4-5", "Stereo2 ADC MIX" },
+ { "TDM Data Mux", "Slot 6-7", "IF2 DAC" },
+
+ { "DSP UL Mux", "Bypass", "TDM Data Mux" },
+ { "DSP UL Mux", NULL, "I2S DSP" },
+ { "DSP DL Mux", "Bypass", "RxDP Mux" },
+ { "DSP DL Mux", NULL, "I2S DSP" },
+
+ { "TxDP_ADC_L", NULL, "DSP UL Mux" },
+ { "TxDP_ADC_R", NULL, "DSP UL Mux" },
+ { "TxDC_DAC", NULL, "DSP DL Mux" },
+
+ { "TxDP_ADC", NULL, "TxDP_ADC_L" },
+ { "TxDP_ADC", NULL, "TxDP_ADC_R" },
+
+ { "IF1 ADC", NULL, "I2S1" },
+ { "IF1 ADC", NULL, "IF1_ADC1" },
+ { "IF1 ADC", NULL, "IF1_ADC2" },
+ { "IF1 ADC", NULL, "IF_ADC3" },
+ { "IF1 ADC", NULL, "TxDP_ADC" },
+
+ { "IF2 ADC Mux", "IF_ADC1", "IF_ADC1" },
+ { "IF2 ADC Mux", "IF_ADC2", "IF_ADC2" },
+ { "IF2 ADC Mux", "IF_ADC3", "IF_ADC3" },
+ { "IF2 ADC Mux", "TxDC_DAC", "TxDC_DAC" },
+ { "IF2 ADC Mux", "TxDP_ADC", "TxDP_ADC" },
+ { "IF2 ADC Mux", "VAD_ADC", "VAD_ADC" },
+
+ { "IF2 ADC L", NULL, "IF2 ADC Mux" },
+ { "IF2 ADC R", NULL, "IF2 ADC Mux" },
+
+ { "IF2 ADC", NULL, "I2S2" },
+ { "IF2 ADC", NULL, "IF2 ADC L" },
+ { "IF2 ADC", NULL, "IF2 ADC R" },
+
+ { "AIF1TX", NULL, "IF1 ADC" },
+ { "AIF2TX", NULL, "IF2 ADC" },
+
+ { "IF1 DAC1", NULL, "AIF1RX" },
+ { "IF1 DAC2", NULL, "AIF1RX" },
+ { "IF2 DAC", NULL, "AIF2RX" },
+
+ { "IF1 DAC1", NULL, "I2S1" },
+ { "IF1 DAC2", NULL, "I2S1" },
+ { "IF2 DAC", NULL, "I2S2" },
+
+ { "IF1 DAC2 L", NULL, "IF1 DAC2" },
+ { "IF1 DAC2 R", NULL, "IF1 DAC2" },
+ { "IF1 DAC1 L", NULL, "IF1 DAC1" },
+ { "IF1 DAC1 R", NULL, "IF1 DAC1" },
+ { "IF2 DAC L", NULL, "IF2 DAC" },
+ { "IF2 DAC R", NULL, "IF2 DAC" },
+
+ { "DAC1 L Mux", "IF1 DAC", "IF1 DAC1 L" },
+ { "DAC1 L Mux", "IF2 DAC", "IF2 DAC L" },
+
+ { "DAC1 R Mux", "IF1 DAC", "IF1 DAC1 R" },
+ { "DAC1 R Mux", "IF2 DAC", "IF2 DAC R" },
+
+ { "DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL" },
+ { "DAC1 MIXL", "DAC1 Switch", "DAC1 L Mux" },
+ { "DAC1 MIXL", NULL, "DAC Stereo1 Filter" },
+ { "DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR" },
+ { "DAC1 MIXR", "DAC1 Switch", "DAC1 R Mux" },
+ { "DAC1 MIXR", NULL, "DAC Stereo1 Filter" },
+
+ { "DAC MIX", NULL, "DAC1 MIXL" },
+ { "DAC MIX", NULL, "DAC1 MIXR" },
+
+ { "Audio DSP", NULL, "DAC1 MIXL" },
+ { "Audio DSP", NULL, "DAC1 MIXR" },
+
+ { "DAC L2 Mux", "IF1 DAC", "IF1 DAC2 L" },
+ { "DAC L2 Mux", "IF2 DAC", "IF2 DAC L" },
+ { "DAC L2 Mux", "TxDC DAC", "TxDC_DAC" },
+ { "DAC L2 Mux", "VAD_ADC", "VAD_ADC" },
+ { "DAC L2 Volume", NULL, "DAC L2 Mux" },
+ { "DAC L2 Volume", NULL, "DAC Mono Left Filter" },
+
+ { "DAC R2 Mux", "IF1 DAC", "IF1 DAC2 R" },
+ { "DAC R2 Mux", "IF2 DAC", "IF2 DAC R" },
+ { "DAC R2 Mux", "TxDC DAC", "TxDC_DAC" },
+ { "DAC R2 Mux", "TxDP ADC", "TxDP_ADC" },
+ { "DAC R2 Volume", NULL, "DAC R2 Mux" },
+ { "DAC R2 Volume", NULL, "DAC Mono Right Filter" },
+
+ { "Stereo DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" },
+ { "Stereo DAC MIXL", "DAC R1 Switch", "DAC1 MIXR" },
+ { "Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" },
+ { "Stereo DAC MIXL", NULL, "DAC Stereo1 Filter" },
+ { "Stereo DAC MIXL", NULL, "DAC L1 Power" },
+ { "Stereo DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" },
+ { "Stereo DAC MIXR", "DAC L1 Switch", "DAC1 MIXL" },
+ { "Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" },
+ { "Stereo DAC MIXR", NULL, "DAC Stereo1 Filter" },
+ { "Stereo DAC MIXR", NULL, "DAC R1 Power" },
+
+ { "Mono DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" },
+ { "Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" },
+ { "Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" },
+ { "Mono DAC MIXL", NULL, "DAC Mono Left Filter" },
+ { "Mono DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" },
+ { "Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" },
+ { "Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" },
+ { "Mono DAC MIXR", NULL, "DAC Mono Right Filter" },
+
+ { "DAC MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" },
+ { "DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" },
+ { "DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" },
+ { "DAC MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" },
+ { "DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" },
+ { "DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" },
+
+ { "DAC L1", NULL, "DAC L1 Power" },
+ { "DAC L1", NULL, "Stereo DAC MIXL" },
+ { "DAC L1", NULL, "PLL1", is_sys_clk_from_pll },
+ { "DAC R1", NULL, "DAC R1 Power" },
+ { "DAC R1", NULL, "Stereo DAC MIXR" },
+ { "DAC R1", NULL, "PLL1", is_sys_clk_from_pll },
+ { "DAC L2", NULL, "Mono DAC MIXL" },
+ { "DAC L2", NULL, "PLL1", is_sys_clk_from_pll },
+ { "DAC R2", NULL, "Mono DAC MIXR" },
+ { "DAC R2", NULL, "PLL1", is_sys_clk_from_pll },
+
+ { "OUT MIXL", "BST1 Switch", "BST1" },
+ { "OUT MIXL", "INL Switch", "INL VOL" },
+ { "OUT MIXL", "DAC L2 Switch", "DAC L2" },
+ { "OUT MIXL", "DAC L1 Switch", "DAC L1" },
+
+ { "OUT MIXR", "BST2 Switch", "BST2" },
+ { "OUT MIXR", "INR Switch", "INR VOL" },
+ { "OUT MIXR", "DAC R2 Switch", "DAC R2" },
+ { "OUT MIXR", "DAC R1 Switch", "DAC R1" },
+
+ { "HPOVOL MIXL", "DAC1 Switch", "DAC L1" },
+ { "HPOVOL MIXL", "INL Switch", "INL VOL" },
+ { "HPOVOL MIXR", "DAC1 Switch", "DAC R1" },
+ { "HPOVOL MIXR", "INR Switch", "INR VOL" },
+
+ { "DAC 2", NULL, "DAC L2" },
+ { "DAC 2", NULL, "DAC R2" },
+ { "DAC 1", NULL, "DAC L1" },
+ { "DAC 1", NULL, "DAC R1" },
+ { "HPOVOL", NULL, "HPOVOL MIXL" },
+ { "HPOVOL", NULL, "HPOVOL MIXR" },
+ { "HPO MIX", "DAC1 Switch", "DAC 1" },
+ { "HPO MIX", "HPVOL Switch", "HPOVOL" },
+
+ { "LOUT MIX", "DAC L1 Switch", "DAC L1" },
+ { "LOUT MIX", "DAC R1 Switch", "DAC R1" },
+ { "LOUT MIX", "OUTMIX L Switch", "OUT MIXL" },
+ { "LOUT MIX", "OUTMIX R Switch", "OUT MIXR" },
+
+ { "PDM1 L Mux", "Stereo DAC", "Stereo DAC MIXL" },
+ { "PDM1 L Mux", "Mono DAC", "Mono DAC MIXL" },
+ { "PDM1 L Mux", NULL, "PDM1 Power" },
+ { "PDM1 R Mux", "Stereo DAC", "Stereo DAC MIXR" },
+ { "PDM1 R Mux", "Mono DAC", "Mono DAC MIXR" },
+ { "PDM1 R Mux", NULL, "PDM1 Power" },
+ { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" },
+ { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" },
+ { "PDM2 L Mux", NULL, "PDM2 Power" },
+ { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" },
+ { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" },
+ { "PDM2 R Mux", NULL, "PDM2 Power" },
+
+ { "HP Amp", NULL, "HPO MIX" },
+ { "HP Amp", NULL, "Mic Det Power" },
+ { "HPOL", NULL, "HP Amp" },
+ { "HPOL", NULL, "HP L Amp" },
+ { "HPOL", NULL, "Improve HP Amp Drv" },
+ { "HPOR", NULL, "HP Amp" },
+ { "HPOR", NULL, "HP R Amp" },
+ { "HPOR", NULL, "Improve HP Amp Drv" },
+
+ { "LOUT Amp", NULL, "LOUT MIX" },
+ { "LOUT L Playback", "Switch", "LOUT Amp" },
+ { "LOUT R Playback", "Switch", "LOUT Amp" },
+ { "LOUTL", NULL, "LOUT L Playback" },
+ { "LOUTR", NULL, "LOUT R Playback" },
+ { "LOUTL", NULL, "Improve HP Amp Drv" },
+ { "LOUTR", NULL, "Improve HP Amp Drv" },
+
+ { "PDM1L", NULL, "PDM1 L Mux" },
+ { "PDM1R", NULL, "PDM1 R Mux" },
+ { "PDM2L", NULL, "PDM2 L Mux" },
+ { "PDM2R", NULL, "PDM2 R Mux" },
+};
+
+static int rt5670_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 rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
+ unsigned int val_len = 0, val_clk, mask_clk;
+ int pre_div, bclk_ms, frame_size;
+
+ rt5670->lrck[dai->id] = params_rate(params);
+ pre_div = rl6231_get_clk_info(rt5670->sysclk, rt5670->lrck[dai->id]);
+ if (pre_div < 0) {
+ dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n",
+ rt5670->lrck[dai->id], dai->id);
+ return -EINVAL;
+ }
+ frame_size = snd_soc_params_to_frame_size(params);
+ if (frame_size < 0) {
+ dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size);
+ return -EINVAL;
+ }
+ bclk_ms = frame_size > 32;
+ rt5670->bclk[dai->id] = rt5670->lrck[dai->id] * (32 << bclk_ms);
+
+ dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n",
+ rt5670->bclk[dai->id], rt5670->lrck[dai->id]);
+ dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n",
+ bclk_ms, pre_div, dai->id);
+
+ switch (params_width(params)) {
+ case 16:
+ break;
+ case 20:
+ val_len |= RT5670_I2S_DL_20;
+ break;
+ case 24:
+ val_len |= RT5670_I2S_DL_24;
+ break;
+ case 8:
+ val_len |= RT5670_I2S_DL_8;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (dai->id) {
+ case RT5670_AIF1:
+ mask_clk = RT5670_I2S_BCLK_MS1_MASK | RT5670_I2S_PD1_MASK;
+ val_clk = bclk_ms << RT5670_I2S_BCLK_MS1_SFT |
+ pre_div << RT5670_I2S_PD1_SFT;
+ snd_soc_update_bits(codec, RT5670_I2S1_SDP,
+ RT5670_I2S_DL_MASK, val_len);
+ snd_soc_update_bits(codec, RT5670_ADDA_CLK1, mask_clk, val_clk);
+ break;
+ case RT5670_AIF2:
+ mask_clk = RT5670_I2S_BCLK_MS2_MASK | RT5670_I2S_PD2_MASK;
+ val_clk = bclk_ms << RT5670_I2S_BCLK_MS2_SFT |
+ pre_div << RT5670_I2S_PD2_SFT;
+ snd_soc_update_bits(codec, RT5670_I2S2_SDP,
+ RT5670_I2S_DL_MASK, val_len);
+ snd_soc_update_bits(codec, RT5670_ADDA_CLK1, mask_clk, val_clk);
+ break;
+ default:
+ dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int rt5670_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
+ unsigned int reg_val = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ rt5670->master[dai->id] = 1;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ reg_val |= RT5670_I2S_MS_S;
+ rt5670->master[dai->id] = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ reg_val |= RT5670_I2S_BP_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ reg_val |= RT5670_I2S_DF_LEFT;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ reg_val |= RT5670_I2S_DF_PCM_A;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ reg_val |= RT5670_I2S_DF_PCM_B;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (dai->id) {
+ case RT5670_AIF1:
+ snd_soc_update_bits(codec, RT5670_I2S1_SDP,
+ RT5670_I2S_MS_MASK | RT5670_I2S_BP_MASK |
+ RT5670_I2S_DF_MASK, reg_val);
+ break;
+ case RT5670_AIF2:
+ snd_soc_update_bits(codec, RT5670_I2S2_SDP,
+ RT5670_I2S_MS_MASK | RT5670_I2S_BP_MASK |
+ RT5670_I2S_DF_MASK, reg_val);
+ break;
+ default:
+ dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
+ unsigned int reg_val = 0;
+
+ if (freq == rt5670->sysclk && clk_id == rt5670->sysclk_src)
+ return 0;
+
+ switch (clk_id) {
+ case RT5670_SCLK_S_MCLK:
+ reg_val |= RT5670_SCLK_SRC_MCLK;
+ break;
+ case RT5670_SCLK_S_PLL1:
+ reg_val |= RT5670_SCLK_SRC_PLL1;
+ break;
+ case RT5670_SCLK_S_RCCLK:
+ reg_val |= RT5670_SCLK_SRC_RCCLK;
+ break;
+ default:
+ dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id);
+ return -EINVAL;
+ }
+ snd_soc_update_bits(codec, RT5670_GLB_CLK,
+ RT5670_SCLK_SRC_MASK, reg_val);
+ rt5670->sysclk = freq;
+ rt5670->sysclk_src = clk_id;
+
+ dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id);
+
+ return 0;
+}
+
+static int rt5670_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
+ unsigned int freq_in, unsigned int freq_out)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
+ struct rl6231_pll_code pll_code;
+ int ret;
+
+ if (source == rt5670->pll_src && freq_in == rt5670->pll_in &&
+ freq_out == rt5670->pll_out)
+ return 0;
+
+ if (!freq_in || !freq_out) {
+ dev_dbg(codec->dev, "PLL disabled\n");
+
+ rt5670->pll_in = 0;
+ rt5670->pll_out = 0;
+ snd_soc_update_bits(codec, RT5670_GLB_CLK,
+ RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_MCLK);
+ return 0;
+ }
+
+ switch (source) {
+ case RT5670_PLL1_S_MCLK:
+ snd_soc_update_bits(codec, RT5670_GLB_CLK,
+ RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_MCLK);
+ break;
+ case RT5670_PLL1_S_BCLK1:
+ case RT5670_PLL1_S_BCLK2:
+ case RT5670_PLL1_S_BCLK3:
+ case RT5670_PLL1_S_BCLK4:
+ switch (dai->id) {
+ case RT5670_AIF1:
+ snd_soc_update_bits(codec, RT5670_GLB_CLK,
+ RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_BCLK1);
+ break;
+ case RT5670_AIF2:
+ snd_soc_update_bits(codec, RT5670_GLB_CLK,
+ RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_BCLK2);
+ break;
+ default:
+ dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id);
+ return -EINVAL;
+ }
+ break;
+ default:
+ dev_err(codec->dev, "Unknown PLL source %d\n", source);
+ return -EINVAL;
+ }
+
+ ret = rl6231_pll_calc(freq_in, freq_out, &pll_code);
+ if (ret < 0) {
+ dev_err(codec->dev, "Unsupport input clock %d\n", freq_in);
+ return ret;
+ }
+
+ dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n",
+ pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code),
+ pll_code.n_code, pll_code.k_code);
+
+ snd_soc_write(codec, RT5670_PLL_CTRL1,
+ pll_code.n_code << RT5670_PLL_N_SFT | pll_code.k_code);
+ snd_soc_write(codec, RT5670_PLL_CTRL2,
+ (pll_code.m_bp ? 0 : pll_code.m_code) << RT5670_PLL_M_SFT |
+ pll_code.m_bp << RT5670_PLL_M_BP_SFT);
+
+ rt5670->pll_in = freq_in;
+ rt5670->pll_out = freq_out;
+ rt5670->pll_src = source;
+
+ return 0;
+}
+
+static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+ unsigned int rx_mask, int slots, int slot_width)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned int val = 0;
+
+ if (rx_mask || tx_mask)
+ val |= (1 << 14);
+
+ switch (slots) {
+ case 4:
+ val |= (1 << 12);
+ break;
+ case 6:
+ val |= (2 << 12);
+ break;
+ case 8:
+ val |= (3 << 12);
+ break;
+ case 2:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (slot_width) {
+ case 20:
+ val |= (1 << 10);
+ break;
+ case 24:
+ val |= (2 << 10);
+ break;
+ case 32:
+ val |= (3 << 10);
+ break;
+ case 16:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, RT5670_TDM_CTRL_1, 0x7c00, val);
+
+ return 0;
+}
+
+static int rt5670_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) {
+ snd_soc_update_bits(codec, RT5670_PWR_ANLG1,
+ RT5670_PWR_VREF1 | RT5670_PWR_MB |
+ RT5670_PWR_BG | RT5670_PWR_VREF2,
+ RT5670_PWR_VREF1 | RT5670_PWR_MB |
+ RT5670_PWR_BG | RT5670_PWR_VREF2);
+ mdelay(10);
+ snd_soc_update_bits(codec, RT5670_PWR_ANLG1,
+ RT5670_PWR_FV1 | RT5670_PWR_FV2,
+ RT5670_PWR_FV1 | RT5670_PWR_FV2);
+ snd_soc_update_bits(codec, RT5670_CHARGE_PUMP,
+ RT5670_OSW_L_MASK | RT5670_OSW_R_MASK,
+ RT5670_OSW_L_DIS | RT5670_OSW_R_DIS);
+ snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x1);
+ snd_soc_update_bits(codec, RT5670_PWR_ANLG1,
+ RT5670_LDO_SEL_MASK, 0x3);
+ }
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ snd_soc_write(codec, RT5670_PWR_DIG1, 0x0000);
+ snd_soc_write(codec, RT5670_PWR_DIG2, 0x0001);
+ snd_soc_write(codec, RT5670_PWR_VOL, 0x0000);
+ snd_soc_write(codec, RT5670_PWR_MIXER, 0x0001);
+ snd_soc_write(codec, RT5670_PWR_ANLG1, 0x2800);
+ snd_soc_write(codec, RT5670_PWR_ANLG2, 0x0004);
+ snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0);
+ snd_soc_update_bits(codec, RT5670_PWR_ANLG1,
+ RT5670_LDO_SEL_MASK, 0x1);
+ break;
+
+ default:
+ break;
+ }
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+static int rt5670_probe(struct snd_soc_codec *codec)
+{
+ struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
+
+ rt5670->codec = codec;
+
+ return 0;
+}
+
+static int rt5670_remove(struct snd_soc_codec *codec)
+{
+ struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
+
+ regmap_write(rt5670->regmap, RT5670_RESET, 0);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int rt5670_suspend(struct snd_soc_codec *codec)
+{
+ struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
+
+ regcache_cache_only(rt5670->regmap, true);
+ regcache_mark_dirty(rt5670->regmap);
+ return 0;
+}
+
+static int rt5670_resume(struct snd_soc_codec *codec)
+{
+ struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
+
+ regcache_cache_only(rt5670->regmap, false);
+ regcache_sync(rt5670->regmap);
+
+ return 0;
+}
+#else
+#define rt5670_suspend NULL
+#define rt5670_resume NULL
+#endif
+
+#define RT5670_STEREO_RATES SNDRV_PCM_RATE_8000_96000
+#define RT5670_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
+
+static struct snd_soc_dai_ops rt5670_aif_dai_ops = {
+ .hw_params = rt5670_hw_params,
+ .set_fmt = rt5670_set_dai_fmt,
+ .set_sysclk = rt5670_set_dai_sysclk,
+ .set_tdm_slot = rt5670_set_tdm_slot,
+ .set_pll = rt5670_set_dai_pll,
+};
+
+static struct snd_soc_dai_driver rt5670_dai[] = {
+ {
+ .name = "rt5670-aif1",
+ .id = RT5670_AIF1,
+ .playback = {
+ .stream_name = "AIF1 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT5670_STEREO_RATES,
+ .formats = RT5670_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF1 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT5670_STEREO_RATES,
+ .formats = RT5670_FORMATS,
+ },
+ .ops = &rt5670_aif_dai_ops,
+ },
+ {
+ .name = "rt5670-aif2",
+ .id = RT5670_AIF2,
+ .playback = {
+ .stream_name = "AIF2 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT5670_STEREO_RATES,
+ .formats = RT5670_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF2 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT5670_STEREO_RATES,
+ .formats = RT5670_FORMATS,
+ },
+ .ops = &rt5670_aif_dai_ops,
+ },
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_rt5670 = {
+ .probe = rt5670_probe,
+ .remove = rt5670_remove,
+ .suspend = rt5670_suspend,
+ .resume = rt5670_resume,
+ .set_bias_level = rt5670_set_bias_level,
+ .idle_bias_off = true,
+ .controls = rt5670_snd_controls,
+ .num_controls = ARRAY_SIZE(rt5670_snd_controls),
+ .dapm_widgets = rt5670_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(rt5670_dapm_widgets),
+ .dapm_routes = rt5670_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(rt5670_dapm_routes),
+};
+
+static const struct regmap_config rt5670_regmap = {
+ .reg_bits = 8,
+ .val_bits = 16,
+ .max_register = RT5670_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5670_ranges) *
+ RT5670_PR_SPACING),
+ .volatile_reg = rt5670_volatile_register,
+ .readable_reg = rt5670_readable_register,
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults = rt5670_reg,
+ .num_reg_defaults = ARRAY_SIZE(rt5670_reg),
+ .ranges = rt5670_ranges,
+ .num_ranges = ARRAY_SIZE(rt5670_ranges),
+};
+
+static const struct i2c_device_id rt5670_i2c_id[] = {
+ { "rt5670", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id);
+
+static int rt5670_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct rt5670_platform_data *pdata = dev_get_platdata(&i2c->dev);
+ struct rt5670_priv *rt5670;
+ int ret;
+ unsigned int val;
+
+ rt5670 = devm_kzalloc(&i2c->dev,
+ sizeof(struct rt5670_priv),
+ GFP_KERNEL);
+ if (NULL == rt5670)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, rt5670);
+
+ if (pdata)
+ rt5670->pdata = *pdata;
+
+ rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap);
+ if (IS_ERR(rt5670->regmap)) {
+ ret = PTR_ERR(rt5670->regmap);
+ dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+ ret);
+ return ret;
+ }
+
+ regmap_read(rt5670->regmap, RT5670_VENDOR_ID2, &val);
+ if (val != RT5670_DEVICE_ID) {
+ dev_err(&i2c->dev,
+ "Device with ID register %x is not rt5670/72\n", val);
+ return -ENODEV;
+ }
+
+ regmap_write(rt5670->regmap, RT5670_RESET, 0);
+ regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1,
+ RT5670_PWR_HP_L | RT5670_PWR_HP_R |
+ RT5670_PWR_VREF2, RT5670_PWR_VREF2);
+ msleep(100);
+
+ regmap_write(rt5670->regmap, RT5670_RESET, 0);
+
+ ret = regmap_register_patch(rt5670->regmap, init_list,
+ ARRAY_SIZE(init_list));
+ if (ret != 0)
+ dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret);
+
+ if (rt5670->pdata.in2_diff)
+ regmap_update_bits(rt5670->regmap, RT5670_IN2,
+ RT5670_IN_DF2, RT5670_IN_DF2);
+
+ if (i2c->irq) {
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1,
+ RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ);
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+ RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT);
+
+ }
+
+ if (rt5670->pdata.jd_mode) {
+ regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1,
+ RT5670_PWR_MB, RT5670_PWR_MB);
+ regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG2,
+ RT5670_PWR_JD1, RT5670_PWR_JD1);
+ regmap_update_bits(rt5670->regmap, RT5670_IRQ_CTRL1,
+ RT5670_JD1_1_EN_MASK, RT5670_JD1_1_EN);
+ regmap_update_bits(rt5670->regmap, RT5670_JD_CTRL3,
+ RT5670_JD_TRI_CBJ_SEL_MASK |
+ RT5670_JD_TRI_HPO_SEL_MASK,
+ RT5670_JD_CBJ_JD1_1 | RT5670_JD_HPO_JD1_1);
+ switch (rt5670->pdata.jd_mode) {
+ case 1:
+ regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1,
+ RT5670_JD1_MODE_MASK,
+ RT5670_JD1_MODE_0);
+ break;
+ case 2:
+ regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1,
+ RT5670_JD1_MODE_MASK,
+ RT5670_JD1_MODE_1);
+ break;
+ case 3:
+ regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1,
+ RT5670_JD1_MODE_MASK,
+ RT5670_JD1_MODE_2);
+ break;
+ default:
+ break;
+ }
+ }
+
+ if (rt5670->pdata.dmic_en) {
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1,
+ RT5670_GP2_PIN_MASK,
+ RT5670_GP2_PIN_DMIC1_SCL);
+
+ switch (rt5670->pdata.dmic1_data_pin) {
+ case RT5670_DMIC_DATA_IN2P:
+ regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1,
+ RT5670_DMIC_1_DP_MASK,
+ RT5670_DMIC_1_DP_IN2P);
+ break;
+
+ case RT5670_DMIC_DATA_GPIO6:
+ regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1,
+ RT5670_DMIC_1_DP_MASK,
+ RT5670_DMIC_1_DP_GPIO6);
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1,
+ RT5670_GP6_PIN_MASK,
+ RT5670_GP6_PIN_DMIC1_SDA);
+ break;
+
+ case RT5670_DMIC_DATA_GPIO7:
+ regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1,
+ RT5670_DMIC_1_DP_MASK,
+ RT5670_DMIC_1_DP_GPIO7);
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1,
+ RT5670_GP7_PIN_MASK,
+ RT5670_GP7_PIN_DMIC1_SDA);
+ break;
+
+ default:
+ break;
+ }
+
+ switch (rt5670->pdata.dmic2_data_pin) {
+ case RT5670_DMIC_DATA_IN3N:
+ regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1,
+ RT5670_DMIC_2_DP_MASK,
+ RT5670_DMIC_2_DP_IN3N);
+ break;
+
+ case RT5670_DMIC_DATA_GPIO8:
+ regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1,
+ RT5670_DMIC_2_DP_MASK,
+ RT5670_DMIC_2_DP_GPIO8);
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1,
+ RT5670_GP8_PIN_MASK,
+ RT5670_GP8_PIN_DMIC2_SDA);
+ break;
+
+ default:
+ break;
+ }
+
+ switch (rt5670->pdata.dmic3_data_pin) {
+ case RT5670_DMIC_DATA_GPIO5:
+ regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL2,
+ RT5670_DMIC_3_DP_MASK,
+ RT5670_DMIC_3_DP_GPIO5);
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1,
+ RT5670_GP5_PIN_MASK,
+ RT5670_GP5_PIN_DMIC3_SDA);
+ break;
+
+ case RT5670_DMIC_DATA_GPIO9:
+ case RT5670_DMIC_DATA_GPIO10:
+ dev_err(&i2c->dev,
+ "Always use GPIO5 as DMIC3 data pin\n");
+ break;
+
+ default:
+ break;
+ }
+
+ }
+
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670,
+ rt5670_dai, ARRAY_SIZE(rt5670_dai));
+ if (ret < 0)
+ goto err;
+
+ return 0;
+err:
+ return ret;
+}
+
+static int rt5670_i2c_remove(struct i2c_client *i2c)
+{
+ snd_soc_unregister_codec(&i2c->dev);
+
+ return 0;
+}
+
+static struct i2c_driver rt5670_i2c_driver = {
+ .driver = {
+ .name = "rt5670",
+ .owner = THIS_MODULE,
+ },
+ .probe = rt5670_i2c_probe,
+ .remove = rt5670_i2c_remove,
+ .id_table = rt5670_i2c_id,
+};
+
+module_i2c_driver(rt5670_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC RT5670 driver");
+MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
new file mode 100644
index 000000000000..a0b5c855b492
--- /dev/null
+++ b/sound/soc/codecs/rt5670.h
@@ -0,0 +1,2000 @@
+/*
+ * rt5670.h -- RT5670 ALSA SoC audio driver
+ *
+ * Copyright 2014 Realtek Microelectronics
+ * Author: Bard Liao <bardliao@realtek.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 __RT5670_H__
+#define __RT5670_H__
+
+#include <sound/rt5670.h>
+
+/* Info */
+#define RT5670_RESET 0x00
+#define RT5670_VENDOR_ID 0xfd
+#define RT5670_VENDOR_ID1 0xfe
+#define RT5670_VENDOR_ID2 0xff
+/* I/O - Output */
+#define RT5670_HP_VOL 0x02
+#define RT5670_LOUT1 0x03
+/* I/O - Input */
+#define RT5670_CJ_CTRL1 0x0a
+#define RT5670_CJ_CTRL2 0x0b
+#define RT5670_CJ_CTRL3 0x0c
+#define RT5670_IN2 0x0e
+#define RT5670_INL1_INR1_VOL 0x0f
+/* I/O - ADC/DAC/DMIC */
+#define RT5670_DAC1_DIG_VOL 0x19
+#define RT5670_DAC2_DIG_VOL 0x1a
+#define RT5670_DAC_CTRL 0x1b
+#define RT5670_STO1_ADC_DIG_VOL 0x1c
+#define RT5670_MONO_ADC_DIG_VOL 0x1d
+#define RT5670_ADC_BST_VOL1 0x1e
+#define RT5670_STO2_ADC_DIG_VOL 0x1f
+/* Mixer - D-D */
+#define RT5670_ADC_BST_VOL2 0x20
+#define RT5670_STO2_ADC_MIXER 0x26
+#define RT5670_STO1_ADC_MIXER 0x27
+#define RT5670_MONO_ADC_MIXER 0x28
+#define RT5670_AD_DA_MIXER 0x29
+#define RT5670_STO_DAC_MIXER 0x2a
+#define RT5670_DD_MIXER 0x2b
+#define RT5670_DIG_MIXER 0x2c
+#define RT5670_DSP_PATH1 0x2d
+#define RT5670_DSP_PATH2 0x2e
+#define RT5670_DIG_INF1_DATA 0x2f
+#define RT5670_DIG_INF2_DATA 0x30
+/* Mixer - PDM */
+#define RT5670_PDM_OUT_CTRL 0x31
+#define RT5670_PDM_DATA_CTRL1 0x32
+#define RT5670_PDM1_DATA_CTRL2 0x33
+#define RT5670_PDM1_DATA_CTRL3 0x34
+#define RT5670_PDM1_DATA_CTRL4 0x35
+#define RT5670_PDM2_DATA_CTRL2 0x36
+#define RT5670_PDM2_DATA_CTRL3 0x37
+#define RT5670_PDM2_DATA_CTRL4 0x38
+/* Mixer - ADC */
+#define RT5670_REC_L1_MIXER 0x3b
+#define RT5670_REC_L2_MIXER 0x3c
+#define RT5670_REC_R1_MIXER 0x3d
+#define RT5670_REC_R2_MIXER 0x3e
+/* Mixer - DAC */
+#define RT5670_HPO_MIXER 0x45
+#define RT5670_MONO_MIXER 0x4c
+#define RT5670_OUT_L1_MIXER 0x4f
+#define RT5670_OUT_R1_MIXER 0x52
+#define RT5670_LOUT_MIXER 0x53
+/* Power */
+#define RT5670_PWR_DIG1 0x61
+#define RT5670_PWR_DIG2 0x62
+#define RT5670_PWR_ANLG1 0x63
+#define RT5670_PWR_ANLG2 0x64
+#define RT5670_PWR_MIXER 0x65
+#define RT5670_PWR_VOL 0x66
+/* Private Register Control */
+#define RT5670_PRIV_INDEX 0x6a
+#define RT5670_PRIV_DATA 0x6c
+/* Format - ADC/DAC */
+#define RT5670_I2S4_SDP 0x6f
+#define RT5670_I2S1_SDP 0x70
+#define RT5670_I2S2_SDP 0x71
+#define RT5670_I2S3_SDP 0x72
+#define RT5670_ADDA_CLK1 0x73
+#define RT5670_ADDA_CLK2 0x74
+#define RT5670_DMIC_CTRL1 0x75
+#define RT5670_DMIC_CTRL2 0x76
+/* Format - TDM Control */
+#define RT5670_TDM_CTRL_1 0x77
+#define RT5670_TDM_CTRL_2 0x78
+#define RT5670_TDM_CTRL_3 0x79
+
+/* Function - Analog */
+#define RT5670_DSP_CLK 0x7f
+#define RT5670_GLB_CLK 0x80
+#define RT5670_PLL_CTRL1 0x81
+#define RT5670_PLL_CTRL2 0x82
+#define RT5670_ASRC_1 0x83
+#define RT5670_ASRC_2 0x84
+#define RT5670_ASRC_3 0x85
+#define RT5670_ASRC_4 0x86
+#define RT5670_ASRC_5 0x87
+#define RT5670_ASRC_7 0x89
+#define RT5670_ASRC_8 0x8a
+#define RT5670_ASRC_9 0x8b
+#define RT5670_ASRC_10 0x8c
+#define RT5670_ASRC_11 0x8d
+#define RT5670_DEPOP_M1 0x8e
+#define RT5670_DEPOP_M2 0x8f
+#define RT5670_DEPOP_M3 0x90
+#define RT5670_CHARGE_PUMP 0x91
+#define RT5670_MICBIAS 0x93
+#define RT5670_A_JD_CTRL1 0x94
+#define RT5670_A_JD_CTRL2 0x95
+#define RT5670_ASRC_12 0x97
+#define RT5670_ASRC_13 0x98
+#define RT5670_ASRC_14 0x99
+#define RT5670_VAD_CTRL1 0x9a
+#define RT5670_VAD_CTRL2 0x9b
+#define RT5670_VAD_CTRL3 0x9c
+#define RT5670_VAD_CTRL4 0x9d
+#define RT5670_VAD_CTRL5 0x9e
+/* Function - Digital */
+#define RT5670_ADC_EQ_CTRL1 0xae
+#define RT5670_ADC_EQ_CTRL2 0xaf
+#define RT5670_EQ_CTRL1 0xb0
+#define RT5670_EQ_CTRL2 0xb1
+#define RT5670_ALC_DRC_CTRL1 0xb2
+#define RT5670_ALC_DRC_CTRL2 0xb3
+#define RT5670_ALC_CTRL_1 0xb4
+#define RT5670_ALC_CTRL_2 0xb5
+#define RT5670_ALC_CTRL_3 0xb6
+#define RT5670_ALC_CTRL_4 0xb7
+#define RT5670_JD_CTRL 0xbb
+#define RT5670_IRQ_CTRL1 0xbd
+#define RT5670_IRQ_CTRL2 0xbe
+#define RT5670_INT_IRQ_ST 0xbf
+#define RT5670_GPIO_CTRL1 0xc0
+#define RT5670_GPIO_CTRL2 0xc1
+#define RT5670_GPIO_CTRL3 0xc2
+#define RT5670_SCRABBLE_FUN 0xcd
+#define RT5670_SCRABBLE_CTRL 0xce
+#define RT5670_BASE_BACK 0xcf
+#define RT5670_MP3_PLUS1 0xd0
+#define RT5670_MP3_PLUS2 0xd1
+#define RT5670_ADJ_HPF1 0xd3
+#define RT5670_ADJ_HPF2 0xd4
+#define RT5670_HP_CALIB_AMP_DET 0xd6
+#define RT5670_SV_ZCD1 0xd9
+#define RT5670_SV_ZCD2 0xda
+#define RT5670_IL_CMD 0xdb
+#define RT5670_IL_CMD2 0xdc
+#define RT5670_IL_CMD3 0xdd
+#define RT5670_DRC_HL_CTRL1 0xe6
+#define RT5670_DRC_HL_CTRL2 0xe7
+#define RT5670_ADC_MONO_HP_CTRL1 0xec
+#define RT5670_ADC_MONO_HP_CTRL2 0xed
+#define RT5670_ADC_STO2_HP_CTRL1 0xee
+#define RT5670_ADC_STO2_HP_CTRL2 0xef
+#define RT5670_JD_CTRL3 0xf8
+#define RT5670_JD_CTRL4 0xf9
+/* General Control */
+#define RT5670_DIG_MISC 0xfa
+#define RT5670_GEN_CTRL2 0xfb
+#define RT5670_GEN_CTRL3 0xfc
+
+
+/* Index of Codec Private Register definition */
+#define RT5670_DIG_VOL 0x00
+#define RT5670_PR_ALC_CTRL_1 0x01
+#define RT5670_PR_ALC_CTRL_2 0x02
+#define RT5670_PR_ALC_CTRL_3 0x03
+#define RT5670_PR_ALC_CTRL_4 0x04
+#define RT5670_PR_ALC_CTRL_5 0x05
+#define RT5670_PR_ALC_CTRL_6 0x06
+#define RT5670_BIAS_CUR1 0x12
+#define RT5670_BIAS_CUR3 0x14
+#define RT5670_CLSD_INT_REG1 0x1c
+#define RT5670_MAMP_INT_REG2 0x37
+#define RT5670_CHOP_DAC_ADC 0x3d
+#define RT5670_MIXER_INT_REG 0x3f
+#define RT5670_3D_SPK 0x63
+#define RT5670_WND_1 0x6c
+#define RT5670_WND_2 0x6d
+#define RT5670_WND_3 0x6e
+#define RT5670_WND_4 0x6f
+#define RT5670_WND_5 0x70
+#define RT5670_WND_8 0x73
+#define RT5670_DIP_SPK_INF 0x75
+#define RT5670_HP_DCC_INT1 0x77
+#define RT5670_EQ_BW_LOP 0xa0
+#define RT5670_EQ_GN_LOP 0xa1
+#define RT5670_EQ_FC_BP1 0xa2
+#define RT5670_EQ_BW_BP1 0xa3
+#define RT5670_EQ_GN_BP1 0xa4
+#define RT5670_EQ_FC_BP2 0xa5
+#define RT5670_EQ_BW_BP2 0xa6
+#define RT5670_EQ_GN_BP2 0xa7
+#define RT5670_EQ_FC_BP3 0xa8
+#define RT5670_EQ_BW_BP3 0xa9
+#define RT5670_EQ_GN_BP3 0xaa
+#define RT5670_EQ_FC_BP4 0xab
+#define RT5670_EQ_BW_BP4 0xac
+#define RT5670_EQ_GN_BP4 0xad
+#define RT5670_EQ_FC_HIP1 0xae
+#define RT5670_EQ_GN_HIP1 0xaf
+#define RT5670_EQ_FC_HIP2 0xb0
+#define RT5670_EQ_BW_HIP2 0xb1
+#define RT5670_EQ_GN_HIP2 0xb2
+#define RT5670_EQ_PRE_VOL 0xb3
+#define RT5670_EQ_PST_VOL 0xb4
+
+
+/* global definition */
+#define RT5670_L_MUTE (0x1 << 15)
+#define RT5670_L_MUTE_SFT 15
+#define RT5670_VOL_L_MUTE (0x1 << 14)
+#define RT5670_VOL_L_SFT 14
+#define RT5670_R_MUTE (0x1 << 7)
+#define RT5670_R_MUTE_SFT 7
+#define RT5670_VOL_R_MUTE (0x1 << 6)
+#define RT5670_VOL_R_SFT 6
+#define RT5670_L_VOL_MASK (0x3f << 8)
+#define RT5670_L_VOL_SFT 8
+#define RT5670_R_VOL_MASK (0x3f)
+#define RT5670_R_VOL_SFT 0
+
+/* Combo Jack Control 1 (0x0a) */
+#define RT5670_CBJ_BST1_MASK (0xf << 12)
+#define RT5670_CBJ_BST1_SFT (12)
+#define RT5670_CBJ_JD_HP_EN (0x1 << 9)
+#define RT5670_CBJ_JD_MIC_EN (0x1 << 8)
+#define RT5670_CBJ_BST1_EN (0x1 << 2)
+
+/* Combo Jack Control 1 (0x0b) */
+#define RT5670_CBJ_MN_JD (0x1 << 12)
+#define RT5670_CAPLESS_EN (0x1 << 11)
+#define RT5670_CBJ_DET_MODE (0x1 << 7)
+
+/* IN2 Control (0x0e) */
+#define RT5670_BST_MASK1 (0xf<<12)
+#define RT5670_BST_SFT1 12
+#define RT5670_BST_MASK2 (0xf<<8)
+#define RT5670_BST_SFT2 8
+#define RT5670_IN_DF1 (0x1 << 7)
+#define RT5670_IN_SFT1 7
+#define RT5670_IN_DF2 (0x1 << 6)
+#define RT5670_IN_SFT2 6
+
+/* INL and INR Volume Control (0x0f) */
+#define RT5670_INL_SEL_MASK (0x1 << 15)
+#define RT5670_INL_SEL_SFT 15
+#define RT5670_INL_SEL_IN4P (0x0 << 15)
+#define RT5670_INL_SEL_MONOP (0x1 << 15)
+#define RT5670_INL_VOL_MASK (0x1f << 8)
+#define RT5670_INL_VOL_SFT 8
+#define RT5670_INR_SEL_MASK (0x1 << 7)
+#define RT5670_INR_SEL_SFT 7
+#define RT5670_INR_SEL_IN4N (0x0 << 7)
+#define RT5670_INR_SEL_MONON (0x1 << 7)
+#define RT5670_INR_VOL_MASK (0x1f)
+#define RT5670_INR_VOL_SFT 0
+
+/* Sidetone Control (0x18) */
+#define RT5670_ST_SEL_MASK (0x7 << 9)
+#define RT5670_ST_SEL_SFT 9
+#define RT5670_M_ST_DACR2 (0x1 << 8)
+#define RT5670_M_ST_DACR2_SFT 8
+#define RT5670_M_ST_DACL2 (0x1 << 7)
+#define RT5670_M_ST_DACL2_SFT 7
+#define RT5670_ST_EN (0x1 << 6)
+#define RT5670_ST_EN_SFT 6
+
+/* DAC1 Digital Volume (0x19) */
+#define RT5670_DAC_L1_VOL_MASK (0xff << 8)
+#define RT5670_DAC_L1_VOL_SFT 8
+#define RT5670_DAC_R1_VOL_MASK (0xff)
+#define RT5670_DAC_R1_VOL_SFT 0
+
+/* DAC2 Digital Volume (0x1a) */
+#define RT5670_DAC_L2_VOL_MASK (0xff << 8)
+#define RT5670_DAC_L2_VOL_SFT 8
+#define RT5670_DAC_R2_VOL_MASK (0xff)
+#define RT5670_DAC_R2_VOL_SFT 0
+
+/* DAC2 Control (0x1b) */
+#define RT5670_M_DAC_L2_VOL (0x1 << 13)
+#define RT5670_M_DAC_L2_VOL_SFT 13
+#define RT5670_M_DAC_R2_VOL (0x1 << 12)
+#define RT5670_M_DAC_R2_VOL_SFT 12
+#define RT5670_DAC2_L_SEL_MASK (0x7 << 4)
+#define RT5670_DAC2_L_SEL_SFT 4
+#define RT5670_DAC2_R_SEL_MASK (0x7 << 0)
+#define RT5670_DAC2_R_SEL_SFT 0
+
+/* ADC Digital Volume Control (0x1c) */
+#define RT5670_ADC_L_VOL_MASK (0x7f << 8)
+#define RT5670_ADC_L_VOL_SFT 8
+#define RT5670_ADC_R_VOL_MASK (0x7f)
+#define RT5670_ADC_R_VOL_SFT 0
+
+/* Mono ADC Digital Volume Control (0x1d) */
+#define RT5670_MONO_ADC_L_VOL_MASK (0x7f << 8)
+#define RT5670_MONO_ADC_L_VOL_SFT 8
+#define RT5670_MONO_ADC_R_VOL_MASK (0x7f)
+#define RT5670_MONO_ADC_R_VOL_SFT 0
+
+/* ADC Boost Volume Control (0x1e) */
+#define RT5670_STO1_ADC_L_BST_MASK (0x3 << 14)
+#define RT5670_STO1_ADC_L_BST_SFT 14
+#define RT5670_STO1_ADC_R_BST_MASK (0x3 << 12)
+#define RT5670_STO1_ADC_R_BST_SFT 12
+#define RT5670_STO1_ADC_COMP_MASK (0x3 << 10)
+#define RT5670_STO1_ADC_COMP_SFT 10
+#define RT5670_STO2_ADC_L_BST_MASK (0x3 << 8)
+#define RT5670_STO2_ADC_L_BST_SFT 8
+#define RT5670_STO2_ADC_R_BST_MASK (0x3 << 6)
+#define RT5670_STO2_ADC_R_BST_SFT 6
+#define RT5670_STO2_ADC_COMP_MASK (0x3 << 4)
+#define RT5670_STO2_ADC_COMP_SFT 4
+
+/* Stereo2 ADC Mixer Control (0x26) */
+#define RT5670_STO2_ADC_SRC_MASK (0x1 << 15)
+#define RT5670_STO2_ADC_SRC_SFT 15
+
+/* Stereo ADC Mixer Control (0x26 0x27) */
+#define RT5670_M_ADC_L1 (0x1 << 14)
+#define RT5670_M_ADC_L1_SFT 14
+#define RT5670_M_ADC_L2 (0x1 << 13)
+#define RT5670_M_ADC_L2_SFT 13
+#define RT5670_ADC_1_SRC_MASK (0x1 << 12)
+#define RT5670_ADC_1_SRC_SFT 12
+#define RT5670_ADC_1_SRC_ADC (0x1 << 12)
+#define RT5670_ADC_1_SRC_DACMIX (0x0 << 12)
+#define RT5670_ADC_2_SRC_MASK (0x1 << 11)
+#define RT5670_ADC_2_SRC_SFT 11
+#define RT5670_ADC_SRC_MASK (0x1 << 10)
+#define RT5670_ADC_SRC_SFT 10
+#define RT5670_DMIC_SRC_MASK (0x3 << 8)
+#define RT5670_DMIC_SRC_SFT 8
+#define RT5670_M_ADC_R1 (0x1 << 6)
+#define RT5670_M_ADC_R1_SFT 6
+#define RT5670_M_ADC_R2 (0x1 << 5)
+#define RT5670_M_ADC_R2_SFT 5
+#define RT5670_DMIC3_SRC_MASK (0x1 << 1)
+#define RT5670_DMIC3_SRC_SFT 0
+
+/* Mono ADC Mixer Control (0x28) */
+#define RT5670_M_MONO_ADC_L1 (0x1 << 14)
+#define RT5670_M_MONO_ADC_L1_SFT 14
+#define RT5670_M_MONO_ADC_L2 (0x1 << 13)
+#define RT5670_M_MONO_ADC_L2_SFT 13
+#define RT5670_MONO_ADC_L1_SRC_MASK (0x1 << 12)
+#define RT5670_MONO_ADC_L1_SRC_SFT 12
+#define RT5670_MONO_ADC_L1_SRC_DACMIXL (0x0 << 12)
+#define RT5670_MONO_ADC_L1_SRC_ADCL (0x1 << 12)
+#define RT5670_MONO_ADC_L2_SRC_MASK (0x1 << 11)
+#define RT5670_MONO_ADC_L2_SRC_SFT 11
+#define RT5670_MONO_ADC_L_SRC_MASK (0x1 << 10)
+#define RT5670_MONO_ADC_L_SRC_SFT 10
+#define RT5670_MONO_DMIC_L_SRC_MASK (0x3 << 8)
+#define RT5670_MONO_DMIC_L_SRC_SFT 8
+#define RT5670_M_MONO_ADC_R1 (0x1 << 6)
+#define RT5670_M_MONO_ADC_R1_SFT 6
+#define RT5670_M_MONO_ADC_R2 (0x1 << 5)
+#define RT5670_M_MONO_ADC_R2_SFT 5
+#define RT5670_MONO_ADC_R1_SRC_MASK (0x1 << 4)
+#define RT5670_MONO_ADC_R1_SRC_SFT 4
+#define RT5670_MONO_ADC_R1_SRC_ADCR (0x1 << 4)
+#define RT5670_MONO_ADC_R1_SRC_DACMIXR (0x0 << 4)
+#define RT5670_MONO_ADC_R2_SRC_MASK (0x1 << 3)
+#define RT5670_MONO_ADC_R2_SRC_SFT 3
+#define RT5670_MONO_DMIC_R_SRC_MASK (0x3)
+#define RT5670_MONO_DMIC_R_SRC_SFT 0
+
+/* ADC Mixer to DAC Mixer Control (0x29) */
+#define RT5670_M_ADCMIX_L (0x1 << 15)
+#define RT5670_M_ADCMIX_L_SFT 15
+#define RT5670_M_DAC1_L (0x1 << 14)
+#define RT5670_M_DAC1_L_SFT 14
+#define RT5670_DAC1_R_SEL_MASK (0x3 << 10)
+#define RT5670_DAC1_R_SEL_SFT 10
+#define RT5670_DAC1_R_SEL_IF1 (0x0 << 10)
+#define RT5670_DAC1_R_SEL_IF2 (0x1 << 10)
+#define RT5670_DAC1_R_SEL_IF3 (0x2 << 10)
+#define RT5670_DAC1_R_SEL_IF4 (0x3 << 10)
+#define RT5670_DAC1_L_SEL_MASK (0x3 << 8)
+#define RT5670_DAC1_L_SEL_SFT 8
+#define RT5670_DAC1_L_SEL_IF1 (0x0 << 8)
+#define RT5670_DAC1_L_SEL_IF2 (0x1 << 8)
+#define RT5670_DAC1_L_SEL_IF3 (0x2 << 8)
+#define RT5670_DAC1_L_SEL_IF4 (0x3 << 8)
+#define RT5670_M_ADCMIX_R (0x1 << 7)
+#define RT5670_M_ADCMIX_R_SFT 7
+#define RT5670_M_DAC1_R (0x1 << 6)
+#define RT5670_M_DAC1_R_SFT 6
+
+/* Stereo DAC Mixer Control (0x2a) */
+#define RT5670_M_DAC_L1 (0x1 << 14)
+#define RT5670_M_DAC_L1_SFT 14
+#define RT5670_DAC_L1_STO_L_VOL_MASK (0x1 << 13)
+#define RT5670_DAC_L1_STO_L_VOL_SFT 13
+#define RT5670_M_DAC_L2 (0x1 << 12)
+#define RT5670_M_DAC_L2_SFT 12
+#define RT5670_DAC_L2_STO_L_VOL_MASK (0x1 << 11)
+#define RT5670_DAC_L2_STO_L_VOL_SFT 11
+#define RT5670_M_DAC_R1_STO_L (0x1 << 9)
+#define RT5670_M_DAC_R1_STO_L_SFT 9
+#define RT5670_DAC_R1_STO_L_VOL_MASK (0x1 << 8)
+#define RT5670_DAC_R1_STO_L_VOL_SFT 8
+#define RT5670_M_DAC_R1 (0x1 << 6)
+#define RT5670_M_DAC_R1_SFT 6
+#define RT5670_DAC_R1_STO_R_VOL_MASK (0x1 << 5)
+#define RT5670_DAC_R1_STO_R_VOL_SFT 5
+#define RT5670_M_DAC_R2 (0x1 << 4)
+#define RT5670_M_DAC_R2_SFT 4
+#define RT5670_DAC_R2_STO_R_VOL_MASK (0x1 << 3)
+#define RT5670_DAC_R2_STO_R_VOL_SFT 3
+#define RT5670_M_DAC_L1_STO_R (0x1 << 1)
+#define RT5670_M_DAC_L1_STO_R_SFT 1
+#define RT5670_DAC_L1_STO_R_VOL_MASK (0x1)
+#define RT5670_DAC_L1_STO_R_VOL_SFT 0
+
+/* Mono DAC Mixer Control (0x2b) */
+#define RT5670_M_DAC_L1_MONO_L (0x1 << 14)
+#define RT5670_M_DAC_L1_MONO_L_SFT 14
+#define RT5670_DAC_L1_MONO_L_VOL_MASK (0x1 << 13)
+#define RT5670_DAC_L1_MONO_L_VOL_SFT 13
+#define RT5670_M_DAC_L2_MONO_L (0x1 << 12)
+#define RT5670_M_DAC_L2_MONO_L_SFT 12
+#define RT5670_DAC_L2_MONO_L_VOL_MASK (0x1 << 11)
+#define RT5670_DAC_L2_MONO_L_VOL_SFT 11
+#define RT5670_M_DAC_R2_MONO_L (0x1 << 10)
+#define RT5670_M_DAC_R2_MONO_L_SFT 10
+#define RT5670_DAC_R2_MONO_L_VOL_MASK (0x1 << 9)
+#define RT5670_DAC_R2_MONO_L_VOL_SFT 9
+#define RT5670_M_DAC_R1_MONO_R (0x1 << 6)
+#define RT5670_M_DAC_R1_MONO_R_SFT 6
+#define RT5670_DAC_R1_MONO_R_VOL_MASK (0x1 << 5)
+#define RT5670_DAC_R1_MONO_R_VOL_SFT 5
+#define RT5670_M_DAC_R2_MONO_R (0x1 << 4)
+#define RT5670_M_DAC_R2_MONO_R_SFT 4
+#define RT5670_DAC_R2_MONO_R_VOL_MASK (0x1 << 3)
+#define RT5670_DAC_R2_MONO_R_VOL_SFT 3
+#define RT5670_M_DAC_L2_MONO_R (0x1 << 2)
+#define RT5670_M_DAC_L2_MONO_R_SFT 2
+#define RT5670_DAC_L2_MONO_R_VOL_MASK (0x1 << 1)
+#define RT5670_DAC_L2_MONO_R_VOL_SFT 1
+
+/* Digital Mixer Control (0x2c) */
+#define RT5670_M_STO_L_DAC_L (0x1 << 15)
+#define RT5670_M_STO_L_DAC_L_SFT 15
+#define RT5670_STO_L_DAC_L_VOL_MASK (0x1 << 14)
+#define RT5670_STO_L_DAC_L_VOL_SFT 14
+#define RT5670_M_DAC_L2_DAC_L (0x1 << 13)
+#define RT5670_M_DAC_L2_DAC_L_SFT 13
+#define RT5670_DAC_L2_DAC_L_VOL_MASK (0x1 << 12)
+#define RT5670_DAC_L2_DAC_L_VOL_SFT 12
+#define RT5670_M_STO_R_DAC_R (0x1 << 11)
+#define RT5670_M_STO_R_DAC_R_SFT 11
+#define RT5670_STO_R_DAC_R_VOL_MASK (0x1 << 10)
+#define RT5670_STO_R_DAC_R_VOL_SFT 10
+#define RT5670_M_DAC_R2_DAC_R (0x1 << 9)
+#define RT5670_M_DAC_R2_DAC_R_SFT 9
+#define RT5670_DAC_R2_DAC_R_VOL_MASK (0x1 << 8)
+#define RT5670_DAC_R2_DAC_R_VOL_SFT 8
+#define RT5670_M_DAC_R2_DAC_L (0x1 << 7)
+#define RT5670_M_DAC_R2_DAC_L_SFT 7
+#define RT5670_DAC_R2_DAC_L_VOL_MASK (0x1 << 6)
+#define RT5670_DAC_R2_DAC_L_VOL_SFT 6
+#define RT5670_M_DAC_L2_DAC_R (0x1 << 5)
+#define RT5670_M_DAC_L2_DAC_R_SFT 5
+#define RT5670_DAC_L2_DAC_R_VOL_MASK (0x1 << 4)
+#define RT5670_DAC_L2_DAC_R_VOL_SFT 4
+
+/* DSP Path Control 1 (0x2d) */
+#define RT5670_RXDP_SEL_MASK (0x7 << 13)
+#define RT5670_RXDP_SEL_SFT 13
+#define RT5670_RXDP_SRC_MASK (0x3 << 11)
+#define RT5670_RXDP_SRC_SFT 11
+#define RT5670_RXDP_SRC_NOR (0x0 << 11)
+#define RT5670_RXDP_SRC_DIV2 (0x1 << 11)
+#define RT5670_RXDP_SRC_DIV3 (0x2 << 11)
+#define RT5670_TXDP_SRC_MASK (0x3 << 4)
+#define RT5670_TXDP_SRC_SFT 4
+#define RT5670_TXDP_SRC_NOR (0x0 << 4)
+#define RT5670_TXDP_SRC_DIV2 (0x1 << 4)
+#define RT5670_TXDP_SRC_DIV3 (0x2 << 4)
+#define RT5670_TXDP_SLOT_SEL_MASK (0x3 << 2)
+#define RT5670_TXDP_SLOT_SEL_SFT 2
+#define RT5670_DSP_UL_SEL (0x1 << 1)
+#define RT5670_DSP_UL_SFT 1
+#define RT5670_DSP_DL_SEL 0x1
+#define RT5670_DSP_DL_SFT 0
+
+/* DSP Path Control 2 (0x2e) */
+#define RT5670_TXDP_L_VOL_MASK (0x7f << 8)
+#define RT5670_TXDP_L_VOL_SFT 8
+#define RT5670_TXDP_R_VOL_MASK (0x7f)
+#define RT5670_TXDP_R_VOL_SFT 0
+
+/* Digital Interface Data Control (0x2f) */
+#define RT5670_IF1_ADC2_IN_SEL (0x1 << 15)
+#define RT5670_IF1_ADC2_IN_SFT 15
+#define RT5670_IF2_ADC_IN_MASK (0x7 << 12)
+#define RT5670_IF2_ADC_IN_SFT 12
+#define RT5670_IF2_DAC_SEL_MASK (0x3 << 10)
+#define RT5670_IF2_DAC_SEL_SFT 10
+#define RT5670_IF2_ADC_SEL_MASK (0x3 << 8)
+#define RT5670_IF2_ADC_SEL_SFT 8
+
+/* Digital Interface Data Control (0x30) */
+#define RT5670_IF4_ADC_IN_MASK (0x3 << 4)
+#define RT5670_IF4_ADC_IN_SFT 4
+
+/* PDM Output Control (0x31) */
+#define RT5670_PDM1_L_MASK (0x1 << 15)
+#define RT5670_PDM1_L_SFT 15
+#define RT5670_M_PDM1_L (0x1 << 14)
+#define RT5670_M_PDM1_L_SFT 14
+#define RT5670_PDM1_R_MASK (0x1 << 13)
+#define RT5670_PDM1_R_SFT 13
+#define RT5670_M_PDM1_R (0x1 << 12)
+#define RT5670_M_PDM1_R_SFT 12
+#define RT5670_PDM2_L_MASK (0x1 << 11)
+#define RT5670_PDM2_L_SFT 11
+#define RT5670_M_PDM2_L (0x1 << 10)
+#define RT5670_M_PDM2_L_SFT 10
+#define RT5670_PDM2_R_MASK (0x1 << 9)
+#define RT5670_PDM2_R_SFT 9
+#define RT5670_M_PDM2_R (0x1 << 8)
+#define RT5670_M_PDM2_R_SFT 8
+#define RT5670_PDM2_BUSY (0x1 << 7)
+#define RT5670_PDM1_BUSY (0x1 << 6)
+#define RT5670_PDM_PATTERN (0x1 << 5)
+#define RT5670_PDM_GAIN (0x1 << 4)
+#define RT5670_PDM_DIV_MASK (0x3)
+
+/* REC Left Mixer Control 1 (0x3b) */
+#define RT5670_G_HP_L_RM_L_MASK (0x7 << 13)
+#define RT5670_G_HP_L_RM_L_SFT 13
+#define RT5670_G_IN_L_RM_L_MASK (0x7 << 10)
+#define RT5670_G_IN_L_RM_L_SFT 10
+#define RT5670_G_BST4_RM_L_MASK (0x7 << 7)
+#define RT5670_G_BST4_RM_L_SFT 7
+#define RT5670_G_BST3_RM_L_MASK (0x7 << 4)
+#define RT5670_G_BST3_RM_L_SFT 4
+#define RT5670_G_BST2_RM_L_MASK (0x7 << 1)
+#define RT5670_G_BST2_RM_L_SFT 1
+
+/* REC Left Mixer Control 2 (0x3c) */
+#define RT5670_G_BST1_RM_L_MASK (0x7 << 13)
+#define RT5670_G_BST1_RM_L_SFT 13
+#define RT5670_M_IN_L_RM_L (0x1 << 5)
+#define RT5670_M_IN_L_RM_L_SFT 5
+#define RT5670_M_BST2_RM_L (0x1 << 3)
+#define RT5670_M_BST2_RM_L_SFT 3
+#define RT5670_M_BST1_RM_L (0x1 << 1)
+#define RT5670_M_BST1_RM_L_SFT 1
+
+/* REC Right Mixer Control 1 (0x3d) */
+#define RT5670_G_HP_R_RM_R_MASK (0x7 << 13)
+#define RT5670_G_HP_R_RM_R_SFT 13
+#define RT5670_G_IN_R_RM_R_MASK (0x7 << 10)
+#define RT5670_G_IN_R_RM_R_SFT 10
+#define RT5670_G_BST4_RM_R_MASK (0x7 << 7)
+#define RT5670_G_BST4_RM_R_SFT 7
+#define RT5670_G_BST3_RM_R_MASK (0x7 << 4)
+#define RT5670_G_BST3_RM_R_SFT 4
+#define RT5670_G_BST2_RM_R_MASK (0x7 << 1)
+#define RT5670_G_BST2_RM_R_SFT 1
+
+/* REC Right Mixer Control 2 (0x3e) */
+#define RT5670_G_BST1_RM_R_MASK (0x7 << 13)
+#define RT5670_G_BST1_RM_R_SFT 13
+#define RT5670_M_IN_R_RM_R (0x1 << 5)
+#define RT5670_M_IN_R_RM_R_SFT 5
+#define RT5670_M_BST2_RM_R (0x1 << 3)
+#define RT5670_M_BST2_RM_R_SFT 3
+#define RT5670_M_BST1_RM_R (0x1 << 1)
+#define RT5670_M_BST1_RM_R_SFT 1
+
+/* HPMIX Control (0x45) */
+#define RT5670_M_DAC2_HM (0x1 << 15)
+#define RT5670_M_DAC2_HM_SFT 15
+#define RT5670_M_HPVOL_HM (0x1 << 14)
+#define RT5670_M_HPVOL_HM_SFT 14
+#define RT5670_M_DAC1_HM (0x1 << 13)
+#define RT5670_M_DAC1_HM_SFT 13
+#define RT5670_G_HPOMIX_MASK (0x1 << 12)
+#define RT5670_G_HPOMIX_SFT 12
+#define RT5670_M_INR1_HMR (0x1 << 3)
+#define RT5670_M_INR1_HMR_SFT 3
+#define RT5670_M_DACR1_HMR (0x1 << 2)
+#define RT5670_M_DACR1_HMR_SFT 2
+#define RT5670_M_INL1_HML (0x1 << 1)
+#define RT5670_M_INL1_HML_SFT 1
+#define RT5670_M_DACL1_HML (0x1)
+#define RT5670_M_DACL1_HML_SFT 0
+
+/* Mono Output Mixer Control (0x4c) */
+#define RT5670_M_DAC_R2_MA (0x1 << 15)
+#define RT5670_M_DAC_R2_MA_SFT 15
+#define RT5670_M_DAC_L2_MA (0x1 << 14)
+#define RT5670_M_DAC_L2_MA_SFT 14
+#define RT5670_M_OV_R_MM (0x1 << 13)
+#define RT5670_M_OV_R_MM_SFT 13
+#define RT5670_M_OV_L_MM (0x1 << 12)
+#define RT5670_M_OV_L_MM_SFT 12
+#define RT5670_G_MONOMIX_MASK (0x1 << 10)
+#define RT5670_G_MONOMIX_SFT 10
+#define RT5670_M_DAC_R2_MM (0x1 << 9)
+#define RT5670_M_DAC_R2_MM_SFT 9
+#define RT5670_M_DAC_L2_MM (0x1 << 8)
+#define RT5670_M_DAC_L2_MM_SFT 8
+#define RT5670_M_BST4_MM (0x1 << 7)
+#define RT5670_M_BST4_MM_SFT 7
+
+/* Output Left Mixer Control 1 (0x4d) */
+#define RT5670_G_BST3_OM_L_MASK (0x7 << 13)
+#define RT5670_G_BST3_OM_L_SFT 13
+#define RT5670_G_BST2_OM_L_MASK (0x7 << 10)
+#define RT5670_G_BST2_OM_L_SFT 10
+#define RT5670_G_BST1_OM_L_MASK (0x7 << 7)
+#define RT5670_G_BST1_OM_L_SFT 7
+#define RT5670_G_IN_L_OM_L_MASK (0x7 << 4)
+#define RT5670_G_IN_L_OM_L_SFT 4
+#define RT5670_G_RM_L_OM_L_MASK (0x7 << 1)
+#define RT5670_G_RM_L_OM_L_SFT 1
+
+/* Output Left Mixer Control 2 (0x4e) */
+#define RT5670_G_DAC_R2_OM_L_MASK (0x7 << 13)
+#define RT5670_G_DAC_R2_OM_L_SFT 13
+#define RT5670_G_DAC_L2_OM_L_MASK (0x7 << 10)
+#define RT5670_G_DAC_L2_OM_L_SFT 10
+#define RT5670_G_DAC_L1_OM_L_MASK (0x7 << 7)
+#define RT5670_G_DAC_L1_OM_L_SFT 7
+
+/* Output Left Mixer Control 3 (0x4f) */
+#define RT5670_M_BST1_OM_L (0x1 << 5)
+#define RT5670_M_BST1_OM_L_SFT 5
+#define RT5670_M_IN_L_OM_L (0x1 << 4)
+#define RT5670_M_IN_L_OM_L_SFT 4
+#define RT5670_M_DAC_L2_OM_L (0x1 << 1)
+#define RT5670_M_DAC_L2_OM_L_SFT 1
+#define RT5670_M_DAC_L1_OM_L (0x1)
+#define RT5670_M_DAC_L1_OM_L_SFT 0
+
+/* Output Right Mixer Control 1 (0x50) */
+#define RT5670_G_BST4_OM_R_MASK (0x7 << 13)
+#define RT5670_G_BST4_OM_R_SFT 13
+#define RT5670_G_BST2_OM_R_MASK (0x7 << 10)
+#define RT5670_G_BST2_OM_R_SFT 10
+#define RT5670_G_BST1_OM_R_MASK (0x7 << 7)
+#define RT5670_G_BST1_OM_R_SFT 7
+#define RT5670_G_IN_R_OM_R_MASK (0x7 << 4)
+#define RT5670_G_IN_R_OM_R_SFT 4
+#define RT5670_G_RM_R_OM_R_MASK (0x7 << 1)
+#define RT5670_G_RM_R_OM_R_SFT 1
+
+/* Output Right Mixer Control 2 (0x51) */
+#define RT5670_G_DAC_L2_OM_R_MASK (0x7 << 13)
+#define RT5670_G_DAC_L2_OM_R_SFT 13
+#define RT5670_G_DAC_R2_OM_R_MASK (0x7 << 10)
+#define RT5670_G_DAC_R2_OM_R_SFT 10
+#define RT5670_G_DAC_R1_OM_R_MASK (0x7 << 7)
+#define RT5670_G_DAC_R1_OM_R_SFT 7
+
+/* Output Right Mixer Control 3 (0x52) */
+#define RT5670_M_BST2_OM_R (0x1 << 6)
+#define RT5670_M_BST2_OM_R_SFT 6
+#define RT5670_M_IN_R_OM_R (0x1 << 4)
+#define RT5670_M_IN_R_OM_R_SFT 4
+#define RT5670_M_DAC_R2_OM_R (0x1 << 1)
+#define RT5670_M_DAC_R2_OM_R_SFT 1
+#define RT5670_M_DAC_R1_OM_R (0x1)
+#define RT5670_M_DAC_R1_OM_R_SFT 0
+
+/* LOUT Mixer Control (0x53) */
+#define RT5670_M_DAC_L1_LM (0x1 << 15)
+#define RT5670_M_DAC_L1_LM_SFT 15
+#define RT5670_M_DAC_R1_LM (0x1 << 14)
+#define RT5670_M_DAC_R1_LM_SFT 14
+#define RT5670_M_OV_L_LM (0x1 << 13)
+#define RT5670_M_OV_L_LM_SFT 13
+#define RT5670_M_OV_R_LM (0x1 << 12)
+#define RT5670_M_OV_R_LM_SFT 12
+#define RT5670_G_LOUTMIX_MASK (0x1 << 11)
+#define RT5670_G_LOUTMIX_SFT 11
+
+/* Power Management for Digital 1 (0x61) */
+#define RT5670_PWR_I2S1 (0x1 << 15)
+#define RT5670_PWR_I2S1_BIT 15
+#define RT5670_PWR_I2S2 (0x1 << 14)
+#define RT5670_PWR_I2S2_BIT 14
+#define RT5670_PWR_DAC_L1 (0x1 << 12)
+#define RT5670_PWR_DAC_L1_BIT 12
+#define RT5670_PWR_DAC_R1 (0x1 << 11)
+#define RT5670_PWR_DAC_R1_BIT 11
+#define RT5670_PWR_DAC_L2 (0x1 << 7)
+#define RT5670_PWR_DAC_L2_BIT 7
+#define RT5670_PWR_DAC_R2 (0x1 << 6)
+#define RT5670_PWR_DAC_R2_BIT 6
+#define RT5670_PWR_ADC_L (0x1 << 2)
+#define RT5670_PWR_ADC_L_BIT 2
+#define RT5670_PWR_ADC_R (0x1 << 1)
+#define RT5670_PWR_ADC_R_BIT 1
+#define RT5670_PWR_CLS_D (0x1)
+#define RT5670_PWR_CLS_D_BIT 0
+
+/* Power Management for Digital 2 (0x62) */
+#define RT5670_PWR_ADC_S1F (0x1 << 15)
+#define RT5670_PWR_ADC_S1F_BIT 15
+#define RT5670_PWR_ADC_MF_L (0x1 << 14)
+#define RT5670_PWR_ADC_MF_L_BIT 14
+#define RT5670_PWR_ADC_MF_R (0x1 << 13)
+#define RT5670_PWR_ADC_MF_R_BIT 13
+#define RT5670_PWR_I2S_DSP (0x1 << 12)
+#define RT5670_PWR_I2S_DSP_BIT 12
+#define RT5670_PWR_DAC_S1F (0x1 << 11)
+#define RT5670_PWR_DAC_S1F_BIT 11
+#define RT5670_PWR_DAC_MF_L (0x1 << 10)
+#define RT5670_PWR_DAC_MF_L_BIT 10
+#define RT5670_PWR_DAC_MF_R (0x1 << 9)
+#define RT5670_PWR_DAC_MF_R_BIT 9
+#define RT5670_PWR_ADC_S2F (0x1 << 8)
+#define RT5670_PWR_ADC_S2F_BIT 8
+#define RT5670_PWR_PDM1 (0x1 << 7)
+#define RT5670_PWR_PDM1_BIT 7
+#define RT5670_PWR_PDM2 (0x1 << 6)
+#define RT5670_PWR_PDM2_BIT 6
+
+/* Power Management for Analog 1 (0x63) */
+#define RT5670_PWR_VREF1 (0x1 << 15)
+#define RT5670_PWR_VREF1_BIT 15
+#define RT5670_PWR_FV1 (0x1 << 14)
+#define RT5670_PWR_FV1_BIT 14
+#define RT5670_PWR_MB (0x1 << 13)
+#define RT5670_PWR_MB_BIT 13
+#define RT5670_PWR_LM (0x1 << 12)
+#define RT5670_PWR_LM_BIT 12
+#define RT5670_PWR_BG (0x1 << 11)
+#define RT5670_PWR_BG_BIT 11
+#define RT5670_PWR_HP_L (0x1 << 7)
+#define RT5670_PWR_HP_L_BIT 7
+#define RT5670_PWR_HP_R (0x1 << 6)
+#define RT5670_PWR_HP_R_BIT 6
+#define RT5670_PWR_HA (0x1 << 5)
+#define RT5670_PWR_HA_BIT 5
+#define RT5670_PWR_VREF2 (0x1 << 4)
+#define RT5670_PWR_VREF2_BIT 4
+#define RT5670_PWR_FV2 (0x1 << 3)
+#define RT5670_PWR_FV2_BIT 3
+#define RT5670_LDO_SEL_MASK (0x3)
+#define RT5670_LDO_SEL_SFT 0
+
+/* Power Management for Analog 2 (0x64) */
+#define RT5670_PWR_BST1 (0x1 << 15)
+#define RT5670_PWR_BST1_BIT 15
+#define RT5670_PWR_BST2 (0x1 << 13)
+#define RT5670_PWR_BST2_BIT 13
+#define RT5670_PWR_MB1 (0x1 << 11)
+#define RT5670_PWR_MB1_BIT 11
+#define RT5670_PWR_MB2 (0x1 << 10)
+#define RT5670_PWR_MB2_BIT 10
+#define RT5670_PWR_PLL (0x1 << 9)
+#define RT5670_PWR_PLL_BIT 9
+#define RT5670_PWR_BST1_P (0x1 << 6)
+#define RT5670_PWR_BST1_P_BIT 6
+#define RT5670_PWR_BST2_P (0x1 << 4)
+#define RT5670_PWR_BST2_P_BIT 4
+#define RT5670_PWR_JD1 (0x1 << 2)
+#define RT5670_PWR_JD1_BIT 2
+#define RT5670_PWR_JD (0x1 << 1)
+#define RT5670_PWR_JD_BIT 1
+
+/* Power Management for Mixer (0x65) */
+#define RT5670_PWR_OM_L (0x1 << 15)
+#define RT5670_PWR_OM_L_BIT 15
+#define RT5670_PWR_OM_R (0x1 << 14)
+#define RT5670_PWR_OM_R_BIT 14
+#define RT5670_PWR_RM_L (0x1 << 11)
+#define RT5670_PWR_RM_L_BIT 11
+#define RT5670_PWR_RM_R (0x1 << 10)
+#define RT5670_PWR_RM_R_BIT 10
+
+/* Power Management for Volume (0x66) */
+#define RT5670_PWR_HV_L (0x1 << 11)
+#define RT5670_PWR_HV_L_BIT 11
+#define RT5670_PWR_HV_R (0x1 << 10)
+#define RT5670_PWR_HV_R_BIT 10
+#define RT5670_PWR_IN_L (0x1 << 9)
+#define RT5670_PWR_IN_L_BIT 9
+#define RT5670_PWR_IN_R (0x1 << 8)
+#define RT5670_PWR_IN_R_BIT 8
+#define RT5670_PWR_MIC_DET (0x1 << 5)
+#define RT5670_PWR_MIC_DET_BIT 5
+
+/* I2S1/2/3 Audio Serial Data Port Control (0x70 0x71 0x72) */
+#define RT5670_I2S_MS_MASK (0x1 << 15)
+#define RT5670_I2S_MS_SFT 15
+#define RT5670_I2S_MS_M (0x0 << 15)
+#define RT5670_I2S_MS_S (0x1 << 15)
+#define RT5670_I2S_IF_MASK (0x7 << 12)
+#define RT5670_I2S_IF_SFT 12
+#define RT5670_I2S_O_CP_MASK (0x3 << 10)
+#define RT5670_I2S_O_CP_SFT 10
+#define RT5670_I2S_O_CP_OFF (0x0 << 10)
+#define RT5670_I2S_O_CP_U_LAW (0x1 << 10)
+#define RT5670_I2S_O_CP_A_LAW (0x2 << 10)
+#define RT5670_I2S_I_CP_MASK (0x3 << 8)
+#define RT5670_I2S_I_CP_SFT 8
+#define RT5670_I2S_I_CP_OFF (0x0 << 8)
+#define RT5670_I2S_I_CP_U_LAW (0x1 << 8)
+#define RT5670_I2S_I_CP_A_LAW (0x2 << 8)
+#define RT5670_I2S_BP_MASK (0x1 << 7)
+#define RT5670_I2S_BP_SFT 7
+#define RT5670_I2S_BP_NOR (0x0 << 7)
+#define RT5670_I2S_BP_INV (0x1 << 7)
+#define RT5670_I2S_DL_MASK (0x3 << 2)
+#define RT5670_I2S_DL_SFT 2
+#define RT5670_I2S_DL_16 (0x0 << 2)
+#define RT5670_I2S_DL_20 (0x1 << 2)
+#define RT5670_I2S_DL_24 (0x2 << 2)
+#define RT5670_I2S_DL_8 (0x3 << 2)
+#define RT5670_I2S_DF_MASK (0x3)
+#define RT5670_I2S_DF_SFT 0
+#define RT5670_I2S_DF_I2S (0x0)
+#define RT5670_I2S_DF_LEFT (0x1)
+#define RT5670_I2S_DF_PCM_A (0x2)
+#define RT5670_I2S_DF_PCM_B (0x3)
+
+/* I2S2 Audio Serial Data Port Control (0x71) */
+#define RT5670_I2S2_SDI_MASK (0x1 << 6)
+#define RT5670_I2S2_SDI_SFT 6
+#define RT5670_I2S2_SDI_I2S1 (0x0 << 6)
+#define RT5670_I2S2_SDI_I2S2 (0x1 << 6)
+
+/* ADC/DAC Clock Control 1 (0x73) */
+#define RT5670_I2S_BCLK_MS1_MASK (0x1 << 15)
+#define RT5670_I2S_BCLK_MS1_SFT 15
+#define RT5670_I2S_BCLK_MS1_32 (0x0 << 15)
+#define RT5670_I2S_BCLK_MS1_64 (0x1 << 15)
+#define RT5670_I2S_PD1_MASK (0x7 << 12)
+#define RT5670_I2S_PD1_SFT 12
+#define RT5670_I2S_PD1_1 (0x0 << 12)
+#define RT5670_I2S_PD1_2 (0x1 << 12)
+#define RT5670_I2S_PD1_3 (0x2 << 12)
+#define RT5670_I2S_PD1_4 (0x3 << 12)
+#define RT5670_I2S_PD1_6 (0x4 << 12)
+#define RT5670_I2S_PD1_8 (0x5 << 12)
+#define RT5670_I2S_PD1_12 (0x6 << 12)
+#define RT5670_I2S_PD1_16 (0x7 << 12)
+#define RT5670_I2S_BCLK_MS2_MASK (0x1 << 11)
+#define RT5670_I2S_BCLK_MS2_SFT 11
+#define RT5670_I2S_BCLK_MS2_32 (0x0 << 11)
+#define RT5670_I2S_BCLK_MS2_64 (0x1 << 11)
+#define RT5670_I2S_PD2_MASK (0x7 << 8)
+#define RT5670_I2S_PD2_SFT 8
+#define RT5670_I2S_PD2_1 (0x0 << 8)
+#define RT5670_I2S_PD2_2 (0x1 << 8)
+#define RT5670_I2S_PD2_3 (0x2 << 8)
+#define RT5670_I2S_PD2_4 (0x3 << 8)
+#define RT5670_I2S_PD2_6 (0x4 << 8)
+#define RT5670_I2S_PD2_8 (0x5 << 8)
+#define RT5670_I2S_PD2_12 (0x6 << 8)
+#define RT5670_I2S_PD2_16 (0x7 << 8)
+#define RT5670_I2S_BCLK_MS3_MASK (0x1 << 7)
+#define RT5670_I2S_BCLK_MS3_SFT 7
+#define RT5670_I2S_BCLK_MS3_32 (0x0 << 7)
+#define RT5670_I2S_BCLK_MS3_64 (0x1 << 7)
+#define RT5670_I2S_PD3_MASK (0x7 << 4)
+#define RT5670_I2S_PD3_SFT 4
+#define RT5670_I2S_PD3_1 (0x0 << 4)
+#define RT5670_I2S_PD3_2 (0x1 << 4)
+#define RT5670_I2S_PD3_3 (0x2 << 4)
+#define RT5670_I2S_PD3_4 (0x3 << 4)
+#define RT5670_I2S_PD3_6 (0x4 << 4)
+#define RT5670_I2S_PD3_8 (0x5 << 4)
+#define RT5670_I2S_PD3_12 (0x6 << 4)
+#define RT5670_I2S_PD3_16 (0x7 << 4)
+#define RT5670_DAC_OSR_MASK (0x3 << 2)
+#define RT5670_DAC_OSR_SFT 2
+#define RT5670_DAC_OSR_128 (0x0 << 2)
+#define RT5670_DAC_OSR_64 (0x1 << 2)
+#define RT5670_DAC_OSR_32 (0x2 << 2)
+#define RT5670_DAC_OSR_16 (0x3 << 2)
+#define RT5670_ADC_OSR_MASK (0x3)
+#define RT5670_ADC_OSR_SFT 0
+#define RT5670_ADC_OSR_128 (0x0)
+#define RT5670_ADC_OSR_64 (0x1)
+#define RT5670_ADC_OSR_32 (0x2)
+#define RT5670_ADC_OSR_16 (0x3)
+
+/* ADC/DAC Clock Control 2 (0x74) */
+#define RT5670_DAC_L_OSR_MASK (0x3 << 14)
+#define RT5670_DAC_L_OSR_SFT 14
+#define RT5670_DAC_L_OSR_128 (0x0 << 14)
+#define RT5670_DAC_L_OSR_64 (0x1 << 14)
+#define RT5670_DAC_L_OSR_32 (0x2 << 14)
+#define RT5670_DAC_L_OSR_16 (0x3 << 14)
+#define RT5670_ADC_R_OSR_MASK (0x3 << 12)
+#define RT5670_ADC_R_OSR_SFT 12
+#define RT5670_ADC_R_OSR_128 (0x0 << 12)
+#define RT5670_ADC_R_OSR_64 (0x1 << 12)
+#define RT5670_ADC_R_OSR_32 (0x2 << 12)
+#define RT5670_ADC_R_OSR_16 (0x3 << 12)
+#define RT5670_DAHPF_EN (0x1 << 11)
+#define RT5670_DAHPF_EN_SFT 11
+#define RT5670_ADHPF_EN (0x1 << 10)
+#define RT5670_ADHPF_EN_SFT 10
+
+/* Digital Microphone Control (0x75) */
+#define RT5670_DMIC_1_EN_MASK (0x1 << 15)
+#define RT5670_DMIC_1_EN_SFT 15
+#define RT5670_DMIC_1_DIS (0x0 << 15)
+#define RT5670_DMIC_1_EN (0x1 << 15)
+#define RT5670_DMIC_2_EN_MASK (0x1 << 14)
+#define RT5670_DMIC_2_EN_SFT 14
+#define RT5670_DMIC_2_DIS (0x0 << 14)
+#define RT5670_DMIC_2_EN (0x1 << 14)
+#define RT5670_DMIC_1L_LH_MASK (0x1 << 13)
+#define RT5670_DMIC_1L_LH_SFT 13
+#define RT5670_DMIC_1L_LH_FALLING (0x0 << 13)
+#define RT5670_DMIC_1L_LH_RISING (0x1 << 13)
+#define RT5670_DMIC_1R_LH_MASK (0x1 << 12)
+#define RT5670_DMIC_1R_LH_SFT 12
+#define RT5670_DMIC_1R_LH_FALLING (0x0 << 12)
+#define RT5670_DMIC_1R_LH_RISING (0x1 << 12)
+#define RT5670_DMIC_2_DP_MASK (0x1 << 10)
+#define RT5670_DMIC_2_DP_SFT 10
+#define RT5670_DMIC_2_DP_GPIO8 (0x0 << 10)
+#define RT5670_DMIC_2_DP_IN3N (0x1 << 10)
+#define RT5670_DMIC_2L_LH_MASK (0x1 << 9)
+#define RT5670_DMIC_2L_LH_SFT 9
+#define RT5670_DMIC_2L_LH_FALLING (0x0 << 9)
+#define RT5670_DMIC_2L_LH_RISING (0x1 << 9)
+#define RT5670_DMIC_2R_LH_MASK (0x1 << 8)
+#define RT5670_DMIC_2R_LH_SFT 8
+#define RT5670_DMIC_2R_LH_FALLING (0x0 << 8)
+#define RT5670_DMIC_2R_LH_RISING (0x1 << 8)
+#define RT5670_DMIC_CLK_MASK (0x7 << 5)
+#define RT5670_DMIC_CLK_SFT 5
+#define RT5670_DMIC_3_EN_MASK (0x1 << 4)
+#define RT5670_DMIC_3_EN_SFT 4
+#define RT5670_DMIC_3_DIS (0x0 << 4)
+#define RT5670_DMIC_3_EN (0x1 << 4)
+#define RT5670_DMIC_1_DP_MASK (0x3 << 0)
+#define RT5670_DMIC_1_DP_SFT 0
+#define RT5670_DMIC_1_DP_GPIO6 (0x0 << 0)
+#define RT5670_DMIC_1_DP_IN2P (0x1 << 0)
+#define RT5670_DMIC_1_DP_GPIO7 (0x2 << 0)
+
+/* Digital Microphone Control2 (0x76) */
+#define RT5670_DMIC_3_DP_MASK (0x3 << 6)
+#define RT5670_DMIC_3_DP_SFT 6
+#define RT5670_DMIC_3_DP_GPIO9 (0x0 << 6)
+#define RT5670_DMIC_3_DP_GPIO10 (0x1 << 6)
+#define RT5670_DMIC_3_DP_GPIO5 (0x2 << 6)
+
+/* Global Clock Control (0x80) */
+#define RT5670_SCLK_SRC_MASK (0x3 << 14)
+#define RT5670_SCLK_SRC_SFT 14
+#define RT5670_SCLK_SRC_MCLK (0x0 << 14)
+#define RT5670_SCLK_SRC_PLL1 (0x1 << 14)
+#define RT5670_SCLK_SRC_RCCLK (0x2 << 14) /* 15MHz */
+#define RT5670_PLL1_SRC_MASK (0x3 << 12)
+#define RT5670_PLL1_SRC_SFT 12
+#define RT5670_PLL1_SRC_MCLK (0x0 << 12)
+#define RT5670_PLL1_SRC_BCLK1 (0x1 << 12)
+#define RT5670_PLL1_SRC_BCLK2 (0x2 << 12)
+#define RT5670_PLL1_SRC_BCLK3 (0x3 << 12)
+#define RT5670_PLL1_PD_MASK (0x1 << 3)
+#define RT5670_PLL1_PD_SFT 3
+#define RT5670_PLL1_PD_1 (0x0 << 3)
+#define RT5670_PLL1_PD_2 (0x1 << 3)
+
+#define RT5670_PLL_INP_MAX 40000000
+#define RT5670_PLL_INP_MIN 256000
+/* PLL M/N/K Code Control 1 (0x81) */
+#define RT5670_PLL_N_MAX 0x1ff
+#define RT5670_PLL_N_MASK (RT5670_PLL_N_MAX << 7)
+#define RT5670_PLL_N_SFT 7
+#define RT5670_PLL_K_MAX 0x1f
+#define RT5670_PLL_K_MASK (RT5670_PLL_K_MAX)
+#define RT5670_PLL_K_SFT 0
+
+/* PLL M/N/K Code Control 2 (0x82) */
+#define RT5670_PLL_M_MAX 0xf
+#define RT5670_PLL_M_MASK (RT5670_PLL_M_MAX << 12)
+#define RT5670_PLL_M_SFT 12
+#define RT5670_PLL_M_BP (0x1 << 11)
+#define RT5670_PLL_M_BP_SFT 11
+
+/* ASRC Control 1 (0x83) */
+#define RT5670_STO_T_MASK (0x1 << 15)
+#define RT5670_STO_T_SFT 15
+#define RT5670_STO_T_SCLK (0x0 << 15)
+#define RT5670_STO_T_LRCK1 (0x1 << 15)
+#define RT5670_M1_T_MASK (0x1 << 14)
+#define RT5670_M1_T_SFT 14
+#define RT5670_M1_T_I2S2 (0x0 << 14)
+#define RT5670_M1_T_I2S2_D3 (0x1 << 14)
+#define RT5670_I2S2_F_MASK (0x1 << 12)
+#define RT5670_I2S2_F_SFT 12
+#define RT5670_I2S2_F_I2S2_D2 (0x0 << 12)
+#define RT5670_I2S2_F_I2S1_TCLK (0x1 << 12)
+#define RT5670_DMIC_1_M_MASK (0x1 << 9)
+#define RT5670_DMIC_1_M_SFT 9
+#define RT5670_DMIC_1_M_NOR (0x0 << 9)
+#define RT5670_DMIC_1_M_ASYN (0x1 << 9)
+#define RT5670_DMIC_2_M_MASK (0x1 << 8)
+#define RT5670_DMIC_2_M_SFT 8
+#define RT5670_DMIC_2_M_NOR (0x0 << 8)
+#define RT5670_DMIC_2_M_ASYN (0x1 << 8)
+
+/* ASRC Control 2 (0x84) */
+#define RT5670_MDA_L_M_MASK (0x1 << 15)
+#define RT5670_MDA_L_M_SFT 15
+#define RT5670_MDA_L_M_NOR (0x0 << 15)
+#define RT5670_MDA_L_M_ASYN (0x1 << 15)
+#define RT5670_MDA_R_M_MASK (0x1 << 14)
+#define RT5670_MDA_R_M_SFT 14
+#define RT5670_MDA_R_M_NOR (0x0 << 14)
+#define RT5670_MDA_R_M_ASYN (0x1 << 14)
+#define RT5670_MAD_L_M_MASK (0x1 << 13)
+#define RT5670_MAD_L_M_SFT 13
+#define RT5670_MAD_L_M_NOR (0x0 << 13)
+#define RT5670_MAD_L_M_ASYN (0x1 << 13)
+#define RT5670_MAD_R_M_MASK (0x1 << 12)
+#define RT5670_MAD_R_M_SFT 12
+#define RT5670_MAD_R_M_NOR (0x0 << 12)
+#define RT5670_MAD_R_M_ASYN (0x1 << 12)
+#define RT5670_ADC_M_MASK (0x1 << 11)
+#define RT5670_ADC_M_SFT 11
+#define RT5670_ADC_M_NOR (0x0 << 11)
+#define RT5670_ADC_M_ASYN (0x1 << 11)
+#define RT5670_STO_DAC_M_MASK (0x1 << 5)
+#define RT5670_STO_DAC_M_SFT 5
+#define RT5670_STO_DAC_M_NOR (0x0 << 5)
+#define RT5670_STO_DAC_M_ASYN (0x1 << 5)
+#define RT5670_I2S1_R_D_MASK (0x1 << 4)
+#define RT5670_I2S1_R_D_SFT 4
+#define RT5670_I2S1_R_D_DIS (0x0 << 4)
+#define RT5670_I2S1_R_D_EN (0x1 << 4)
+#define RT5670_I2S2_R_D_MASK (0x1 << 3)
+#define RT5670_I2S2_R_D_SFT 3
+#define RT5670_I2S2_R_D_DIS (0x0 << 3)
+#define RT5670_I2S2_R_D_EN (0x1 << 3)
+#define RT5670_PRE_SCLK_MASK (0x3)
+#define RT5670_PRE_SCLK_SFT 0
+#define RT5670_PRE_SCLK_512 (0x0)
+#define RT5670_PRE_SCLK_1024 (0x1)
+#define RT5670_PRE_SCLK_2048 (0x2)
+
+/* ASRC Control 3 (0x85) */
+#define RT5670_I2S1_RATE_MASK (0xf << 12)
+#define RT5670_I2S1_RATE_SFT 12
+#define RT5670_I2S2_RATE_MASK (0xf << 8)
+#define RT5670_I2S2_RATE_SFT 8
+
+/* ASRC Control 4 (0x89) */
+#define RT5670_I2S1_PD_MASK (0x7 << 12)
+#define RT5670_I2S1_PD_SFT 12
+#define RT5670_I2S2_PD_MASK (0x7 << 8)
+#define RT5670_I2S2_PD_SFT 8
+
+/* HPOUT Over Current Detection (0x8b) */
+#define RT5670_HP_OVCD_MASK (0x1 << 10)
+#define RT5670_HP_OVCD_SFT 10
+#define RT5670_HP_OVCD_DIS (0x0 << 10)
+#define RT5670_HP_OVCD_EN (0x1 << 10)
+#define RT5670_HP_OC_TH_MASK (0x3 << 8)
+#define RT5670_HP_OC_TH_SFT 8
+#define RT5670_HP_OC_TH_90 (0x0 << 8)
+#define RT5670_HP_OC_TH_105 (0x1 << 8)
+#define RT5670_HP_OC_TH_120 (0x2 << 8)
+#define RT5670_HP_OC_TH_135 (0x3 << 8)
+
+/* Class D Over Current Control (0x8c) */
+#define RT5670_CLSD_OC_MASK (0x1 << 9)
+#define RT5670_CLSD_OC_SFT 9
+#define RT5670_CLSD_OC_PU (0x0 << 9)
+#define RT5670_CLSD_OC_PD (0x1 << 9)
+#define RT5670_AUTO_PD_MASK (0x1 << 8)
+#define RT5670_AUTO_PD_SFT 8
+#define RT5670_AUTO_PD_DIS (0x0 << 8)
+#define RT5670_AUTO_PD_EN (0x1 << 8)
+#define RT5670_CLSD_OC_TH_MASK (0x3f)
+#define RT5670_CLSD_OC_TH_SFT 0
+
+/* Class D Output Control (0x8d) */
+#define RT5670_CLSD_RATIO_MASK (0xf << 12)
+#define RT5670_CLSD_RATIO_SFT 12
+#define RT5670_CLSD_OM_MASK (0x1 << 11)
+#define RT5670_CLSD_OM_SFT 11
+#define RT5670_CLSD_OM_MONO (0x0 << 11)
+#define RT5670_CLSD_OM_STO (0x1 << 11)
+#define RT5670_CLSD_SCH_MASK (0x1 << 10)
+#define RT5670_CLSD_SCH_SFT 10
+#define RT5670_CLSD_SCH_L (0x0 << 10)
+#define RT5670_CLSD_SCH_S (0x1 << 10)
+
+/* Depop Mode Control 1 (0x8e) */
+#define RT5670_SMT_TRIG_MASK (0x1 << 15)
+#define RT5670_SMT_TRIG_SFT 15
+#define RT5670_SMT_TRIG_DIS (0x0 << 15)
+#define RT5670_SMT_TRIG_EN (0x1 << 15)
+#define RT5670_HP_L_SMT_MASK (0x1 << 9)
+#define RT5670_HP_L_SMT_SFT 9
+#define RT5670_HP_L_SMT_DIS (0x0 << 9)
+#define RT5670_HP_L_SMT_EN (0x1 << 9)
+#define RT5670_HP_R_SMT_MASK (0x1 << 8)
+#define RT5670_HP_R_SMT_SFT 8
+#define RT5670_HP_R_SMT_DIS (0x0 << 8)
+#define RT5670_HP_R_SMT_EN (0x1 << 8)
+#define RT5670_HP_CD_PD_MASK (0x1 << 7)
+#define RT5670_HP_CD_PD_SFT 7
+#define RT5670_HP_CD_PD_DIS (0x0 << 7)
+#define RT5670_HP_CD_PD_EN (0x1 << 7)
+#define RT5670_RSTN_MASK (0x1 << 6)
+#define RT5670_RSTN_SFT 6
+#define RT5670_RSTN_DIS (0x0 << 6)
+#define RT5670_RSTN_EN (0x1 << 6)
+#define RT5670_RSTP_MASK (0x1 << 5)
+#define RT5670_RSTP_SFT 5
+#define RT5670_RSTP_DIS (0x0 << 5)
+#define RT5670_RSTP_EN (0x1 << 5)
+#define RT5670_HP_CO_MASK (0x1 << 4)
+#define RT5670_HP_CO_SFT 4
+#define RT5670_HP_CO_DIS (0x0 << 4)
+#define RT5670_HP_CO_EN (0x1 << 4)
+#define RT5670_HP_CP_MASK (0x1 << 3)
+#define RT5670_HP_CP_SFT 3
+#define RT5670_HP_CP_PD (0x0 << 3)
+#define RT5670_HP_CP_PU (0x1 << 3)
+#define RT5670_HP_SG_MASK (0x1 << 2)
+#define RT5670_HP_SG_SFT 2
+#define RT5670_HP_SG_DIS (0x0 << 2)
+#define RT5670_HP_SG_EN (0x1 << 2)
+#define RT5670_HP_DP_MASK (0x1 << 1)
+#define RT5670_HP_DP_SFT 1
+#define RT5670_HP_DP_PD (0x0 << 1)
+#define RT5670_HP_DP_PU (0x1 << 1)
+#define RT5670_HP_CB_MASK (0x1)
+#define RT5670_HP_CB_SFT 0
+#define RT5670_HP_CB_PD (0x0)
+#define RT5670_HP_CB_PU (0x1)
+
+/* Depop Mode Control 2 (0x8f) */
+#define RT5670_DEPOP_MASK (0x1 << 13)
+#define RT5670_DEPOP_SFT 13
+#define RT5670_DEPOP_AUTO (0x0 << 13)
+#define RT5670_DEPOP_MAN (0x1 << 13)
+#define RT5670_RAMP_MASK (0x1 << 12)
+#define RT5670_RAMP_SFT 12
+#define RT5670_RAMP_DIS (0x0 << 12)
+#define RT5670_RAMP_EN (0x1 << 12)
+#define RT5670_BPS_MASK (0x1 << 11)
+#define RT5670_BPS_SFT 11
+#define RT5670_BPS_DIS (0x0 << 11)
+#define RT5670_BPS_EN (0x1 << 11)
+#define RT5670_FAST_UPDN_MASK (0x1 << 10)
+#define RT5670_FAST_UPDN_SFT 10
+#define RT5670_FAST_UPDN_DIS (0x0 << 10)
+#define RT5670_FAST_UPDN_EN (0x1 << 10)
+#define RT5670_MRES_MASK (0x3 << 8)
+#define RT5670_MRES_SFT 8
+#define RT5670_MRES_15MO (0x0 << 8)
+#define RT5670_MRES_25MO (0x1 << 8)
+#define RT5670_MRES_35MO (0x2 << 8)
+#define RT5670_MRES_45MO (0x3 << 8)
+#define RT5670_VLO_MASK (0x1 << 7)
+#define RT5670_VLO_SFT 7
+#define RT5670_VLO_3V (0x0 << 7)
+#define RT5670_VLO_32V (0x1 << 7)
+#define RT5670_DIG_DP_MASK (0x1 << 6)
+#define RT5670_DIG_DP_SFT 6
+#define RT5670_DIG_DP_DIS (0x0 << 6)
+#define RT5670_DIG_DP_EN (0x1 << 6)
+#define RT5670_DP_TH_MASK (0x3 << 4)
+#define RT5670_DP_TH_SFT 4
+
+/* Depop Mode Control 3 (0x90) */
+#define RT5670_CP_SYS_MASK (0x7 << 12)
+#define RT5670_CP_SYS_SFT 12
+#define RT5670_CP_FQ1_MASK (0x7 << 8)
+#define RT5670_CP_FQ1_SFT 8
+#define RT5670_CP_FQ2_MASK (0x7 << 4)
+#define RT5670_CP_FQ2_SFT 4
+#define RT5670_CP_FQ3_MASK (0x7)
+#define RT5670_CP_FQ3_SFT 0
+#define RT5670_CP_FQ_1_5_KHZ 0
+#define RT5670_CP_FQ_3_KHZ 1
+#define RT5670_CP_FQ_6_KHZ 2
+#define RT5670_CP_FQ_12_KHZ 3
+#define RT5670_CP_FQ_24_KHZ 4
+#define RT5670_CP_FQ_48_KHZ 5
+#define RT5670_CP_FQ_96_KHZ 6
+#define RT5670_CP_FQ_192_KHZ 7
+
+/* HPOUT charge pump (0x91) */
+#define RT5670_OSW_L_MASK (0x1 << 11)
+#define RT5670_OSW_L_SFT 11
+#define RT5670_OSW_L_DIS (0x0 << 11)
+#define RT5670_OSW_L_EN (0x1 << 11)
+#define RT5670_OSW_R_MASK (0x1 << 10)
+#define RT5670_OSW_R_SFT 10
+#define RT5670_OSW_R_DIS (0x0 << 10)
+#define RT5670_OSW_R_EN (0x1 << 10)
+#define RT5670_PM_HP_MASK (0x3 << 8)
+#define RT5670_PM_HP_SFT 8
+#define RT5670_PM_HP_LV (0x0 << 8)
+#define RT5670_PM_HP_MV (0x1 << 8)
+#define RT5670_PM_HP_HV (0x2 << 8)
+#define RT5670_IB_HP_MASK (0x3 << 6)
+#define RT5670_IB_HP_SFT 6
+#define RT5670_IB_HP_125IL (0x0 << 6)
+#define RT5670_IB_HP_25IL (0x1 << 6)
+#define RT5670_IB_HP_5IL (0x2 << 6)
+#define RT5670_IB_HP_1IL (0x3 << 6)
+
+/* PV detection and SPK gain control (0x92) */
+#define RT5670_PVDD_DET_MASK (0x1 << 15)
+#define RT5670_PVDD_DET_SFT 15
+#define RT5670_PVDD_DET_DIS (0x0 << 15)
+#define RT5670_PVDD_DET_EN (0x1 << 15)
+#define RT5670_SPK_AG_MASK (0x1 << 14)
+#define RT5670_SPK_AG_SFT 14
+#define RT5670_SPK_AG_DIS (0x0 << 14)
+#define RT5670_SPK_AG_EN (0x1 << 14)
+
+/* Micbias Control (0x93) */
+#define RT5670_MIC1_BS_MASK (0x1 << 15)
+#define RT5670_MIC1_BS_SFT 15
+#define RT5670_MIC1_BS_9AV (0x0 << 15)
+#define RT5670_MIC1_BS_75AV (0x1 << 15)
+#define RT5670_MIC2_BS_MASK (0x1 << 14)
+#define RT5670_MIC2_BS_SFT 14
+#define RT5670_MIC2_BS_9AV (0x0 << 14)
+#define RT5670_MIC2_BS_75AV (0x1 << 14)
+#define RT5670_MIC1_CLK_MASK (0x1 << 13)
+#define RT5670_MIC1_CLK_SFT 13
+#define RT5670_MIC1_CLK_DIS (0x0 << 13)
+#define RT5670_MIC1_CLK_EN (0x1 << 13)
+#define RT5670_MIC2_CLK_MASK (0x1 << 12)
+#define RT5670_MIC2_CLK_SFT 12
+#define RT5670_MIC2_CLK_DIS (0x0 << 12)
+#define RT5670_MIC2_CLK_EN (0x1 << 12)
+#define RT5670_MIC1_OVCD_MASK (0x1 << 11)
+#define RT5670_MIC1_OVCD_SFT 11
+#define RT5670_MIC1_OVCD_DIS (0x0 << 11)
+#define RT5670_MIC1_OVCD_EN (0x1 << 11)
+#define RT5670_MIC1_OVTH_MASK (0x3 << 9)
+#define RT5670_MIC1_OVTH_SFT 9
+#define RT5670_MIC1_OVTH_600UA (0x0 << 9)
+#define RT5670_MIC1_OVTH_1500UA (0x1 << 9)
+#define RT5670_MIC1_OVTH_2000UA (0x2 << 9)
+#define RT5670_MIC2_OVCD_MASK (0x1 << 8)
+#define RT5670_MIC2_OVCD_SFT 8
+#define RT5670_MIC2_OVCD_DIS (0x0 << 8)
+#define RT5670_MIC2_OVCD_EN (0x1 << 8)
+#define RT5670_MIC2_OVTH_MASK (0x3 << 6)
+#define RT5670_MIC2_OVTH_SFT 6
+#define RT5670_MIC2_OVTH_600UA (0x0 << 6)
+#define RT5670_MIC2_OVTH_1500UA (0x1 << 6)
+#define RT5670_MIC2_OVTH_2000UA (0x2 << 6)
+#define RT5670_PWR_MB_MASK (0x1 << 5)
+#define RT5670_PWR_MB_SFT 5
+#define RT5670_PWR_MB_PD (0x0 << 5)
+#define RT5670_PWR_MB_PU (0x1 << 5)
+#define RT5670_PWR_CLK25M_MASK (0x1 << 4)
+#define RT5670_PWR_CLK25M_SFT 4
+#define RT5670_PWR_CLK25M_PD (0x0 << 4)
+#define RT5670_PWR_CLK25M_PU (0x1 << 4)
+
+/* Analog JD Control 1 (0x94) */
+#define RT5670_JD1_MODE_MASK (0x3 << 0)
+#define RT5670_JD1_MODE_0 (0x0 << 0)
+#define RT5670_JD1_MODE_1 (0x1 << 0)
+#define RT5670_JD1_MODE_2 (0x2 << 0)
+
+/* VAD Control 4 (0x9d) */
+#define RT5670_VAD_SEL_MASK (0x3 << 8)
+#define RT5670_VAD_SEL_SFT 8
+
+/* EQ Control 1 (0xb0) */
+#define RT5670_EQ_SRC_MASK (0x1 << 15)
+#define RT5670_EQ_SRC_SFT 15
+#define RT5670_EQ_SRC_DAC (0x0 << 15)
+#define RT5670_EQ_SRC_ADC (0x1 << 15)
+#define RT5670_EQ_UPD (0x1 << 14)
+#define RT5670_EQ_UPD_BIT 14
+#define RT5670_EQ_CD_MASK (0x1 << 13)
+#define RT5670_EQ_CD_SFT 13
+#define RT5670_EQ_CD_DIS (0x0 << 13)
+#define RT5670_EQ_CD_EN (0x1 << 13)
+#define RT5670_EQ_DITH_MASK (0x3 << 8)
+#define RT5670_EQ_DITH_SFT 8
+#define RT5670_EQ_DITH_NOR (0x0 << 8)
+#define RT5670_EQ_DITH_LSB (0x1 << 8)
+#define RT5670_EQ_DITH_LSB_1 (0x2 << 8)
+#define RT5670_EQ_DITH_LSB_2 (0x3 << 8)
+
+/* EQ Control 2 (0xb1) */
+#define RT5670_EQ_HPF1_M_MASK (0x1 << 8)
+#define RT5670_EQ_HPF1_M_SFT 8
+#define RT5670_EQ_HPF1_M_HI (0x0 << 8)
+#define RT5670_EQ_HPF1_M_1ST (0x1 << 8)
+#define RT5670_EQ_LPF1_M_MASK (0x1 << 7)
+#define RT5670_EQ_LPF1_M_SFT 7
+#define RT5670_EQ_LPF1_M_LO (0x0 << 7)
+#define RT5670_EQ_LPF1_M_1ST (0x1 << 7)
+#define RT5670_EQ_HPF2_MASK (0x1 << 6)
+#define RT5670_EQ_HPF2_SFT 6
+#define RT5670_EQ_HPF2_DIS (0x0 << 6)
+#define RT5670_EQ_HPF2_EN (0x1 << 6)
+#define RT5670_EQ_HPF1_MASK (0x1 << 5)
+#define RT5670_EQ_HPF1_SFT 5
+#define RT5670_EQ_HPF1_DIS (0x0 << 5)
+#define RT5670_EQ_HPF1_EN (0x1 << 5)
+#define RT5670_EQ_BPF4_MASK (0x1 << 4)
+#define RT5670_EQ_BPF4_SFT 4
+#define RT5670_EQ_BPF4_DIS (0x0 << 4)
+#define RT5670_EQ_BPF4_EN (0x1 << 4)
+#define RT5670_EQ_BPF3_MASK (0x1 << 3)
+#define RT5670_EQ_BPF3_SFT 3
+#define RT5670_EQ_BPF3_DIS (0x0 << 3)
+#define RT5670_EQ_BPF3_EN (0x1 << 3)
+#define RT5670_EQ_BPF2_MASK (0x1 << 2)
+#define RT5670_EQ_BPF2_SFT 2
+#define RT5670_EQ_BPF2_DIS (0x0 << 2)
+#define RT5670_EQ_BPF2_EN (0x1 << 2)
+#define RT5670_EQ_BPF1_MASK (0x1 << 1)
+#define RT5670_EQ_BPF1_SFT 1
+#define RT5670_EQ_BPF1_DIS (0x0 << 1)
+#define RT5670_EQ_BPF1_EN (0x1 << 1)
+#define RT5670_EQ_LPF_MASK (0x1)
+#define RT5670_EQ_LPF_SFT 0
+#define RT5670_EQ_LPF_DIS (0x0)
+#define RT5670_EQ_LPF_EN (0x1)
+#define RT5670_EQ_CTRL_MASK (0x7f)
+
+/* Memory Test (0xb2) */
+#define RT5670_MT_MASK (0x1 << 15)
+#define RT5670_MT_SFT 15
+#define RT5670_MT_DIS (0x0 << 15)
+#define RT5670_MT_EN (0x1 << 15)
+
+/* DRC/AGC Control 1 (0xb4) */
+#define RT5670_DRC_AGC_P_MASK (0x1 << 15)
+#define RT5670_DRC_AGC_P_SFT 15
+#define RT5670_DRC_AGC_P_DAC (0x0 << 15)
+#define RT5670_DRC_AGC_P_ADC (0x1 << 15)
+#define RT5670_DRC_AGC_MASK (0x1 << 14)
+#define RT5670_DRC_AGC_SFT 14
+#define RT5670_DRC_AGC_DIS (0x0 << 14)
+#define RT5670_DRC_AGC_EN (0x1 << 14)
+#define RT5670_DRC_AGC_UPD (0x1 << 13)
+#define RT5670_DRC_AGC_UPD_BIT 13
+#define RT5670_DRC_AGC_AR_MASK (0x1f << 8)
+#define RT5670_DRC_AGC_AR_SFT 8
+#define RT5670_DRC_AGC_R_MASK (0x7 << 5)
+#define RT5670_DRC_AGC_R_SFT 5
+#define RT5670_DRC_AGC_R_48K (0x1 << 5)
+#define RT5670_DRC_AGC_R_96K (0x2 << 5)
+#define RT5670_DRC_AGC_R_192K (0x3 << 5)
+#define RT5670_DRC_AGC_R_441K (0x5 << 5)
+#define RT5670_DRC_AGC_R_882K (0x6 << 5)
+#define RT5670_DRC_AGC_R_1764K (0x7 << 5)
+#define RT5670_DRC_AGC_RC_MASK (0x1f)
+#define RT5670_DRC_AGC_RC_SFT 0
+
+/* DRC/AGC Control 2 (0xb5) */
+#define RT5670_DRC_AGC_POB_MASK (0x3f << 8)
+#define RT5670_DRC_AGC_POB_SFT 8
+#define RT5670_DRC_AGC_CP_MASK (0x1 << 7)
+#define RT5670_DRC_AGC_CP_SFT 7
+#define RT5670_DRC_AGC_CP_DIS (0x0 << 7)
+#define RT5670_DRC_AGC_CP_EN (0x1 << 7)
+#define RT5670_DRC_AGC_CPR_MASK (0x3 << 5)
+#define RT5670_DRC_AGC_CPR_SFT 5
+#define RT5670_DRC_AGC_CPR_1_1 (0x0 << 5)
+#define RT5670_DRC_AGC_CPR_1_2 (0x1 << 5)
+#define RT5670_DRC_AGC_CPR_1_3 (0x2 << 5)
+#define RT5670_DRC_AGC_CPR_1_4 (0x3 << 5)
+#define RT5670_DRC_AGC_PRB_MASK (0x1f)
+#define RT5670_DRC_AGC_PRB_SFT 0
+
+/* DRC/AGC Control 3 (0xb6) */
+#define RT5670_DRC_AGC_NGB_MASK (0xf << 12)
+#define RT5670_DRC_AGC_NGB_SFT 12
+#define RT5670_DRC_AGC_TAR_MASK (0x1f << 7)
+#define RT5670_DRC_AGC_TAR_SFT 7
+#define RT5670_DRC_AGC_NG_MASK (0x1 << 6)
+#define RT5670_DRC_AGC_NG_SFT 6
+#define RT5670_DRC_AGC_NG_DIS (0x0 << 6)
+#define RT5670_DRC_AGC_NG_EN (0x1 << 6)
+#define RT5670_DRC_AGC_NGH_MASK (0x1 << 5)
+#define RT5670_DRC_AGC_NGH_SFT 5
+#define RT5670_DRC_AGC_NGH_DIS (0x0 << 5)
+#define RT5670_DRC_AGC_NGH_EN (0x1 << 5)
+#define RT5670_DRC_AGC_NGT_MASK (0x1f)
+#define RT5670_DRC_AGC_NGT_SFT 0
+
+/* Jack Detect Control (0xbb) */
+#define RT5670_JD_MASK (0x7 << 13)
+#define RT5670_JD_SFT 13
+#define RT5670_JD_DIS (0x0 << 13)
+#define RT5670_JD_GPIO1 (0x1 << 13)
+#define RT5670_JD_JD1_IN4P (0x2 << 13)
+#define RT5670_JD_JD2_IN4N (0x3 << 13)
+#define RT5670_JD_GPIO2 (0x4 << 13)
+#define RT5670_JD_GPIO3 (0x5 << 13)
+#define RT5670_JD_GPIO4 (0x6 << 13)
+#define RT5670_JD_HP_MASK (0x1 << 11)
+#define RT5670_JD_HP_SFT 11
+#define RT5670_JD_HP_DIS (0x0 << 11)
+#define RT5670_JD_HP_EN (0x1 << 11)
+#define RT5670_JD_HP_TRG_MASK (0x1 << 10)
+#define RT5670_JD_HP_TRG_SFT 10
+#define RT5670_JD_HP_TRG_LO (0x0 << 10)
+#define RT5670_JD_HP_TRG_HI (0x1 << 10)
+#define RT5670_JD_SPL_MASK (0x1 << 9)
+#define RT5670_JD_SPL_SFT 9
+#define RT5670_JD_SPL_DIS (0x0 << 9)
+#define RT5670_JD_SPL_EN (0x1 << 9)
+#define RT5670_JD_SPL_TRG_MASK (0x1 << 8)
+#define RT5670_JD_SPL_TRG_SFT 8
+#define RT5670_JD_SPL_TRG_LO (0x0 << 8)
+#define RT5670_JD_SPL_TRG_HI (0x1 << 8)
+#define RT5670_JD_SPR_MASK (0x1 << 7)
+#define RT5670_JD_SPR_SFT 7
+#define RT5670_JD_SPR_DIS (0x0 << 7)
+#define RT5670_JD_SPR_EN (0x1 << 7)
+#define RT5670_JD_SPR_TRG_MASK (0x1 << 6)
+#define RT5670_JD_SPR_TRG_SFT 6
+#define RT5670_JD_SPR_TRG_LO (0x0 << 6)
+#define RT5670_JD_SPR_TRG_HI (0x1 << 6)
+#define RT5670_JD_MO_MASK (0x1 << 5)
+#define RT5670_JD_MO_SFT 5
+#define RT5670_JD_MO_DIS (0x0 << 5)
+#define RT5670_JD_MO_EN (0x1 << 5)
+#define RT5670_JD_MO_TRG_MASK (0x1 << 4)
+#define RT5670_JD_MO_TRG_SFT 4
+#define RT5670_JD_MO_TRG_LO (0x0 << 4)
+#define RT5670_JD_MO_TRG_HI (0x1 << 4)
+#define RT5670_JD_LO_MASK (0x1 << 3)
+#define RT5670_JD_LO_SFT 3
+#define RT5670_JD_LO_DIS (0x0 << 3)
+#define RT5670_JD_LO_EN (0x1 << 3)
+#define RT5670_JD_LO_TRG_MASK (0x1 << 2)
+#define RT5670_JD_LO_TRG_SFT 2
+#define RT5670_JD_LO_TRG_LO (0x0 << 2)
+#define RT5670_JD_LO_TRG_HI (0x1 << 2)
+#define RT5670_JD1_IN4P_MASK (0x1 << 1)
+#define RT5670_JD1_IN4P_SFT 1
+#define RT5670_JD1_IN4P_DIS (0x0 << 1)
+#define RT5670_JD1_IN4P_EN (0x1 << 1)
+#define RT5670_JD2_IN4N_MASK (0x1)
+#define RT5670_JD2_IN4N_SFT 0
+#define RT5670_JD2_IN4N_DIS (0x0)
+#define RT5670_JD2_IN4N_EN (0x1)
+
+/* IRQ Control 1 (0xbd) */
+#define RT5670_IRQ_JD_MASK (0x1 << 15)
+#define RT5670_IRQ_JD_SFT 15
+#define RT5670_IRQ_JD_BP (0x0 << 15)
+#define RT5670_IRQ_JD_NOR (0x1 << 15)
+#define RT5670_IRQ_OT_MASK (0x1 << 14)
+#define RT5670_IRQ_OT_SFT 14
+#define RT5670_IRQ_OT_BP (0x0 << 14)
+#define RT5670_IRQ_OT_NOR (0x1 << 14)
+#define RT5670_JD_STKY_MASK (0x1 << 13)
+#define RT5670_JD_STKY_SFT 13
+#define RT5670_JD_STKY_DIS (0x0 << 13)
+#define RT5670_JD_STKY_EN (0x1 << 13)
+#define RT5670_OT_STKY_MASK (0x1 << 12)
+#define RT5670_OT_STKY_SFT 12
+#define RT5670_OT_STKY_DIS (0x0 << 12)
+#define RT5670_OT_STKY_EN (0x1 << 12)
+#define RT5670_JD_P_MASK (0x1 << 11)
+#define RT5670_JD_P_SFT 11
+#define RT5670_JD_P_NOR (0x0 << 11)
+#define RT5670_JD_P_INV (0x1 << 11)
+#define RT5670_OT_P_MASK (0x1 << 10)
+#define RT5670_OT_P_SFT 10
+#define RT5670_OT_P_NOR (0x0 << 10)
+#define RT5670_OT_P_INV (0x1 << 10)
+#define RT5670_JD1_1_EN_MASK (0x1 << 9)
+#define RT5670_JD1_1_EN_SFT 9
+#define RT5670_JD1_1_DIS (0x0 << 9)
+#define RT5670_JD1_1_EN (0x1 << 9)
+
+/* IRQ Control 2 (0xbe) */
+#define RT5670_IRQ_MB1_OC_MASK (0x1 << 15)
+#define RT5670_IRQ_MB1_OC_SFT 15
+#define RT5670_IRQ_MB1_OC_BP (0x0 << 15)
+#define RT5670_IRQ_MB1_OC_NOR (0x1 << 15)
+#define RT5670_IRQ_MB2_OC_MASK (0x1 << 14)
+#define RT5670_IRQ_MB2_OC_SFT 14
+#define RT5670_IRQ_MB2_OC_BP (0x0 << 14)
+#define RT5670_IRQ_MB2_OC_NOR (0x1 << 14)
+#define RT5670_MB1_OC_STKY_MASK (0x1 << 11)
+#define RT5670_MB1_OC_STKY_SFT 11
+#define RT5670_MB1_OC_STKY_DIS (0x0 << 11)
+#define RT5670_MB1_OC_STKY_EN (0x1 << 11)
+#define RT5670_MB2_OC_STKY_MASK (0x1 << 10)
+#define RT5670_MB2_OC_STKY_SFT 10
+#define RT5670_MB2_OC_STKY_DIS (0x0 << 10)
+#define RT5670_MB2_OC_STKY_EN (0x1 << 10)
+#define RT5670_MB1_OC_P_MASK (0x1 << 7)
+#define RT5670_MB1_OC_P_SFT 7
+#define RT5670_MB1_OC_P_NOR (0x0 << 7)
+#define RT5670_MB1_OC_P_INV (0x1 << 7)
+#define RT5670_MB2_OC_P_MASK (0x1 << 6)
+#define RT5670_MB2_OC_P_SFT 6
+#define RT5670_MB2_OC_P_NOR (0x0 << 6)
+#define RT5670_MB2_OC_P_INV (0x1 << 6)
+#define RT5670_MB1_OC_CLR (0x1 << 3)
+#define RT5670_MB1_OC_CLR_SFT 3
+#define RT5670_MB2_OC_CLR (0x1 << 2)
+#define RT5670_MB2_OC_CLR_SFT 2
+
+/* GPIO Control 1 (0xc0) */
+#define RT5670_GP1_PIN_MASK (0x1 << 15)
+#define RT5670_GP1_PIN_SFT 15
+#define RT5670_GP1_PIN_GPIO1 (0x0 << 15)
+#define RT5670_GP1_PIN_IRQ (0x1 << 15)
+#define RT5670_GP2_PIN_MASK (0x1 << 14)
+#define RT5670_GP2_PIN_SFT 14
+#define RT5670_GP2_PIN_GPIO2 (0x0 << 14)
+#define RT5670_GP2_PIN_DMIC1_SCL (0x1 << 14)
+#define RT5670_GP3_PIN_MASK (0x3 << 12)
+#define RT5670_GP3_PIN_SFT 12
+#define RT5670_GP3_PIN_GPIO3 (0x0 << 12)
+#define RT5670_GP3_PIN_DMIC1_SDA (0x1 << 12)
+#define RT5670_GP3_PIN_IRQ (0x2 << 12)
+#define RT5670_GP4_PIN_MASK (0x1 << 11)
+#define RT5670_GP4_PIN_SFT 11
+#define RT5670_GP4_PIN_GPIO4 (0x0 << 11)
+#define RT5670_GP4_PIN_DMIC2_SDA (0x1 << 11)
+#define RT5670_DP_SIG_MASK (0x1 << 10)
+#define RT5670_DP_SIG_SFT 10
+#define RT5670_DP_SIG_TEST (0x0 << 10)
+#define RT5670_DP_SIG_AP (0x1 << 10)
+#define RT5670_GPIO_M_MASK (0x1 << 9)
+#define RT5670_GPIO_M_SFT 9
+#define RT5670_GPIO_M_FLT (0x0 << 9)
+#define RT5670_GPIO_M_PH (0x1 << 9)
+#define RT5670_I2S2_PIN_MASK (0x1 << 8)
+#define RT5670_I2S2_PIN_SFT 8
+#define RT5670_I2S2_PIN_I2S (0x0 << 8)
+#define RT5670_I2S2_PIN_GPIO (0x1 << 8)
+#define RT5670_GP5_PIN_MASK (0x1 << 7)
+#define RT5670_GP5_PIN_SFT 7
+#define RT5670_GP5_PIN_GPIO5 (0x0 << 7)
+#define RT5670_GP5_PIN_DMIC3_SDA (0x1 << 7)
+#define RT5670_GP6_PIN_MASK (0x1 << 6)
+#define RT5670_GP6_PIN_SFT 6
+#define RT5670_GP6_PIN_GPIO6 (0x0 << 6)
+#define RT5670_GP6_PIN_DMIC1_SDA (0x1 << 6)
+#define RT5670_GP7_PIN_MASK (0x3 << 4)
+#define RT5670_GP7_PIN_SFT 4
+#define RT5670_GP7_PIN_GPIO7 (0x0 << 4)
+#define RT5670_GP7_PIN_DMIC1_SDA (0x1 << 4)
+#define RT5670_GP7_PIN_PDM_SCL2 (0x2 << 4)
+#define RT5670_GP8_PIN_MASK (0x1 << 3)
+#define RT5670_GP8_PIN_SFT 3
+#define RT5670_GP8_PIN_GPIO8 (0x0 << 3)
+#define RT5670_GP8_PIN_DMIC2_SDA (0x1 << 3)
+#define RT5670_GP9_PIN_MASK (0x1 << 2)
+#define RT5670_GP9_PIN_SFT 2
+#define RT5670_GP9_PIN_GPIO9 (0x0 << 2)
+#define RT5670_GP9_PIN_DMIC3_SDA (0x1 << 2)
+#define RT5670_GP10_PIN_MASK (0x3)
+#define RT5670_GP10_PIN_SFT 0
+#define RT5670_GP10_PIN_GPIO9 (0x0)
+#define RT5670_GP10_PIN_DMIC3_SDA (0x1)
+#define RT5670_GP10_PIN_PDM_ADT2 (0x2)
+
+/* GPIO Control 2 (0xc1) */
+#define RT5670_GP4_PF_MASK (0x1 << 11)
+#define RT5670_GP4_PF_SFT 11
+#define RT5670_GP4_PF_IN (0x0 << 11)
+#define RT5670_GP4_PF_OUT (0x1 << 11)
+#define RT5670_GP4_OUT_MASK (0x1 << 10)
+#define RT5670_GP4_OUT_SFT 10
+#define RT5670_GP4_OUT_LO (0x0 << 10)
+#define RT5670_GP4_OUT_HI (0x1 << 10)
+#define RT5670_GP4_P_MASK (0x1 << 9)
+#define RT5670_GP4_P_SFT 9
+#define RT5670_GP4_P_NOR (0x0 << 9)
+#define RT5670_GP4_P_INV (0x1 << 9)
+#define RT5670_GP3_PF_MASK (0x1 << 8)
+#define RT5670_GP3_PF_SFT 8
+#define RT5670_GP3_PF_IN (0x0 << 8)
+#define RT5670_GP3_PF_OUT (0x1 << 8)
+#define RT5670_GP3_OUT_MASK (0x1 << 7)
+#define RT5670_GP3_OUT_SFT 7
+#define RT5670_GP3_OUT_LO (0x0 << 7)
+#define RT5670_GP3_OUT_HI (0x1 << 7)
+#define RT5670_GP3_P_MASK (0x1 << 6)
+#define RT5670_GP3_P_SFT 6
+#define RT5670_GP3_P_NOR (0x0 << 6)
+#define RT5670_GP3_P_INV (0x1 << 6)
+#define RT5670_GP2_PF_MASK (0x1 << 5)
+#define RT5670_GP2_PF_SFT 5
+#define RT5670_GP2_PF_IN (0x0 << 5)
+#define RT5670_GP2_PF_OUT (0x1 << 5)
+#define RT5670_GP2_OUT_MASK (0x1 << 4)
+#define RT5670_GP2_OUT_SFT 4
+#define RT5670_GP2_OUT_LO (0x0 << 4)
+#define RT5670_GP2_OUT_HI (0x1 << 4)
+#define RT5670_GP2_P_MASK (0x1 << 3)
+#define RT5670_GP2_P_SFT 3
+#define RT5670_GP2_P_NOR (0x0 << 3)
+#define RT5670_GP2_P_INV (0x1 << 3)
+#define RT5670_GP1_PF_MASK (0x1 << 2)
+#define RT5670_GP1_PF_SFT 2
+#define RT5670_GP1_PF_IN (0x0 << 2)
+#define RT5670_GP1_PF_OUT (0x1 << 2)
+#define RT5670_GP1_OUT_MASK (0x1 << 1)
+#define RT5670_GP1_OUT_SFT 1
+#define RT5670_GP1_OUT_LO (0x0 << 1)
+#define RT5670_GP1_OUT_HI (0x1 << 1)
+#define RT5670_GP1_P_MASK (0x1)
+#define RT5670_GP1_P_SFT 0
+#define RT5670_GP1_P_NOR (0x0)
+#define RT5670_GP1_P_INV (0x1)
+
+/* Scramble Function (0xcd) */
+#define RT5670_SCB_KEY_MASK (0xff)
+#define RT5670_SCB_KEY_SFT 0
+
+/* Scramble Control (0xce) */
+#define RT5670_SCB_SWAP_MASK (0x1 << 15)
+#define RT5670_SCB_SWAP_SFT 15
+#define RT5670_SCB_SWAP_DIS (0x0 << 15)
+#define RT5670_SCB_SWAP_EN (0x1 << 15)
+#define RT5670_SCB_MASK (0x1 << 14)
+#define RT5670_SCB_SFT 14
+#define RT5670_SCB_DIS (0x0 << 14)
+#define RT5670_SCB_EN (0x1 << 14)
+
+/* Baseback Control (0xcf) */
+#define RT5670_BB_MASK (0x1 << 15)
+#define RT5670_BB_SFT 15
+#define RT5670_BB_DIS (0x0 << 15)
+#define RT5670_BB_EN (0x1 << 15)
+#define RT5670_BB_CT_MASK (0x7 << 12)
+#define RT5670_BB_CT_SFT 12
+#define RT5670_BB_CT_A (0x0 << 12)
+#define RT5670_BB_CT_B (0x1 << 12)
+#define RT5670_BB_CT_C (0x2 << 12)
+#define RT5670_BB_CT_D (0x3 << 12)
+#define RT5670_M_BB_L_MASK (0x1 << 9)
+#define RT5670_M_BB_L_SFT 9
+#define RT5670_M_BB_R_MASK (0x1 << 8)
+#define RT5670_M_BB_R_SFT 8
+#define RT5670_M_BB_HPF_L_MASK (0x1 << 7)
+#define RT5670_M_BB_HPF_L_SFT 7
+#define RT5670_M_BB_HPF_R_MASK (0x1 << 6)
+#define RT5670_M_BB_HPF_R_SFT 6
+#define RT5670_G_BB_BST_MASK (0x3f)
+#define RT5670_G_BB_BST_SFT 0
+
+/* MP3 Plus Control 1 (0xd0) */
+#define RT5670_M_MP3_L_MASK (0x1 << 15)
+#define RT5670_M_MP3_L_SFT 15
+#define RT5670_M_MP3_R_MASK (0x1 << 14)
+#define RT5670_M_MP3_R_SFT 14
+#define RT5670_M_MP3_MASK (0x1 << 13)
+#define RT5670_M_MP3_SFT 13
+#define RT5670_M_MP3_DIS (0x0 << 13)
+#define RT5670_M_MP3_EN (0x1 << 13)
+#define RT5670_EG_MP3_MASK (0x1f << 8)
+#define RT5670_EG_MP3_SFT 8
+#define RT5670_MP3_HLP_MASK (0x1 << 7)
+#define RT5670_MP3_HLP_SFT 7
+#define RT5670_MP3_HLP_DIS (0x0 << 7)
+#define RT5670_MP3_HLP_EN (0x1 << 7)
+#define RT5670_M_MP3_ORG_L_MASK (0x1 << 6)
+#define RT5670_M_MP3_ORG_L_SFT 6
+#define RT5670_M_MP3_ORG_R_MASK (0x1 << 5)
+#define RT5670_M_MP3_ORG_R_SFT 5
+
+/* MP3 Plus Control 2 (0xd1) */
+#define RT5670_MP3_WT_MASK (0x1 << 13)
+#define RT5670_MP3_WT_SFT 13
+#define RT5670_MP3_WT_1_4 (0x0 << 13)
+#define RT5670_MP3_WT_1_2 (0x1 << 13)
+#define RT5670_OG_MP3_MASK (0x1f << 8)
+#define RT5670_OG_MP3_SFT 8
+#define RT5670_HG_MP3_MASK (0x3f)
+#define RT5670_HG_MP3_SFT 0
+
+/* 3D HP Control 1 (0xd2) */
+#define RT5670_3D_CF_MASK (0x1 << 15)
+#define RT5670_3D_CF_SFT 15
+#define RT5670_3D_CF_DIS (0x0 << 15)
+#define RT5670_3D_CF_EN (0x1 << 15)
+#define RT5670_3D_HP_MASK (0x1 << 14)
+#define RT5670_3D_HP_SFT 14
+#define RT5670_3D_HP_DIS (0x0 << 14)
+#define RT5670_3D_HP_EN (0x1 << 14)
+#define RT5670_3D_BT_MASK (0x1 << 13)
+#define RT5670_3D_BT_SFT 13
+#define RT5670_3D_BT_DIS (0x0 << 13)
+#define RT5670_3D_BT_EN (0x1 << 13)
+#define RT5670_3D_1F_MIX_MASK (0x3 << 11)
+#define RT5670_3D_1F_MIX_SFT 11
+#define RT5670_3D_HP_M_MASK (0x1 << 10)
+#define RT5670_3D_HP_M_SFT 10
+#define RT5670_3D_HP_M_SUR (0x0 << 10)
+#define RT5670_3D_HP_M_FRO (0x1 << 10)
+#define RT5670_M_3D_HRTF_MASK (0x1 << 9)
+#define RT5670_M_3D_HRTF_SFT 9
+#define RT5670_M_3D_D2H_MASK (0x1 << 8)
+#define RT5670_M_3D_D2H_SFT 8
+#define RT5670_M_3D_D2R_MASK (0x1 << 7)
+#define RT5670_M_3D_D2R_SFT 7
+#define RT5670_M_3D_REVB_MASK (0x1 << 6)
+#define RT5670_M_3D_REVB_SFT 6
+
+/* Adjustable high pass filter control 1 (0xd3) */
+#define RT5670_2ND_HPF_MASK (0x1 << 15)
+#define RT5670_2ND_HPF_SFT 15
+#define RT5670_2ND_HPF_DIS (0x0 << 15)
+#define RT5670_2ND_HPF_EN (0x1 << 15)
+#define RT5670_HPF_CF_L_MASK (0x7 << 12)
+#define RT5670_HPF_CF_L_SFT 12
+#define RT5670_1ST_HPF_MASK (0x1 << 11)
+#define RT5670_1ST_HPF_SFT 11
+#define RT5670_1ST_HPF_DIS (0x0 << 11)
+#define RT5670_1ST_HPF_EN (0x1 << 11)
+#define RT5670_HPF_CF_R_MASK (0x7 << 8)
+#define RT5670_HPF_CF_R_SFT 8
+#define RT5670_ZD_T_MASK (0x3 << 6)
+#define RT5670_ZD_T_SFT 6
+#define RT5670_ZD_F_MASK (0x3 << 4)
+#define RT5670_ZD_F_SFT 4
+#define RT5670_ZD_F_IM (0x0 << 4)
+#define RT5670_ZD_F_ZC_IM (0x1 << 4)
+#define RT5670_ZD_F_ZC_IOD (0x2 << 4)
+#define RT5670_ZD_F_UN (0x3 << 4)
+
+/* HP calibration control and Amp detection (0xd6) */
+#define RT5670_SI_DAC_MASK (0x1 << 11)
+#define RT5670_SI_DAC_SFT 11
+#define RT5670_SI_DAC_AUTO (0x0 << 11)
+#define RT5670_SI_DAC_TEST (0x1 << 11)
+#define RT5670_DC_CAL_M_MASK (0x1 << 10)
+#define RT5670_DC_CAL_M_SFT 10
+#define RT5670_DC_CAL_M_CAL (0x0 << 10)
+#define RT5670_DC_CAL_M_NOR (0x1 << 10)
+#define RT5670_DC_CAL_MASK (0x1 << 9)
+#define RT5670_DC_CAL_SFT 9
+#define RT5670_DC_CAL_DIS (0x0 << 9)
+#define RT5670_DC_CAL_EN (0x1 << 9)
+#define RT5670_HPD_RCV_MASK (0x7 << 6)
+#define RT5670_HPD_RCV_SFT 6
+#define RT5670_HPD_PS_MASK (0x1 << 5)
+#define RT5670_HPD_PS_SFT 5
+#define RT5670_HPD_PS_DIS (0x0 << 5)
+#define RT5670_HPD_PS_EN (0x1 << 5)
+#define RT5670_CAL_M_MASK (0x1 << 4)
+#define RT5670_CAL_M_SFT 4
+#define RT5670_CAL_M_DEP (0x0 << 4)
+#define RT5670_CAL_M_CAL (0x1 << 4)
+#define RT5670_CAL_MASK (0x1 << 3)
+#define RT5670_CAL_SFT 3
+#define RT5670_CAL_DIS (0x0 << 3)
+#define RT5670_CAL_EN (0x1 << 3)
+#define RT5670_CAL_TEST_MASK (0x1 << 2)
+#define RT5670_CAL_TEST_SFT 2
+#define RT5670_CAL_TEST_DIS (0x0 << 2)
+#define RT5670_CAL_TEST_EN (0x1 << 2)
+#define RT5670_CAL_P_MASK (0x3)
+#define RT5670_CAL_P_SFT 0
+#define RT5670_CAL_P_NONE (0x0)
+#define RT5670_CAL_P_CAL (0x1)
+#define RT5670_CAL_P_DAC_CAL (0x2)
+
+/* Soft volume and zero cross control 1 (0xd9) */
+#define RT5670_SV_MASK (0x1 << 15)
+#define RT5670_SV_SFT 15
+#define RT5670_SV_DIS (0x0 << 15)
+#define RT5670_SV_EN (0x1 << 15)
+#define RT5670_SPO_SV_MASK (0x1 << 14)
+#define RT5670_SPO_SV_SFT 14
+#define RT5670_SPO_SV_DIS (0x0 << 14)
+#define RT5670_SPO_SV_EN (0x1 << 14)
+#define RT5670_OUT_SV_MASK (0x1 << 13)
+#define RT5670_OUT_SV_SFT 13
+#define RT5670_OUT_SV_DIS (0x0 << 13)
+#define RT5670_OUT_SV_EN (0x1 << 13)
+#define RT5670_HP_SV_MASK (0x1 << 12)
+#define RT5670_HP_SV_SFT 12
+#define RT5670_HP_SV_DIS (0x0 << 12)
+#define RT5670_HP_SV_EN (0x1 << 12)
+#define RT5670_ZCD_DIG_MASK (0x1 << 11)
+#define RT5670_ZCD_DIG_SFT 11
+#define RT5670_ZCD_DIG_DIS (0x0 << 11)
+#define RT5670_ZCD_DIG_EN (0x1 << 11)
+#define RT5670_ZCD_MASK (0x1 << 10)
+#define RT5670_ZCD_SFT 10
+#define RT5670_ZCD_PD (0x0 << 10)
+#define RT5670_ZCD_PU (0x1 << 10)
+#define RT5670_M_ZCD_MASK (0x3f << 4)
+#define RT5670_M_ZCD_SFT 4
+#define RT5670_M_ZCD_RM_L (0x1 << 9)
+#define RT5670_M_ZCD_RM_R (0x1 << 8)
+#define RT5670_M_ZCD_SM_L (0x1 << 7)
+#define RT5670_M_ZCD_SM_R (0x1 << 6)
+#define RT5670_M_ZCD_OM_L (0x1 << 5)
+#define RT5670_M_ZCD_OM_R (0x1 << 4)
+#define RT5670_SV_DLY_MASK (0xf)
+#define RT5670_SV_DLY_SFT 0
+
+/* Soft volume and zero cross control 2 (0xda) */
+#define RT5670_ZCD_HP_MASK (0x1 << 15)
+#define RT5670_ZCD_HP_SFT 15
+#define RT5670_ZCD_HP_DIS (0x0 << 15)
+#define RT5670_ZCD_HP_EN (0x1 << 15)
+
+
+/* Codec Private Register definition */
+/* 3D Speaker Control (0x63) */
+#define RT5670_3D_SPK_MASK (0x1 << 15)
+#define RT5670_3D_SPK_SFT 15
+#define RT5670_3D_SPK_DIS (0x0 << 15)
+#define RT5670_3D_SPK_EN (0x1 << 15)
+#define RT5670_3D_SPK_M_MASK (0x3 << 13)
+#define RT5670_3D_SPK_M_SFT 13
+#define RT5670_3D_SPK_CG_MASK (0x1f << 8)
+#define RT5670_3D_SPK_CG_SFT 8
+#define RT5670_3D_SPK_SG_MASK (0x1f)
+#define RT5670_3D_SPK_SG_SFT 0
+
+/* Wind Noise Detection Control 1 (0x6c) */
+#define RT5670_WND_MASK (0x1 << 15)
+#define RT5670_WND_SFT 15
+#define RT5670_WND_DIS (0x0 << 15)
+#define RT5670_WND_EN (0x1 << 15)
+
+/* Wind Noise Detection Control 2 (0x6d) */
+#define RT5670_WND_FC_NW_MASK (0x3f << 10)
+#define RT5670_WND_FC_NW_SFT 10
+#define RT5670_WND_FC_WK_MASK (0x3f << 4)
+#define RT5670_WND_FC_WK_SFT 4
+
+/* Wind Noise Detection Control 3 (0x6e) */
+#define RT5670_HPF_FC_MASK (0x3f << 6)
+#define RT5670_HPF_FC_SFT 6
+#define RT5670_WND_FC_ST_MASK (0x3f)
+#define RT5670_WND_FC_ST_SFT 0
+
+/* Wind Noise Detection Control 4 (0x6f) */
+#define RT5670_WND_TH_LO_MASK (0x3ff)
+#define RT5670_WND_TH_LO_SFT 0
+
+/* Wind Noise Detection Control 5 (0x70) */
+#define RT5670_WND_TH_HI_MASK (0x3ff)
+#define RT5670_WND_TH_HI_SFT 0
+
+/* Wind Noise Detection Control 8 (0x73) */
+#define RT5670_WND_WIND_MASK (0x1 << 13) /* Read-Only */
+#define RT5670_WND_WIND_SFT 13
+#define RT5670_WND_STRONG_MASK (0x1 << 12) /* Read-Only */
+#define RT5670_WND_STRONG_SFT 12
+enum {
+ RT5670_NO_WIND,
+ RT5670_BREEZE,
+ RT5670_STORM,
+};
+
+/* Dipole Speaker Interface (0x75) */
+#define RT5670_DP_ATT_MASK (0x3 << 14)
+#define RT5670_DP_ATT_SFT 14
+#define RT5670_DP_SPK_MASK (0x1 << 10)
+#define RT5670_DP_SPK_SFT 10
+#define RT5670_DP_SPK_DIS (0x0 << 10)
+#define RT5670_DP_SPK_EN (0x1 << 10)
+
+/* EQ Pre Volume Control (0xb3) */
+#define RT5670_EQ_PRE_VOL_MASK (0xffff)
+#define RT5670_EQ_PRE_VOL_SFT 0
+
+/* EQ Post Volume Control (0xb4) */
+#define RT5670_EQ_PST_VOL_MASK (0xffff)
+#define RT5670_EQ_PST_VOL_SFT 0
+
+/* Jack Detect Control 3 (0xf8) */
+#define RT5670_CMP_MIC_IN_DET_MASK (0x7 << 12)
+#define RT5670_JD_CBJ_EN (0x1 << 7)
+#define RT5670_JD_CBJ_POL (0x1 << 6)
+#define RT5670_JD_TRI_CBJ_SEL_MASK (0x7 << 3)
+#define RT5670_JD_TRI_CBJ_SEL_SFT (3)
+#define RT5670_JD_CBJ_GPIO_JD1 (0x0 << 3)
+#define RT5670_JD_CBJ_JD1_1 (0x1 << 3)
+#define RT5670_JD_CBJ_JD1_2 (0x2 << 3)
+#define RT5670_JD_CBJ_JD2 (0x3 << 3)
+#define RT5670_JD_CBJ_JD3 (0x4 << 3)
+#define RT5670_JD_CBJ_GPIO_JD2 (0x5 << 3)
+#define RT5670_JD_CBJ_MX0B_12 (0x6 << 3)
+#define RT5670_JD_TRI_HPO_SEL_MASK (0x7 << 3)
+#define RT5670_JD_TRI_HPO_SEL_SFT (0)
+#define RT5670_JD_HPO_GPIO_JD1 (0x0)
+#define RT5670_JD_HPO_JD1_1 (0x1)
+#define RT5670_JD_HPO_JD1_2 (0x2)
+#define RT5670_JD_HPO_JD2 (0x3)
+#define RT5670_JD_HPO_JD3 (0x4)
+#define RT5670_JD_HPO_GPIO_JD2 (0x5)
+#define RT5670_JD_HPO_MX0B_12 (0x6)
+
+/* Digital Misc Control (0xfa) */
+#define RT5670_RST_DSP (0x1 << 13)
+#define RT5670_IF1_ADC1_IN1_SEL (0x1 << 12)
+#define RT5670_IF1_ADC1_IN1_SFT 12
+#define RT5670_IF1_ADC1_IN2_SEL (0x1 << 11)
+#define RT5670_IF1_ADC1_IN2_SFT 11
+#define RT5670_IF1_ADC2_IN1_SEL (0x1 << 10)
+#define RT5670_IF1_ADC2_IN1_SFT 10
+
+/* General Control2 (0xfb) */
+#define RT5670_RXDC_SRC_MASK (0x1 << 7)
+#define RT5670_RXDC_SRC_STO (0x0 << 7)
+#define RT5670_RXDC_SRC_MONO (0x1 << 7)
+#define RT5670_RXDC_SRC_SFT (7)
+#define RT5670_RXDP2_SEL_MASK (0x1 << 3)
+#define RT5670_RXDP2_SEL_IF2 (0x0 << 3)
+#define RT5670_RXDP2_SEL_ADC (0x1 << 3)
+#define RT5670_RXDP2_SEL_SFT (3)
+
+/* System Clock Source */
+enum {
+ RT5670_SCLK_S_MCLK,
+ RT5670_SCLK_S_PLL1,
+ RT5670_SCLK_S_RCCLK,
+};
+
+/* PLL1 Source */
+enum {
+ RT5670_PLL1_S_MCLK,
+ RT5670_PLL1_S_BCLK1,
+ RT5670_PLL1_S_BCLK2,
+ RT5670_PLL1_S_BCLK3,
+ RT5670_PLL1_S_BCLK4,
+};
+
+enum {
+ RT5670_AIF1,
+ RT5670_AIF2,
+ RT5670_AIF3,
+ RT5670_AIF4,
+ RT5670_AIFS,
+};
+
+enum {
+ RT5670_DMIC_DATA_GPIO6,
+ RT5670_DMIC_DATA_IN2P,
+ RT5670_DMIC_DATA_GPIO7,
+};
+
+enum {
+ RT5670_DMIC_DATA_GPIO8,
+ RT5670_DMIC_DATA_IN3N,
+};
+
+enum {
+ RT5670_DMIC_DATA_GPIO9,
+ RT5670_DMIC_DATA_GPIO10,
+ RT5670_DMIC_DATA_GPIO5,
+};
+
+struct rt5670_priv {
+ struct snd_soc_codec *codec;
+ struct rt5670_platform_data pdata;
+ struct regmap *regmap;
+
+ int sysclk;
+ int sysclk_src;
+ int lrck[RT5670_AIFS];
+ int bclk[RT5670_AIFS];
+ int master[RT5670_AIFS];
+
+ int pll_src;
+ int pll_in;
+ int pll_out;
+
+ int dsp_sw; /* expected parameter setting */
+ int dsp_rate;
+ int jack_type;
+};
+
+#endif /* __RT5670_H__ */
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 833231e27340..67f14556462f 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -27,6 +27,7 @@
#include <sound/initval.h>
#include <sound/tlv.h>
+#include "rl6231.h"
#include "rt5677.h"
#define RT5677_DEVICE_ID 0x6327
@@ -604,19 +605,19 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = {
adc_vol_tlv),
/* ADC Boost Volume Control */
- SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5677_STO1_2_ADC_BST,
+ SOC_DOUBLE_TLV("STO1 ADC Boost Volume", RT5677_STO1_2_ADC_BST,
RT5677_STO1_ADC_L_BST_SFT, RT5677_STO1_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
- SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5677_STO1_2_ADC_BST,
+ SOC_DOUBLE_TLV("STO2 ADC Boost Volume", RT5677_STO1_2_ADC_BST,
RT5677_STO2_ADC_L_BST_SFT, RT5677_STO2_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
- SOC_DOUBLE_TLV("STO3 ADC Boost Gain", RT5677_STO3_4_ADC_BST,
+ SOC_DOUBLE_TLV("STO3 ADC Boost Volume", RT5677_STO3_4_ADC_BST,
RT5677_STO3_ADC_L_BST_SFT, RT5677_STO3_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
- SOC_DOUBLE_TLV("STO4 ADC Boost Gain", RT5677_STO3_4_ADC_BST,
+ SOC_DOUBLE_TLV("STO4 ADC Boost Volume", RT5677_STO3_4_ADC_BST,
RT5677_STO4_ADC_L_BST_SFT, RT5677_STO4_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
- SOC_DOUBLE_TLV("Mono ADC Boost Gain", RT5677_ADC_BST_CTRL2,
+ SOC_DOUBLE_TLV("Mono ADC Boost Volume", RT5677_ADC_BST_CTRL2,
RT5677_MONO_ADC_L_BST_SFT, RT5677_MONO_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
};
@@ -636,21 +637,7 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w,
{
struct snd_soc_codec *codec = w->codec;
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
- int div[] = {2, 3, 4, 6, 8, 12}, idx = -EINVAL, i;
- int rate, red, bound, temp;
-
- rate = rt5677->sysclk;
- red = 3000000 * 12;
- for (i = 0; i < ARRAY_SIZE(div); i++) {
- bound = div[i] * 3000000;
- if (rate > bound)
- continue;
- temp = bound - rate;
- if (temp < red) {
- red = temp;
- idx = i;
- }
- }
+ int idx = rl6231_calc_dmic_clk(rt5677->sysclk);
if (idx < 0)
dev_err(codec->dev, "Failed to set DMIC clock\n");
@@ -951,7 +938,7 @@ static const struct snd_kcontrol_new rt5677_ob_7_mix[] = {
/* Mux */
-/* DAC1 L/R source */ /* MX-29 [10:8] */
+/* DAC1 L/R Source */ /* MX-29 [10:8] */
static const char * const rt5677_dac1_src[] = {
"IF1 DAC 01", "IF2 DAC 01", "IF3 DAC LR", "IF4 DAC LR", "SLB DAC 01",
"OB 01"
@@ -962,9 +949,9 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_DAC1_L_SEL_SFT, rt5677_dac1_src);
static const struct snd_kcontrol_new rt5677_dac1_mux =
- SOC_DAPM_ENUM("DAC1 source", rt5677_dac1_enum);
+ SOC_DAPM_ENUM("DAC1 Source", rt5677_dac1_enum);
-/* ADDA1 L/R source */ /* MX-29 [1:0] */
+/* ADDA1 L/R Source */ /* MX-29 [1:0] */
static const char * const rt5677_adda1_src[] = {
"STO1 ADC MIX", "STO2 ADC MIX", "OB 67",
};
@@ -974,10 +961,10 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_ADDA1_SEL_SFT, rt5677_adda1_src);
static const struct snd_kcontrol_new rt5677_adda1_mux =
- SOC_DAPM_ENUM("ADDA1 source", rt5677_adda1_enum);
+ SOC_DAPM_ENUM("ADDA1 Source", rt5677_adda1_enum);
-/*DAC2 L/R source*/ /* MX-1B [6:4] [2:0] */
+/*DAC2 L/R Source*/ /* MX-1B [6:4] [2:0] */
static const char * const rt5677_dac2l_src[] = {
"IF1 DAC 2", "IF2 DAC 2", "IF3 DAC L", "IF4 DAC L", "SLB DAC 2",
"OB 2",
@@ -988,7 +975,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_DAC2_L_SRC_SFT, rt5677_dac2l_src);
static const struct snd_kcontrol_new rt5677_dac2_l_mux =
- SOC_DAPM_ENUM("DAC2 L source", rt5677_dac2l_enum);
+ SOC_DAPM_ENUM("DAC2 L Source", rt5677_dac2l_enum);
static const char * const rt5677_dac2r_src[] = {
"IF1 DAC 3", "IF2 DAC 3", "IF3 DAC R", "IF4 DAC R", "SLB DAC 3",
@@ -1000,9 +987,9 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_DAC2_R_SRC_SFT, rt5677_dac2r_src);
static const struct snd_kcontrol_new rt5677_dac2_r_mux =
- SOC_DAPM_ENUM("DAC2 R source", rt5677_dac2r_enum);
+ SOC_DAPM_ENUM("DAC2 R Source", rt5677_dac2r_enum);
-/*DAC3 L/R source*/ /* MX-16 [6:4] [2:0] */
+/*DAC3 L/R Source*/ /* MX-16 [6:4] [2:0] */
static const char * const rt5677_dac3l_src[] = {
"IF1 DAC 4", "IF2 DAC 4", "IF3 DAC L", "IF4 DAC L",
"SLB DAC 4", "OB 4"
@@ -1013,7 +1000,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_DAC3_L_SRC_SFT, rt5677_dac3l_src);
static const struct snd_kcontrol_new rt5677_dac3_l_mux =
- SOC_DAPM_ENUM("DAC3 L source", rt5677_dac3l_enum);
+ SOC_DAPM_ENUM("DAC3 L Source", rt5677_dac3l_enum);
static const char * const rt5677_dac3r_src[] = {
"IF1 DAC 5", "IF2 DAC 5", "IF3 DAC R", "IF4 DAC R",
@@ -1025,9 +1012,9 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_DAC3_R_SRC_SFT, rt5677_dac3r_src);
static const struct snd_kcontrol_new rt5677_dac3_r_mux =
- SOC_DAPM_ENUM("DAC3 R source", rt5677_dac3r_enum);
+ SOC_DAPM_ENUM("DAC3 R Source", rt5677_dac3r_enum);
-/*DAC4 L/R source*/ /* MX-16 [14:12] [10:8] */
+/*DAC4 L/R Source*/ /* MX-16 [14:12] [10:8] */
static const char * const rt5677_dac4l_src[] = {
"IF1 DAC 6", "IF2 DAC 6", "IF3 DAC L", "IF4 DAC L",
"SLB DAC 6", "OB 6"
@@ -1038,7 +1025,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_DAC4_L_SRC_SFT, rt5677_dac4l_src);
static const struct snd_kcontrol_new rt5677_dac4_l_mux =
- SOC_DAPM_ENUM("DAC4 L source", rt5677_dac4l_enum);
+ SOC_DAPM_ENUM("DAC4 L Source", rt5677_dac4l_enum);
static const char * const rt5677_dac4r_src[] = {
"IF1 DAC 7", "IF2 DAC 7", "IF3 DAC R", "IF4 DAC R",
@@ -1050,7 +1037,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_DAC4_R_SRC_SFT, rt5677_dac4r_src);
static const struct snd_kcontrol_new rt5677_dac4_r_mux =
- SOC_DAPM_ENUM("DAC4 R source", rt5677_dac4r_enum);
+ SOC_DAPM_ENUM("DAC4 R Source", rt5677_dac4r_enum);
/* In/OutBound Source Pass SRC */ /* MX-A5 [3] [4] [0] [1] [2] */
static const char * const rt5677_iob_bypass_src[] = {
@@ -1062,35 +1049,35 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_SRC_OB01_SFT, rt5677_iob_bypass_src);
static const struct snd_kcontrol_new rt5677_ob01_bypass_src_mux =
- SOC_DAPM_ENUM("OB01 Bypass source", rt5677_ob01_bypass_src_enum);
+ SOC_DAPM_ENUM("OB01 Bypass Source", rt5677_ob01_bypass_src_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_ob23_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL,
RT5677_SEL_SRC_OB23_SFT, rt5677_iob_bypass_src);
static const struct snd_kcontrol_new rt5677_ob23_bypass_src_mux =
- SOC_DAPM_ENUM("OB23 Bypass source", rt5677_ob23_bypass_src_enum);
+ SOC_DAPM_ENUM("OB23 Bypass Source", rt5677_ob23_bypass_src_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_ib01_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL,
RT5677_SEL_SRC_IB01_SFT, rt5677_iob_bypass_src);
static const struct snd_kcontrol_new rt5677_ib01_bypass_src_mux =
- SOC_DAPM_ENUM("IB01 Bypass source", rt5677_ib01_bypass_src_enum);
+ SOC_DAPM_ENUM("IB01 Bypass Source", rt5677_ib01_bypass_src_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_ib23_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL,
RT5677_SEL_SRC_IB23_SFT, rt5677_iob_bypass_src);
static const struct snd_kcontrol_new rt5677_ib23_bypass_src_mux =
- SOC_DAPM_ENUM("IB23 Bypass source", rt5677_ib23_bypass_src_enum);
+ SOC_DAPM_ENUM("IB23 Bypass Source", rt5677_ib23_bypass_src_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_ib45_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL,
RT5677_SEL_SRC_IB45_SFT, rt5677_iob_bypass_src);
static const struct snd_kcontrol_new rt5677_ib45_bypass_src_mux =
- SOC_DAPM_ENUM("IB45 Bypass source", rt5677_ib45_bypass_src_enum);
+ SOC_DAPM_ENUM("IB45 Bypass Source", rt5677_ib45_bypass_src_enum);
/* Stereo ADC Source 2 */ /* MX-27 MX26 MX25 [11:10] */
static const char * const rt5677_stereo_adc2_src[] = {
@@ -1102,21 +1089,21 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_STO1_ADC2_SFT, rt5677_stereo_adc2_src);
static const struct snd_kcontrol_new rt5677_sto1_adc2_mux =
- SOC_DAPM_ENUM("Stereo1 ADC2 source", rt5677_stereo1_adc2_enum);
+ SOC_DAPM_ENUM("Stereo1 ADC2 Source", rt5677_stereo1_adc2_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo2_adc2_enum, RT5677_STO2_ADC_MIXER,
RT5677_SEL_STO2_ADC2_SFT, rt5677_stereo_adc2_src);
static const struct snd_kcontrol_new rt5677_sto2_adc2_mux =
- SOC_DAPM_ENUM("Stereo2 ADC2 source", rt5677_stereo2_adc2_enum);
+ SOC_DAPM_ENUM("Stereo2 ADC2 Source", rt5677_stereo2_adc2_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo3_adc2_enum, RT5677_STO3_ADC_MIXER,
RT5677_SEL_STO3_ADC2_SFT, rt5677_stereo_adc2_src);
static const struct snd_kcontrol_new rt5677_sto3_adc2_mux =
- SOC_DAPM_ENUM("Stereo3 ADC2 source", rt5677_stereo3_adc2_enum);
+ SOC_DAPM_ENUM("Stereo3 ADC2 Source", rt5677_stereo3_adc2_enum);
/* DMIC Source */ /* MX-28 [9:8][1:0] MX-27 MX-26 MX-25 MX-24 [9:8] */
static const char * const rt5677_dmic_src[] = {
@@ -1128,44 +1115,44 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_MONO_DMIC_L_SFT, rt5677_dmic_src);
static const struct snd_kcontrol_new rt5677_mono_dmic_l_mux =
- SOC_DAPM_ENUM("Mono DMIC L source", rt5677_mono_dmic_l_enum);
+ SOC_DAPM_ENUM("Mono DMIC L Source", rt5677_mono_dmic_l_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_mono_dmic_r_enum, RT5677_MONO_ADC_MIXER,
RT5677_SEL_MONO_DMIC_R_SFT, rt5677_dmic_src);
static const struct snd_kcontrol_new rt5677_mono_dmic_r_mux =
- SOC_DAPM_ENUM("Mono DMIC R source", rt5677_mono_dmic_r_enum);
+ SOC_DAPM_ENUM("Mono DMIC R Source", rt5677_mono_dmic_r_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo1_dmic_enum, RT5677_STO1_ADC_MIXER,
RT5677_SEL_STO1_DMIC_SFT, rt5677_dmic_src);
static const struct snd_kcontrol_new rt5677_sto1_dmic_mux =
- SOC_DAPM_ENUM("Stereo1 DMIC source", rt5677_stereo1_dmic_enum);
+ SOC_DAPM_ENUM("Stereo1 DMIC Source", rt5677_stereo1_dmic_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo2_dmic_enum, RT5677_STO2_ADC_MIXER,
RT5677_SEL_STO2_DMIC_SFT, rt5677_dmic_src);
static const struct snd_kcontrol_new rt5677_sto2_dmic_mux =
- SOC_DAPM_ENUM("Stereo2 DMIC source", rt5677_stereo2_dmic_enum);
+ SOC_DAPM_ENUM("Stereo2 DMIC Source", rt5677_stereo2_dmic_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo3_dmic_enum, RT5677_STO3_ADC_MIXER,
RT5677_SEL_STO3_DMIC_SFT, rt5677_dmic_src);
static const struct snd_kcontrol_new rt5677_sto3_dmic_mux =
- SOC_DAPM_ENUM("Stereo3 DMIC source", rt5677_stereo3_dmic_enum);
+ SOC_DAPM_ENUM("Stereo3 DMIC Source", rt5677_stereo3_dmic_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo4_dmic_enum, RT5677_STO4_ADC_MIXER,
RT5677_SEL_STO4_DMIC_SFT, rt5677_dmic_src);
static const struct snd_kcontrol_new rt5677_sto4_dmic_mux =
- SOC_DAPM_ENUM("Stereo4 DMIC source", rt5677_stereo4_dmic_enum);
+ SOC_DAPM_ENUM("Stereo4 DMIC Source", rt5677_stereo4_dmic_enum);
-/* Stereo2 ADC source */ /* MX-26 [0] */
+/* Stereo2 ADC Source */ /* MX-26 [0] */
static const char * const rt5677_stereo2_adc_lr_src[] = {
"L", "LR"
};
@@ -1175,7 +1162,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_STO2_LR_MIX_SFT, rt5677_stereo2_adc_lr_src);
static const struct snd_kcontrol_new rt5677_sto2_adc_lr_mux =
- SOC_DAPM_ENUM("Stereo2 ADC LR source", rt5677_stereo2_adc_lr_enum);
+ SOC_DAPM_ENUM("Stereo2 ADC LR Source", rt5677_stereo2_adc_lr_enum);
/* Stereo1 ADC Source 1 */ /* MX-27 MX26 MX25 [13:12] */
static const char * const rt5677_stereo_adc1_src[] = {
@@ -1187,23 +1174,23 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_STO1_ADC1_SFT, rt5677_stereo_adc1_src);
static const struct snd_kcontrol_new rt5677_sto1_adc1_mux =
- SOC_DAPM_ENUM("Stereo1 ADC1 source", rt5677_stereo1_adc1_enum);
+ SOC_DAPM_ENUM("Stereo1 ADC1 Source", rt5677_stereo1_adc1_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo2_adc1_enum, RT5677_STO2_ADC_MIXER,
RT5677_SEL_STO2_ADC1_SFT, rt5677_stereo_adc1_src);
static const struct snd_kcontrol_new rt5677_sto2_adc1_mux =
- SOC_DAPM_ENUM("Stereo2 ADC1 source", rt5677_stereo2_adc1_enum);
+ SOC_DAPM_ENUM("Stereo2 ADC1 Source", rt5677_stereo2_adc1_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo3_adc1_enum, RT5677_STO3_ADC_MIXER,
RT5677_SEL_STO3_ADC1_SFT, rt5677_stereo_adc1_src);
static const struct snd_kcontrol_new rt5677_sto3_adc1_mux =
- SOC_DAPM_ENUM("Stereo3 ADC1 source", rt5677_stereo3_adc1_enum);
+ SOC_DAPM_ENUM("Stereo3 ADC1 Source", rt5677_stereo3_adc1_enum);
-/* Mono ADC Left source 2 */ /* MX-28 [11:10] */
+/* Mono ADC Left Source 2 */ /* MX-28 [11:10] */
static const char * const rt5677_mono_adc2_l_src[] = {
"DD MIX1L", "DMIC", "MONO DAC MIXL"
};
@@ -1213,9 +1200,9 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_MONO_ADC_L2_SFT, rt5677_mono_adc2_l_src);
static const struct snd_kcontrol_new rt5677_mono_adc2_l_mux =
- SOC_DAPM_ENUM("Mono ADC2 L source", rt5677_mono_adc2_l_enum);
+ SOC_DAPM_ENUM("Mono ADC2 L Source", rt5677_mono_adc2_l_enum);
-/* Mono ADC Left source 1 */ /* MX-28 [13:12] */
+/* Mono ADC Left Source 1 */ /* MX-28 [13:12] */
static const char * const rt5677_mono_adc1_l_src[] = {
"DD MIX1L", "ADC1", "MONO DAC MIXL"
};
@@ -1225,9 +1212,9 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_MONO_ADC_L1_SFT, rt5677_mono_adc1_l_src);
static const struct snd_kcontrol_new rt5677_mono_adc1_l_mux =
- SOC_DAPM_ENUM("Mono ADC1 L source", rt5677_mono_adc1_l_enum);
+ SOC_DAPM_ENUM("Mono ADC1 L Source", rt5677_mono_adc1_l_enum);
-/* Mono ADC Right source 2 */ /* MX-28 [3:2] */
+/* Mono ADC Right Source 2 */ /* MX-28 [3:2] */
static const char * const rt5677_mono_adc2_r_src[] = {
"DD MIX1R", "DMIC", "MONO DAC MIXR"
};
@@ -1237,9 +1224,9 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_MONO_ADC_R2_SFT, rt5677_mono_adc2_r_src);
static const struct snd_kcontrol_new rt5677_mono_adc2_r_mux =
- SOC_DAPM_ENUM("Mono ADC2 R source", rt5677_mono_adc2_r_enum);
+ SOC_DAPM_ENUM("Mono ADC2 R Source", rt5677_mono_adc2_r_enum);
-/* Mono ADC Right source 1 */ /* MX-28 [5:4] */
+/* Mono ADC Right Source 1 */ /* MX-28 [5:4] */
static const char * const rt5677_mono_adc1_r_src[] = {
"DD MIX1R", "ADC2", "MONO DAC MIXR"
};
@@ -1249,7 +1236,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_MONO_ADC_R1_SFT, rt5677_mono_adc1_r_src);
static const struct snd_kcontrol_new rt5677_mono_adc1_r_mux =
- SOC_DAPM_ENUM("Mono ADC1 R source", rt5677_mono_adc1_r_enum);
+ SOC_DAPM_ENUM("Mono ADC1 R Source", rt5677_mono_adc1_r_enum);
/* Stereo4 ADC Source 2 */ /* MX-24 [11:10] */
static const char * const rt5677_stereo4_adc2_src[] = {
@@ -1261,7 +1248,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_STO4_ADC2_SFT, rt5677_stereo4_adc2_src);
static const struct snd_kcontrol_new rt5677_sto4_adc2_mux =
- SOC_DAPM_ENUM("Stereo4 ADC2 source", rt5677_stereo4_adc2_enum);
+ SOC_DAPM_ENUM("Stereo4 ADC2 Source", rt5677_stereo4_adc2_enum);
/* Stereo4 ADC Source 1 */ /* MX-24 [13:12] */
@@ -1274,7 +1261,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_STO4_ADC1_SFT, rt5677_stereo4_adc1_src);
static const struct snd_kcontrol_new rt5677_sto4_adc1_mux =
- SOC_DAPM_ENUM("Stereo4 ADC1 source", rt5677_stereo4_adc1_enum);
+ SOC_DAPM_ENUM("Stereo4 ADC1 Source", rt5677_stereo4_adc1_enum);
/* InBound0/1 Source */ /* MX-A3 [14:12] */
static const char * const rt5677_inbound01_src[] = {
@@ -1416,7 +1403,7 @@ static SOC_ENUM_SINGLE_DECL(
static const struct snd_kcontrol_new rt5677_dac3_mux =
SOC_DAPM_ENUM("Analog DAC3 Source", rt5677_dac3_enum);
-/* PDM channel source */ /* MX-31 [13:12][9:8][5:4][1:0] */
+/* PDM channel Source */ /* MX-31 [13:12][9:8][5:4][1:0] */
static const char * const rt5677_pdm_src[] = {
"STO1 DAC MIX", "MONO DAC MIX", "DD MIX1", "DD MIX2"
};
@@ -1426,28 +1413,28 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_PDM1_L_SFT, rt5677_pdm_src);
static const struct snd_kcontrol_new rt5677_pdm1_l_mux =
- SOC_DAPM_ENUM("PDM1 source", rt5677_pdm1_l_enum);
+ SOC_DAPM_ENUM("PDM1 Source", rt5677_pdm1_l_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_pdm2_l_enum, RT5677_PDM_OUT_CTRL,
RT5677_SEL_PDM2_L_SFT, rt5677_pdm_src);
static const struct snd_kcontrol_new rt5677_pdm2_l_mux =
- SOC_DAPM_ENUM("PDM2 source", rt5677_pdm2_l_enum);
+ SOC_DAPM_ENUM("PDM2 Source", rt5677_pdm2_l_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_pdm1_r_enum, RT5677_PDM_OUT_CTRL,
RT5677_SEL_PDM1_R_SFT, rt5677_pdm_src);
static const struct snd_kcontrol_new rt5677_pdm1_r_mux =
- SOC_DAPM_ENUM("PDM1 source", rt5677_pdm1_r_enum);
+ SOC_DAPM_ENUM("PDM1 Source", rt5677_pdm1_r_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_pdm2_r_enum, RT5677_PDM_OUT_CTRL,
RT5677_SEL_PDM2_R_SFT, rt5677_pdm_src);
static const struct snd_kcontrol_new rt5677_pdm2_r_mux =
- SOC_DAPM_ENUM("PDM2 source", rt5677_pdm2_r_enum);
+ SOC_DAPM_ENUM("PDM2 Source", rt5677_pdm2_r_enum);
/* TDM IF1/2 SLB ADC1 Data Selection */ /* MX-3C MX-41 [5:4] MX-08 [1:0]*/
static const char * const rt5677_if12_adc1_src[] = {
@@ -1459,21 +1446,21 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_IF1_ADC1_SFT, rt5677_if12_adc1_src);
static const struct snd_kcontrol_new rt5677_if1_adc1_mux =
- SOC_DAPM_ENUM("IF1 ADC1 source", rt5677_if1_adc1_enum);
+ SOC_DAPM_ENUM("IF1 ADC1 Source", rt5677_if1_adc1_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_if2_adc1_enum, RT5677_TDM2_CTRL2,
RT5677_IF2_ADC1_SFT, rt5677_if12_adc1_src);
static const struct snd_kcontrol_new rt5677_if2_adc1_mux =
- SOC_DAPM_ENUM("IF2 ADC1 source", rt5677_if2_adc1_enum);
+ SOC_DAPM_ENUM("IF2 ADC1 Source", rt5677_if2_adc1_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_slb_adc1_enum, RT5677_SLIMBUS_RX,
RT5677_SLB_ADC1_SFT, rt5677_if12_adc1_src);
static const struct snd_kcontrol_new rt5677_slb_adc1_mux =
- SOC_DAPM_ENUM("SLB ADC1 source", rt5677_slb_adc1_enum);
+ SOC_DAPM_ENUM("SLB ADC1 Source", rt5677_slb_adc1_enum);
/* TDM IF1/2 SLB ADC2 Data Selection */ /* MX-3C MX-41 [7:6] MX-08 [3:2] */
static const char * const rt5677_if12_adc2_src[] = {
@@ -1485,21 +1472,21 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_IF1_ADC2_SFT, rt5677_if12_adc2_src);
static const struct snd_kcontrol_new rt5677_if1_adc2_mux =
- SOC_DAPM_ENUM("IF1 ADC2 source", rt5677_if1_adc2_enum);
+ SOC_DAPM_ENUM("IF1 ADC2 Source", rt5677_if1_adc2_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_if2_adc2_enum, RT5677_TDM2_CTRL2,
RT5677_IF2_ADC2_SFT, rt5677_if12_adc2_src);
static const struct snd_kcontrol_new rt5677_if2_adc2_mux =
- SOC_DAPM_ENUM("IF2 ADC2 source", rt5677_if2_adc2_enum);
+ SOC_DAPM_ENUM("IF2 ADC2 Source", rt5677_if2_adc2_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_slb_adc2_enum, RT5677_SLIMBUS_RX,
RT5677_SLB_ADC2_SFT, rt5677_if12_adc2_src);
static const struct snd_kcontrol_new rt5677_slb_adc2_mux =
- SOC_DAPM_ENUM("SLB ADC2 source", rt5677_slb_adc2_enum);
+ SOC_DAPM_ENUM("SLB ADC2 Source", rt5677_slb_adc2_enum);
/* TDM IF1/2 SLB ADC3 Data Selection */ /* MX-3C MX-41 [9:8] MX-08 [5:4] */
static const char * const rt5677_if12_adc3_src[] = {
@@ -1511,21 +1498,21 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_IF1_ADC3_SFT, rt5677_if12_adc3_src);
static const struct snd_kcontrol_new rt5677_if1_adc3_mux =
- SOC_DAPM_ENUM("IF1 ADC3 source", rt5677_if1_adc3_enum);
+ SOC_DAPM_ENUM("IF1 ADC3 Source", rt5677_if1_adc3_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_if2_adc3_enum, RT5677_TDM2_CTRL2,
RT5677_IF2_ADC3_SFT, rt5677_if12_adc3_src);
static const struct snd_kcontrol_new rt5677_if2_adc3_mux =
- SOC_DAPM_ENUM("IF2 ADC3 source", rt5677_if2_adc3_enum);
+ SOC_DAPM_ENUM("IF2 ADC3 Source", rt5677_if2_adc3_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_slb_adc3_enum, RT5677_SLIMBUS_RX,
RT5677_SLB_ADC3_SFT, rt5677_if12_adc3_src);
static const struct snd_kcontrol_new rt5677_slb_adc3_mux =
- SOC_DAPM_ENUM("SLB ADC3 source", rt5677_slb_adc3_enum);
+ SOC_DAPM_ENUM("SLB ADC3 Source", rt5677_slb_adc3_enum);
/* TDM IF1/2 SLB ADC4 Data Selection */ /* MX-3C MX-41 [11:10] MX-08 [7:6] */
static const char * const rt5677_if12_adc4_src[] = {
@@ -1537,21 +1524,21 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_IF1_ADC4_SFT, rt5677_if12_adc4_src);
static const struct snd_kcontrol_new rt5677_if1_adc4_mux =
- SOC_DAPM_ENUM("IF1 ADC4 source", rt5677_if1_adc4_enum);
+ SOC_DAPM_ENUM("IF1 ADC4 Source", rt5677_if1_adc4_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_if2_adc4_enum, RT5677_TDM2_CTRL2,
RT5677_IF2_ADC4_SFT, rt5677_if12_adc4_src);
static const struct snd_kcontrol_new rt5677_if2_adc4_mux =
- SOC_DAPM_ENUM("IF2 ADC4 source", rt5677_if2_adc4_enum);
+ SOC_DAPM_ENUM("IF2 ADC4 Source", rt5677_if2_adc4_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_slb_adc4_enum, RT5677_SLIMBUS_RX,
RT5677_SLB_ADC4_SFT, rt5677_if12_adc4_src);
static const struct snd_kcontrol_new rt5677_slb_adc4_mux =
- SOC_DAPM_ENUM("SLB ADC4 source", rt5677_slb_adc4_enum);
+ SOC_DAPM_ENUM("SLB ADC4 Source", rt5677_slb_adc4_enum);
/* Interface3/4 ADC Data Input */ /* MX-2F [3:0] MX-30 [7:4]*/
static const char * const rt5677_if34_adc_src[] = {
@@ -1564,14 +1551,14 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_IF3_ADC_IN_SFT, rt5677_if34_adc_src);
static const struct snd_kcontrol_new rt5677_if3_adc_mux =
- SOC_DAPM_ENUM("IF3 ADC source", rt5677_if3_adc_enum);
+ SOC_DAPM_ENUM("IF3 ADC Source", rt5677_if3_adc_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_if4_adc_enum, RT5677_IF4_DATA,
RT5677_IF4_ADC_IN_SFT, rt5677_if34_adc_src);
static const struct snd_kcontrol_new rt5677_if4_adc_mux =
- SOC_DAPM_ENUM("IF4 ADC source", rt5677_if4_adc_enum);
+ SOC_DAPM_ENUM("IF4 ADC Source", rt5677_if4_adc_enum);
static int rt5677_bst1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
@@ -1670,6 +1657,13 @@ static int rt5677_set_micbias1_event(struct snd_soc_dapm_widget *w,
RT5677_PWR_CLK_MB, RT5677_PWR_CLK_MB1 |
RT5677_PWR_PP_MB1 | RT5677_PWR_CLK_MB);
break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2,
+ RT5677_PWR_CLK_MB1 | RT5677_PWR_PP_MB1 |
+ RT5677_PWR_CLK_MB, 0);
+ break;
+
default:
return 0;
}
@@ -1685,8 +1679,9 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
/* Input Side */
/* micbias */
- SND_SOC_DAPM_SUPPLY("micbias1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT,
- 0, rt5677_set_micbias1_event, SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT,
+ 0, rt5677_set_micbias1_event, SND_SOC_DAPM_PRE_PMD |
+ SND_SOC_DAPM_POST_PMU),
/* Input Lines */
SND_SOC_DAPM_INPUT("DMIC L1"),
@@ -2798,21 +2793,6 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "PDM2R", NULL, "PDM2 R Mux" },
};
-static int get_clk_info(int sclk, int rate)
-{
- int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16};
-
- if (sclk <= 0 || rate <= 0)
- return -EINVAL;
-
- rate = rate << 8;
- for (i = 0; i < ARRAY_SIZE(pd); i++)
- if (sclk == rate * pd[i])
- return i;
-
- return -EINVAL;
-}
-
static int rt5677_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
@@ -2822,7 +2802,7 @@ static int rt5677_hw_params(struct snd_pcm_substream *substream,
int pre_div, bclk_ms, frame_size;
rt5677->lrck[dai->id] = params_rate(params);
- pre_div = get_clk_info(rt5677->sysclk, rt5677->lrck[dai->id]);
+ pre_div = rl6231_get_clk_info(rt5677->sysclk, rt5677->lrck[dai->id]);
if (pre_div < 0) {
dev_err(codec->dev, "Unsupported clock setting\n");
return -EINVAL;
@@ -3016,62 +2996,12 @@ static int rt5677_set_dai_sysclk(struct snd_soc_dai *dai,
* Returns 0 for success or negative error code.
*/
static int rt5677_pll_calc(const unsigned int freq_in,
- const unsigned int freq_out, struct rt5677_pll_code *pll_code)
+ const unsigned int freq_out, struct rl6231_pll_code *pll_code)
{
- int max_n = RT5677_PLL_N_MAX, max_m = RT5677_PLL_M_MAX;
- int k, red, n_t, pll_out, in_t;
- int n = 0, m = 0, m_t = 0;
- int out_t, red_t = abs(freq_out - freq_in);
- bool m_bp = false, k_bp = false;
-
- if (RT5677_PLL_INP_MAX < freq_in || RT5677_PLL_INP_MIN > freq_in)
+ if (RT5677_PLL_INP_MIN > freq_in)
return -EINVAL;
- k = 100000000 / freq_out - 2;
- if (k > RT5677_PLL_K_MAX)
- k = RT5677_PLL_K_MAX;
- for (n_t = 0; n_t <= max_n; n_t++) {
- in_t = freq_in / (k + 2);
- pll_out = freq_out / (n_t + 2);
- if (in_t < 0)
- continue;
- if (in_t == pll_out) {
- m_bp = true;
- n = n_t;
- goto code_find;
- }
- red = abs(in_t - pll_out);
- if (red < red_t) {
- m_bp = true;
- n = n_t;
- m = m_t;
- if (red == 0)
- goto code_find;
- red_t = red;
- }
- for (m_t = 0; m_t <= max_m; m_t++) {
- out_t = in_t / (m_t + 2);
- red = abs(out_t - pll_out);
- if (red < red_t) {
- m_bp = false;
- n = n_t;
- m = m_t;
- if (red == 0)
- goto code_find;
- red_t = red;
- }
- }
- }
- pr_debug("Only get approximation about PLL\n");
-
-code_find:
-
- pll_code->m_bp = m_bp;
- pll_code->k_bp = k_bp;
- pll_code->m_code = m;
- pll_code->n_code = n;
- pll_code->k_code = k;
- return 0;
+ return rl6231_pll_calc(freq_in, freq_out, pll_code);
}
static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
@@ -3079,7 +3009,7 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
{
struct snd_soc_codec *codec = dai->codec;
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
- struct rt5677_pll_code pll_code;
+ struct rl6231_pll_code pll_code;
int ret;
if (source == rt5677->pll_src && freq_in == rt5677->pll_in &&
@@ -3137,15 +3067,12 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
return ret;
}
- dev_dbg(codec->dev, "m_bypass=%d k_bypass=%d m=%d n=%d k=%d\n",
- pll_code.m_bp, pll_code.k_bp,
- (pll_code.m_bp ? 0 : pll_code.m_code), pll_code.n_code,
- (pll_code.k_bp ? 0 : pll_code.k_code));
+ dev_dbg(codec->dev, "m_bypass=%d m=%d n=%d k=%d\n",
+ pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code),
+ pll_code.n_code, pll_code.k_code);
regmap_write(rt5677->regmap, RT5677_PLL1_CTRL1,
- pll_code.n_code << RT5677_PLL_N_SFT |
- pll_code.k_bp << RT5677_PLL_K_BP_SFT |
- (pll_code.k_bp ? 0 : pll_code.k_code));
+ pll_code.n_code << RT5677_PLL_N_SFT | pll_code.k_code);
regmap_write(rt5677->regmap, RT5677_PLL1_CTRL2,
(pll_code.m_bp ? 0 : pll_code.m_code) << RT5677_PLL_M_SFT |
pll_code.m_bp << RT5677_PLL_M_BP_SFT);
@@ -3197,7 +3124,7 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec,
regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x1, 0x0);
regmap_write(rt5677->regmap, RT5677_PWR_DIG1, 0x0000);
regmap_write(rt5677->regmap, RT5677_PWR_DIG2, 0x0000);
- regmap_write(rt5677->regmap, RT5677_PWR_ANLG1, 0x0000);
+ regmap_write(rt5677->regmap, RT5677_PWR_ANLG1, 0x0022);
regmap_write(rt5677->regmap, RT5677_PWR_ANLG2, 0x0000);
regmap_update_bits(rt5677->regmap,
RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0000);
@@ -3454,14 +3381,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
regmap_update_bits(rt5677->regmap, RT5677_IN1,
RT5677_IN_DF2, RT5677_IN_DF2);
- ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677,
- rt5677_dai, ARRAY_SIZE(rt5677_dai));
- if (ret < 0)
- goto err;
-
- return 0;
-err:
- return ret;
+ return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677,
+ rt5677_dai, ARRAY_SIZE(rt5677_dai));
}
static int rt5677_i2c_remove(struct i2c_client *i2c)
@@ -3480,18 +3401,7 @@ static struct i2c_driver rt5677_i2c_driver = {
.remove = rt5677_i2c_remove,
.id_table = rt5677_i2c_id,
};
-
-static int __init rt5677_modinit(void)
-{
- return i2c_add_driver(&rt5677_i2c_driver);
-}
-module_init(rt5677_modinit);
-
-static void __exit rt5677_modexit(void)
-{
- i2c_del_driver(&rt5677_i2c_driver);
-}
-module_exit(rt5677_modexit);
+module_i2c_driver(rt5677_i2c_driver);
MODULE_DESCRIPTION("ASoC RT5677 driver");
MODULE_AUTHOR("Oder Chiou <oder_chiou@realtek.com>");
diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h
index af4e9c797408..863393e62096 100644
--- a/sound/soc/codecs/rt5677.h
+++ b/sound/soc/codecs/rt5677.h
@@ -1393,13 +1393,6 @@
#define RT5677_DSP_IB_9_L (0x1 << 1)
#define RT5677_DSP_IB_9_L_SFT 1
-/* Debug String Length */
-#define RT5677_REG_DISP_LEN 23
-
-#define RT5677_NO_JACK BIT(0)
-#define RT5677_HEADSET_DET BIT(1)
-#define RT5677_HEADPHO_DET BIT(2)
-
/* System Clock Source */
enum {
RT5677_SCLK_S_MCLK,
@@ -1425,14 +1418,6 @@ enum {
RT5677_AIFS,
};
-struct rt5677_pll_code {
- bool m_bp; /* Indicates bypass m code or not. */
- bool k_bp; /* Indicates bypass k code or not. */
- int m_code;
- int n_code;
- int k_code;
-};
-
struct rt5677_priv {
struct snd_soc_codec *codec;
struct rt5677_platform_data pdata;
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 3d39f0b5b4a8..e997d271728d 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -724,25 +724,25 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream,
return ret;
/* set i2s data format */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J)
return -EINVAL;
i2s_ctl |= SGTL5000_I2S_DLEN_16 << SGTL5000_I2S_DLEN_SHIFT;
i2s_ctl |= SGTL5000_I2S_SCLKFREQ_32FS <<
SGTL5000_I2S_SCLKFREQ_SHIFT;
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
i2s_ctl |= SGTL5000_I2S_DLEN_20 << SGTL5000_I2S_DLEN_SHIFT;
i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS <<
SGTL5000_I2S_SCLKFREQ_SHIFT;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
i2s_ctl |= SGTL5000_I2S_DLEN_24 << SGTL5000_I2S_DLEN_SHIFT;
i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS <<
SGTL5000_I2S_SCLKFREQ_SHIFT;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J)
return -EINVAL;
i2s_ctl |= SGTL5000_I2S_DLEN_32 << SGTL5000_I2S_DLEN_SHIFT;
@@ -843,10 +843,8 @@ static int ldo_regulator_register(struct snd_soc_codec *codec,
ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL);
- if (!ldo) {
- dev_err(codec->dev, "failed to allocate ldo_regulator\n");
+ if (!ldo)
return -ENOMEM;
- }
ldo->desc.name = kstrdup(dev_name(codec->dev), GFP_KERNEL);
if (!ldo->desc.name) {
@@ -1277,7 +1275,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
return ret;
}
- ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
+ ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
sgtl5000->supplies);
if (ret)
goto err_ldo_remove;
@@ -1285,13 +1283,16 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
sgtl5000->supplies);
if (ret)
- goto err_ldo_remove;
+ goto err_regulator_free;
/* wait for all power rails bring up */
udelay(10);
return 0;
+err_regulator_free:
+ regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
err_ldo_remove:
if (!external_vddd)
ldo_regulator_remove(codec);
@@ -1361,6 +1362,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
err:
regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
sgtl5000->supplies);
+ regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
ldo_regulator_remove(codec);
return ret;
@@ -1374,6 +1377,8 @@ static int sgtl5000_remove(struct snd_soc_codec *codec)
regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
sgtl5000->supplies);
+ regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
ldo_regulator_remove(codec);
return 0;
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
index f26befb0c297..cdf882fa7716 100644
--- a/sound/soc/codecs/si476x.c
+++ b/sound/soc/codecs/si476x.c
@@ -167,17 +167,17 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S8:
+ switch (params_width(params)) {
+ case 8:
width = SI476X_PCM_FORMAT_S8;
break;
- case SNDRV_PCM_FORMAT_S16_LE:
+ case 16:
width = SI476X_PCM_FORMAT_S16_LE;
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
width = SI476X_PCM_FORMAT_S20_3LE;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
width = SI476X_PCM_FORMAT_S24_LE;
break;
default:
diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c
index d90cb0fafcb2..06ba4923fd5a 100644
--- a/sound/soc/codecs/sirf-audio-codec.c
+++ b/sound/soc/codecs/sirf-audio-codec.c
@@ -471,8 +471,8 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev)
mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
base = devm_ioremap_resource(&pdev->dev, mem_res);
- if (base == NULL)
- return -ENOMEM;
+ if (IS_ERR(base))
+ return PTR_ERR(base);
sirf_audio_codec->regmap = devm_regmap_init_mmio(&pdev->dev, base,
&sirf_audio_codec_regmap_config);
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 42dff26b3a2a..cf8fa40662f0 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -661,12 +661,12 @@ static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream,
{
unsigned int format, rate;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
format = BIT(4)|BIT(5);
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
format = 0;
break;
default:
diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c
index a078aa31052a..e0df537dd4b7 100644
--- a/sound/soc/codecs/spdif_transmitter.c
+++ b/sound/soc/codecs/spdif_transmitter.c
@@ -24,7 +24,7 @@
#define DRV_NAME "spdif-dit"
-#define STUB_RATES SNDRV_PCM_RATE_8000_96000
+#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)
diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index 56adb3e2def9..e8680bea5f86 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -361,11 +361,11 @@ static int ssm2518_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
if (ssm2518->right_j) {
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_16BIT;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT;
break;
default:
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 97b0454eb346..484b3bbe8624 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -275,17 +275,17 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
regmap_write(ssm2602->regmap, SSM2602_SRATE, srate);
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
iface = 0x0;
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface = 0x4;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface = 0x8;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
iface = 0xc;
break;
default:
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 0579d187135b..48740855566d 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -678,15 +678,11 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream,
confb = snd_soc_read(codec, STA32X_CONFB);
confb &= ~(STA32X_CONFB_SAI_MASK | STA32X_CONFB_SAIFB);
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S24_LE:
- case SNDRV_PCM_FORMAT_S24_BE:
- case SNDRV_PCM_FORMAT_S24_3LE:
- case SNDRV_PCM_FORMAT_S24_3BE:
+ switch (params_width(params)) {
+ case 24:
pr_debug("24bit\n");
/* fall through */
- case SNDRV_PCM_FORMAT_S32_LE:
- case SNDRV_PCM_FORMAT_S32_BE:
+ case 32:
pr_debug("24bit or 32bit\n");
switch (sta32x->format) {
case SND_SOC_DAIFMT_I2S:
@@ -701,8 +697,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream,
}
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
- case SNDRV_PCM_FORMAT_S20_3BE:
+ case 20:
pr_debug("20bit\n");
switch (sta32x->format) {
case SND_SOC_DAIFMT_I2S:
@@ -717,8 +712,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream,
}
break;
- case SNDRV_PCM_FORMAT_S18_3LE:
- case SNDRV_PCM_FORMAT_S18_3BE:
+ case 18:
pr_debug("18bit\n");
switch (sta32x->format) {
case SND_SOC_DAIFMT_I2S:
@@ -733,8 +727,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream,
}
break;
- case SNDRV_PCM_FORMAT_S16_LE:
- case SNDRV_PCM_FORMAT_S16_BE:
+ case 16:
pr_debug("16bit\n");
switch (sta32x->format) {
case SND_SOC_DAIFMT_I2S:
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index a40c4b0196a3..9aa1323fb2ab 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -197,16 +197,16 @@ static int sta529_hw_params(struct snd_pcm_substream *substream,
int pdata, play_freq_val, record_freq_val;
int bclk_to_fs_ratio;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
pdata = 1;
bclk_to_fs_ratio = 0;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
pdata = 2;
bclk_to_fs_ratio = 1;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
pdata = 3;
bclk_to_fs_ratio = 2;
break;
@@ -380,10 +380,8 @@ static int sta529_i2c_probe(struct i2c_client *i2c,
return -EINVAL;
sta529 = devm_kzalloc(&i2c->dev, sizeof(struct sta529), GFP_KERNEL);
- if (sta529 == NULL) {
- dev_err(&i2c->dev, "Can not allocate memory\n");
+ if (!sta529)
return -ENOMEM;
- }
sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap);
if (IS_ERR(sta529->regmap)) {
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
new file mode 100644
index 000000000000..23b32960ff1d
--- /dev/null
+++ b/sound/soc/codecs/tas2552.c
@@ -0,0 +1,544 @@
+/*
+ * tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier
+ *
+ * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
+ *
+ * Author: Dan Murphy <dmurphy@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.
+ */
+
+#include <linux/module.h>
+#include <linux/errno.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+
+#include <linux/gpio/consumer.h>
+#include <linux/regulator/consumer.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/tas2552-plat.h>
+
+#include "tas2552.h"
+
+static struct reg_default tas2552_reg_defs[] = {
+ {TAS2552_CFG_1, 0x22},
+ {TAS2552_CFG_3, 0x80},
+ {TAS2552_DOUT, 0x00},
+ {TAS2552_OUTPUT_DATA, 0xc0},
+ {TAS2552_PDM_CFG, 0x01},
+ {TAS2552_PGA_GAIN, 0x00},
+ {TAS2552_BOOST_PT_CTRL, 0x0f},
+ {TAS2552_RESERVED_0D, 0x00},
+ {TAS2552_LIMIT_RATE_HYS, 0x08},
+ {TAS2552_CFG_2, 0xef},
+ {TAS2552_SER_CTRL_1, 0x00},
+ {TAS2552_SER_CTRL_2, 0x00},
+ {TAS2552_PLL_CTRL_1, 0x10},
+ {TAS2552_PLL_CTRL_2, 0x00},
+ {TAS2552_PLL_CTRL_3, 0x00},
+ {TAS2552_BTIP, 0x8f},
+ {TAS2552_BTS_CTRL, 0x80},
+ {TAS2552_LIMIT_RELEASE, 0x04},
+ {TAS2552_LIMIT_INT_COUNT, 0x00},
+ {TAS2552_EDGE_RATE_CTRL, 0x40},
+ {TAS2552_VBAT_DATA, 0x00},
+};
+
+#define TAS2552_NUM_SUPPLIES 3
+static const char *tas2552_supply_names[TAS2552_NUM_SUPPLIES] = {
+ "vbat", /* vbat voltage */
+ "iovdd", /* I/O Voltage */
+ "avdd", /* Analog DAC Voltage */
+};
+
+struct tas2552_data {
+ struct snd_soc_codec *codec;
+ struct regmap *regmap;
+ struct i2c_client *tas2552_client;
+ struct regulator_bulk_data supplies[TAS2552_NUM_SUPPLIES];
+ struct gpio_desc *enable_gpio;
+ unsigned char regs[TAS2552_VBAT_DATA];
+ unsigned int mclk;
+};
+
+static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown)
+{
+ u8 cfg1_reg;
+
+ if (sw_shutdown)
+ cfg1_reg = 0;
+ else
+ cfg1_reg = TAS2552_SWS_MASK;
+
+ snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1,
+ TAS2552_SWS_MASK, cfg1_reg);
+}
+
+static int tas2552_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 tas2552_data *tas2552 = dev_get_drvdata(codec->dev);
+ int sample_rate, pll_clk;
+ int d;
+ u8 p, j;
+
+ /* Turn on Class D amplifier */
+ snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK,
+ TAS2552_CLASSD_EN);
+
+ if (!tas2552->mclk)
+ return -EINVAL;
+
+ snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0);
+
+ if (tas2552->mclk == TAS2552_245MHZ_CLK ||
+ tas2552->mclk == TAS2552_225MHZ_CLK) {
+ /* By pass the PLL configuration */
+ snd_soc_update_bits(codec, TAS2552_PLL_CTRL_2,
+ TAS2552_PLL_BYPASS_MASK,
+ TAS2552_PLL_BYPASS);
+ } else {
+ /* Fill in the PLL control registers for J & D
+ * PLL_CLK = (.5 * freq * J.D) / 2^p
+ * Need to fill in J and D here based on incoming freq
+ */
+ p = snd_soc_read(codec, TAS2552_PLL_CTRL_1);
+ p = (p >> 7);
+ sample_rate = params_rate(params);
+
+ if (sample_rate == 48000)
+ pll_clk = TAS2552_245MHZ_CLK;
+ else if (sample_rate == 44100)
+ pll_clk = TAS2552_225MHZ_CLK;
+ else {
+ dev_vdbg(codec->dev, "Substream sample rate is not found %i\n",
+ params_rate(params));
+ return -EINVAL;
+ }
+
+ j = (pll_clk * 2 * (1 << p)) / tas2552->mclk;
+ d = (pll_clk * 2 * (1 << p)) % tas2552->mclk;
+
+ snd_soc_update_bits(codec, TAS2552_PLL_CTRL_1,
+ TAS2552_PLL_J_MASK, j);
+ snd_soc_write(codec, TAS2552_PLL_CTRL_2,
+ (d >> 7) & TAS2552_PLL_D_UPPER_MASK);
+ snd_soc_write(codec, TAS2552_PLL_CTRL_3,
+ d & TAS2552_PLL_D_LOWER_MASK);
+
+ }
+
+ snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE,
+ TAS2552_PLL_ENABLE);
+
+ return 0;
+}
+
+static int tas2552_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 serial_format;
+ u8 serial_control_mask;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ serial_format = 0x00;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ serial_format = TAS2552_WORD_CLK_MASK;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ serial_format = TAS2552_BIT_CLK_MASK;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ serial_format = (TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK);
+ break;
+ default:
+ dev_vdbg(codec->dev, "DAI Format master is not found\n");
+ return -EINVAL;
+ }
+
+ serial_control_mask = TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK;
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ serial_format &= TAS2552_DAIFMT_I2S_MASK;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ serial_format |= TAS2552_DAIFMT_DSP;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ serial_format |= TAS2552_DAIFMT_RIGHT_J;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ serial_format |= TAS2552_DAIFMT_LEFT_J;
+ break;
+ default:
+ dev_vdbg(codec->dev, "DAI Format is not found\n");
+ return -EINVAL;
+ }
+
+ if (fmt & SND_SOC_DAIFMT_FORMAT_MASK)
+ serial_control_mask |= TAS2552_DATA_FORMAT_MASK;
+
+ snd_soc_update_bits(codec, TAS2552_SER_CTRL_1, serial_control_mask,
+ serial_format);
+
+ return 0;
+}
+
+static int tas2552_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct tas2552_data *tas2552 = dev_get_drvdata(codec->dev);
+
+ tas2552->mclk = freq;
+
+ return 0;
+}
+
+static int tas2552_mute(struct snd_soc_dai *dai, int mute)
+{
+ u8 cfg1_reg;
+ struct snd_soc_codec *codec = dai->codec;
+
+ if (mute)
+ cfg1_reg = TAS2552_MUTE_MASK;
+ else
+ cfg1_reg = ~TAS2552_MUTE_MASK;
+
+ snd_soc_update_bits(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK, cfg1_reg);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM_RUNTIME
+static int tas2552_runtime_suspend(struct device *dev)
+{
+ struct tas2552_data *tas2552 = dev_get_drvdata(dev);
+
+ tas2552_sw_shutdown(tas2552, 0);
+
+ regcache_cache_only(tas2552->regmap, true);
+ regcache_mark_dirty(tas2552->regmap);
+
+ if (tas2552->enable_gpio)
+ gpiod_set_value(tas2552->enable_gpio, 0);
+
+ return 0;
+}
+
+static int tas2552_runtime_resume(struct device *dev)
+{
+ struct tas2552_data *tas2552 = dev_get_drvdata(dev);
+
+ if (tas2552->enable_gpio)
+ gpiod_set_value(tas2552->enable_gpio, 1);
+
+ tas2552_sw_shutdown(tas2552, 1);
+
+ regcache_cache_only(tas2552->regmap, false);
+ regcache_sync(tas2552->regmap);
+
+ return 0;
+}
+#endif
+
+static const struct dev_pm_ops tas2552_pm = {
+ SET_RUNTIME_PM_OPS(tas2552_runtime_suspend, tas2552_runtime_resume,
+ NULL)
+};
+
+static void tas2552_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0);
+}
+
+static struct snd_soc_dai_ops tas2552_speaker_dai_ops = {
+ .hw_params = tas2552_hw_params,
+ .set_sysclk = tas2552_set_dai_sysclk,
+ .set_fmt = tas2552_set_dai_fmt,
+ .shutdown = tas2552_shutdown,
+ .digital_mute = tas2552_mute,
+};
+
+/* Formats supported by TAS2552 driver. */
+#define TAS2552_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+/* TAS2552 dai structure. */
+static struct snd_soc_dai_driver tas2552_dai[] = {
+ {
+ .name = "tas2552-amplifier",
+ .playback = {
+ .stream_name = "Speaker",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = TAS2552_FORMATS,
+ },
+ .ops = &tas2552_speaker_dai_ops,
+ },
+};
+
+/*
+ * DAC digital volumes. From -7 to 24 dB in 1 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24);
+
+static const struct snd_kcontrol_new tas2552_snd_controls[] = {
+ SOC_SINGLE_TLV("Speaker Driver Playback Volume",
+ TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv),
+};
+
+static const struct reg_default tas2552_init_regs[] = {
+ { TAS2552_RESERVED_0D, 0xc0 },
+};
+
+static int tas2552_codec_probe(struct snd_soc_codec *codec)
+{
+ struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ tas2552->codec = codec;
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies),
+ tas2552->supplies);
+
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to enable supplies: %d\n",
+ ret);
+ return ret;
+ }
+
+ if (tas2552->enable_gpio)
+ gpiod_set_value(tas2552->enable_gpio, 1);
+
+ ret = pm_runtime_get_sync(codec->dev);
+ if (ret < 0) {
+ dev_err(codec->dev, "Enabling device failed: %d\n",
+ ret);
+ goto probe_fail;
+ }
+
+ snd_soc_write(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK |
+ TAS2552_PLL_SRC_BCLK);
+ snd_soc_write(codec, TAS2552_CFG_3, TAS2552_I2S_OUT_SEL |
+ TAS2552_DIN_SRC_SEL_AVG_L_R | TAS2552_88_96KHZ);
+ snd_soc_write(codec, TAS2552_DOUT, TAS2552_PDM_DATA_I);
+ snd_soc_write(codec, TAS2552_OUTPUT_DATA, TAS2552_PDM_DATA_V_I | 0x8);
+ snd_soc_write(codec, TAS2552_PDM_CFG, TAS2552_PDM_BCLK_SEL);
+ snd_soc_write(codec, TAS2552_BOOST_PT_CTRL, TAS2552_APT_DELAY_200 |
+ TAS2552_APT_THRESH_2_1_7);
+
+ ret = regmap_register_patch(tas2552->regmap, tas2552_init_regs,
+ ARRAY_SIZE(tas2552_init_regs));
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to write init registers: %d\n",
+ ret);
+ goto patch_fail;
+ }
+
+ snd_soc_write(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN |
+ TAS2552_BOOST_EN | TAS2552_APT_EN |
+ TAS2552_LIM_EN);
+ return 0;
+
+patch_fail:
+ pm_runtime_put(codec->dev);
+probe_fail:
+ if (tas2552->enable_gpio)
+ gpiod_set_value(tas2552->enable_gpio, 0);
+
+ regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies),
+ tas2552->supplies);
+ return -EIO;
+}
+
+static int tas2552_codec_remove(struct snd_soc_codec *codec)
+{
+ struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec);
+
+ pm_runtime_put(codec->dev);
+
+ if (tas2552->enable_gpio)
+ gpiod_set_value(tas2552->enable_gpio, 0);
+
+ return 0;
+};
+
+#ifdef CONFIG_PM
+static int tas2552_suspend(struct snd_soc_codec *codec)
+{
+ struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ ret = regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies),
+ tas2552->supplies);
+
+ if (ret != 0)
+ dev_err(codec->dev, "Failed to disable supplies: %d\n",
+ ret);
+ return 0;
+}
+
+static int tas2552_resume(struct snd_soc_codec *codec)
+{
+ struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies),
+ tas2552->supplies);
+
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to enable supplies: %d\n",
+ ret);
+ }
+
+ return 0;
+}
+#else
+#define tas2552_suspend NULL
+#define tas2552_resume NULL
+#endif
+
+static struct snd_soc_codec_driver soc_codec_dev_tas2552 = {
+ .probe = tas2552_codec_probe,
+ .remove = tas2552_codec_remove,
+ .suspend = tas2552_suspend,
+ .resume = tas2552_resume,
+ .controls = tas2552_snd_controls,
+ .num_controls = ARRAY_SIZE(tas2552_snd_controls),
+};
+
+static const struct regmap_config tas2552_regmap_config = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = TAS2552_MAX_REG,
+ .reg_defaults = tas2552_reg_defs,
+ .num_reg_defaults = ARRAY_SIZE(tas2552_reg_defs),
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static int tas2552_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct device *dev;
+ struct tas2552_data *data;
+ int ret;
+ int i;
+
+ dev = &client->dev;
+ data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL);
+ if (data == NULL)
+ return -ENOMEM;
+
+ data->enable_gpio = devm_gpiod_get(dev, "enable");
+ if (IS_ERR(data->enable_gpio)) {
+ ret = PTR_ERR(data->enable_gpio);
+ if (ret != -ENOENT && ret != -ENOSYS)
+ return ret;
+
+ data->enable_gpio = NULL;
+ } else {
+ gpiod_direction_output(data->enable_gpio, 0);
+ }
+
+ data->tas2552_client = client;
+ data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config);
+ if (IS_ERR(data->regmap)) {
+ ret = PTR_ERR(data->regmap);
+ dev_err(&client->dev, "Failed to allocate register map: %d\n",
+ ret);
+ return ret;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
+ data->supplies[i].supply = tas2552_supply_names[i];
+
+ ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies),
+ data->supplies);
+ if (ret != 0) {
+ dev_err(dev, "Failed to request supplies: %d\n", ret);
+ return ret;
+ }
+
+ pm_runtime_set_active(&client->dev);
+ pm_runtime_set_autosuspend_delay(&client->dev, 1000);
+ pm_runtime_use_autosuspend(&client->dev);
+ pm_runtime_enable(&client->dev);
+ pm_runtime_mark_last_busy(&client->dev);
+ pm_runtime_put_sync_autosuspend(&client->dev);
+
+ dev_set_drvdata(&client->dev, data);
+
+ ret = snd_soc_register_codec(&client->dev,
+ &soc_codec_dev_tas2552,
+ tas2552_dai, ARRAY_SIZE(tas2552_dai));
+ if (ret < 0)
+ dev_err(&client->dev, "Failed to register codec: %d\n", ret);
+
+ return ret;
+}
+
+static int tas2552_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static const struct i2c_device_id tas2552_id[] = {
+ { "tas2552", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, tas2552_id);
+
+#if IS_ENABLED(CONFIG_OF)
+static const struct of_device_id tas2552_of_match[] = {
+ { .compatible = "ti,tas2552", },
+ {},
+};
+MODULE_DEVICE_TABLE(of, tas2552_of_match);
+#endif
+
+static struct i2c_driver tas2552_i2c_driver = {
+ .driver = {
+ .name = "tas2552",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(tas2552_of_match),
+ .pm = &tas2552_pm,
+ },
+ .probe = tas2552_probe,
+ .remove = tas2552_i2c_remove,
+ .id_table = tas2552_id,
+};
+
+module_i2c_driver(tas2552_i2c_driver);
+
+MODULE_AUTHOR("Dan Muprhy <dmurphy@ti.com>");
+MODULE_DESCRIPTION("TAS2552 Audio amplifier driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tas2552.h b/sound/soc/codecs/tas2552.h
new file mode 100644
index 000000000000..6cea8f31bf88
--- /dev/null
+++ b/sound/soc/codecs/tas2552.h
@@ -0,0 +1,129 @@
+/*
+ * tas2552.h - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier
+ *
+ * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
+ *
+ * Author: Dan Murphy <dmurphy@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.
+ */
+
+#ifndef __TAS2552_H__
+#define __TAS2552_H__
+
+/* Register Address Map */
+#define TAS2552_DEVICE_STATUS 0x00
+#define TAS2552_CFG_1 0x01
+#define TAS2552_CFG_2 0x02
+#define TAS2552_CFG_3 0x03
+#define TAS2552_DOUT 0x04
+#define TAS2552_SER_CTRL_1 0x05
+#define TAS2552_SER_CTRL_2 0x06
+#define TAS2552_OUTPUT_DATA 0x07
+#define TAS2552_PLL_CTRL_1 0x08
+#define TAS2552_PLL_CTRL_2 0x09
+#define TAS2552_PLL_CTRL_3 0x0a
+#define TAS2552_BTIP 0x0b
+#define TAS2552_BTS_CTRL 0x0c
+#define TAS2552_RESERVED_0D 0x0d
+#define TAS2552_LIMIT_RATE_HYS 0x0e
+#define TAS2552_LIMIT_RELEASE 0x0f
+#define TAS2552_LIMIT_INT_COUNT 0x10
+#define TAS2552_PDM_CFG 0x11
+#define TAS2552_PGA_GAIN 0x12
+#define TAS2552_EDGE_RATE_CTRL 0x13
+#define TAS2552_BOOST_PT_CTRL 0x14
+#define TAS2552_VER_NUM 0x16
+#define TAS2552_VBAT_DATA 0x19
+#define TAS2552_MAX_REG 0x20
+
+/* CFG1 Register Masks */
+#define TAS2552_MUTE_MASK (1 << 2)
+#define TAS2552_SWS_MASK (1 << 1)
+#define TAS2552_WCLK_MASK 0x07
+#define TAS2552_CLASSD_EN_MASK (1 << 7)
+
+/* CFG2 Register Masks */
+#define TAS2552_CLASSD_EN (1 << 7)
+#define TAS2552_BOOST_EN (1 << 6)
+#define TAS2552_APT_EN (1 << 5)
+#define TAS2552_PLL_ENABLE (1 << 3)
+#define TAS2552_LIM_EN (1 << 2)
+#define TAS2552_IVSENSE_EN (1 << 1)
+
+/* CFG3 Register Masks */
+#define TAS2552_WORD_CLK_MASK (1 << 7)
+#define TAS2552_BIT_CLK_MASK (1 << 6)
+#define TAS2552_DATA_FORMAT_MASK (0x11 << 2)
+
+#define TAS2552_DAIFMT_I2S_MASK 0xf3
+#define TAS2552_DAIFMT_DSP (1 << 3)
+#define TAS2552_DAIFMT_RIGHT_J (1 << 4)
+#define TAS2552_DAIFMT_LEFT_J (0x11 << 3)
+
+#define TAS2552_PLL_SRC_MCLK 0x00
+#define TAS2552_PLL_SRC_BCLK (1 << 3)
+#define TAS2552_PLL_SRC_IVCLKIN (1 << 4)
+#define TAS2552_PLL_SRC_1_8_FIXED (0x11 << 3)
+
+#define TAS2552_DIN_SRC_SEL_MUTED 0x00
+#define TAS2552_DIN_SRC_SEL_LEFT (1 << 4)
+#define TAS2552_DIN_SRC_SEL_RIGHT (1 << 5)
+#define TAS2552_DIN_SRC_SEL_AVG_L_R (0x11 << 4)
+
+#define TAS2552_PDM_IN_SEL (1 << 5)
+#define TAS2552_I2S_OUT_SEL (1 << 6)
+#define TAS2552_ANALOG_IN_SEL (1 << 7)
+
+/* CFG3 WCLK Dividers */
+#define TAS2552_8KHZ 0x00
+#define TAS2552_11_12KHZ (1 << 1)
+#define TAS2552_16KHZ (1 << 2)
+#define TAS2552_22_24KHZ (1 << 3)
+#define TAS2552_32KHZ (1 << 4)
+#define TAS2552_44_48KHZ (1 << 5)
+#define TAS2552_88_96KHZ (1 << 6)
+#define TAS2552_176_192KHZ (1 << 7)
+
+/* OUTPUT_DATA register */
+#define TAS2552_PDM_DATA_I 0x00
+#define TAS2552_PDM_DATA_V (1 << 6)
+#define TAS2552_PDM_DATA_I_V (1 << 7)
+#define TAS2552_PDM_DATA_V_I (0x11 << 6)
+
+/* PDM CFG Register */
+#define TAS2552_PDM_DATA_ES_RISE 0x4
+
+#define TAS2552_PDM_PLL_CLK_SEL 0x00
+#define TAS2552_PDM_IV_CLK_SEL (1 << 1)
+#define TAS2552_PDM_BCLK_SEL (1 << 2)
+#define TAS2552_PDM_MCLK_SEL (1 << 3)
+
+/* Boost pass-through register */
+#define TAS2552_APT_DELAY_50 0x00
+#define TAS2552_APT_DELAY_75 (1 << 1)
+#define TAS2552_APT_DELAY_125 (1 << 2)
+#define TAS2552_APT_DELAY_200 (1 << 3)
+
+#define TAS2552_APT_THRESH_2_5 0x00
+#define TAS2552_APT_THRESH_1_7 (1 << 3)
+#define TAS2552_APT_THRESH_1_4_1_1 (1 << 4)
+#define TAS2552_APT_THRESH_2_1_7 (0x11 << 2)
+
+/* PLL Control Register */
+#define TAS2552_245MHZ_CLK 24576000
+#define TAS2552_225MHZ_CLK 22579200
+#define TAS2552_PLL_J_MASK 0x7f
+#define TAS2552_PLL_D_UPPER_MASK 0x3f
+#define TAS2552_PLL_D_LOWER_MASK 0xff
+#define TAS2552_PLL_BYPASS_MASK 0x80
+#define TAS2552_PLL_BYPASS 0x80
+
+#endif
diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c
index d48491a4a19d..249ef5c4c762 100644
--- a/sound/soc/codecs/tas5086.c
+++ b/sound/soc/codecs/tas5086.c
@@ -36,6 +36,7 @@
#include <linux/gpio.h>
#include <linux/i2c.h>
#include <linux/regmap.h>
+#include <linux/regulator/consumer.h>
#include <linux/spi/spi.h>
#include <linux/of.h>
#include <linux/of_device.h>
@@ -240,6 +241,10 @@ static int tas5086_reg_read(void *context, unsigned int reg,
return 0;
}
+static const char * const supply_names[] = {
+ "dvdd", "avdd"
+};
+
struct tas5086_private {
struct regmap *regmap;
unsigned int mclk, sclk;
@@ -251,6 +256,7 @@ struct tas5086_private {
int rate;
/* GPIO driving Reset pin, if any */
int gpio_nreset;
+ struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
};
static int tas5086_deemph[] = { 0, 32000, 44100, 48000 };
@@ -419,14 +425,14 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream,
}
/* ... then add the offset for the sample bit depth. */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
val += 0;
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
val += 1;
break;
- case SNDRV_PCM_FORMAT_S24_3LE:
+ case 24:
val += 2;
break;
default:
@@ -773,6 +779,8 @@ static int tas5086_soc_suspend(struct snd_soc_codec *codec)
if (ret < 0)
return ret;
+ regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+
return 0;
}
@@ -781,6 +789,10 @@ static int tas5086_soc_resume(struct snd_soc_codec *codec)
struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
int ret;
+ ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+ if (ret < 0)
+ return ret;
+
tas5086_reset(priv);
regcache_mark_dirty(priv->regmap);
@@ -812,6 +824,12 @@ static int tas5086_probe(struct snd_soc_codec *codec)
struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
int i, ret;
+ ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to enable regulators: %d\n", ret);
+ return ret;
+ }
+
priv->pwm_start_mid_z = 0;
priv->charge_period = 1300000; /* hardware default is 1300 ms */
@@ -832,16 +850,22 @@ static int tas5086_probe(struct snd_soc_codec *codec)
}
}
+ tas5086_reset(priv);
ret = tas5086_init(codec->dev, priv);
if (ret < 0)
- return ret;
+ goto exit_disable_regulators;
/* set master volume to 0 dB */
ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30);
if (ret < 0)
- return ret;
+ goto exit_disable_regulators;
return 0;
+
+exit_disable_regulators:
+ regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+
+ return ret;
}
static int tas5086_remove(struct snd_soc_codec *codec)
@@ -852,6 +876,8 @@ static int tas5086_remove(struct snd_soc_codec *codec)
/* Set codec to the reset state */
gpio_set_value(priv->gpio_nreset, 0);
+ regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+
return 0;
};
@@ -900,6 +926,16 @@ static int tas5086_i2c_probe(struct i2c_client *i2c,
if (!priv)
return -ENOMEM;
+ for (i = 0; i < ARRAY_SIZE(supply_names); i++)
+ priv->supplies[i].supply = supply_names[i];
+
+ ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies),
+ priv->supplies);
+ if (ret < 0) {
+ dev_err(dev, "Failed to get regulators: %d\n", ret);
+ return ret;
+ }
+
priv->regmap = devm_regmap_init(dev, NULL, i2c, &tas5086_regmap);
if (IS_ERR(priv->regmap)) {
ret = PTR_ERR(priv->regmap);
@@ -919,21 +955,34 @@ static int tas5086_i2c_probe(struct i2c_client *i2c,
gpio_nreset = -EINVAL;
priv->gpio_nreset = gpio_nreset;
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+ if (ret < 0) {
+ dev_err(dev, "Failed to enable regulators: %d\n", ret);
+ return ret;
+ }
+
tas5086_reset(priv);
/* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */
ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i);
- if (ret < 0)
- return ret;
-
- if (i != 0x3) {
+ if (ret == 0 && i != 0x3) {
dev_err(dev,
"Failed to identify TAS5086 codec (got %02x)\n", i);
- return -ENODEV;
+ ret = -ENODEV;
}
- return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086,
- &tas5086_dai, 1);
+ /*
+ * The chip has been identified, so we can turn off the power
+ * again until the dai link is set up.
+ */
+ regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+
+ if (ret == 0)
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086,
+ &tas5086_dai, 1);
+
+ return ret;
}
static int tas5086_i2c_remove(struct i2c_client *i2c)
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 686b8b85b956..d67167920c2f 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -364,16 +364,16 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface_reg |= (0x01 << 2);
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface_reg |= (0x02 << 2);
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
iface_reg |= (0x03 << 2);
break;
}
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 43069de3d56a..620ab9ea1ef0 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -71,8 +71,8 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
dev_dbg(&aic26->spi->dev, "aic26_hw_params(substream=%p, params=%p)\n",
substream, params);
- dev_dbg(&aic26->spi->dev, "rate=%i format=%i\n", params_rate(params),
- params_format(params));
+ dev_dbg(&aic26->spi->dev, "rate=%i width=%d\n", params_rate(params),
+ params_width(params));
switch (params_rate(params)) {
case 8000: fsref = 48000; divisor = AIC26_DIV_6; break;
@@ -89,11 +89,11 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
}
/* select data word length */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S8: wlen = AIC26_WLEN_16; break;
- case SNDRV_PCM_FORMAT_S16_BE: wlen = AIC26_WLEN_16; break;
- case SNDRV_PCM_FORMAT_S24_BE: wlen = AIC26_WLEN_24; break;
- case SNDRV_PCM_FORMAT_S32_BE: wlen = AIC26_WLEN_32; break;
+ switch (params_width(params)) {
+ case 8: wlen = AIC26_WLEN_16; break;
+ case 16: wlen = AIC26_WLEN_16; break;
+ case 24: wlen = AIC26_WLEN_24; break;
+ case 32: wlen = AIC26_WLEN_32; break;
default:
dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
}
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 23419109ecac..0f64c7890eed 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -249,17 +249,16 @@ static const char * const mic_select_text[] = {
"Off", "FFR 10 Ohm", "FFR 20 Ohm", "FFR 40 Ohm"
};
-static const
-SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, mic_select_text);
-static const
-SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, mic_select_text);
-static const
-SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text);
-
-static const
-SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text);
-static const
-SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text);
+static SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6,
+ mic_select_text);
+static SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4,
+ mic_select_text);
+static SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2,
+ mic_select_text);
+
+static SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text);
+static SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4,
+ mic_select_text);
static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0);
static const DECLARE_TLV_DB_SCALE(adc_fgain_tlv, 0, 10, 0);
@@ -329,6 +328,7 @@ static int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg,
unsigned int bits;
int counter = count;
int ret = regmap_read(aic31xx->regmap, reg, &bits);
+
while ((bits & mask) != wbits && counter && !ret) {
usleep_range(sleep, sleep * 2);
ret = regmap_read(aic31xx->regmap, reg, &bits);
@@ -435,6 +435,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
{
struct snd_soc_codec *codec = w->codec;
struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+
switch (event) {
case SND_SOC_DAPM_POST_PMU:
/* change mic bias voltage to user defined */
@@ -759,8 +760,8 @@ static int aic31xx_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = dai->codec;
u8 data = 0;
- dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n",
- __func__, params_format(params), params_width(params),
+ dev_dbg(codec->dev, "## %s: width %d rate %d\n",
+ __func__, params_width(params),
params_rate(params));
switch (params_width(params)) {
@@ -779,8 +780,8 @@ static int aic31xx_hw_params(struct snd_pcm_substream *substream,
AIC31XX_IFACE1_DATALEN_SHIFT);
break;
default:
- dev_err(codec->dev, "%s: Unsupported format %d\n",
- __func__, params_format(params));
+ dev_err(codec->dev, "%s: Unsupported width %d\n",
+ __func__, params_width(params));
return -EINVAL;
}
@@ -1178,7 +1179,7 @@ static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx)
}
#endif /* CONFIG_OF */
-static void aic31xx_device_init(struct aic31xx_priv *aic31xx)
+static int aic31xx_device_init(struct aic31xx_priv *aic31xx)
{
int ret, i;
@@ -1197,7 +1198,7 @@ static void aic31xx_device_init(struct aic31xx_priv *aic31xx)
"aic31xx-reset-pin");
if (ret < 0) {
dev_err(aic31xx->dev, "not able to acquire gpio\n");
- return;
+ return ret;
}
}
@@ -1210,6 +1211,7 @@ static void aic31xx_device_init(struct aic31xx_priv *aic31xx)
if (ret != 0)
dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret);
+ return ret;
}
static int aic31xx_i2c_probe(struct i2c_client *i2c,
@@ -1239,7 +1241,9 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
aic31xx->pdata.codec_type = id->driver_data;
- aic31xx_device_init(aic31xx);
+ ret = aic31xx_device_init(aic31xx);
+ if (ret)
+ return ret;
return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx,
aic31xx_dai_driver,
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 1d9b117345a3..6ea662db2410 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -450,16 +450,16 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream,
data = snd_soc_read(codec, AIC32X4_IFACE1);
data = data & ~(3 << 4);
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
data |= (AIC32X4_WORD_LEN_20BITS << AIC32X4_DOSRMSB_SHIFT);
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
data |= (AIC32X4_WORD_LEN_24BITS << AIC32X4_DOSRMSB_SHIFT);
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
data |= (AIC32X4_WORD_LEN_32BITS << AIC32X4_DOSRMSB_SHIFT);
break;
}
@@ -626,32 +626,33 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN |
AIC32X4_MICBIAS_2075V);
}
- if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) {
+ if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE)
snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE);
- }
tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ?
AIC32X4_LDOCTLEN : 0;
snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg);
tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE);
- if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) {
+ if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36)
tmp_reg |= AIC32X4_LDOIN_18_36;
- }
- if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) {
+ if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED)
tmp_reg |= AIC32X4_LDOIN2HP;
- }
snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg);
/* Mic PGA routing */
if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K)
- snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K);
+ snd_soc_write(codec, AIC32X4_LMICPGANIN,
+ AIC32X4_LMICPGANIN_IN2R_10K);
else
- snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_CM1L_10K);
+ snd_soc_write(codec, AIC32X4_LMICPGANIN,
+ AIC32X4_LMICPGANIN_CM1L_10K);
if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K)
- snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K);
+ snd_soc_write(codec, AIC32X4_RMICPGANIN,
+ AIC32X4_RMICPGANIN_IN1L_10K);
else
- snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_CM1R_10K);
+ snd_soc_write(codec, AIC32X4_RMICPGANIN,
+ AIC32X4_RMICPGANIN_CM1R_10K);
aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index e12fafbb1e09..64f179ee9834 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -873,16 +873,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
/* select data word length */
data = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
data |= (0x01 << 4);
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
data |= (0x02 << 4);
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
data |= (0x03 << 4);
break;
}
@@ -1194,7 +1194,8 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
#define AIC3X_RATES SNDRV_PCM_RATE_8000_96000
#define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
- SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
static const struct snd_soc_dai_ops aic3x_dai_ops = {
.hw_params = aic3x_hw_params,
@@ -1477,10 +1478,8 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
u32 value;
aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL);
- if (aic3x == NULL) {
- dev_err(&i2c->dev, "failed to create private data\n");
+ if (!aic3x)
return -ENOMEM;
- }
aic3x->regmap = devm_regmap_init_i2c(i2c, &aic3x_regmap);
if (IS_ERR(aic3x->regmap)) {
@@ -1498,10 +1497,8 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
} else if (np) {
ai3x_setup = devm_kzalloc(&i2c->dev, sizeof(*ai3x_setup),
GFP_KERNEL);
- if (ai3x_setup == NULL) {
- dev_err(&i2c->dev, "failed to create private data\n");
+ if (!ai3x_setup)
return -ENOMEM;
- }
ret = of_get_named_gpio(np, "gpio-reset", 0);
if (ret >= 0)
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index df3a7506c023..e21ed934bdbf 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -832,18 +832,18 @@ static int dac33_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
dac33->fifo_size = DAC33_FIFO_SIZE_16BIT;
dac33->burst_rate = CALC_BURST_RATE(dac33->burst_bclkdiv, 32);
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
dac33->fifo_size = DAC33_FIFO_SIZE_24BIT;
dac33->burst_rate = CALC_BURST_RATE(dac33->burst_bclkdiv, 64);
break;
default:
- dev_err(codec->dev, "unsupported format %d\n",
- params_format(params));
+ dev_err(codec->dev, "unsupported width %d\n",
+ params_width(params));
return -EINVAL;
}
@@ -1404,7 +1404,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
if (dac33->irq >= 0) {
ret = request_irq(dac33->irq, dac33_interrupt_handler,
IRQF_TRIGGER_RISING,
- codec->name, codec);
+ codec->component.name, codec);
if (ret < 0) {
dev_err(codec->dev, "Could not request IRQ%d (%d)\n",
dac33->irq, ret);
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 8fc5a647453b..6fac9e034c48 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -381,10 +381,8 @@ static int tpa6130a2_probe(struct i2c_client *client,
dev = &client->dev;
data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL);
- if (data == NULL) {
- dev_err(dev, "Can not allocate memory\n");
+ if (!data)
return -ENOMEM;
- }
if (pdata) {
data->power_gpio = pdata->power_gpio;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 69e12a311ba2..b6b0cb399599 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -344,17 +344,16 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
{
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
- int status = -1;
if (enable) {
twl4030->apll_enabled++;
if (twl4030->apll_enabled == 1)
- status = twl4030_audio_enable_resource(
+ twl4030_audio_enable_resource(
TWL4030_AUDIO_RES_APLL);
} else {
twl4030->apll_enabled--;
if (!twl4030->apll_enabled)
- status = twl4030_audio_disable_resource(
+ twl4030_audio_disable_resource(
TWL4030_AUDIO_RES_APLL);
}
}
@@ -1764,16 +1763,16 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
old_format = twl4030_read(codec, TWL4030_REG_AUDIO_IF);
format = old_format;
format &= ~TWL4030_DATA_WIDTH;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
format |= TWL4030_DATA_WIDTH_16S_16W;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
format |= TWL4030_DATA_WIDTH_32S_24W;
break;
default:
- dev_err(codec->dev, "%s: unknown format %d\n", __func__,
- params_format(params));
+ dev_err(codec->dev, "%s: unsupported bits/sample %d\n",
+ __func__, params_width(params));
return -EINVAL;
}
@@ -2162,10 +2161,8 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec)
twl4030 = devm_kzalloc(codec->dev, sizeof(struct twl4030_priv),
GFP_KERNEL);
- if (twl4030 == NULL) {
- dev_err(codec->dev, "Can not allocate memory\n");
+ if (!twl4030)
return -ENOMEM;
- }
snd_soc_codec_set_drvdata(codec, twl4030);
/* Set the defaults, and power up the codec */
twl4030->sysclk = twl4030_audio_get_mclk() / 1000;
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index edf27acc1d77..32b2f78aa62c 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -243,14 +243,14 @@ static int uda134x_hw_params(struct snd_pcm_substream *substream,
case SND_SOC_DAIFMT_I2S:
break;
case SND_SOC_DAIFMT_RIGHT_J:
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
hw_params |= (1<<1);
break;
- case SNDRV_PCM_FORMAT_S18_3LE:
+ case 18:
hw_params |= (1<<2);
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
hw_params |= ((1<<2) | (1<<1));
break;
default:
@@ -479,7 +479,7 @@ static struct snd_soc_dai_driver uda134x_dai = {
static int uda134x_soc_probe(struct snd_soc_codec *codec)
{
struct uda134x_priv *uda134x;
- struct uda134x_platform_data *pd = codec->card->dev->platform_data;
+ struct uda134x_platform_data *pd = codec->component.card->dev->platform_data;
const struct snd_soc_dapm_widget *widgets;
unsigned num_widgets;
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index 4ead0dc02b87..f3d4e88d0b7b 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -341,8 +341,9 @@ static int wl1273_hw_params(struct snd_pcm_substream *substream,
struct wl1273_core *core = wl1273->core;
unsigned int rate, width, r;
- if (params_format(params) != SNDRV_PCM_FORMAT_S16_LE) {
- pr_err("Only SNDRV_PCM_FORMAT_S16_LE supported.\n");
+ if (params_width(params) != 16) {
+ dev_err(dai->dev, "%d bits/sample not supported\n",
+ params_width(params));
return -EINVAL;
}
@@ -461,10 +462,8 @@ static int wl1273_probe(struct snd_soc_codec *codec)
}
wl1273 = kzalloc(sizeof(struct wl1273_priv), GFP_KERNEL);
- if (wl1273 == NULL) {
- dev_err(codec->dev, "Cannot allocate memory.\n");
+ if (!wl1273)
return -ENOMEM;
- }
wl1273->mode = WL1273_MODE_BT;
wl1273->core = *core;
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index 71ce3159a62e..f37989ec7cba 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -144,7 +144,7 @@ static const struct snd_soc_dapm_route wm0010_dapm_routes[] = {
static const char *wm0010_state_to_str(enum wm0010_state state)
{
- const char *state_to_str[] = {
+ static const char * const state_to_str[] = {
"Power off",
"Out of reset",
"Boot ROM",
@@ -413,7 +413,6 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec)
xfer = kzalloc(sizeof(*xfer), GFP_KERNEL);
if (!xfer) {
- dev_err(codec->dev, "Failed to allocate xfer\n");
ret = -ENOMEM;
goto abort;
}
@@ -423,8 +422,6 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec)
out = kzalloc(len, GFP_KERNEL | GFP_DMA);
if (!out) {
- dev_err(codec->dev,
- "Failed to allocate RX buffer\n");
ret = -ENOMEM;
goto abort1;
}
@@ -432,8 +429,6 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec)
img = kzalloc(len, GFP_KERNEL | GFP_DMA);
if (!img) {
- dev_err(codec->dev,
- "Failed to allocate image buffer\n");
ret = -ENOMEM;
goto abort1;
}
@@ -526,14 +521,12 @@ static int wm0010_stage2_load(struct snd_soc_codec *codec)
/* Copy to local buffer first as vmalloc causes problems for dma */
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 | GFP_DMA);
if (!out) {
- dev_err(codec->dev, "Failed to allocate output buffer\n");
ret = -ENOMEM;
goto abort1;
}
@@ -679,11 +672,8 @@ static int wm0010_boot(struct snd_soc_codec *codec)
}
img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA);
- if (!img_swap) {
- dev_err(codec->dev,
- "Failed to allocate image buffer\n");
+ if (!img_swap)
goto abort;
- }
/* We need to re-order for 0010 */
byte_swap_64((u64 *)&pll_rec, img_swap, len);
diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c
index 6e6b93d4696e..8011f75fb6cb 100644
--- a/sound/soc/codecs/wm1250-ev1.c
+++ b/sound/soc/codecs/wm1250-ev1.c
@@ -164,7 +164,6 @@ static int wm1250_ev1_pdata(struct i2c_client *i2c)
wm1250 = devm_kzalloc(&i2c->dev, sizeof(*wm1250), GFP_KERNEL);
if (!wm1250) {
- dev_err(&i2c->dev, "Unable to allocate private data\n");
ret = -ENOMEM;
goto err;
}
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index a4c352cc3464..34ef65c52a7d 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -826,10 +826,8 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
wm2000 = devm_kzalloc(&i2c->dev, sizeof(struct wm2000_priv),
GFP_KERNEL);
- if (wm2000 == NULL) {
- dev_err(&i2c->dev, "Unable to allocate private data\n");
+ if (!wm2000)
return -ENOMEM;
- }
mutex_init(&wm2000->lock);
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 91a9ea2a2056..7bb0d36d4c54 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -735,8 +735,7 @@ WM5100_MIXER_CONTROLS("LHPF4", WM5100_HPLP4MIX_INPUT_1_SOURCE),
static void wm5100_seq_notifier(struct snd_soc_dapm_context *dapm,
enum snd_soc_dapm_type event, int subseq)
{
- struct snd_soc_codec *codec = container_of(dapm,
- struct snd_soc_codec, dapm);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec);
u16 val, expect, i;
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 289b64d89abd..f60234962527 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -612,6 +612,62 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w,
return 0;
}
+static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+ uint16_t data;
+
+ mutex_lock(&codec->mutex);
+ data = cpu_to_be16(arizona->dac_comp_coeff);
+ memcpy(ucontrol->value.bytes.data, &data, sizeof(data));
+ mutex_unlock(&codec->mutex);
+
+ return 0;
+}
+
+static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+
+ mutex_lock(&codec->mutex);
+ memcpy(&arizona->dac_comp_coeff, ucontrol->value.bytes.data,
+ sizeof(arizona->dac_comp_coeff));
+ arizona->dac_comp_coeff = be16_to_cpu(arizona->dac_comp_coeff);
+ mutex_unlock(&codec->mutex);
+
+ return 0;
+}
+
+static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+
+ mutex_lock(&codec->mutex);
+ ucontrol->value.integer.value[0] = arizona->dac_comp_enabled;
+ mutex_unlock(&codec->mutex);
+
+ return 0;
+}
+
+static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+
+ mutex_lock(&codec->mutex);
+ arizona->dac_comp_enabled = ucontrol->value.integer.value[0];
+ mutex_unlock(&codec->mutex);
+
+ return 0;
+}
+
static const char *wm5102_osr_text[] = {
"Low power", "Normal", "High performance",
};
@@ -843,6 +899,12 @@ 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),
+SND_SOC_BYTES_EXT("Output Compensation Coefficient", 2,
+ wm5102_out_comp_coeff_get, wm5102_out_comp_coeff_put),
+
+SOC_SINGLE_EXT("Output Compensation Switch", 0, 0, 1, 0,
+ wm5102_out_comp_switch_get, wm5102_out_comp_switch_put),
+
WM5102_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L),
WM5102_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R),
WM5102_NG_SRC("HPOUT2L", ARIZONA_NOISE_GATE_SELECT_2L),
@@ -1653,6 +1715,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = {
},
.ops = &arizona_dai_ops,
.symmetric_rates = 1,
+ .symmetric_samplebits = 1,
},
{
.name = "wm5102-aif2",
@@ -1674,6 +1737,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = {
},
.ops = &arizona_dai_ops,
.symmetric_rates = 1,
+ .symmetric_samplebits = 1,
},
{
.name = "wm5102-aif3",
@@ -1695,6 +1759,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = {
},
.ops = &arizona_dai_ops,
.symmetric_rates = 1,
+ .symmetric_samplebits = 1,
},
{
.name = "wm5102-slim1",
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 2e5fcb559e90..2f2ec26d831c 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -1485,6 +1485,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = {
},
.ops = &arizona_dai_ops,
.symmetric_rates = 1,
+ .symmetric_samplebits = 1,
},
{
.name = "wm5110-aif2",
@@ -1506,6 +1507,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = {
},
.ops = &arizona_dai_ops,
.symmetric_rates = 1,
+ .symmetric_samplebits = 1,
},
{
.name = "wm5110-aif3",
@@ -1527,6 +1529,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = {
},
.ops = &arizona_dai_ops,
.symmetric_rates = 1,
+ .symmetric_samplebits = 1,
},
{
.name = "wm5110-slim1",
@@ -1596,6 +1599,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
arizona_init_spk(codec);
arizona_init_gpio(codec);
+ arizona_init_mono(codec);
ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8);
if (ret != 0)
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 392285edb595..3dfdcc4197fa 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -918,16 +918,16 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
~WM8350_AIF_WL_MASK;
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface |= 0x1 << 10;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface |= 0x2 << 10;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
iface |= 0x3 << 10;
break;
}
@@ -1341,21 +1341,18 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
{
struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
struct wm8350 *wm8350 = priv->wm8350;
- int irq;
int ena;
switch (which) {
case WM8350_JDL:
priv->hpl.jack = jack;
priv->hpl.report = report;
- irq = WM8350_IRQ_CODEC_JCK_DET_L;
ena = WM8350_JDL_ENA;
break;
case WM8350_JDR:
priv->hpr.jack = jack;
priv->hpr.report = report;
- irq = WM8350_IRQ_CODEC_JCK_DET_R;
ena = WM8350_JDR_ENA;
break;
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 06e913d3fea1..72471bef2e9a 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1095,16 +1095,16 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream,
audio1 &= ~WM8400_AIF_WL_MASK;
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
audio1 |= WM8400_AIF_WL_20BITS;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
audio1 |= WM8400_AIF_WL_24BITS;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
audio1 |= WM8400_AIF_WL_32BITS;
break;
}
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 1c1e328feeb8..e11127f9069e 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -449,16 +449,16 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1;
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface |= 0x0020;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface |= 0x0040;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
iface |= 0x0060;
break;
}
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 601ee8178af1..ec1f5740dbd0 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -163,16 +163,16 @@ static int wm8523_hw_params(struct snd_pcm_substream *substream,
aifctrl2 |= lrclk_ratios[i].value;
aifctrl1 &= ~WM8523_WL_MASK;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
aifctrl1 |= 0x8;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
aifctrl1 |= 0x10;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
aifctrl1 |= 0x18;
break;
}
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 7665ff6aea6d..911605ee25b0 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -511,19 +511,19 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
int i, ratio, osr;
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
paifa |= 0x8;
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
paifa |= 0x0;
paifb |= WM8580_AIF_LENGTH_20;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
paifa |= 0x0;
paifb |= WM8580_AIF_LENGTH_24;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
paifa |= 0x0;
paifb |= WM8580_AIF_LENGTH_32;
break;
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index b0fbcb377baf..32187e739b4f 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -169,13 +169,13 @@ static int wm8711_hw_params(struct snd_pcm_substream *substream,
snd_soc_write(codec, WM8711_SRATE, srate);
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface |= 0x0004;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface |= 0x0008;
break;
}
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index bac7fc28fe71..38ff826f589a 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -94,13 +94,13 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream,
dac &= ~0x18;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
dac |= 0x10;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
dac |= 0x08;
break;
default:
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 5ada61611324..eebb3280bfad 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -348,13 +348,13 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
snd_soc_write(codec, WM8731_SRATE, srate);
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface |= 0x0004;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface |= 0x0008;
break;
}
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index b27f26cdc049..744a422ecb05 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -367,16 +367,16 @@ static int wm8737_hw_params(struct snd_pcm_substream *substream,
clocking |= coeff_div[i].usb | (coeff_div[i].sr << WM8737_SR_SHIFT);
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
af |= 0x8;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
af |= 0x10;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
af |= 0x18;
break;
default:
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index b33542a04607..a237f1627f61 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -241,26 +241,26 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream,
}
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface |= 0x0001;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface |= 0x0002;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
iface |= 0x0003;
break;
default:
dev_dbg(codec->dev, "wm8741_hw_params: Unsupported bit size param = %d",
- params_format(params));
+ params_width(params));
return -EINVAL;
}
dev_dbg(codec->dev, "wm8741_hw_params: bit size param = %d",
- params_format(params));
+ params_width(params));
snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface);
return 0;
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 33990b63d214..67653a2db223 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -586,16 +586,16 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
int coeff = get_coeff(wm8750->sysclk, params_rate(params));
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface |= 0x0004;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface |= 0x0008;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
iface |= 0x000c;
break;
}
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 53e57b4049a8..e54e097f4fcb 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -937,16 +937,16 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x017f;
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
voice |= 0x0004;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
voice |= 0x0008;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
voice |= 0x000c;
break;
}
@@ -1176,16 +1176,16 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
coeff_div[coeff].usb);
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
hifi |= 0x0004;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
hifi |= 0x0008;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
hifi |= 0x000c;
break;
}
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index c61aeb38efb8..180e7a098726 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -426,16 +426,16 @@ static int wm8770_hw_params(struct snd_pcm_substream *substream,
wm8770 = snd_soc_codec_get_drvdata(codec);
iface = 0;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface |= 0x10;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface |= 0x20;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
iface |= 0x30;
break;
}
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index d96e5963ee35..0ea01dfcb6e1 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -270,19 +270,19 @@ static int wm8804_hw_params(struct snd_pcm_substream *substream,
codec = dai->codec;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
blen = 0x0;
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
blen = 0x1;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
blen = 0x2;
break;
default:
dev_err(dai->dev, "Unsupported word length: %u\n",
- params_format(params));
+ params_width(params));
return -EINVAL;
}
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index d09fdce57f5a..44a5f1511f0f 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -640,16 +640,16 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream,
reg = snd_soc_read(codec, WM8900_REG_AUDIO1) & ~0x60;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
reg |= 0x20;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
reg |= 0x40;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
reg |= 0x60;
break;
default:
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index b84940c359a1..aa0984864e76 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -281,8 +281,7 @@ static int wm8903_dcs_event(struct snd_soc_dapm_widget *w,
static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm,
enum snd_soc_dapm_type event, int subseq)
{
- struct snd_soc_codec *codec = container_of(dapm,
- struct snd_soc_codec, dapm);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
int dcs_mode = WM8903_DCS_MODE_WRITE_STOP;
int i, val;
@@ -1477,19 +1476,19 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
aif1 &= ~WM8903_AIF_WL_MASK;
bclk = 2 * fs;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
bclk *= 16;
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
bclk *= 20;
aif1 |= 0x4;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
bclk *= 24;
aif1 |= 0x8;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
bclk *= 32;
aif1 |= 0xc;
break;
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index f7c549949c54..4d2d2b1380d5 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -11,6 +11,7 @@
* published by the Free Software Foundation.
*/
+#include <linux/clk.h>
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
@@ -49,6 +50,7 @@ static const char *wm8904_supply_names[WM8904_NUM_SUPPLIES] = {
/* codec private data */
struct wm8904_priv {
struct regmap *regmap;
+ struct clk *mclk;
enum wm8904_type devtype;
@@ -1290,16 +1292,16 @@ static int wm8904_hw_params(struct snd_pcm_substream *substream,
wm8904->bclk = snd_soc_params_to_bclk(params);
}
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
aif1 |= 0x40;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
aif1 |= 0x80;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
aif1 |= 0xc0;
break;
default:
@@ -1828,6 +1830,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
switch (level) {
case SND_SOC_BIAS_ON:
+ clk_prepare_enable(wm8904->mclk);
break;
case SND_SOC_BIAS_PREPARE:
@@ -1894,6 +1897,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies),
wm8904->supplies);
+ clk_disable_unprepare(wm8904->mclk);
break;
}
codec->dapm.bias_level = level;
@@ -2013,12 +2017,8 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec)
/* We need an array of texts for the enum API */
wm8904->drc_texts = kmalloc(sizeof(char *)
* pdata->num_drc_cfgs, GFP_KERNEL);
- if (!wm8904->drc_texts) {
- dev_err(codec->dev,
- "Failed to allocate %d DRC config texts\n",
- pdata->num_drc_cfgs);
+ if (!wm8904->drc_texts)
return;
- }
for (i = 0; i < pdata->num_drc_cfgs; i++)
wm8904->drc_texts[i] = pdata->drc_cfgs[i].name;
@@ -2110,6 +2110,13 @@ static int wm8904_i2c_probe(struct i2c_client *i2c,
if (wm8904 == NULL)
return -ENOMEM;
+ wm8904->mclk = devm_clk_get(&i2c->dev, "mclk");
+ if (IS_ERR(wm8904->mclk)) {
+ ret = PTR_ERR(wm8904->mclk);
+ dev_err(&i2c->dev, "Failed to get MCLK\n");
+ return ret;
+ }
+
wm8904->regmap = devm_regmap_init_i2c(i2c, &wm8904_regmap);
if (IS_ERR(wm8904->regmap)) {
ret = PTR_ERR(wm8904->regmap);
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index fc6eec9ad66b..52011043e54c 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -430,19 +430,19 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream,
if (ret)
goto error_ret;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S8:
+ switch (params_width(params)) {
+ case 8:
companding = companding | (1 << 5);
break;
- case SNDRV_PCM_FORMAT_S16_LE:
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface |= (1 << 5);
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface |= (2 << 5);
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
iface |= (3 << 5);
break;
}
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 2a35108f233d..09d91d9dc4ee 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -597,17 +597,17 @@ static int wm8955_hw_params(struct snd_pcm_substream *substream,
int ret;
int wl;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
wl = 0;
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
wl = 0x4;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
wl = 0x8;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
wl = 0xc;
break;
default:
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index b2ebb104d879..0dada7f0105e 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -934,12 +934,8 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec)
/* We need an array of texts for the enum API */
wm8994->mbc_texts = kmalloc(sizeof(char *)
* pdata->num_mbc_cfgs, GFP_KERNEL);
- if (!wm8994->mbc_texts) {
- dev_err(wm8994->hubs.codec->dev,
- "Failed to allocate %d MBC config texts\n",
- pdata->num_mbc_cfgs);
+ if (!wm8994->mbc_texts)
return;
- }
for (i = 0; i < pdata->num_mbc_cfgs; i++)
wm8994->mbc_texts[i] = pdata->mbc_cfgs[i].name;
@@ -963,12 +959,8 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec)
/* We need an array of texts for the enum API */
wm8994->vss_texts = kmalloc(sizeof(char *)
* pdata->num_vss_cfgs, GFP_KERNEL);
- if (!wm8994->vss_texts) {
- dev_err(wm8994->hubs.codec->dev,
- "Failed to allocate %d VSS config texts\n",
- pdata->num_vss_cfgs);
+ if (!wm8994->vss_texts)
return;
- }
for (i = 0; i < pdata->num_vss_cfgs; i++)
wm8994->vss_texts[i] = pdata->vss_cfgs[i].name;
@@ -993,12 +985,8 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec)
/* We need an array of texts for the enum API */
wm8994->vss_hpf_texts = kmalloc(sizeof(char *)
* pdata->num_vss_hpf_cfgs, GFP_KERNEL);
- if (!wm8994->vss_hpf_texts) {
- dev_err(wm8994->hubs.codec->dev,
- "Failed to allocate %d VSS HPF config texts\n",
- pdata->num_vss_hpf_cfgs);
+ if (!wm8994->vss_hpf_texts)
return;
- }
for (i = 0; i < pdata->num_vss_hpf_cfgs; i++)
wm8994->vss_hpf_texts[i] = pdata->vss_hpf_cfgs[i].name;
@@ -1024,12 +1012,8 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec)
/* We need an array of texts for the enum API */
wm8994->enh_eq_texts = kmalloc(sizeof(char *)
* pdata->num_enh_eq_cfgs, GFP_KERNEL);
- if (!wm8994->enh_eq_texts) {
- dev_err(wm8994->hubs.codec->dev,
- "Failed to allocate %d enhanced EQ config texts\n",
- pdata->num_enh_eq_cfgs);
+ if (!wm8994->enh_eq_texts)
return;
- }
for (i = 0; i < pdata->num_enh_eq_cfgs; i++)
wm8994->enh_eq_texts[i] = pdata->enh_eq_cfgs[i].name;
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index a145d0431b63..4dc4e85116cd 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -472,7 +472,7 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec)
* list each time to find the desired power state do so now
* and save the result.
*/
- list_for_each_entry(w, &codec->card->widgets, list) {
+ list_for_each_entry(w, &codec->component.card->widgets, list) {
if (w->dapm != &codec->dapm)
continue;
if (strcmp(w->name, "LOUT1 PGA") == 0)
@@ -567,24 +567,21 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = dai->codec;
struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3;
- snd_pcm_format_t format = params_format(params);
int i;
/* bit size */
- switch (format) {
- case SNDRV_PCM_FORMAT_S16_LE:
- case SNDRV_PCM_FORMAT_S16_BE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
- case SNDRV_PCM_FORMAT_S20_3BE:
+ case 20:
iface |= 0x0004;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
- case SNDRV_PCM_FORMAT_S24_BE:
+ case 24:
iface |= 0x0008;
break;
default:
- dev_err(codec->dev, "unsupported format %i\n", format);
+ dev_err(codec->dev, "unsupported width %d\n",
+ params_width(params));
return -EINVAL;
}
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 9c88f04442b3..41d23e920ad5 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -565,16 +565,16 @@ static int wm8961_hw_params(struct snd_pcm_substream *substream,
reg = snd_soc_read(codec, WM8961_AUDIO_INTERFACE_0);
reg &= ~WM8961_WL_MASK;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
reg |= 1 << WM8961_WL_SHIFT;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
reg |= 2 << WM8961_WL_SHIFT;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
reg |= 3 << WM8961_WL_SHIFT;
break;
default:
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index ca2fda9d72be..1098ae32f1f9 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -14,6 +14,7 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
+#include <linux/clk.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/gcd.h>
@@ -2586,16 +2587,16 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream,
if (wm8962->lrclk % 8000 == 0)
adctl3 |= WM8962_SAMPLE_RATE_INT_MODE;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
aif0 |= 0x4;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
aif0 |= 0x8;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
aif0 |= 0xc;
break;
default:
@@ -3541,6 +3542,8 @@ static int wm8962_set_pdata_from_of(struct i2c_client *i2c,
pdata->gpio_init[i] = 0x0;
}
+ pdata->mclk = devm_clk_get(&i2c->dev, NULL);
+
return 0;
}
@@ -3572,6 +3575,14 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
return ret;
}
+ /* Mark the mclk pointer to NULL if no mclk assigned */
+ if (IS_ERR(wm8962->pdata.mclk)) {
+ /* But do not ignore the request for probe defer */
+ if (PTR_ERR(wm8962->pdata.mclk) == -EPROBE_DEFER)
+ return -EPROBE_DEFER;
+ wm8962->pdata.mclk = NULL;
+ }
+
for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++)
wm8962->supplies[i].supply = wm8962_supply_names[i];
@@ -3780,6 +3791,12 @@ static int wm8962_runtime_resume(struct device *dev)
struct wm8962_priv *wm8962 = dev_get_drvdata(dev);
int ret;
+ ret = clk_prepare_enable(wm8962->pdata.mclk);
+ if (ret) {
+ dev_err(dev, "Failed to enable MCLK: %d\n", ret);
+ return ret;
+ }
+
ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies),
wm8962->supplies);
if (ret != 0) {
@@ -3839,6 +3856,8 @@ static int wm8962_runtime_suspend(struct device *dev)
regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies),
wm8962->supplies);
+ clk_disable_unprepare(wm8962->pdata.mclk);
+
return 0;
}
#endif
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 09b7b4200221..0499cd4cfb71 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -517,16 +517,16 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream,
int coeff = get_coeff(wm8971->sysclk, params_rate(params));
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface |= 0x0004;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface |= 0x0008;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
iface |= 0x000c;
break;
}
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 0627c56fa44e..682e9eda1019 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -445,16 +445,16 @@ static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream,
u16 adn = snd_soc_read(codec, WM8974_ADD) & 0x1f1;
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface |= 0x0020;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface |= 0x0040;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
iface |= 0x0060;
break;
}
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 28ef46c91f62..ee2ba574952b 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -736,16 +736,16 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface_ctl |= 0x20;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface_ctl |= 0x40;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
iface_ctl |= 0x60;
break;
}
@@ -817,8 +817,8 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream,
wm8978->sysclk == WM8978_MCLK ?
", consider using PLL" : "");
- dev_dbg(codec->dev, "%s: fmt %d, rate %u, MCLK divisor #%d\n", __func__,
- params_format(params), params_rate(params), best);
+ dev_dbg(codec->dev, "%s: width %d, rate %u, MCLK divisor #%d\n", __func__,
+ params_width(params), params_rate(params), best);
/* MCLK divisor mask = 0xe0 */
snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, best << 5);
diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c
index 19d5baa38f5c..ac5defda8824 100644
--- a/sound/soc/codecs/wm8983.c
+++ b/sound/soc/codecs/wm8983.c
@@ -719,22 +719,22 @@ static int wm8983_hw_params(struct snd_pcm_substream *substream,
wm8983->bclk = ret;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
blen = 0x0;
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
blen = 0x1;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
blen = 0x2;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
blen = 0x3;
break;
default:
dev_err(dai->dev, "Unsupported word length %u\n",
- params_format(params));
+ params_width(params));
return -EINVAL;
}
diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c
index 0f5780c09f3a..ee380190399f 100644
--- a/sound/soc/codecs/wm8985.c
+++ b/sound/soc/codecs/wm8985.c
@@ -698,22 +698,22 @@ static int wm8985_hw_params(struct snd_pcm_substream *substream,
if ((int)wm8985->bclk < 0)
return wm8985->bclk;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
blen = 0x0;
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
blen = 0x1;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
blen = 0x2;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
blen = 0x3;
break;
default:
dev_err(dai->dev, "Unsupported word length %u\n",
- params_format(params));
+ params_width(params));
return -EINVAL;
}
@@ -980,9 +980,6 @@ static int wm8985_resume(struct snd_soc_codec *codec)
static int wm8985_remove(struct snd_soc_codec *codec)
{
- struct wm8985_priv *wm8985;
-
- wm8985 = snd_soc_codec_get_drvdata(codec);
wm8985_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index d3fea46d58e8..a5130d965146 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -687,16 +687,16 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream,
}
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
iface |= 0x0004;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
iface |= 0x0008;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
iface |= 0x000c;
break;
}
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index b5c1f0f07058..03e43e3f395e 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1073,16 +1073,16 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream,
audio1 &= ~WM8990_AIF_WL_MASK;
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
audio1 |= WM8990_AIF_WL_20BITS;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
audio1 |= WM8990_AIF_WL_24BITS;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
audio1 |= WM8990_AIF_WL_32BITS;
break;
}
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index b8fd284fc0c0..d0be89731cdb 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -1081,16 +1081,16 @@ static int wm8991_hw_params(struct snd_pcm_substream *substream,
audio1 &= ~WM8991_AIF_WL_MASK;
/* bit size */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
audio1 |= WM8991_AIF_WL_20BITS;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
audio1 |= WM8991_AIF_WL_24BITS;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
audio1 |= WM8991_AIF_WL_32BITS;
break;
}
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index f825dc04ebe1..93b14eda355a 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1214,19 +1214,19 @@ static int wm8993_hw_params(struct snd_pcm_substream *substream,
wm8993->tdm_slots, wm8993->tdm_width);
wm8993->bclk *= wm8993->tdm_width * wm8993->tdm_slots;
} else {
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
wm8993->bclk *= 16;
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
wm8993->bclk *= 20;
aif1 |= 0x8;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
wm8993->bclk *= 24;
aif1 |= 0x10;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
wm8993->bclk *= 32;
aif1 |= 0x18;
break;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 247b39013fba..6cc0566dc29a 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2815,19 +2815,19 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
}
bclk_rate = params_rate(params);
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
bclk_rate *= 16;
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
bclk_rate *= 20;
aif1 |= 0x20;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
bclk_rate *= 24;
aif1 |= 0x40;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
bclk_rate *= 32;
aif1 |= 0x60;
break;
@@ -2966,16 +2966,16 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream,
return 0;
}
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
aif1 |= 0x20;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
aif1 |= 0x40;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
aif1 |= 0x60;
break;
default:
@@ -3296,12 +3296,8 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994)
/* We need an array of texts for the enum API */
wm8994->drc_texts = devm_kzalloc(wm8994->hubs.codec->dev,
sizeof(char *) * pdata->num_drc_cfgs, GFP_KERNEL);
- if (!wm8994->drc_texts) {
- dev_err(wm8994->hubs.codec->dev,
- "Failed to allocate %d DRC config texts\n",
- pdata->num_drc_cfgs);
+ if (!wm8994->drc_texts)
return;
- }
for (i = 0; i < pdata->num_drc_cfgs; i++)
wm8994->drc_texts[i] = pdata->drc_cfgs[i].name;
@@ -3505,6 +3501,7 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data)
return IRQ_HANDLED;
}
+/* Should be called with accdet_lock held */
static void wm1811_micd_stop(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
@@ -3512,14 +3509,10 @@ static void wm1811_micd_stop(struct snd_soc_codec *codec)
if (!wm8994->jackdet)
return;
- mutex_lock(&wm8994->accdet_lock);
-
snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0);
wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_JACK);
- mutex_unlock(&wm8994->accdet_lock);
-
if (wm8994->wm8994->pdata.jd_ext_cap)
snd_soc_dapm_disable_pin(&codec->dapm,
"MICBIAS2");
@@ -3560,10 +3553,10 @@ static void wm8958_open_circuit_work(struct work_struct *work)
open_circuit_work.work);
struct device *dev = wm8994->wm8994->dev;
- wm1811_micd_stop(wm8994->hubs.codec);
-
mutex_lock(&wm8994->accdet_lock);
+ wm1811_micd_stop(wm8994->hubs.codec);
+
dev_dbg(dev, "Reporting open circuit\n");
wm8994->jack_mic = false;
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 863a2c38bcb5..cae4ac5a5730 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1597,21 +1597,21 @@ static int wm8995_hw_params(struct snd_pcm_substream *substream,
return bclk_rate;
aif1 = 0;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
aif1 |= (0x1 << WM8995_AIF1_WL_SHIFT);
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
aif1 |= (0x2 << WM8995_AIF1_WL_SHIFT);
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
aif1 |= (0x3 << WM8995_AIF1_WL_SHIFT);
break;
default:
dev_err(dai->dev, "Unsupported word length %u\n",
- params_format(params));
+ params_width(params));
return -EINVAL;
}
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 69266332760e..f16ff4f56923 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -620,15 +620,12 @@ static int bg_event(struct snd_soc_dapm_widget *w,
static int cp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- int ret = 0;
-
switch (event) {
case SND_SOC_DAPM_POST_PMU:
msleep(5);
break;
default:
WARN(1, "Invalid event %d\n", event);
- ret = -EINVAL;
}
return 0;
@@ -690,8 +687,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask)
static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm,
enum snd_soc_dapm_type event, int subseq)
{
- struct snd_soc_codec *codec = container_of(dapm,
- struct snd_soc_codec, dapm);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
u16 val, mask;
diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c
index bb9b47b956aa..ab33fe596519 100644
--- a/sound/soc/codecs/wm8997.c
+++ b/sound/soc/codecs/wm8997.c
@@ -967,6 +967,7 @@ static struct snd_soc_dai_driver wm8997_dai[] = {
},
.ops = &arizona_dai_ops,
.symmetric_rates = 1,
+ .symmetric_samplebits = 1,
},
{
.name = "wm8997-aif2",
@@ -988,6 +989,7 @@ static struct snd_soc_dai_driver wm8997_dai[] = {
},
.ops = &arizona_dai_ops,
.symmetric_rates = 1,
+ .symmetric_samplebits = 1,
},
{
.name = "wm8997-slim1",
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 185eb97769e7..0cdc9e2184ab 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1029,19 +1029,19 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream,
/* Otherwise work out a BCLK from the sample size */
wm9081->bclk = 2 * wm9081->fs;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
wm9081->bclk *= 16;
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
wm9081->bclk *= 20;
aif2 |= 0x4;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
wm9081->bclk *= 24;
aif2 |= 0x8;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
wm9081->bclk *= 32;
aif2 |= 0xc;
break;
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 87934171f063..a13f0725611a 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -613,10 +613,8 @@ static int wm9090_i2c_probe(struct i2c_client *i2c,
int ret;
wm9090 = devm_kzalloc(&i2c->dev, sizeof(*wm9090), GFP_KERNEL);
- if (wm9090 == NULL) {
- dev_err(&i2c->dev, "Can not allocate memory\n");
+ if (!wm9090)
return -ENOMEM;
- }
wm9090->regmap = devm_regmap_init_i2c(i2c, &wm9090_regmap);
if (IS_ERR(wm9090->regmap)) {
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 2a9c6d11330c..bddee30a4bc7 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -953,16 +953,16 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = dai->codec;
u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
+ switch (params_width(params)) {
+ case 16:
break;
- case SNDRV_PCM_FORMAT_S20_3LE:
+ case 20:
reg |= 0x0004;
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case 24:
reg |= 0x0008;
break;
- case SNDRV_PCM_FORMAT_S32_LE:
+ case 32:
reg |= 0x000c;
break;
}
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 060027182dcb..f412a9911a75 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1382,7 +1382,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
int ret;
int val;
- dsp->card = codec->card;
+ dsp->card = codec->component.card;
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -1617,7 +1617,7 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w,
struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
struct wm_adsp *dsp = &dsps[w->shift];
- dsp->card = codec->card;
+ dsp->card = codec->component.card;
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
@@ -1758,3 +1758,5 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs)
return 0;
}
EXPORT_SYMBOL_GPL(wm_adsp2_init);
+
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 916817fe6632..374537d5e179 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -183,10 +183,8 @@ static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg)
return;
cache = devm_kzalloc(codec->dev, sizeof(*cache), GFP_KERNEL);
- if (!cache) {
- dev_err(codec->dev, "Failed to allocate DCS cache entry\n");
+ if (!cache)
return;
- }
cache->left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME);
cache->left &= WM8993_HPOUT1L_VOL_MASK;
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 50a098749b9e..d69510c53239 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -1,12 +1,29 @@
config SND_DAVINCI_SOC
- tristate "SoC Audio for TI DAVINCI or AM33XX/AM43XX chips"
- depends on ARCH_DAVINCI || SOC_AM33XX || SOC_AM43XX
+ tristate "SoC Audio for TI DAVINCI"
+ depends on ARCH_DAVINCI
+
+config SND_EDMA_SOC
+ tristate "SoC Audio for Texas Instruments chips using eDMA (AM33XX/43XX)"
+ depends on SOC_AM33XX || SOC_AM43XX
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+ help
+ Say Y or M here if you want audio support for TI SoC which uses eDMA.
+ The following line of SoCs are supported by this platform driver:
+ - AM335x
+ - AM437x/AM438x
config SND_DAVINCI_SOC_I2S
tristate
config SND_DAVINCI_SOC_MCASP
- tristate
+ tristate "Multichannel Audio Serial Port (McASP) support"
+ depends on SND_DAVINCI_SOC || SND_OMAP_SOC || SND_EDMA_SOC
+ help
+ Say Y or M here if you want to have support for McASP IP found in
+ various Texas Instruments SoCs like:
+ - daVinci devices
+ - Sitara line of SoCs (AM335x, AM438x, etc)
+ - DRA7x devices
config SND_DAVINCI_SOC_VCIF
tristate
@@ -18,7 +35,7 @@ config SND_DAVINCI_SOC_GENERIC_EVM
config SND_AM33XX_SOC_EVM
tristate "SoC Audio for the AM33XX chip based boards"
- depends on SND_DAVINCI_SOC && SOC_AM33XX && I2C
+ depends on SND_EDMA_SOC && SOC_AM33XX && I2C
select SND_DAVINCI_SOC_GENERIC_EVM
help
Say Y or M if you want to add support for SoC audio on AM33XX
diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile
index 744d4d9a0184..09bf2ba92d38 100644
--- a/sound/soc/davinci/Makefile
+++ b/sound/soc/davinci/Makefile
@@ -1,10 +1,12 @@
# DAVINCI Platform Support
snd-soc-davinci-objs := davinci-pcm.o
+snd-soc-edma-objs := edma-pcm.o
snd-soc-davinci-i2s-objs := davinci-i2s.o
snd-soc-davinci-mcasp-objs:= davinci-mcasp.o
snd-soc-davinci-vcif-objs:= davinci-vcif.o
obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o
+obj-$(CONFIG_SND_EDMA_SOC) += snd-soc-edma.o
obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o
obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o
obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 9afb14629a17..c28508da34cf 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -27,6 +27,7 @@
#include <linux/of_platform.h>
#include <linux/of_device.h>
+#include <sound/asoundef.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -36,6 +37,7 @@
#include <sound/omap-pcm.h>
#include "davinci-pcm.h"
+#include "edma-pcm.h"
#include "davinci-mcasp.h"
#define MCASP_MAX_AFIFO_DEPTH 64
@@ -63,6 +65,7 @@ struct davinci_mcasp {
u8 num_serializer;
u8 *serial_dir;
u8 version;
+ u8 bclk_div;
u16 bclk_lrclk_ratio;
int streams;
@@ -417,6 +420,7 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div
ACLKXDIV(div - 1), ACLKXDIV_MASK);
mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG,
ACLKRDIV(div - 1), ACLKRDIV_MASK);
+ mcasp->bclk_div = div;
break;
case 2: /* BCLK/LRCLK ratio */
@@ -637,8 +641,12 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream)
}
/* S/PDIF */
-static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp)
+static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp,
+ unsigned int rate)
{
+ u32 cs_value = 0;
+ u8 *cs_bytes = (u8*) &cs_value;
+
/* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0
and LSB first */
mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15));
@@ -660,6 +668,46 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp)
/* Enable the DIT */
mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN);
+ /* Set S/PDIF channel status bits */
+ cs_bytes[0] = IEC958_AES0_CON_NOT_COPYRIGHT;
+ cs_bytes[1] = IEC958_AES1_CON_PCM_CODER;
+
+ switch (rate) {
+ case 22050:
+ cs_bytes[3] |= IEC958_AES3_CON_FS_22050;
+ break;
+ case 24000:
+ cs_bytes[3] |= IEC958_AES3_CON_FS_24000;
+ break;
+ case 32000:
+ cs_bytes[3] |= IEC958_AES3_CON_FS_32000;
+ break;
+ case 44100:
+ cs_bytes[3] |= IEC958_AES3_CON_FS_44100;
+ break;
+ case 48000:
+ cs_bytes[3] |= IEC958_AES3_CON_FS_48000;
+ break;
+ case 88200:
+ cs_bytes[3] |= IEC958_AES3_CON_FS_88200;
+ break;
+ case 96000:
+ cs_bytes[3] |= IEC958_AES3_CON_FS_96000;
+ break;
+ case 176400:
+ cs_bytes[3] |= IEC958_AES3_CON_FS_176400;
+ break;
+ case 192000:
+ cs_bytes[3] |= IEC958_AES3_CON_FS_192000;
+ break;
+ default:
+ printk(KERN_WARNING "unsupported sampling rate: %d\n", rate);
+ return -EINVAL;
+ }
+
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, cs_value);
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, cs_value);
+
return 0;
}
@@ -675,15 +723,22 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
int period_size = params_period_size(params);
int ret;
- /* If mcasp is BCLK master we need to set BCLK divider */
- if (mcasp->bclk_master) {
+ /*
+ * If mcasp is BCLK master, and a BCLK divider was not provided by
+ * the machine driver, we need to calculate the ratio.
+ */
+ if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) {
unsigned int bclk_freq = snd_soc_params_to_bclk(params);
+ unsigned int div = mcasp->sysclk_freq / bclk_freq;
if (mcasp->sysclk_freq % bclk_freq != 0) {
- dev_err(mcasp->dev, "Can't produce required BCLK\n");
- return -EINVAL;
+ if (((mcasp->sysclk_freq / div) - bclk_freq) >
+ (bclk_freq - (mcasp->sysclk_freq / (div+1))))
+ div++;
+ dev_warn(mcasp->dev,
+ "Inaccurate BCLK: %u Hz / %u != %u Hz\n",
+ mcasp->sysclk_freq, div, bclk_freq);
}
- davinci_mcasp_set_clkdiv(
- cpu_dai, 1, mcasp->sysclk_freq / bclk_freq);
+ davinci_mcasp_set_clkdiv(cpu_dai, 1, div);
}
ret = mcasp_common_hw_param(mcasp, substream->stream,
@@ -692,7 +747,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
return ret;
if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
- ret = mcasp_dit_hw_param(mcasp);
+ ret = mcasp_dit_hw_param(mcasp, params_rate(params));
else
ret = mcasp_i2s_hw_param(mcasp, substream->stream);
@@ -720,6 +775,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
case SNDRV_PCM_FORMAT_U24_LE:
case SNDRV_PCM_FORMAT_S24_LE:
+ dma_params->data_type = 4;
+ word_length = 24;
+ break;
+
case SNDRV_PCM_FORMAT_U32_LE:
case SNDRV_PCM_FORMAT_S32_LE:
dma_params->data_type = 4;
@@ -778,7 +837,7 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai)
{
struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
- if (mcasp->version == MCASP_VERSION_4) {
+ if (mcasp->version >= MCASP_VERSION_3) {
/* Using dmaengine PCM */
dai->playback_dma_data =
&mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
@@ -1223,14 +1282,28 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
goto err;
switch (mcasp->version) {
+#if IS_BUILTIN(CONFIG_SND_DAVINCI_SOC) || \
+ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \
+ IS_MODULE(CONFIG_SND_DAVINCI_SOC))
case MCASP_VERSION_1:
case MCASP_VERSION_2:
- case MCASP_VERSION_3:
ret = davinci_soc_platform_register(&pdev->dev);
break;
+#endif
+#if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \
+ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \
+ IS_MODULE(CONFIG_SND_EDMA_SOC))
+ case MCASP_VERSION_3:
+ ret = edma_pcm_platform_register(&pdev->dev);
+ break;
+#endif
+#if IS_BUILTIN(CONFIG_SND_OMAP_SOC) || \
+ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \
+ IS_MODULE(CONFIG_SND_OMAP_SOC))
case MCASP_VERSION_4:
ret = omap_pcm_platform_register(&pdev->dev);
break;
+#endif
default:
dev_err(&pdev->dev, "Invalid McASP version: %d\n",
mcasp->version);
diff --git a/sound/soc/davinci/edma-pcm.c b/sound/soc/davinci/edma-pcm.c
index d38afb1c61ae..605e643133db 100644
--- a/sound/soc/davinci/edma-pcm.c
+++ b/sound/soc/davinci/edma-pcm.c
@@ -28,8 +28,8 @@
static const struct snd_pcm_hardware edma_pcm_hardware = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_BATCH |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_NO_PERIOD_WAKEUP |
SNDRV_PCM_INFO_INTERLEAVED,
.buffer_bytes_max = 128 * 1024,
.period_bytes_min = 32,
diff --git a/sound/soc/davinci/edma-pcm.h b/sound/soc/davinci/edma-pcm.h
index 894c378c0f74..b0957744851c 100644
--- a/sound/soc/davinci/edma-pcm.h
+++ b/sound/soc/davinci/edma-pcm.h
@@ -20,6 +20,13 @@
#ifndef __EDMA_PCM_H__
#define __EDMA_PCM_H__
+#if IS_ENABLED(CONFIG_SND_EDMA_SOC)
int edma_pcm_platform_register(struct device *dev);
+#else
+static inline int edma_pcm_platform_register(struct device *dev)
+{
+ return 0;
+}
+#endif /* CONFIG_SND_EDMA_SOC */
#endif /* __EDMA_PCM_H__ */
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 37933629cbed..f54a8fc99291 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -2,9 +2,20 @@ menu "SoC Audio for Freescale CPUs"
comment "Common SoC Audio options for Freescale CPUs:"
+config SND_SOC_FSL_ASRC
+ tristate "Asynchronous Sample Rate Converter (ASRC) module support"
+ select REGMAP_MMIO
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+ help
+ Say Y if you want to add Asynchronous Sample Rate Converter (ASRC)
+ support for the Freescale CPUs.
+ This option is only useful for out-of-tree drivers since
+ in-tree drivers select it automatically.
+
config SND_SOC_FSL_SAI
tristate "Synchronous Audio Interface (SAI) module support"
select REGMAP_MMIO
+ select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
select SND_SOC_GENERIC_DMAENGINE_PCM
help
Say Y if you want to add Synchronous Audio Interface (SAI)
@@ -15,7 +26,7 @@ config SND_SOC_FSL_SAI
config SND_SOC_FSL_SSI
tristate "Synchronous Serial Interface module support"
select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
- select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC
+ select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC)
select REGMAP_MMIO
help
Say Y if you want to add Synchronous Serial Interface (SSI)
@@ -27,7 +38,7 @@ config SND_SOC_FSL_SPDIF
tristate "Sony/Philips Digital Interface module support"
select REGMAP_MMIO
select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
- select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC
+ select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC)
help
Say Y if you want to add Sony/Philips Digital Interface (SPDIF)
support for the Freescale CPUs.
@@ -37,6 +48,7 @@ config SND_SOC_FSL_SPDIF
config SND_SOC_FSL_ESAI
tristate "Enhanced Serial Audio Interface (ESAI) module support"
select REGMAP_MMIO
+ select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
select SND_SOC_FSL_UTILS
help
Say Y if you want to add Enhanced Synchronous Audio Interface
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index db254e358c18..9ff59267eac9 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -11,6 +11,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o
obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
# Freescale SSI/DMA/SAI/SPDIF Support
+snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o
snd-soc-fsl-sai-objs := fsl_sai.o
snd-soc-fsl-ssi-y := fsl_ssi.o
snd-soc-fsl-ssi-$(CONFIG_DEBUG_FS) += fsl_ssi_dbg.o
@@ -18,6 +19,7 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o
snd-soc-fsl-esai-objs := fsl_esai.o
snd-soc-fsl-utils-objs := fsl_utils.o
snd-soc-fsl-dma-objs := fsl_dma.o
+obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o
obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
new file mode 100644
index 000000000000..822110420b71
--- /dev/null
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -0,0 +1,995 @@
+/*
+ * Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <nicoleotsuka@gmail.com>
+ *
+ * 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/clk.h>
+#include <linux/delay.h>
+#include <linux/dma-mapping.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/platform_data/dma-imx.h>
+#include <linux/pm_runtime.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_asrc.h"
+
+#define IDEAL_RATIO_DECIMAL_DEPTH 26
+
+#define pair_err(fmt, ...) \
+ dev_err(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__)
+
+#define pair_dbg(fmt, ...) \
+ dev_dbg(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__)
+
+/* Sample rates are aligned with that defined in pcm.h file */
+static const u8 process_option[][8][2] = {
+ /* 32kHz 44.1kHz 48kHz 64kHz 88.2kHz 96kHz 176kHz 192kHz */
+ {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 5512Hz */
+ {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 8kHz */
+ {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 11025Hz */
+ {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 16kHz */
+ {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 22050Hz */
+ {{0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0},}, /* 32kHz */
+ {{0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 44.1kHz */
+ {{0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 48kHz */
+ {{1, 2}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0},}, /* 64kHz */
+ {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 88.2kHz */
+ {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 96kHz */
+ {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 176kHz */
+ {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 192kHz */
+};
+
+/* Corresponding to process_option */
+static int supported_input_rate[] = {
+ 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200,
+ 96000, 176400, 192000,
+};
+
+static int supported_asrc_rate[] = {
+ 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000,
+};
+
+/**
+ * The following tables map the relationship between asrc_inclk/asrc_outclk in
+ * fsl_asrc.h and the registers of ASRCSR
+ */
+static unsigned char input_clk_map_imx35[] = {
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf,
+};
+
+static unsigned char output_clk_map_imx35[] = {
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf,
+};
+
+/* i.MX53 uses the same map for input and output */
+static unsigned char input_clk_map_imx53[] = {
+/* 0x0 0x1 0x2 0x3 0x4 0x5 0x6 0x7 0x8 0x9 0xa 0xb 0xc 0xd 0xe 0xf */
+ 0x0, 0x1, 0x2, 0x7, 0x4, 0x5, 0x6, 0x3, 0x8, 0x9, 0xa, 0xb, 0xc, 0xf, 0xe, 0xd,
+};
+
+static unsigned char output_clk_map_imx53[] = {
+/* 0x0 0x1 0x2 0x3 0x4 0x5 0x6 0x7 0x8 0x9 0xa 0xb 0xc 0xd 0xe 0xf */
+ 0x8, 0x9, 0xa, 0x7, 0xc, 0x5, 0x6, 0xb, 0x0, 0x1, 0x2, 0x3, 0x4, 0xf, 0xe, 0xd,
+};
+
+static unsigned char *clk_map[2];
+
+/**
+ * Request ASRC pair
+ *
+ * It assigns pair by the order of A->C->B because allocation of pair B,
+ * within range [ANCA, ANCA+ANCB-1], depends on the channels of pair A
+ * while pair A and pair C are comparatively independent.
+ */
+static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair)
+{
+ enum asrc_pair_index index = ASRC_INVALID_PAIR;
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ struct device *dev = &asrc_priv->pdev->dev;
+ unsigned long lock_flags;
+ int i, ret = 0;
+
+ spin_lock_irqsave(&asrc_priv->lock, lock_flags);
+
+ for (i = ASRC_PAIR_A; i < ASRC_PAIR_MAX_NUM; i++) {
+ if (asrc_priv->pair[i] != NULL)
+ continue;
+
+ index = i;
+
+ if (i != ASRC_PAIR_B)
+ break;
+ }
+
+ if (index == ASRC_INVALID_PAIR) {
+ dev_err(dev, "all pairs are busy now\n");
+ ret = -EBUSY;
+ } else if (asrc_priv->channel_avail < channels) {
+ dev_err(dev, "can't afford required channels: %d\n", channels);
+ ret = -EINVAL;
+ } else {
+ asrc_priv->channel_avail -= channels;
+ asrc_priv->pair[index] = pair;
+ pair->channels = channels;
+ pair->index = index;
+ }
+
+ spin_unlock_irqrestore(&asrc_priv->lock, lock_flags);
+
+ return ret;
+}
+
+/**
+ * Release ASRC pair
+ *
+ * It clears the resource from asrc_priv and releases the occupied channels.
+ */
+static void fsl_asrc_release_pair(struct fsl_asrc_pair *pair)
+{
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ enum asrc_pair_index index = pair->index;
+ unsigned long lock_flags;
+
+ /* Make sure the pair is disabled */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_ASRCEi_MASK(index), 0);
+
+ spin_lock_irqsave(&asrc_priv->lock, lock_flags);
+
+ asrc_priv->channel_avail += pair->channels;
+ asrc_priv->pair[index] = NULL;
+ pair->error = 0;
+
+ spin_unlock_irqrestore(&asrc_priv->lock, lock_flags);
+}
+
+/**
+ * Configure input and output thresholds
+ */
+static void fsl_asrc_set_watermarks(struct fsl_asrc_pair *pair, u32 in, u32 out)
+{
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ enum asrc_pair_index index = pair->index;
+
+ regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index),
+ ASRMCRi_EXTTHRSHi_MASK |
+ ASRMCRi_INFIFO_THRESHOLD_MASK |
+ ASRMCRi_OUTFIFO_THRESHOLD_MASK,
+ ASRMCRi_EXTTHRSHi |
+ ASRMCRi_INFIFO_THRESHOLD(in) |
+ ASRMCRi_OUTFIFO_THRESHOLD(out));
+}
+
+/**
+ * Calculate the total divisor between asrck clock rate and sample rate
+ *
+ * It follows the formula clk_rate = samplerate * (2 ^ prescaler) * divider
+ */
+static u32 fsl_asrc_cal_asrck_divisor(struct fsl_asrc_pair *pair, u32 div)
+{
+ u32 ps;
+
+ /* Calculate the divisors: prescaler [2^0, 2^7], divder [1, 8] */
+ for (ps = 0; div > 8; ps++)
+ div >>= 1;
+
+ return ((div - 1) << ASRCDRi_AxCPi_WIDTH) | ps;
+}
+
+/**
+ * Calculate and set the ratio for Ideal Ratio mode only
+ *
+ * The ratio is a 32-bit fixed point value with 26 fractional bits.
+ */
+static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair,
+ int inrate, int outrate)
+{
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ enum asrc_pair_index index = pair->index;
+ unsigned long ratio;
+ int i;
+
+ if (!outrate) {
+ pair_err("output rate should not be zero\n");
+ return -EINVAL;
+ }
+
+ /* Calculate the intergal part of the ratio */
+ ratio = (inrate / outrate) << IDEAL_RATIO_DECIMAL_DEPTH;
+
+ /* ... and then the 26 depth decimal part */
+ inrate %= outrate;
+
+ for (i = 1; i <= IDEAL_RATIO_DECIMAL_DEPTH; i++) {
+ inrate <<= 1;
+
+ if (inrate < outrate)
+ continue;
+
+ ratio |= 1 << (IDEAL_RATIO_DECIMAL_DEPTH - i);
+ inrate -= outrate;
+
+ if (!inrate)
+ break;
+ }
+
+ regmap_write(asrc_priv->regmap, REG_ASRIDRL(index), ratio);
+ regmap_write(asrc_priv->regmap, REG_ASRIDRH(index), ratio >> 24);
+
+ return 0;
+}
+
+/**
+ * Configure the assigned ASRC pair
+ *
+ * It configures those ASRC registers according to a configuration instance
+ * of struct asrc_config which includes in/output sample rate, width, channel
+ * and clock settings.
+ */
+static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair)
+{
+ struct asrc_config *config = pair->config;
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ enum asrc_pair_index index = pair->index;
+ u32 inrate, outrate, indiv, outdiv;
+ u32 clk_index[2], div[2];
+ int in, out, channels;
+ struct clk *clk;
+ bool ideal;
+
+ if (!config) {
+ pair_err("invalid pair config\n");
+ return -EINVAL;
+ }
+
+ /* Validate channels */
+ if (config->channel_num < 1 || config->channel_num > 10) {
+ pair_err("does not support %d channels\n", config->channel_num);
+ return -EINVAL;
+ }
+
+ /* Validate output width */
+ if (config->output_word_width == ASRC_WIDTH_8_BIT) {
+ pair_err("does not support 8bit width output\n");
+ return -EINVAL;
+ }
+
+ inrate = config->input_sample_rate;
+ outrate = config->output_sample_rate;
+ ideal = config->inclk == INCLK_NONE;
+
+ /* Validate input and output sample rates */
+ for (in = 0; in < ARRAY_SIZE(supported_input_rate); in++)
+ if (inrate == supported_input_rate[in])
+ break;
+
+ if (in == ARRAY_SIZE(supported_input_rate)) {
+ pair_err("unsupported input sample rate: %dHz\n", inrate);
+ return -EINVAL;
+ }
+
+ for (out = 0; out < ARRAY_SIZE(supported_asrc_rate); out++)
+ if (outrate == supported_asrc_rate[out])
+ break;
+
+ if (out == ARRAY_SIZE(supported_asrc_rate)) {
+ pair_err("unsupported output sample rate: %dHz\n", outrate);
+ return -EINVAL;
+ }
+
+ /* Validate input and output clock sources */
+ clk_index[IN] = clk_map[IN][config->inclk];
+ clk_index[OUT] = clk_map[OUT][config->outclk];
+
+ /* We only have output clock for ideal ratio mode */
+ clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]];
+
+ div[IN] = clk_get_rate(clk) / inrate;
+ if (div[IN] == 0) {
+ pair_err("failed to support input sample rate %dHz by asrck_%x\n",
+ inrate, clk_index[ideal ? OUT : IN]);
+ return -EINVAL;
+ }
+
+ clk = asrc_priv->asrck_clk[clk_index[OUT]];
+
+ /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */
+ if (ideal)
+ div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE;
+ else
+ div[OUT] = clk_get_rate(clk) / outrate;
+
+ if (div[OUT] == 0) {
+ pair_err("failed to support output sample rate %dHz by asrck_%x\n",
+ outrate, clk_index[OUT]);
+ return -EINVAL;
+ }
+
+ /* Set the channel number */
+ channels = config->channel_num;
+
+ if (asrc_priv->channel_bits < 4)
+ channels /= 2;
+
+ /* Update channels for current pair */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCNCR,
+ ASRCNCR_ANCi_MASK(index, asrc_priv->channel_bits),
+ ASRCNCR_ANCi(index, channels, asrc_priv->channel_bits));
+
+ /* Default setting: Automatic selection for processing mode */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_ATSi_MASK(index), ASRCTR_ATS(index));
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_USRi_MASK(index), 0);
+
+ /* Set the input and output clock sources */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCSR,
+ ASRCSR_AICSi_MASK(index) | ASRCSR_AOCSi_MASK(index),
+ ASRCSR_AICS(index, clk_index[IN]) |
+ ASRCSR_AOCS(index, clk_index[OUT]));
+
+ /* Calculate the input clock divisors */
+ indiv = fsl_asrc_cal_asrck_divisor(pair, div[IN]);
+ outdiv = fsl_asrc_cal_asrck_divisor(pair, div[OUT]);
+
+ /* Suppose indiv and outdiv includes prescaler, so add its MASK too */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCDR(index),
+ ASRCDRi_AOCPi_MASK(index) | ASRCDRi_AICPi_MASK(index) |
+ ASRCDRi_AOCDi_MASK(index) | ASRCDRi_AICDi_MASK(index),
+ ASRCDRi_AOCP(index, outdiv) | ASRCDRi_AICP(index, indiv));
+
+ /* Implement word_width configurations */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRMCR1(index),
+ ASRMCR1i_OW16_MASK | ASRMCR1i_IWD_MASK,
+ ASRMCR1i_OW16(config->output_word_width) |
+ ASRMCR1i_IWD(config->input_word_width));
+
+ /* Enable BUFFER STALL */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index),
+ ASRMCRi_BUFSTALLi_MASK, ASRMCRi_BUFSTALLi);
+
+ /* Set default thresholds for input and output FIFO */
+ fsl_asrc_set_watermarks(pair, ASRC_INPUTFIFO_THRESHOLD,
+ ASRC_INPUTFIFO_THRESHOLD);
+
+ /* Configure the followings only for Ideal Ratio mode */
+ if (!ideal)
+ return 0;
+
+ /* Clear ASTSx bit to use Ideal Ratio mode */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_ATSi_MASK(index), 0);
+
+ /* Enable Ideal Ratio mode */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_IDRi_MASK(index) | ASRCTR_USRi_MASK(index),
+ ASRCTR_IDR(index) | ASRCTR_USR(index));
+
+ /* Apply configurations for pre- and post-processing */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCFG,
+ ASRCFG_PREMODi_MASK(index) | ASRCFG_POSTMODi_MASK(index),
+ ASRCFG_PREMOD(index, process_option[in][out][0]) |
+ ASRCFG_POSTMOD(index, process_option[in][out][1]));
+
+ return fsl_asrc_set_ideal_ratio(pair, inrate, outrate);
+}
+
+/**
+ * Start the assigned ASRC pair
+ *
+ * It enables the assigned pair and makes it stopped at the stall level.
+ */
+static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair)
+{
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ enum asrc_pair_index index = pair->index;
+ int reg, retry = 10, i;
+
+ /* Enable the current pair */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_ASRCEi_MASK(index), ASRCTR_ASRCE(index));
+
+ /* Wait for status of initialization */
+ do {
+ udelay(5);
+ regmap_read(asrc_priv->regmap, REG_ASRCFG, &reg);
+ reg &= ASRCFG_INIRQi_MASK(index);
+ } while (!reg && --retry);
+
+ /* Make the input fifo to ASRC STALL level */
+ regmap_read(asrc_priv->regmap, REG_ASRCNCR, &reg);
+ for (i = 0; i < pair->channels * 4; i++)
+ regmap_write(asrc_priv->regmap, REG_ASRDI(index), 0);
+
+ /* Enable overload interrupt */
+ regmap_write(asrc_priv->regmap, REG_ASRIER, ASRIER_AOLIE);
+}
+
+/**
+ * Stop the assigned ASRC pair
+ */
+static void fsl_asrc_stop_pair(struct fsl_asrc_pair *pair)
+{
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ enum asrc_pair_index index = pair->index;
+
+ /* Stop the current pair */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_ASRCEi_MASK(index), 0);
+}
+
+/**
+ * Get DMA channel according to the pair and direction.
+ */
+struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir)
+{
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ enum asrc_pair_index index = pair->index;
+ char name[4];
+
+ sprintf(name, "%cx%c", dir == IN ? 'r' : 't', index + 'a');
+
+ return dma_request_slave_channel(&asrc_priv->pdev->dev, name);
+}
+EXPORT_SYMBOL_GPL(fsl_asrc_get_dma_channel);
+
+static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai);
+ int width = snd_pcm_format_width(params_format(params));
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+ unsigned int channels = params_channels(params);
+ unsigned int rate = params_rate(params);
+ struct asrc_config config;
+ int word_width, ret;
+
+ ret = fsl_asrc_request_pair(channels, pair);
+ if (ret) {
+ dev_err(dai->dev, "fail to request asrc pair\n");
+ return ret;
+ }
+
+ pair->config = &config;
+
+ if (width == 16)
+ width = ASRC_WIDTH_16_BIT;
+ else
+ width = ASRC_WIDTH_24_BIT;
+
+ if (asrc_priv->asrc_width == 16)
+ word_width = ASRC_WIDTH_16_BIT;
+ else
+ word_width = ASRC_WIDTH_24_BIT;
+
+ config.pair = pair->index;
+ config.channel_num = channels;
+ config.inclk = INCLK_NONE;
+ config.outclk = OUTCLK_ASRCK1_CLK;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ config.input_word_width = width;
+ config.output_word_width = word_width;
+ config.input_sample_rate = rate;
+ config.output_sample_rate = asrc_priv->asrc_rate;
+ } else {
+ config.input_word_width = word_width;
+ config.output_word_width = width;
+ config.input_sample_rate = asrc_priv->asrc_rate;
+ config.output_sample_rate = rate;
+ }
+
+ ret = fsl_asrc_config_pair(pair);
+ if (ret) {
+ dev_err(dai->dev, "fail to config asrc pair\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static int fsl_asrc_dai_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+
+ if (pair)
+ fsl_asrc_release_pair(pair);
+
+ return 0;
+}
+
+static int fsl_asrc_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ fsl_asrc_start_pair(pair);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ fsl_asrc_stop_pair(pair);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops fsl_asrc_dai_ops = {
+ .hw_params = fsl_asrc_dai_hw_params,
+ .hw_free = fsl_asrc_dai_hw_free,
+ .trigger = fsl_asrc_dai_trigger,
+};
+
+static int fsl_asrc_dai_probe(struct snd_soc_dai *dai)
+{
+ struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai);
+
+ snd_soc_dai_init_dma_data(dai, &asrc_priv->dma_params_tx,
+ &asrc_priv->dma_params_rx);
+
+ return 0;
+}
+
+#define FSL_ASRC_RATES SNDRV_PCM_RATE_8000_192000
+#define FSL_ASRC_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE)
+
+static struct snd_soc_dai_driver fsl_asrc_dai = {
+ .probe = fsl_asrc_dai_probe,
+ .playback = {
+ .stream_name = "ASRC-Playback",
+ .channels_min = 1,
+ .channels_max = 10,
+ .rates = FSL_ASRC_RATES,
+ .formats = FSL_ASRC_FORMATS,
+ },
+ .capture = {
+ .stream_name = "ASRC-Capture",
+ .channels_min = 1,
+ .channels_max = 10,
+ .rates = FSL_ASRC_RATES,
+ .formats = FSL_ASRC_FORMATS,
+ },
+ .ops = &fsl_asrc_dai_ops,
+};
+
+static const struct snd_soc_component_driver fsl_asrc_component = {
+ .name = "fsl-asrc-dai",
+};
+
+static bool fsl_asrc_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case REG_ASRCTR:
+ case REG_ASRIER:
+ case REG_ASRCNCR:
+ case REG_ASRCFG:
+ case REG_ASRCSR:
+ case REG_ASRCDR1:
+ case REG_ASRCDR2:
+ case REG_ASRSTR:
+ case REG_ASRPM1:
+ case REG_ASRPM2:
+ case REG_ASRPM3:
+ case REG_ASRPM4:
+ case REG_ASRPM5:
+ case REG_ASRTFR1:
+ case REG_ASRCCR:
+ case REG_ASRDOA:
+ case REG_ASRDOB:
+ case REG_ASRDOC:
+ case REG_ASRIDRHA:
+ case REG_ASRIDRLA:
+ case REG_ASRIDRHB:
+ case REG_ASRIDRLB:
+ case REG_ASRIDRHC:
+ case REG_ASRIDRLC:
+ case REG_ASR76K:
+ case REG_ASR56K:
+ case REG_ASRMCRA:
+ case REG_ASRFSTA:
+ case REG_ASRMCRB:
+ case REG_ASRFSTB:
+ case REG_ASRMCRC:
+ case REG_ASRFSTC:
+ case REG_ASRMCR1A:
+ case REG_ASRMCR1B:
+ case REG_ASRMCR1C:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool fsl_asrc_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case REG_ASRSTR:
+ case REG_ASRDIA:
+ case REG_ASRDIB:
+ case REG_ASRDIC:
+ case REG_ASRDOA:
+ case REG_ASRDOB:
+ case REG_ASRDOC:
+ case REG_ASRFSTA:
+ case REG_ASRFSTB:
+ case REG_ASRFSTC:
+ case REG_ASRCFG:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case REG_ASRCTR:
+ case REG_ASRIER:
+ case REG_ASRCNCR:
+ case REG_ASRCFG:
+ case REG_ASRCSR:
+ case REG_ASRCDR1:
+ case REG_ASRCDR2:
+ case REG_ASRSTR:
+ case REG_ASRPM1:
+ case REG_ASRPM2:
+ case REG_ASRPM3:
+ case REG_ASRPM4:
+ case REG_ASRPM5:
+ case REG_ASRTFR1:
+ case REG_ASRCCR:
+ case REG_ASRDIA:
+ case REG_ASRDIB:
+ case REG_ASRDIC:
+ case REG_ASRIDRHA:
+ case REG_ASRIDRLA:
+ case REG_ASRIDRHB:
+ case REG_ASRIDRLB:
+ case REG_ASRIDRHC:
+ case REG_ASRIDRLC:
+ case REG_ASR76K:
+ case REG_ASR56K:
+ case REG_ASRMCRA:
+ case REG_ASRMCRB:
+ case REG_ASRMCRC:
+ case REG_ASRMCR1A:
+ case REG_ASRMCR1B:
+ case REG_ASRMCR1C:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static struct regmap_config fsl_asrc_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+
+ .max_register = REG_ASRMCR1C,
+ .readable_reg = fsl_asrc_readable_reg,
+ .volatile_reg = fsl_asrc_volatile_reg,
+ .writeable_reg = fsl_asrc_writeable_reg,
+ .cache_type = REGCACHE_RBTREE,
+};
+
+/**
+ * Initialize ASRC registers with a default configurations
+ */
+static int fsl_asrc_init(struct fsl_asrc *asrc_priv)
+{
+ /* Halt ASRC internal FP when input FIFO needs data for pair A, B, C */
+ regmap_write(asrc_priv->regmap, REG_ASRCTR, ASRCTR_ASRCEN);
+
+ /* Disable interrupt by default */
+ regmap_write(asrc_priv->regmap, REG_ASRIER, 0x0);
+
+ /* Apply recommended settings for parameters from Reference Manual */
+ regmap_write(asrc_priv->regmap, REG_ASRPM1, 0x7fffff);
+ regmap_write(asrc_priv->regmap, REG_ASRPM2, 0x255555);
+ regmap_write(asrc_priv->regmap, REG_ASRPM3, 0xff7280);
+ regmap_write(asrc_priv->regmap, REG_ASRPM4, 0xff7280);
+ regmap_write(asrc_priv->regmap, REG_ASRPM5, 0xff7280);
+
+ /* Base address for task queue FIFO. Set to 0x7C */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRTFR1,
+ ASRTFR1_TF_BASE_MASK, ASRTFR1_TF_BASE(0xfc));
+
+ /* Set the processing clock for 76KHz to 133M */
+ regmap_write(asrc_priv->regmap, REG_ASR76K, 0x06D6);
+
+ /* Set the processing clock for 56KHz to 133M */
+ return regmap_write(asrc_priv->regmap, REG_ASR56K, 0x0947);
+}
+
+/**
+ * Interrupt handler for ASRC
+ */
+static irqreturn_t fsl_asrc_isr(int irq, void *dev_id)
+{
+ struct fsl_asrc *asrc_priv = (struct fsl_asrc *)dev_id;
+ struct device *dev = &asrc_priv->pdev->dev;
+ enum asrc_pair_index index;
+ u32 status;
+
+ regmap_read(asrc_priv->regmap, REG_ASRSTR, &status);
+
+ /* Clean overload error */
+ regmap_write(asrc_priv->regmap, REG_ASRSTR, ASRSTR_AOLE);
+
+ /*
+ * We here use dev_dbg() for all exceptions because ASRC itself does
+ * not care if FIFO overflowed or underrun while a warning in the
+ * interrupt would result a ridged conversion.
+ */
+ for (index = ASRC_PAIR_A; index < ASRC_PAIR_MAX_NUM; index++) {
+ if (!asrc_priv->pair[index])
+ continue;
+
+ if (status & ASRSTR_ATQOL) {
+ asrc_priv->pair[index]->error |= ASRC_TASK_Q_OVERLOAD;
+ dev_dbg(dev, "ASRC Task Queue FIFO overload\n");
+ }
+
+ if (status & ASRSTR_AOOL(index)) {
+ asrc_priv->pair[index]->error |= ASRC_OUTPUT_TASK_OVERLOAD;
+ pair_dbg("Output Task Overload\n");
+ }
+
+ if (status & ASRSTR_AIOL(index)) {
+ asrc_priv->pair[index]->error |= ASRC_INPUT_TASK_OVERLOAD;
+ pair_dbg("Input Task Overload\n");
+ }
+
+ if (status & ASRSTR_AODO(index)) {
+ asrc_priv->pair[index]->error |= ASRC_OUTPUT_BUFFER_OVERFLOW;
+ pair_dbg("Output Data Buffer has overflowed\n");
+ }
+
+ if (status & ASRSTR_AIDU(index)) {
+ asrc_priv->pair[index]->error |= ASRC_INPUT_BUFFER_UNDERRUN;
+ pair_dbg("Input Data Buffer has underflowed\n");
+ }
+ }
+
+ return IRQ_HANDLED;
+}
+
+static int fsl_asrc_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct fsl_asrc *asrc_priv;
+ struct resource *res;
+ void __iomem *regs;
+ int irq, ret, i;
+ char tmp[16];
+
+ asrc_priv = devm_kzalloc(&pdev->dev, sizeof(*asrc_priv), GFP_KERNEL);
+ if (!asrc_priv)
+ return -ENOMEM;
+
+ asrc_priv->pdev = pdev;
+ strcpy(asrc_priv->name, np->name);
+
+ /* Get the addresses and IRQ */
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ regs = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ asrc_priv->paddr = res->start;
+
+ /* Register regmap and let it prepare core clock */
+ if (of_property_read_bool(np, "big-endian"))
+ fsl_asrc_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
+
+ asrc_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "mem", regs,
+ &fsl_asrc_regmap_config);
+ if (IS_ERR(asrc_priv->regmap)) {
+ dev_err(&pdev->dev, "failed to init regmap\n");
+ return PTR_ERR(asrc_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, fsl_asrc_isr, 0,
+ asrc_priv->name, asrc_priv);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to claim irq %u: %d\n", irq, ret);
+ return ret;
+ }
+
+ asrc_priv->mem_clk = devm_clk_get(&pdev->dev, "mem");
+ if (IS_ERR(asrc_priv->mem_clk)) {
+ dev_err(&pdev->dev, "failed to get mem clock\n");
+ return PTR_ERR(asrc_priv->mem_clk);
+ }
+
+ asrc_priv->ipg_clk = devm_clk_get(&pdev->dev, "ipg");
+ if (IS_ERR(asrc_priv->ipg_clk)) {
+ dev_err(&pdev->dev, "failed to get ipg clock\n");
+ return PTR_ERR(asrc_priv->ipg_clk);
+ }
+
+ for (i = 0; i < ASRC_CLK_MAX_NUM; i++) {
+ sprintf(tmp, "asrck_%x", i);
+ asrc_priv->asrck_clk[i] = devm_clk_get(&pdev->dev, tmp);
+ if (IS_ERR(asrc_priv->asrck_clk[i])) {
+ dev_err(&pdev->dev, "failed to get %s clock\n", tmp);
+ return PTR_ERR(asrc_priv->asrck_clk[i]);
+ }
+ }
+
+ if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx35-asrc")) {
+ asrc_priv->channel_bits = 3;
+ clk_map[IN] = input_clk_map_imx35;
+ clk_map[OUT] = output_clk_map_imx35;
+ } else {
+ asrc_priv->channel_bits = 4;
+ clk_map[IN] = input_clk_map_imx53;
+ clk_map[OUT] = output_clk_map_imx53;
+ }
+
+ ret = fsl_asrc_init(asrc_priv);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to init asrc %d\n", ret);
+ return -EINVAL;
+ }
+
+ asrc_priv->channel_avail = 10;
+
+ ret = of_property_read_u32(np, "fsl,asrc-rate",
+ &asrc_priv->asrc_rate);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to get output rate\n");
+ return -EINVAL;
+ }
+
+ ret = of_property_read_u32(np, "fsl,asrc-width",
+ &asrc_priv->asrc_width);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to get output width\n");
+ return -EINVAL;
+ }
+
+ if (asrc_priv->asrc_width != 16 && asrc_priv->asrc_width != 24) {
+ dev_warn(&pdev->dev, "unsupported width, switching to 24bit\n");
+ asrc_priv->asrc_width = 24;
+ }
+
+ platform_set_drvdata(pdev, asrc_priv);
+ pm_runtime_enable(&pdev->dev);
+ spin_lock_init(&asrc_priv->lock);
+
+ ret = devm_snd_soc_register_component(&pdev->dev, &fsl_asrc_component,
+ &fsl_asrc_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to register ASoC DAI\n");
+ return ret;
+ }
+
+ ret = devm_snd_soc_register_platform(&pdev->dev, &fsl_asrc_platform);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to register ASoC platform\n");
+ return ret;
+ }
+
+ dev_info(&pdev->dev, "driver registered\n");
+
+ return 0;
+}
+
+#ifdef CONFIG_PM_RUNTIME
+static int fsl_asrc_runtime_resume(struct device *dev)
+{
+ struct fsl_asrc *asrc_priv = dev_get_drvdata(dev);
+ int i;
+
+ clk_prepare_enable(asrc_priv->mem_clk);
+ clk_prepare_enable(asrc_priv->ipg_clk);
+ for (i = 0; i < ASRC_CLK_MAX_NUM; i++)
+ clk_prepare_enable(asrc_priv->asrck_clk[i]);
+
+ return 0;
+}
+
+static int fsl_asrc_runtime_suspend(struct device *dev)
+{
+ struct fsl_asrc *asrc_priv = dev_get_drvdata(dev);
+ int i;
+
+ for (i = 0; i < ASRC_CLK_MAX_NUM; i++)
+ clk_disable_unprepare(asrc_priv->asrck_clk[i]);
+ clk_disable_unprepare(asrc_priv->ipg_clk);
+ clk_disable_unprepare(asrc_priv->mem_clk);
+
+ return 0;
+}
+#endif /* CONFIG_PM_RUNTIME */
+
+#ifdef CONFIG_PM_SLEEP
+static int fsl_asrc_suspend(struct device *dev)
+{
+ struct fsl_asrc *asrc_priv = dev_get_drvdata(dev);
+
+ regcache_cache_only(asrc_priv->regmap, true);
+ regcache_mark_dirty(asrc_priv->regmap);
+
+ return 0;
+}
+
+static int fsl_asrc_resume(struct device *dev)
+{
+ struct fsl_asrc *asrc_priv = dev_get_drvdata(dev);
+ u32 asrctr;
+
+ /* Stop all pairs provisionally */
+ regmap_read(asrc_priv->regmap, REG_ASRCTR, &asrctr);
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_ASRCEi_ALL_MASK, 0);
+
+ /* Restore all registers */
+ regcache_cache_only(asrc_priv->regmap, false);
+ regcache_sync(asrc_priv->regmap);
+
+ /* Restart enabled pairs */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_ASRCEi_ALL_MASK, asrctr);
+
+ return 0;
+}
+#endif /* CONFIG_PM_SLEEP */
+
+static const struct dev_pm_ops fsl_asrc_pm = {
+ SET_RUNTIME_PM_OPS(fsl_asrc_runtime_suspend, fsl_asrc_runtime_resume, NULL)
+ SET_SYSTEM_SLEEP_PM_OPS(fsl_asrc_suspend, fsl_asrc_resume)
+};
+
+static const struct of_device_id fsl_asrc_ids[] = {
+ { .compatible = "fsl,imx35-asrc", },
+ { .compatible = "fsl,imx53-asrc", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, fsl_asrc_ids);
+
+static struct platform_driver fsl_asrc_driver = {
+ .probe = fsl_asrc_probe,
+ .driver = {
+ .name = "fsl-asrc",
+ .of_match_table = fsl_asrc_ids,
+ .pm = &fsl_asrc_pm,
+ },
+};
+module_platform_driver(fsl_asrc_driver);
+
+MODULE_DESCRIPTION("Freescale ASRC ASoC driver");
+MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
+MODULE_ALIAS("platform:fsl-asrc");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h
new file mode 100644
index 000000000000..a3f211f53c23
--- /dev/null
+++ b/sound/soc/fsl/fsl_asrc.h
@@ -0,0 +1,461 @@
+/*
+ * fsl_asrc.h - Freescale ASRC ALSA SoC header file
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <nicoleotsuka@gmail.com>
+ *
+ * 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_ASRC_H
+#define _FSL_ASRC_H
+
+#define IN 0
+#define OUT 1
+
+#define ASRC_DMA_BUFFER_NUM 2
+#define ASRC_INPUTFIFO_THRESHOLD 32
+#define ASRC_OUTPUTFIFO_THRESHOLD 32
+#define ASRC_FIFO_THRESHOLD_MIN 0
+#define ASRC_FIFO_THRESHOLD_MAX 63
+#define ASRC_DMA_BUFFER_SIZE (1024 * 48 * 4)
+#define ASRC_MAX_BUFFER_SIZE (1024 * 48)
+#define ASRC_OUTPUT_LAST_SAMPLE 8
+
+#define IDEAL_RATIO_RATE 1000000
+
+#define REG_ASRCTR 0x00
+#define REG_ASRIER 0x04
+#define REG_ASRCNCR 0x0C
+#define REG_ASRCFG 0x10
+#define REG_ASRCSR 0x14
+
+#define REG_ASRCDR1 0x18
+#define REG_ASRCDR2 0x1C
+#define REG_ASRCDR(i) ((i < 2) ? REG_ASRCDR1 : REG_ASRCDR2)
+
+#define REG_ASRSTR 0x20
+#define REG_ASRRA 0x24
+#define REG_ASRRB 0x28
+#define REG_ASRRC 0x2C
+#define REG_ASRPM1 0x40
+#define REG_ASRPM2 0x44
+#define REG_ASRPM3 0x48
+#define REG_ASRPM4 0x4C
+#define REG_ASRPM5 0x50
+#define REG_ASRTFR1 0x54
+#define REG_ASRCCR 0x5C
+
+#define REG_ASRDIA 0x60
+#define REG_ASRDOA 0x64
+#define REG_ASRDIB 0x68
+#define REG_ASRDOB 0x6C
+#define REG_ASRDIC 0x70
+#define REG_ASRDOC 0x74
+#define REG_ASRDI(i) (REG_ASRDIA + (i << 3))
+#define REG_ASRDO(i) (REG_ASRDOA + (i << 3))
+#define REG_ASRDx(x, i) (x == IN ? REG_ASRDI(i) : REG_ASRDO(i))
+
+#define REG_ASRIDRHA 0x80
+#define REG_ASRIDRLA 0x84
+#define REG_ASRIDRHB 0x88
+#define REG_ASRIDRLB 0x8C
+#define REG_ASRIDRHC 0x90
+#define REG_ASRIDRLC 0x94
+#define REG_ASRIDRH(i) (REG_ASRIDRHA + (i << 3))
+#define REG_ASRIDRL(i) (REG_ASRIDRLA + (i << 3))
+
+#define REG_ASR76K 0x98
+#define REG_ASR56K 0x9C
+
+#define REG_ASRMCRA 0xA0
+#define REG_ASRFSTA 0xA4
+#define REG_ASRMCRB 0xA8
+#define REG_ASRFSTB 0xAC
+#define REG_ASRMCRC 0xB0
+#define REG_ASRFSTC 0xB4
+#define REG_ASRMCR(i) (REG_ASRMCRA + (i << 3))
+#define REG_ASRFST(i) (REG_ASRFSTA + (i << 3))
+
+#define REG_ASRMCR1A 0xC0
+#define REG_ASRMCR1B 0xC4
+#define REG_ASRMCR1C 0xC8
+#define REG_ASRMCR1(i) (REG_ASRMCR1A + (i << 2))
+
+
+/* REG0 0x00 REG_ASRCTR */
+#define ASRCTR_ATSi_SHIFT(i) (20 + i)
+#define ASRCTR_ATSi_MASK(i) (1 << ASRCTR_ATSi_SHIFT(i))
+#define ASRCTR_ATS(i) (1 << ASRCTR_ATSi_SHIFT(i))
+#define ASRCTR_USRi_SHIFT(i) (14 + (i << 1))
+#define ASRCTR_USRi_MASK(i) (1 << ASRCTR_USRi_SHIFT(i))
+#define ASRCTR_USR(i) (1 << ASRCTR_USRi_SHIFT(i))
+#define ASRCTR_IDRi_SHIFT(i) (13 + (i << 1))
+#define ASRCTR_IDRi_MASK(i) (1 << ASRCTR_IDRi_SHIFT(i))
+#define ASRCTR_IDR(i) (1 << ASRCTR_IDRi_SHIFT(i))
+#define ASRCTR_SRST_SHIFT 4
+#define ASRCTR_SRST_MASK (1 << ASRCTR_SRST_SHIFT)
+#define ASRCTR_SRST (1 << ASRCTR_SRST_SHIFT)
+#define ASRCTR_ASRCEi_SHIFT(i) (1 + i)
+#define ASRCTR_ASRCEi_MASK(i) (1 << ASRCTR_ASRCEi_SHIFT(i))
+#define ASRCTR_ASRCE(i) (1 << ASRCTR_ASRCEi_SHIFT(i))
+#define ASRCTR_ASRCEi_ALL_MASK (0x7 << ASRCTR_ASRCEi_SHIFT(0))
+#define ASRCTR_ASRCEN_SHIFT 0
+#define ASRCTR_ASRCEN_MASK (1 << ASRCTR_ASRCEN_SHIFT)
+#define ASRCTR_ASRCEN (1 << ASRCTR_ASRCEN_SHIFT)
+
+/* REG1 0x04 REG_ASRIER */
+#define ASRIER_AFPWE_SHIFT 7
+#define ASRIER_AFPWE_MASK (1 << ASRIER_AFPWE_SHIFT)
+#define ASRIER_AFPWE (1 << ASRIER_AFPWE_SHIFT)
+#define ASRIER_AOLIE_SHIFT 6
+#define ASRIER_AOLIE_MASK (1 << ASRIER_AOLIE_SHIFT)
+#define ASRIER_AOLIE (1 << ASRIER_AOLIE_SHIFT)
+#define ASRIER_ADOEi_SHIFT(i) (3 + i)
+#define ASRIER_ADOEi_MASK(i) (1 << ASRIER_ADOEi_SHIFT(i))
+#define ASRIER_ADOE(i) (1 << ASRIER_ADOEi_SHIFT(i))
+#define ASRIER_ADIEi_SHIFT(i) (0 + i)
+#define ASRIER_ADIEi_MASK(i) (1 << ASRIER_ADIEi_SHIFT(i))
+#define ASRIER_ADIE(i) (1 << ASRIER_ADIEi_SHIFT(i))
+
+/* REG2 0x0C REG_ASRCNCR */
+#define ASRCNCR_ANCi_SHIFT(i, b) (b * i)
+#define ASRCNCR_ANCi_MASK(i, b) (((1 << b) - 1) << ASRCNCR_ANCi_SHIFT(i, b))
+#define ASRCNCR_ANCi(i, v, b) ((v << ASRCNCR_ANCi_SHIFT(i, b)) & ASRCNCR_ANCi_MASK(i, b))
+
+/* REG3 0x10 REG_ASRCFG */
+#define ASRCFG_INIRQi_SHIFT(i) (21 + i)
+#define ASRCFG_INIRQi_MASK(i) (1 << ASRCFG_INIRQi_SHIFT(i))
+#define ASRCFG_INIRQi (1 << ASRCFG_INIRQi_SHIFT(i))
+#define ASRCFG_NDPRi_SHIFT(i) (18 + i)
+#define ASRCFG_NDPRi_MASK(i) (1 << ASRCFG_NDPRi_SHIFT(i))
+#define ASRCFG_NDPRi (1 << ASRCFG_NDPRi_SHIFT(i))
+#define ASRCFG_POSTMODi_SHIFT(i) (8 + (i << 2))
+#define ASRCFG_POSTMODi_WIDTH 2
+#define ASRCFG_POSTMODi_MASK(i) (((1 << ASRCFG_POSTMODi_WIDTH) - 1) << ASRCFG_POSTMODi_SHIFT(i))
+#define ASRCFG_POSTMOD(i, v) ((v) << ASRCFG_POSTMODi_SHIFT(i))
+#define ASRCFG_POSTMODi_UP(i) (0 << ASRCFG_POSTMODi_SHIFT(i))
+#define ASRCFG_POSTMODi_DCON(i) (1 << ASRCFG_POSTMODi_SHIFT(i))
+#define ASRCFG_POSTMODi_DOWN(i) (2 << ASRCFG_POSTMODi_SHIFT(i))
+#define ASRCFG_PREMODi_SHIFT(i) (6 + (i << 2))
+#define ASRCFG_PREMODi_WIDTH 2
+#define ASRCFG_PREMODi_MASK(i) (((1 << ASRCFG_PREMODi_WIDTH) - 1) << ASRCFG_PREMODi_SHIFT(i))
+#define ASRCFG_PREMOD(i, v) ((v) << ASRCFG_PREMODi_SHIFT(i))
+#define ASRCFG_PREMODi_UP(i) (0 << ASRCFG_PREMODi_SHIFT(i))
+#define ASRCFG_PREMODi_DCON(i) (1 << ASRCFG_PREMODi_SHIFT(i))
+#define ASRCFG_PREMODi_DOWN(i) (2 << ASRCFG_PREMODi_SHIFT(i))
+#define ASRCFG_PREMODi_BYPASS(i) (3 << ASRCFG_PREMODi_SHIFT(i))
+
+/* REG4 0x14 REG_ASRCSR */
+#define ASRCSR_AxCSi_WIDTH 4
+#define ASRCSR_AxCSi_MASK ((1 << ASRCSR_AxCSi_WIDTH) - 1)
+#define ASRCSR_AOCSi_SHIFT(i) (12 + (i << 2))
+#define ASRCSR_AOCSi_MASK(i) (((1 << ASRCSR_AxCSi_WIDTH) - 1) << ASRCSR_AOCSi_SHIFT(i))
+#define ASRCSR_AOCS(i, v) ((v) << ASRCSR_AOCSi_SHIFT(i))
+#define ASRCSR_AICSi_SHIFT(i) (i << 2)
+#define ASRCSR_AICSi_MASK(i) (((1 << ASRCSR_AxCSi_WIDTH) - 1) << ASRCSR_AICSi_SHIFT(i))
+#define ASRCSR_AICS(i, v) ((v) << ASRCSR_AICSi_SHIFT(i))
+
+/* REG5&6 0x18 & 0x1C REG_ASRCDR1 & ASRCDR2 */
+#define ASRCDRi_AxCPi_WIDTH 3
+#define ASRCDRi_AICPi_SHIFT(i) (0 + (i % 2) * 6)
+#define ASRCDRi_AICPi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AICPi_SHIFT(i))
+#define ASRCDRi_AICP(i, v) ((v) << ASRCDRi_AICPi_SHIFT(i))
+#define ASRCDRi_AICDi_SHIFT(i) (3 + (i % 2) * 6)
+#define ASRCDRi_AICDi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AICDi_SHIFT(i))
+#define ASRCDRi_AICD(i, v) ((v) << ASRCDRi_AICDi_SHIFT(i))
+#define ASRCDRi_AOCPi_SHIFT(i) ((i < 2) ? 12 + i * 6 : 6)
+#define ASRCDRi_AOCPi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AOCPi_SHIFT(i))
+#define ASRCDRi_AOCP(i, v) ((v) << ASRCDRi_AOCPi_SHIFT(i))
+#define ASRCDRi_AOCDi_SHIFT(i) ((i < 2) ? 15 + i * 6 : 9)
+#define ASRCDRi_AOCDi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AOCDi_SHIFT(i))
+#define ASRCDRi_AOCD(i, v) ((v) << ASRCDRi_AOCDi_SHIFT(i))
+
+/* REG7 0x20 REG_ASRSTR */
+#define ASRSTR_DSLCNT_SHIFT 21
+#define ASRSTR_DSLCNT_MASK (1 << ASRSTR_DSLCNT_SHIFT)
+#define ASRSTR_DSLCNT (1 << ASRSTR_DSLCNT_SHIFT)
+#define ASRSTR_ATQOL_SHIFT 20
+#define ASRSTR_ATQOL_MASK (1 << ASRSTR_ATQOL_SHIFT)
+#define ASRSTR_ATQOL (1 << ASRSTR_ATQOL_SHIFT)
+#define ASRSTR_AOOLi_SHIFT(i) (17 + i)
+#define ASRSTR_AOOLi_MASK(i) (1 << ASRSTR_AOOLi_SHIFT(i))
+#define ASRSTR_AOOL(i) (1 << ASRSTR_AOOLi_SHIFT(i))
+#define ASRSTR_AIOLi_SHIFT(i) (14 + i)
+#define ASRSTR_AIOLi_MASK(i) (1 << ASRSTR_AIOLi_SHIFT(i))
+#define ASRSTR_AIOL(i) (1 << ASRSTR_AIOLi_SHIFT(i))
+#define ASRSTR_AODOi_SHIFT(i) (11 + i)
+#define ASRSTR_AODOi_MASK(i) (1 << ASRSTR_AODOi_SHIFT(i))
+#define ASRSTR_AODO(i) (1 << ASRSTR_AODOi_SHIFT(i))
+#define ASRSTR_AIDUi_SHIFT(i) (8 + i)
+#define ASRSTR_AIDUi_MASK(i) (1 << ASRSTR_AIDUi_SHIFT(i))
+#define ASRSTR_AIDU(i) (1 << ASRSTR_AIDUi_SHIFT(i))
+#define ASRSTR_FPWT_SHIFT 7
+#define ASRSTR_FPWT_MASK (1 << ASRSTR_FPWT_SHIFT)
+#define ASRSTR_FPWT (1 << ASRSTR_FPWT_SHIFT)
+#define ASRSTR_AOLE_SHIFT 6
+#define ASRSTR_AOLE_MASK (1 << ASRSTR_AOLE_SHIFT)
+#define ASRSTR_AOLE (1 << ASRSTR_AOLE_SHIFT)
+#define ASRSTR_AODEi_SHIFT(i) (3 + i)
+#define ASRSTR_AODFi_MASK(i) (1 << ASRSTR_AODEi_SHIFT(i))
+#define ASRSTR_AODF(i) (1 << ASRSTR_AODEi_SHIFT(i))
+#define ASRSTR_AIDEi_SHIFT(i) (0 + i)
+#define ASRSTR_AIDEi_MASK(i) (1 << ASRSTR_AIDEi_SHIFT(i))
+#define ASRSTR_AIDE(i) (1 << ASRSTR_AIDEi_SHIFT(i))
+
+/* REG10 0x54 REG_ASRTFR1 */
+#define ASRTFR1_TF_BASE_WIDTH 7
+#define ASRTFR1_TF_BASE_SHIFT 6
+#define ASRTFR1_TF_BASE_MASK (((1 << ASRTFR1_TF_BASE_WIDTH) - 1) << ASRTFR1_TF_BASE_SHIFT)
+#define ASRTFR1_TF_BASE(i) ((i) << ASRTFR1_TF_BASE_SHIFT)
+
+/*
+ * REG22 0xA0 REG_ASRMCRA
+ * REG24 0xA8 REG_ASRMCRB
+ * REG26 0xB0 REG_ASRMCRC
+ */
+#define ASRMCRi_ZEROBUFi_SHIFT 23
+#define ASRMCRi_ZEROBUFi_MASK (1 << ASRMCRi_ZEROBUFi_SHIFT)
+#define ASRMCRi_ZEROBUFi (1 << ASRMCRi_ZEROBUFi_SHIFT)
+#define ASRMCRi_EXTTHRSHi_SHIFT 22
+#define ASRMCRi_EXTTHRSHi_MASK (1 << ASRMCRi_EXTTHRSHi_SHIFT)
+#define ASRMCRi_EXTTHRSHi (1 << ASRMCRi_EXTTHRSHi_SHIFT)
+#define ASRMCRi_BUFSTALLi_SHIFT 21
+#define ASRMCRi_BUFSTALLi_MASK (1 << ASRMCRi_BUFSTALLi_SHIFT)
+#define ASRMCRi_BUFSTALLi (1 << ASRMCRi_BUFSTALLi_SHIFT)
+#define ASRMCRi_BYPASSPOLYi_SHIFT 20
+#define ASRMCRi_BYPASSPOLYi_MASK (1 << ASRMCRi_BYPASSPOLYi_SHIFT)
+#define ASRMCRi_BYPASSPOLYi (1 << ASRMCRi_BYPASSPOLYi_SHIFT)
+#define ASRMCRi_OUTFIFO_THRESHOLD_WIDTH 6
+#define ASRMCRi_OUTFIFO_THRESHOLD_SHIFT 12
+#define ASRMCRi_OUTFIFO_THRESHOLD_MASK (((1 << ASRMCRi_OUTFIFO_THRESHOLD_WIDTH) - 1) << ASRMCRi_OUTFIFO_THRESHOLD_SHIFT)
+#define ASRMCRi_OUTFIFO_THRESHOLD(v) (((v) << ASRMCRi_OUTFIFO_THRESHOLD_SHIFT) & ASRMCRi_OUTFIFO_THRESHOLD_MASK)
+#define ASRMCRi_RSYNIFi_SHIFT 11
+#define ASRMCRi_RSYNIFi_MASK (1 << ASRMCRi_RSYNIFi_SHIFT)
+#define ASRMCRi_RSYNIFi (1 << ASRMCRi_RSYNIFi_SHIFT)
+#define ASRMCRi_RSYNOFi_SHIFT 10
+#define ASRMCRi_RSYNOFi_MASK (1 << ASRMCRi_RSYNOFi_SHIFT)
+#define ASRMCRi_RSYNOFi (1 << ASRMCRi_RSYNOFi_SHIFT)
+#define ASRMCRi_INFIFO_THRESHOLD_WIDTH 6
+#define ASRMCRi_INFIFO_THRESHOLD_SHIFT 0
+#define ASRMCRi_INFIFO_THRESHOLD_MASK (((1 << ASRMCRi_INFIFO_THRESHOLD_WIDTH) - 1) << ASRMCRi_INFIFO_THRESHOLD_SHIFT)
+#define ASRMCRi_INFIFO_THRESHOLD(v) (((v) << ASRMCRi_INFIFO_THRESHOLD_SHIFT) & ASRMCRi_INFIFO_THRESHOLD_MASK)
+
+/*
+ * REG23 0xA4 REG_ASRFSTA
+ * REG25 0xAC REG_ASRFSTB
+ * REG27 0xB4 REG_ASRFSTC
+ */
+#define ASRFSTi_OAFi_SHIFT 23
+#define ASRFSTi_OAFi_MASK (1 << ASRFSTi_OAFi_SHIFT)
+#define ASRFSTi_OAFi (1 << ASRFSTi_OAFi_SHIFT)
+#define ASRFSTi_OUTPUT_FIFO_WIDTH 7
+#define ASRFSTi_OUTPUT_FIFO_SHIFT 12
+#define ASRFSTi_OUTPUT_FIFO_MASK (((1 << ASRFSTi_OUTPUT_FIFO_WIDTH) - 1) << ASRFSTi_OUTPUT_FIFO_SHIFT)
+#define ASRFSTi_IAEi_SHIFT 11
+#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_OAFi_SHIFT)
+#define ASRFSTi_IAEi (1 << ASRFSTi_OAFi_SHIFT)
+#define ASRFSTi_INPUT_FIFO_WIDTH 7
+#define ASRFSTi_INPUT_FIFO_SHIFT 0
+#define ASRFSTi_INPUT_FIFO_MASK ((1 << ASRFSTi_INPUT_FIFO_WIDTH) - 1)
+
+/* REG28 0xC0 & 0xC4 & 0xC8 REG_ASRMCR1i */
+#define ASRMCR1i_IWD_WIDTH 3
+#define ASRMCR1i_IWD_SHIFT 9
+#define ASRMCR1i_IWD_MASK (((1 << ASRMCR1i_IWD_WIDTH) - 1) << ASRMCR1i_IWD_SHIFT)
+#define ASRMCR1i_IWD(v) ((v) << ASRMCR1i_IWD_SHIFT)
+#define ASRMCR1i_IMSB_SHIFT 8
+#define ASRMCR1i_IMSB_MASK (1 << ASRMCR1i_IMSB_SHIFT)
+#define ASRMCR1i_IMSB_MSB (1 << ASRMCR1i_IMSB_SHIFT)
+#define ASRMCR1i_IMSB_LSB (0 << ASRMCR1i_IMSB_SHIFT)
+#define ASRMCR1i_OMSB_SHIFT 2
+#define ASRMCR1i_OMSB_MASK (1 << ASRMCR1i_OMSB_SHIFT)
+#define ASRMCR1i_OMSB_MSB (1 << ASRMCR1i_OMSB_SHIFT)
+#define ASRMCR1i_OMSB_LSB (0 << ASRMCR1i_OMSB_SHIFT)
+#define ASRMCR1i_OSGN_SHIFT 1
+#define ASRMCR1i_OSGN_MASK (1 << ASRMCR1i_OSGN_SHIFT)
+#define ASRMCR1i_OSGN (1 << ASRMCR1i_OSGN_SHIFT)
+#define ASRMCR1i_OW16_SHIFT 0
+#define ASRMCR1i_OW16_MASK (1 << ASRMCR1i_OW16_SHIFT)
+#define ASRMCR1i_OW16(v) ((v) << ASRMCR1i_OW16_SHIFT)
+
+
+enum asrc_pair_index {
+ ASRC_INVALID_PAIR = -1,
+ ASRC_PAIR_A = 0,
+ ASRC_PAIR_B = 1,
+ ASRC_PAIR_C = 2,
+};
+
+#define ASRC_PAIR_MAX_NUM (ASRC_PAIR_C + 1)
+
+enum asrc_inclk {
+ INCLK_NONE = 0x03,
+ INCLK_ESAI_RX = 0x00,
+ INCLK_SSI1_RX = 0x01,
+ INCLK_SSI2_RX = 0x02,
+ INCLK_SSI3_RX = 0x07,
+ INCLK_SPDIF_RX = 0x04,
+ INCLK_MLB_CLK = 0x05,
+ INCLK_PAD = 0x06,
+ INCLK_ESAI_TX = 0x08,
+ INCLK_SSI1_TX = 0x09,
+ INCLK_SSI2_TX = 0x0a,
+ INCLK_SSI3_TX = 0x0b,
+ INCLK_SPDIF_TX = 0x0c,
+ INCLK_ASRCK1_CLK = 0x0f,
+};
+
+enum asrc_outclk {
+ OUTCLK_NONE = 0x03,
+ OUTCLK_ESAI_TX = 0x00,
+ OUTCLK_SSI1_TX = 0x01,
+ OUTCLK_SSI2_TX = 0x02,
+ OUTCLK_SSI3_TX = 0x07,
+ OUTCLK_SPDIF_TX = 0x04,
+ OUTCLK_MLB_CLK = 0x05,
+ OUTCLK_PAD = 0x06,
+ OUTCLK_ESAI_RX = 0x08,
+ OUTCLK_SSI1_RX = 0x09,
+ OUTCLK_SSI2_RX = 0x0a,
+ OUTCLK_SSI3_RX = 0x0b,
+ OUTCLK_SPDIF_RX = 0x0c,
+ OUTCLK_ASRCK1_CLK = 0x0f,
+};
+
+#define ASRC_CLK_MAX_NUM 16
+
+enum asrc_word_width {
+ ASRC_WIDTH_24_BIT = 0,
+ ASRC_WIDTH_16_BIT = 1,
+ ASRC_WIDTH_8_BIT = 2,
+};
+
+struct asrc_config {
+ enum asrc_pair_index pair;
+ unsigned int channel_num;
+ unsigned int buffer_num;
+ unsigned int dma_buffer_size;
+ unsigned int input_sample_rate;
+ unsigned int output_sample_rate;
+ enum asrc_word_width input_word_width;
+ enum asrc_word_width output_word_width;
+ enum asrc_inclk inclk;
+ enum asrc_outclk outclk;
+};
+
+struct asrc_req {
+ unsigned int chn_num;
+ enum asrc_pair_index index;
+};
+
+struct asrc_querybuf {
+ unsigned int buffer_index;
+ unsigned int input_length;
+ unsigned int output_length;
+ unsigned long input_offset;
+ unsigned long output_offset;
+};
+
+struct asrc_convert_buffer {
+ void *input_buffer_vaddr;
+ void *output_buffer_vaddr;
+ unsigned int input_buffer_length;
+ unsigned int output_buffer_length;
+};
+
+struct asrc_status_flags {
+ enum asrc_pair_index index;
+ unsigned int overload_error;
+};
+
+enum asrc_error_status {
+ ASRC_TASK_Q_OVERLOAD = 0x01,
+ ASRC_OUTPUT_TASK_OVERLOAD = 0x02,
+ ASRC_INPUT_TASK_OVERLOAD = 0x04,
+ ASRC_OUTPUT_BUFFER_OVERFLOW = 0x08,
+ ASRC_INPUT_BUFFER_UNDERRUN = 0x10,
+};
+
+struct dma_block {
+ dma_addr_t dma_paddr;
+ void *dma_vaddr;
+ unsigned int length;
+};
+
+/**
+ * fsl_asrc_pair: ASRC Pair private data
+ *
+ * @asrc_priv: pointer to its parent module
+ * @config: configuration profile
+ * @error: error record
+ * @index: pair index (ASRC_PAIR_A, ASRC_PAIR_B, ASRC_PAIR_C)
+ * @channels: occupied channel number
+ * @desc: input and output dma descriptors
+ * @dma_chan: inputer and output DMA channels
+ * @dma_data: private dma data
+ * @pos: hardware pointer position
+ * @private: pair private area
+ */
+struct fsl_asrc_pair {
+ struct fsl_asrc *asrc_priv;
+ struct asrc_config *config;
+ unsigned int error;
+
+ enum asrc_pair_index index;
+ unsigned int channels;
+
+ struct dma_async_tx_descriptor *desc[2];
+ struct dma_chan *dma_chan[2];
+ struct imx_dma_data dma_data;
+ unsigned int pos;
+
+ void *private;
+};
+
+/**
+ * fsl_asrc_pair: ASRC private data
+ *
+ * @dma_params_rx: DMA parameters for receive channel
+ * @dma_params_tx: DMA parameters for transmit channel
+ * @pdev: platform device pointer
+ * @regmap: regmap handler
+ * @paddr: physical address to the base address of registers
+ * @mem_clk: clock source to access register
+ * @ipg_clk: clock source to drive peripheral
+ * @asrck_clk: clock sources to driver ASRC internal logic
+ * @lock: spin lock for resource protection
+ * @pair: pair pointers
+ * @channel_bits: width of ASRCNCR register for each pair
+ * @channel_avail: non-occupied channel numbers
+ * @asrc_rate: default sample rate for ASoC Back-Ends
+ * @asrc_width: default sample width for ASoC Back-Ends
+ * @name: driver name
+ */
+struct fsl_asrc {
+ struct snd_dmaengine_dai_dma_data dma_params_rx;
+ struct snd_dmaengine_dai_dma_data dma_params_tx;
+ struct platform_device *pdev;
+ struct regmap *regmap;
+ unsigned long paddr;
+ struct clk *mem_clk;
+ struct clk *ipg_clk;
+ struct clk *asrck_clk[ASRC_CLK_MAX_NUM];
+ spinlock_t lock;
+
+ struct fsl_asrc_pair *pair[ASRC_PAIR_MAX_NUM];
+ unsigned int channel_bits;
+ unsigned int channel_avail;
+
+ int asrc_rate;
+ int asrc_width;
+
+ char name[32];
+};
+
+extern struct snd_soc_platform_driver fsl_asrc_platform;
+struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir);
+#endif /* _FSL_ASRC_H */
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
new file mode 100644
index 000000000000..ffc000bc1f15
--- /dev/null
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -0,0 +1,391 @@
+/*
+ * Freescale ASRC ALSA SoC Platform (DMA) driver
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <nicoleotsuka@gmail.com>
+ *
+ * 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/dma-mapping.h>
+#include <linux/module.h>
+#include <linux/platform_data/dma-imx.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_asrc.h"
+
+#define FSL_ASRC_DMABUF_SIZE (256 * 1024)
+
+static struct snd_pcm_hardware snd_imx_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .buffer_bytes_max = FSL_ASRC_DMABUF_SIZE,
+ .period_bytes_min = 128,
+ .period_bytes_max = 65535, /* Limited by SDMA engine */
+ .periods_min = 2,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+ if (!imx_dma_is_general_purpose(chan))
+ return false;
+
+ chan->private = param;
+
+ return true;
+}
+
+static void fsl_asrc_dma_complete(void *arg)
+{
+ struct snd_pcm_substream *substream = arg;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+
+ pair->pos += snd_pcm_lib_period_bytes(substream);
+ if (pair->pos >= snd_pcm_lib_buffer_bytes(substream))
+ pair->pos = 0;
+
+ snd_pcm_period_elapsed(substream);
+}
+
+static int fsl_asrc_dma_prepare_and_submit(struct snd_pcm_substream *substream)
+{
+ u8 dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? OUT : IN;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+ struct device *dev = rtd->platform->dev;
+ unsigned long flags = DMA_CTRL_ACK;
+
+ /* Prepare and submit Front-End DMA channel */
+ if (!substream->runtime->no_period_wakeup)
+ flags |= DMA_PREP_INTERRUPT;
+
+ pair->pos = 0;
+ pair->desc[!dir] = dmaengine_prep_dma_cyclic(
+ pair->dma_chan[!dir], runtime->dma_addr,
+ snd_pcm_lib_buffer_bytes(substream),
+ snd_pcm_lib_period_bytes(substream),
+ dir == OUT ? DMA_TO_DEVICE : DMA_FROM_DEVICE, flags);
+ if (!pair->desc[!dir]) {
+ dev_err(dev, "failed to prepare slave DMA for Front-End\n");
+ return -ENOMEM;
+ }
+
+ pair->desc[!dir]->callback = fsl_asrc_dma_complete;
+ pair->desc[!dir]->callback_param = substream;
+
+ dmaengine_submit(pair->desc[!dir]);
+
+ /* Prepare and submit Back-End DMA channel */
+ pair->desc[dir] = dmaengine_prep_dma_cyclic(
+ pair->dma_chan[dir], 0xffff, 64, 64, DMA_DEV_TO_DEV, 0);
+ if (!pair->desc[dir]) {
+ dev_err(dev, "failed to prepare slave DMA for Back-End\n");
+ return -ENOMEM;
+ }
+
+ dmaengine_submit(pair->desc[dir]);
+
+ return 0;
+}
+
+static int fsl_asrc_dma_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+ int ret;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ret = fsl_asrc_dma_prepare_and_submit(substream);
+ if (ret)
+ return ret;
+ dma_async_issue_pending(pair->dma_chan[IN]);
+ dma_async_issue_pending(pair->dma_chan[OUT]);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ dmaengine_terminate_all(pair->dma_chan[OUT]);
+ dmaengine_terminate_all(pair->dma_chan[IN]);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ struct snd_dmaengine_dai_dma_data *dma_params_fe = NULL;
+ struct snd_dmaengine_dai_dma_data *dma_params_be = NULL;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ struct dma_slave_config config_fe, config_be;
+ enum asrc_pair_index index = pair->index;
+ struct device *dev = rtd->platform->dev;
+ int stream = substream->stream;
+ struct imx_dma_data *tmp_data;
+ struct snd_soc_dpcm *dpcm;
+ struct dma_chan *tmp_chan;
+ struct device *dev_be;
+ u8 dir = tx ? OUT : IN;
+ dma_cap_mask_t mask;
+ int ret;
+
+ /* Fetch the Back-End dma_data from DPCM */
+ list_for_each_entry(dpcm, &rtd->dpcm[stream].be_clients, list_be) {
+ struct snd_soc_pcm_runtime *be = dpcm->be;
+ struct snd_pcm_substream *substream_be;
+ struct snd_soc_dai *dai = be->cpu_dai;
+
+ if (dpcm->fe != rtd)
+ continue;
+
+ substream_be = snd_soc_dpcm_get_substream(be, stream);
+ dma_params_be = snd_soc_dai_get_dma_data(dai, substream_be);
+ dev_be = dai->dev;
+ break;
+ }
+
+ if (!dma_params_be) {
+ dev_err(dev, "failed to get the substream of Back-End\n");
+ return -EINVAL;
+ }
+
+ /* Override dma_data of the Front-End and config its dmaengine */
+ dma_params_fe = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ dma_params_fe->addr = asrc_priv->paddr + REG_ASRDx(!dir, index);
+ dma_params_fe->maxburst = dma_params_be->maxburst;
+
+ pair->dma_chan[!dir] = fsl_asrc_get_dma_channel(pair, !dir);
+ if (!pair->dma_chan[!dir]) {
+ dev_err(dev, "failed to request DMA channel\n");
+ return -EINVAL;
+ }
+
+ memset(&config_fe, 0, sizeof(config_fe));
+ ret = snd_dmaengine_pcm_prepare_slave_config(substream, params, &config_fe);
+ if (ret) {
+ dev_err(dev, "failed to prepare DMA config for Front-End\n");
+ return ret;
+ }
+
+ ret = dmaengine_slave_config(pair->dma_chan[!dir], &config_fe);
+ if (ret) {
+ dev_err(dev, "failed to config DMA channel for Front-End\n");
+ return ret;
+ }
+
+ /* Request and config DMA channel for Back-End */
+ dma_cap_zero(mask);
+ dma_cap_set(DMA_SLAVE, mask);
+ dma_cap_set(DMA_CYCLIC, mask);
+
+ /* Get DMA request of Back-End */
+ tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx");
+ tmp_data = tmp_chan->private;
+ pair->dma_data.dma_request = tmp_data->dma_request;
+ dma_release_channel(tmp_chan);
+
+ /* Get DMA request of Front-End */
+ tmp_chan = fsl_asrc_get_dma_channel(pair, dir);
+ tmp_data = tmp_chan->private;
+ pair->dma_data.dma_request2 = tmp_data->dma_request;
+ pair->dma_data.peripheral_type = tmp_data->peripheral_type;
+ pair->dma_data.priority = tmp_data->priority;
+ dma_release_channel(tmp_chan);
+
+ pair->dma_chan[dir] = dma_request_channel(mask, filter, &pair->dma_data);
+ if (!pair->dma_chan[dir]) {
+ dev_err(dev, "failed to request DMA channel for Back-End\n");
+ return -EINVAL;
+ }
+
+ if (asrc_priv->asrc_width == 16)
+ buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
+ else
+ buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES;
+
+ config_be.direction = DMA_DEV_TO_DEV;
+ config_be.src_addr_width = buswidth;
+ config_be.src_maxburst = dma_params_be->maxburst;
+ config_be.dst_addr_width = buswidth;
+ config_be.dst_maxburst = dma_params_be->maxburst;
+
+ if (tx) {
+ config_be.src_addr = asrc_priv->paddr + REG_ASRDO(index);
+ config_be.dst_addr = dma_params_be->addr;
+ } else {
+ config_be.dst_addr = asrc_priv->paddr + REG_ASRDI(index);
+ config_be.src_addr = dma_params_be->addr;
+ }
+
+ ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be);
+ if (ret) {
+ dev_err(dev, "failed to config DMA channel for Back-End\n");
+ return ret;
+ }
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static int fsl_asrc_dma_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+
+ snd_pcm_set_runtime_buffer(substream, NULL);
+
+ if (pair->dma_chan[IN])
+ dma_release_channel(pair->dma_chan[IN]);
+
+ if (pair->dma_chan[OUT])
+ dma_release_channel(pair->dma_chan[OUT]);
+
+ pair->dma_chan[IN] = NULL;
+ pair->dma_chan[OUT] = NULL;
+
+ return 0;
+}
+
+static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct device *dev = rtd->platform->dev;
+ struct fsl_asrc *asrc_priv = dev_get_drvdata(dev);
+ struct fsl_asrc_pair *pair;
+
+ pair = kzalloc(sizeof(struct fsl_asrc_pair), GFP_KERNEL);
+ if (!pair) {
+ dev_err(dev, "failed to allocate pair\n");
+ return -ENOMEM;
+ }
+
+ pair->asrc_priv = asrc_priv;
+
+ runtime->private_data = pair;
+
+ snd_pcm_hw_constraint_integer(substream->runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);
+
+ return 0;
+}
+
+static int fsl_asrc_dma_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+ struct fsl_asrc *asrc_priv;
+
+ if (!pair)
+ return 0;
+
+ asrc_priv = pair->asrc_priv;
+
+ if (asrc_priv->pair[pair->index] == pair)
+ asrc_priv->pair[pair->index] = NULL;
+
+ kfree(pair);
+
+ return 0;
+}
+
+static snd_pcm_uframes_t fsl_asrc_dma_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+
+ return bytes_to_frames(substream->runtime, pair->pos);
+}
+
+static struct snd_pcm_ops fsl_asrc_dma_pcm_ops = {
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = fsl_asrc_dma_hw_params,
+ .hw_free = fsl_asrc_dma_hw_free,
+ .trigger = fsl_asrc_dma_trigger,
+ .open = fsl_asrc_dma_startup,
+ .close = fsl_asrc_dma_shutdown,
+ .pointer = fsl_asrc_dma_pcm_pointer,
+};
+
+static int fsl_asrc_dma_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret, i;
+
+ ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+ if (ret) {
+ dev_err(card->dev, "failed to set DMA mask\n");
+ return ret;
+ }
+
+ for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) {
+ substream = pcm->streams[i].substream;
+ if (!substream)
+ continue;
+
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev,
+ FSL_ASRC_DMABUF_SIZE, &substream->dma_buffer);
+ if (ret) {
+ dev_err(card->dev, "failed to allocate DMA buffer\n");
+ goto err;
+ }
+ }
+
+ return 0;
+
+err:
+ if (--i == 0 && pcm->streams[i].substream)
+ snd_dma_free_pages(&pcm->streams[i].substream->dma_buffer);
+
+ return ret;
+}
+
+static void fsl_asrc_dma_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ int i;
+
+ for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) {
+ substream = pcm->streams[i].substream;
+ if (!substream)
+ continue;
+
+ snd_dma_free_pages(&substream->dma_buffer);
+ substream->dma_buffer.area = NULL;
+ substream->dma_buffer.addr = 0;
+ }
+}
+
+struct snd_soc_platform_driver fsl_asrc_platform = {
+ .ops = &fsl_asrc_dma_pcm_ops,
+ .pcm_new = fsl_asrc_dma_pcm_new,
+ .pcm_free = fsl_asrc_dma_pcm_free,
+};
+EXPORT_SYMBOL_GPL(fsl_asrc_platform);
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index d719caf26dc2..72d154e7dd03 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -624,12 +624,14 @@ static int fsl_esai_dai_probe(struct snd_soc_dai *dai)
static struct snd_soc_dai_driver fsl_esai_dai = {
.probe = fsl_esai_dai_probe,
.playback = {
+ .stream_name = "CPU-Playback",
.channels_min = 1,
.channels_max = 12,
.rates = FSL_ESAI_RATES,
.formats = FSL_ESAI_FORMATS,
},
.capture = {
+ .stream_name = "CPU-Capture",
.channels_min = 1,
.channels_max = 8,
.rates = FSL_ESAI_RATES,
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index c5a0e8af8226..faa049797897 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -106,7 +106,7 @@ irq_rx:
xcsr &= ~FSL_SAI_CSR_xF_MASK;
if (flags)
- regmap_write(sai->regmap, FSL_SAI_TCSR, flags | xcsr);
+ regmap_write(sai->regmap, FSL_SAI_RCSR, flags | xcsr);
out:
if (irq_none)
@@ -327,7 +327,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
{
struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
- u32 tcsr, rcsr;
+ u32 xcsr, count = 100;
/*
* The transmitter bit clock and frame sync are to be
@@ -338,9 +338,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC,
FSL_SAI_CR2_SYNC);
- regmap_read(sai->regmap, FSL_SAI_TCSR, &tcsr);
- regmap_read(sai->regmap, FSL_SAI_RCSR, &rcsr);
-
/*
* It is recommended that the transmitter is the last enabled
* and the first disabled.
@@ -349,17 +346,16 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) {
- regmap_update_bits(sai->regmap, FSL_SAI_RCSR,
- FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE);
- regmap_update_bits(sai->regmap, FSL_SAI_TCSR,
- FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE);
- }
+ regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx),
+ FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE);
+
+ regmap_update_bits(sai->regmap, FSL_SAI_RCSR,
+ FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE);
+ regmap_update_bits(sai->regmap, FSL_SAI_TCSR,
+ FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE);
regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx),
FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS);
- regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx),
- FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
@@ -370,11 +366,24 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
FSL_SAI_CSR_xIE_MASK, 0);
/* Check if the opposite FRDE is also disabled */
- if (!(tx ? rcsr & FSL_SAI_CSR_FRDE : tcsr & FSL_SAI_CSR_FRDE)) {
+ regmap_read(sai->regmap, FSL_SAI_xCSR(!tx), &xcsr);
+ if (!(xcsr & FSL_SAI_CSR_FRDE)) {
+ /* Disable both directions and reset their FIFOs */
regmap_update_bits(sai->regmap, FSL_SAI_TCSR,
FSL_SAI_CSR_TERE, 0);
regmap_update_bits(sai->regmap, FSL_SAI_RCSR,
FSL_SAI_CSR_TERE, 0);
+
+ /* TERE will remain set till the end of current frame */
+ do {
+ udelay(10);
+ regmap_read(sai->regmap, FSL_SAI_xCSR(tx), &xcsr);
+ } while (--count && xcsr & FSL_SAI_CSR_TERE);
+
+ regmap_update_bits(sai->regmap, FSL_SAI_TCSR,
+ FSL_SAI_CSR_FR, FSL_SAI_CSR_FR);
+ regmap_update_bits(sai->regmap, FSL_SAI_RCSR,
+ FSL_SAI_CSR_FR, FSL_SAI_CSR_FR);
}
break;
default:
@@ -446,12 +455,14 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
static struct snd_soc_dai_driver fsl_sai_dai = {
.probe = fsl_sai_dai_probe,
.playback = {
+ .stream_name = "CPU-Playback",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = FSL_SAI_FORMATS,
},
.capture = {
+ .stream_name = "CPU-Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index d7a60614dd21..70acfe4a9bd5 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -32,10 +32,13 @@
#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)
+#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)
+
+#define SIE_INTR_FOR(tx) (tx ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE)
/* Index list for the values that has if (DPLL Locked) condition */
static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb };
@@ -96,7 +99,7 @@ struct fsl_spdif_priv {
struct platform_device *pdev;
struct regmap *regmap;
bool dpll_locked;
- u16 txrate[SPDIF_TXRATE_MAX];
+ u32 txrate[SPDIF_TXRATE_MAX];
u8 txclk_df[SPDIF_TXRATE_MAX];
u8 sysclk_df[SPDIF_TXRATE_MAX];
u8 txclk_src[SPDIF_TXRATE_MAX];
@@ -137,10 +140,9 @@ static void spdif_irq_sym_error(struct fsl_spdif_priv *spdif_priv)
dev_dbg(&pdev->dev, "isr: receiver found illegal symbol\n");
- if (!spdif_priv->dpll_locked) {
- /* DPLL unlocked seems no audio stream */
+ /* Clear illegal symbol if DPLL unlocked since no audio stream */
+ if (!spdif_priv->dpll_locked)
regmap_update_bits(regmap, REG_SPDIF_SIE, INT_SYM_ERR, 0);
- }
}
/* U/Q Channel receive register full */
@@ -335,8 +337,8 @@ static void spdif_write_channel_status(struct fsl_spdif_priv *spdif_priv)
u32 ch_status;
ch_status = (bitrev8(ctrl->ch_status[0]) << 16) |
- (bitrev8(ctrl->ch_status[1]) << 8) |
- bitrev8(ctrl->ch_status[2]);
+ (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);
@@ -390,6 +392,14 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
rate = SPDIF_TXRATE_48000;
csfs = IEC958_AES3_CON_FS_48000;
break;
+ case 96000:
+ rate = SPDIF_TXRATE_96000;
+ csfs = IEC958_AES3_CON_FS_96000;
+ break;
+ case 192000:
+ rate = SPDIF_TXRATE_192000;
+ csfs = IEC958_AES3_CON_FS_192000;
+ break;
default:
dev_err(&pdev->dev, "unsupported sample rate %d\n", sample_rate);
return -EINVAL;
@@ -433,13 +443,12 @@ clk_set_bypass:
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_DF(txclk_df);
- mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK | STC_TXCLK_DF_MASK;
+ stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) |
+ STC_TXCLK_DF(txclk_df) | STC_SYSCLK_DF(sysclk_df);
+ mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK |
+ STC_TXCLK_DF_MASK | STC_SYSCLK_DF_MASK;
regmap_update_bits(regmap, REG_SPDIF_STC, mask, stc);
- regmap_update_bits(regmap, REG_SPDIF_STC,
- STC_SYSCLK_DF_MASK, STC_SYSCLK_DF(sysclk_df));
-
dev_dbg(&pdev->dev, "set sample rate to %dHz for %dHz playback\n",
spdif_priv->txrate[rate], sample_rate);
@@ -553,7 +562,7 @@ static int fsl_spdif_hw_params(struct snd_pcm_substream *substream,
return ret;
}
spdif_set_cstatus(ctrl, IEC958_AES3_CON_CLOCK,
- IEC958_AES3_CON_CLOCK_1000PPM);
+ IEC958_AES3_CON_CLOCK_1000PPM);
spdif_write_channel_status(spdif_priv);
} else {
/* Setup rx clock source */
@@ -569,9 +578,9 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
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;;
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ u32 intr = SIE_INTR_FOR(tx);
+ u32 dmaen = SCR_DMA_xX_EN(tx);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
@@ -662,9 +671,8 @@ static int fsl_spdif_capture_get(struct snd_kcontrol *kcontrol,
u32 cstatus, val;
regmap_read(regmap, REG_SPDIF_SIS, &val);
- if (!(val & INT_CNEW)) {
+ if (!(val & INT_CNEW))
return -EAGAIN;
- }
regmap_read(regmap, REG_SPDIF_SRCSH, &cstatus);
ucontrol->value.iec958.status[0] = (cstatus >> 16) & 0xFF;
@@ -693,15 +701,14 @@ static int fsl_spdif_subcode_get(struct snd_kcontrol *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;
+ int ret = -EAGAIN;
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;
+ ret = 0;
}
spin_unlock_irqrestore(&ctrl->ctl_lock, flags);
@@ -726,15 +733,14 @@ static int fsl_spdif_qget(struct snd_kcontrol *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;
+ int ret = -EAGAIN;
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;
+ ret = 0;
}
spin_unlock_irqrestore(&ctrl->ctl_lock, flags);
@@ -799,10 +805,10 @@ static int spdif_get_rxclk_rate(struct fsl_spdif_priv *spdif_priv,
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 */
+
+ /* Get bus clock from system */
+ if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED))
busclk_freq = clk_get_rate(spdif_priv->sysclk);
- }
/* FreqMeas_CLK = (BUS_CLK * FreqMeas) / 2 ^ 10 / GAINSEL / 128 */
tmpval64 = (u64) busclk_freq * freqmeas;
@@ -826,12 +832,12 @@ static int fsl_spdif_rxrate_get(struct snd_kcontrol *kcontrol,
{
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);
+ int rate = 0;
if (spdif_priv->dpll_locked)
- ucontrol->value.integer.value[0] = rate;
- else
- ucontrol->value.integer.value[0] = 0;
+ rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL);
+
+ ucontrol->value.integer.value[0] = rate;
return 0;
}
@@ -969,12 +975,14 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai)
static struct snd_soc_dai_driver fsl_spdif_dai = {
.probe = &fsl_spdif_dai_probe,
.playback = {
+ .stream_name = "CPU-Playback",
.channels_min = 2,
.channels_max = 2,
.rates = FSL_SPDIF_RATES_PLAYBACK,
.formats = FSL_SPDIF_FORMATS_PLAYBACK,
},
.capture = {
+ .stream_name = "CPU-Capture",
.channels_min = 2,
.channels_max = 2,
.rates = FSL_SPDIF_RATES_CAPTURE,
@@ -1046,7 +1054,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
struct clk *clk, u64 savesub,
enum spdif_txrate index, bool round)
{
- const u32 rate[] = { 32000, 44100, 48000 };
+ const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 };
bool is_sysclk = clk == spdif_priv->sysclk;
u64 rate_ideal, rate_actual, sub;
u32 sysclk_dfmin, sysclk_dfmax;
@@ -1105,7 +1113,7 @@ out:
static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
enum spdif_txrate index)
{
- const u32 rate[] = { 32000, 44100, 48000 };
+ const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 };
struct platform_device *pdev = spdif_priv->pdev;
struct device *dev = &pdev->dev;
u64 savesub = 100000, ret;
@@ -1238,12 +1246,12 @@ static int fsl_spdif_probe(struct platform_device *pdev)
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[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;
+ ctrl->ch_status[3] = IEC958_AES3_CON_FS_44100 |
+ IEC958_AES3_CON_CLOCK_1000PPM;
spdif_priv->dpll_locked = false;
diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h
index 16fde4b927d3..00bd3514c610 100644
--- a/sound/soc/fsl/fsl_spdif.h
+++ b/sound/soc/fsl/fsl_spdif.h
@@ -93,6 +93,8 @@
#define SCR_USRC_SEL_RECV (0x1 << SCR_USRC_SEL_OFFSET)
#define SCR_USRC_SEL_CHIP (0x3 << SCR_USRC_SEL_OFFSET)
+#define SCR_DMA_xX_EN(tx) (tx ? SCR_DMA_TX_EN : SCR_DMA_RX_EN)
+
/* SPDIF CDText control */
#define SRCD_CD_USER_OFFSET 1
#define SRCD_CD_USER (1 << SRCD_CD_USER_OFFSET)
@@ -164,8 +166,10 @@ enum spdif_txrate {
SPDIF_TXRATE_32000 = 0,
SPDIF_TXRATE_44100,
SPDIF_TXRATE_48000,
+ SPDIF_TXRATE_96000,
+ SPDIF_TXRATE_192000,
};
-#define SPDIF_TXRATE_MAX (SPDIF_TXRATE_48000 + 1)
+#define SPDIF_TXRATE_MAX (SPDIF_TXRATE_192000 + 1)
#define SPDIF_CSTATUS_BYTE 6
@@ -175,7 +179,9 @@ enum spdif_txrate {
#define FSL_SPDIF_RATES_PLAYBACK (SNDRV_PCM_RATE_32000 | \
SNDRV_PCM_RATE_44100 | \
- SNDRV_PCM_RATE_48000)
+ SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_192000)
#define FSL_SPDIF_RATES_CAPTURE (SNDRV_PCM_RATE_16000 | \
SNDRV_PCM_RATE_32000 | \
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 9bfef55d77d1..87eb5776a39b 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -590,8 +590,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
else
clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
- do_div(clkrate, factor);
- afreq = (u32)clkrate / (i + 1);
+ clkrate /= factor;
+ afreq = clkrate / (i + 1);
if (freq == afreq)
sub = 0;
@@ -1032,12 +1032,14 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = {
static struct snd_soc_dai_driver fsl_ssi_dai_template = {
.probe = fsl_ssi_dai_probe,
.playback = {
+ .stream_name = "CPU-Playback",
.channels_min = 1,
.channels_max = 2,
.rates = FSLSSI_I2S_RATES,
.formats = FSLSSI_I2S_FORMATS,
},
.capture = {
+ .stream_name = "CPU-Capture",
.channels_min = 1,
.channels_max = 2,
.rates = FSLSSI_I2S_RATES,
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index 267717aa96c1..46f9beb6b273 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -67,7 +67,7 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf,
{
ssize_t ret;
char *buf;
- int port = (int)file->private_data;
+ uintptr_t port = (uintptr_t)file->private_data;
u32 pdcr, ptcr;
if (audmux_clk) {
@@ -147,7 +147,7 @@ static const struct file_operations audmux_debugfs_fops = {
static void audmux_debugfs_init(void)
{
- int i;
+ uintptr_t i;
char buf[20];
audmux_debugfs_root = debugfs_create_dir("audmux", NULL);
@@ -157,10 +157,10 @@ static void audmux_debugfs_init(void)
}
for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) {
- snprintf(buf, sizeof(buf), "ssi%d", i);
+ snprintf(buf, sizeof(buf), "ssi%lu", i);
if (!debugfs_create_file(buf, 0444, audmux_debugfs_root,
(void *)i, &audmux_debugfs_fops))
- pr_warning("Failed to create AUDMUX port %d debugfs file\n",
+ pr_warning("Failed to create AUDMUX port %lu debugfs file\n",
i);
}
}
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 03a7fdcdf114..159e517fa09a 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -116,6 +116,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
{
struct device_node *node;
struct clk *clk;
+ u32 val;
int ret;
/*
@@ -151,10 +152,8 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
}
dai->sysclk = clk_get_rate(clk);
- } else if (of_property_read_bool(np, "system-clock-frequency")) {
- of_property_read_u32(np,
- "system-clock-frequency",
- &dai->sysclk);
+ } else if (!of_property_read_u32(np, "system-clock-frequency", &val)) {
+ dai->sysclk = val;
} else {
clk = of_clk_get(node, 0);
if (!IS_ERR(clk))
@@ -303,6 +302,7 @@ static int asoc_simple_card_parse_of(struct device_node *node,
{
struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link;
struct simple_dai_props *dai_props = priv->dai_props;
+ u32 val;
int ret;
/* parsing the card name from DT */
@@ -325,8 +325,9 @@ static int asoc_simple_card_parse_of(struct device_node *node,
}
/* Factor to mclk, used in hw_params() */
- of_property_read_u32(node, "simple-audio-card,mclk-fs",
- &priv->mclk_fs);
+ ret = of_property_read_u32(node, "simple-audio-card,mclk-fs", &val);
+ if (ret == 0)
+ priv->mclk_fs = val;
dev_dbg(dev, "New simple-card: %s\n", priv->snd_card.name ?
priv->snd_card.name : "");
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index c30fedb3e149..f5b4a9c79cdf 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -58,3 +58,15 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH
help
This adds audio driver for Intel Baytrail platform based boards
with the MAX98090 audio codec.
+
+config SND_SOC_INTEL_BROADWELL_MACH
+ tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint"
+ depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC
+ select SND_SOC_INTEL_HASWELL
+ select SND_COMPRESS_OFFLOAD
+ select SND_SOC_RT286
+ help
+ This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell
+ Ultrabook platforms.
+ Say Y if you have such a device
+ If unsure select "N".
diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile
index 4bfca79a42ba..7acbfc43a0c6 100644
--- a/sound/soc/intel/Makefile
+++ b/sound/soc/intel/Makefile
@@ -24,7 +24,9 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o
snd-soc-sst-haswell-objs := haswell.o
snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
+snd-soc-sst-broadwell-objs := broadwell.o
obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
+obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c
new file mode 100644
index 000000000000..0e550f14028f
--- /dev/null
+++ b/sound/soc/intel/broadwell.c
@@ -0,0 +1,251 @@
+/*
+ * Intel Broadwell Wildcatpoint SST Audio
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "sst-dsp.h"
+#include "sst-haswell-ipc.h"
+
+#include "../codecs/rt286.h"
+
+static const struct snd_soc_dapm_widget broadwell_widgets[] = {
+ SND_SOC_DAPM_HP("Headphones", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_MIC("DMIC1", NULL),
+ SND_SOC_DAPM_MIC("DMIC2", NULL),
+ SND_SOC_DAPM_LINE("Line Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route broadwell_rt286_map[] = {
+
+ /* speaker */
+ {"Speaker", NULL, "SPOR"},
+ {"Speaker", NULL, "SPOL"},
+
+ /* HP jack connectors - unknown if we have jack deteck */
+ {"Headphones", NULL, "HPO Pin"},
+
+ /* other jacks */
+ {"MIC1", NULL, "Mic Jack"},
+ {"LINE1", NULL, "Line Jack"},
+
+ /* digital mics */
+ {"DMIC1 Pin", NULL, "DMIC1"},
+ {"DMIC2 Pin", NULL, "DMIC2"},
+
+ /* CODEC BE connections */
+ {"SSP0 CODEC IN", NULL, "AIF1 Capture"},
+ {"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
+};
+
+static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* The ADSP will covert the FE rate to 48k, stereo */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set SSP0 to 16 bit */
+ snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+ SNDRV_PCM_HW_PARAM_FIRST_MASK],
+ SNDRV_PCM_FORMAT_S16_LE);
+ return 0;
+}
+
+static int broadwell_rt286_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_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000,
+ SND_SOC_CLOCK_IN);
+
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+ return ret;
+ }
+
+ return ret;
+}
+
+static struct snd_soc_ops broadwell_rt286_ops = {
+ .hw_params = broadwell_rt286_hw_params,
+};
+
+static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev);
+ struct sst_hsw *broadwell = pdata->dsp;
+ int ret;
+
+ /* Set ADSP SSP port settings */
+ ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0,
+ SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
+ SST_HSW_DEVICE_CLOCK_MASTER, 9);
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: failed to set device config\n");
+ return ret;
+ }
+
+ /* always connected - check HP for jack detect */
+ snd_soc_dapm_enable_pin(dapm, "Headphones");
+ snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_enable_pin(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin(dapm, "DMIC1");
+ snd_soc_dapm_enable_pin(dapm, "DMIC2");
+
+ return 0;
+}
+
+/* broadwell digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link broadwell_rt286_dais[] = {
+ /* Front End DAI links */
+ {
+ .name = "System PCM",
+ .stream_name = "System Playback",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .init = broadwell_rtd_init,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Offload0",
+ .stream_name = "Offload0 Playback",
+ .cpu_dai_name = "Offload0 Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Offload1",
+ .stream_name = "Offload1 Playback",
+ .cpu_dai_name = "Offload1 Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Loopback PCM",
+ .stream_name = "Loopback",
+ .cpu_dai_name = "Loopback Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 0,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ },
+ {
+ .name = "Capture PCM",
+ .stream_name = "Capture",
+ .cpu_dai_name = "Capture Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ },
+
+ /* Back End DAI links */
+ {
+ /* SSP0 - Codec */
+ .name = "Codec",
+ .be_id = 0,
+ .cpu_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "snd-soc-dummy",
+ .no_pcm = 1,
+ .codec_name = "i2c-INT343A:00",
+ .codec_dai_name = "rt286-aif1",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = broadwell_ssp0_fixup,
+ .ops = &broadwell_rt286_ops,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+};
+
+/* broadwell audio machine driver for WPT + RT286S */
+static struct snd_soc_card broadwell_rt286 = {
+ .name = "broadwell-rt286",
+ .owner = THIS_MODULE,
+ .dai_link = broadwell_rt286_dais,
+ .num_links = ARRAY_SIZE(broadwell_rt286_dais),
+ .dapm_widgets = broadwell_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets),
+ .dapm_routes = broadwell_rt286_map,
+ .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map),
+ .fully_routed = true,
+};
+
+static int broadwell_audio_probe(struct platform_device *pdev)
+{
+ broadwell_rt286.dev = &pdev->dev;
+
+ return snd_soc_register_card(&broadwell_rt286);
+}
+
+static int broadwell_audio_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&broadwell_rt286);
+ return 0;
+}
+
+static struct platform_driver broadwell_audio = {
+ .probe = broadwell_audio_probe,
+ .remove = broadwell_audio_remove,
+ .driver = {
+ .name = "broadwell-audio",
+ .owner = THIS_MODULE,
+ },
+};
+
+module_platform_driver(broadwell_audio)
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, Xingchao Wang");
+MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:broadwell-audio");
diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c
index 5fc98c64a3f4..b8b8af571ef1 100644
--- a/sound/soc/intel/byt-max98090.c
+++ b/sound/soc/intel/byt-max98090.c
@@ -39,8 +39,7 @@ static const struct snd_soc_dapm_widget byt_max98090_widgets[] = {
static const struct snd_soc_dapm_route byt_max98090_audio_map[] = {
{"IN34", NULL, "Headset Mic"},
- {"IN34", NULL, "MICBIAS"},
- {"MICBIAS", NULL, "Headset Mic"},
+ {"Headset Mic", NULL, "MICBIAS"},
{"DMICL", NULL, "Int Mic"},
{"Headphone", NULL, "HPL"},
{"Headphone", NULL, "HPR"},
@@ -64,14 +63,6 @@ static struct snd_soc_jack_pin hs_jack_pins[] = {
.pin = "Headset Mic",
.mask = SND_JACK_MICROPHONE,
},
- {
- .pin = "Ext Spk",
- .mask = SND_JACK_LINEOUT,
- },
- {
- .pin = "Int Mic",
- .mask = SND_JACK_LINEIN,
- },
};
static struct snd_soc_jack_gpio hs_jack_gpios[] = {
@@ -84,7 +75,8 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = {
{
.name = "mic-gpio",
.idx = 1,
- .report = SND_JACK_MICROPHONE | SND_JACK_LINEIN,
+ .invert = 1,
+ .report = SND_JACK_MICROPHONE,
.debounce_time = 200,
},
};
@@ -108,7 +100,8 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime)
}
/* Enable jack detection */
- ret = snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, jack);
+ ret = snd_soc_jack_new(codec, "Headset",
+ SND_JACK_LINEOUT | SND_JACK_HEADSET, jack);
if (ret)
return ret;
@@ -117,13 +110,9 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime)
if (ret)
return ret;
- ret = snd_soc_jack_add_gpiods(card->dev->parent, jack,
- ARRAY_SIZE(hs_jack_gpios),
- hs_jack_gpios);
- if (ret)
- return ret;
-
- return max98090_mic_detect(codec, jack);
+ return snd_soc_jack_add_gpiods(card->dev->parent, jack,
+ ARRAY_SIZE(hs_jack_gpios),
+ hs_jack_gpios);
}
static struct snd_soc_dai_link byt_max98090_dais[] = {
diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c
index 53d160d39972..234a58de3c53 100644
--- a/sound/soc/intel/byt-rt5640.c
+++ b/sound/soc/intel/byt-rt5640.c
@@ -34,6 +34,7 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = {
};
static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = {
+ {"Headset Mic", NULL, "MICBIAS1"},
{"IN2P", NULL, "Headset Mic"},
{"IN2N", NULL, "Headset Mic"},
{"DMIC1", NULL, "Internal Mic"},
diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h
new file mode 100644
index 000000000000..14063ab8c7c5
--- /dev/null
+++ b/sound/soc/intel/sst-atom-controls.h
@@ -0,0 +1,30 @@
+/*
+ * Copyright (C) 2013-14 Intel Corp
+ * Author: Ramesh Babu <ramesh.babu.koul@intel.com>
+ * Omair M Abdullah <omair.m.abdullah@intel.com>
+ * Samreen Nilofer <samreen.nilofer@intel.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; version 2 of the License.
+ *
+ * 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 __SST_CONTROLS_V2_H__
+#define __SST_CONTROLS_V2_H__
+
+enum {
+ MERR_DPCM_AUDIO = 0,
+ MERR_DPCM_COMPR,
+};
+
+
+#endif
diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c
index d207b22ea330..67673a2c0f41 100644
--- a/sound/soc/intel/sst-baytrail-ipc.c
+++ b/sound/soc/intel/sst-baytrail-ipc.c
@@ -122,6 +122,26 @@ struct sst_byt_tstamp {
u32 channel_peak[8];
} __packed;
+struct sst_byt_fw_version {
+ u8 build;
+ u8 minor;
+ u8 major;
+ u8 type;
+} __packed;
+
+struct sst_byt_fw_build_info {
+ u8 date[16];
+ u8 time[16];
+} __packed;
+
+struct sst_byt_fw_init {
+ struct sst_byt_fw_version fw_version;
+ struct sst_byt_fw_build_info build_info;
+ u16 result;
+ u8 module_id;
+ u8 debug_info;
+} __packed;
+
/* driver internal IPC message structure */
struct ipc_message {
struct list_head list;
@@ -868,6 +888,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata)
{
struct sst_byt *byt;
struct sst_fw *byt_sst_fw;
+ struct sst_byt_fw_init init;
int err;
dev_dbg(dev, "initialising Byt DSP IPC\n");
@@ -929,6 +950,15 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata)
goto boot_err;
}
+ /* show firmware information */
+ sst_dsp_inbox_read(byt->dsp, &init, sizeof(init));
+ dev_info(byt->dev, "FW version: %02x.%02x.%02x.%02x\n",
+ init.fw_version.major, init.fw_version.minor,
+ init.fw_version.build, init.fw_version.type);
+ dev_info(byt->dev, "Build type: %x\n", init.fw_version.type);
+ dev_info(byt->dev, "Build date: %s %s\n",
+ init.build_info.date, init.build_info.time);
+
pdata->dsp = byt;
byt->fw = byt_sst_fw;
diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c
index 8eab97368ea7..599401c0c655 100644
--- a/sound/soc/intel/sst-baytrail-pcm.c
+++ b/sound/soc/intel/sst-baytrail-pcm.c
@@ -32,7 +32,7 @@ static const struct snd_pcm_hardware sst_byt_pcm_hardware = {
SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_RESUME,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FORMAT_S24_LE,
+ SNDRV_PCM_FMTBIT_S24_LE,
.period_bytes_min = 384,
.period_bytes_max = 48000,
.periods_min = 2,
diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c
index 0b715b20a2d7..cd23060a0d86 100644
--- a/sound/soc/intel/sst-dsp.c
+++ b/sound/soc/intel/sst-dsp.c
@@ -224,19 +224,23 @@ EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64);
void sst_dsp_dump(struct sst_dsp *sst)
{
- sst->ops->dump(sst);
+ if (sst->ops->dump)
+ sst->ops->dump(sst);
}
EXPORT_SYMBOL_GPL(sst_dsp_dump);
void sst_dsp_reset(struct sst_dsp *sst)
{
- sst->ops->reset(sst);
+ if (sst->ops->reset)
+ sst->ops->reset(sst);
}
EXPORT_SYMBOL_GPL(sst_dsp_reset);
int sst_dsp_boot(struct sst_dsp *sst)
{
- sst->ops->boot(sst);
+ if (sst->ops->boot)
+ sst->ops->boot(sst);
+
return 0;
}
EXPORT_SYMBOL_GPL(sst_dsp_boot);
diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h
index e44423be66c4..3165dfa97408 100644
--- a/sound/soc/intel/sst-dsp.h
+++ b/sound/soc/intel/sst-dsp.h
@@ -52,7 +52,11 @@
#define SST_CLKCTL 0x78
#define SST_CSR2 0x80
#define SST_LTRC 0xE0
-#define SST_HDMC 0xE8
+#define SST_HMDC 0xE8
+
+#define SST_SHIM_BEGIN SST_CSR
+#define SST_SHIM_END SST_HDMC
+
#define SST_DBGO 0xF0
#define SST_SHIM_SIZE 0x100
@@ -73,6 +77,8 @@
#define SST_CSR_S0IOCS (0x1 << 21)
#define SST_CSR_S1IOCS (0x1 << 23)
#define SST_CSR_LPCS (0x1 << 31)
+#define SST_CSR_24MHZ_LPCS (SST_CSR_SBCS0 | SST_CSR_SBCS1 | SST_CSR_LPCS)
+#define SST_CSR_24MHZ_NO_LPCS (SST_CSR_SBCS0 | SST_CSR_SBCS1)
#define SST_BYT_CSR_RST (0x1 << 0)
#define SST_BYT_CSR_VECTOR_SEL (0x1 << 1)
#define SST_BYT_CSR_STALL (0x1 << 2)
@@ -92,6 +98,14 @@
#define SST_IMRX_DONE (0x1 << 0)
#define SST_BYT_IMRX_REQUEST (0x1 << 1)
+/* IMRD / IMD */
+#define SST_IMRD_DONE (0x1 << 0)
+#define SST_IMRD_BUSY (0x1 << 1)
+#define SST_IMRD_SSP0 (0x1 << 16)
+#define SST_IMRD_DMAC0 (0x1 << 21)
+#define SST_IMRD_DMAC1 (0x1 << 22)
+#define SST_IMRD_DMAC (SST_IMRD_DMAC0 | SST_IMRD_DMAC1)
+
/* IPCX / IPCC */
#define SST_IPCX_DONE (0x1 << 30)
#define SST_IPCX_BUSY (0x1 << 31)
@@ -118,9 +132,21 @@
/* LTRC */
#define SST_LTRC_VAL(x) (x << 0)
-/* HDMC */
-#define SST_HDMC_HDDA0(x) (x << 0)
-#define SST_HDMC_HDDA1(x) (x << 7)
+/* HMDC */
+#define SST_HMDC_HDDA0(x) (x << 0)
+#define SST_HMDC_HDDA1(x) (x << 7)
+#define SST_HMDC_HDDA_E0_CH0 1
+#define SST_HMDC_HDDA_E0_CH1 2
+#define SST_HMDC_HDDA_E0_CH2 4
+#define SST_HMDC_HDDA_E0_CH3 8
+#define SST_HMDC_HDDA_E1_CH0 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH0)
+#define SST_HMDC_HDDA_E1_CH1 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH1)
+#define SST_HMDC_HDDA_E1_CH2 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH2)
+#define SST_HMDC_HDDA_E1_CH3 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH3)
+#define SST_HMDC_HDDA_E0_ALLCH (SST_HMDC_HDDA_E0_CH0 | SST_HMDC_HDDA_E0_CH1 | \
+ SST_HMDC_HDDA_E0_CH2 | SST_HMDC_HDDA_E0_CH3)
+#define SST_HMDC_HDDA_E1_ALLCH (SST_HMDC_HDDA_E1_CH0 | SST_HMDC_HDDA_E1_CH1 | \
+ SST_HMDC_HDDA_E1_CH2 | SST_HMDC_HDDA_E1_CH3)
/* SST Vendor Defined Registers and bits */
@@ -130,11 +156,16 @@
#define SST_VDRTCTL3 0xaC
/* VDRTCTL0 */
+#define SST_VDRTCL0_APLLSE_MASK 1
#define SST_VDRTCL0_DSRAMPGE_SHIFT 16
#define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT)
#define SST_VDRTCL0_ISRAMPGE_SHIFT 6
#define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT)
+/* PMCS */
+#define SST_PMCS 0x84
+#define SST_PMCS_PS_MASK 0x3
+
struct sst_dsp;
/*
diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c
index 535f517629fd..4b6c163c10ff 100644
--- a/sound/soc/intel/sst-haswell-dsp.c
+++ b/sound/soc/intel/sst-haswell-dsp.c
@@ -28,9 +28,6 @@
#include <linux/firmware.h>
#include <linux/pm_runtime.h>
-#include <linux/acpi.h>
-#include <acpi/acpi_bus.h>
-
#include "sst-dsp.h"
#include "sst-dsp-priv.h"
#include "sst-haswell-ipc.h"
@@ -272,9 +269,9 @@ static void hsw_boot(struct sst_dsp *sst)
SST_CSR2_SDFD_SSP1);
/* enable DMA engine 0,1 all channels to access host memory */
- sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC,
- SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff),
- SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff));
+ sst_dsp_shim_update_bits_unlocked(sst, SST_HMDC,
+ SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff),
+ SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff));
/* disable all clock gating */
writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2);
@@ -313,9 +310,7 @@ static const struct sst_adsp_memregion lp_region[] = {
/* wild cat point ADSP mem regions */
static const struct sst_adsp_memregion wpt_region[] = {
- {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */
- {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */
- {0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */
+ {0x00000, 0xA0000, 20, SST_MEM_DRAM}, /* D-SRAM0,D-SRAM1,D-SRAM2 - 20 * 32kB */
{0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */
};
@@ -339,26 +334,56 @@ static int hsw_acpi_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata)
return 0;
}
+struct sst_sram_shift {
+ u32 dev_id; /* SST Device IDs */
+ u32 iram_shift;
+ u32 dram_shift;
+};
+
+static const struct sst_sram_shift sram_shift[] = {
+ {SST_DEV_ID_LYNX_POINT, 6, 16}, /* lp */
+ {SST_DEV_ID_WILDCAT_POINT, 2, 12}, /* wpt */
+};
static u32 hsw_block_get_bit(struct sst_mem_block *block)
{
- u32 bit = 0, shift = 0;
+ u32 bit = 0, shift = 0, index;
+ struct sst_dsp *sst = block->dsp;
- switch (block->type) {
- case SST_MEM_DRAM:
- shift = 16;
- break;
- case SST_MEM_IRAM:
- shift = 6;
- break;
- default:
- return 0;
+ for (index = 0; index < ARRAY_SIZE(sram_shift); index++) {
+ if (sram_shift[index].dev_id == sst->id)
+ break;
}
+ if (index < ARRAY_SIZE(sram_shift)) {
+ switch (block->type) {
+ case SST_MEM_DRAM:
+ shift = sram_shift[index].dram_shift;
+ break;
+ case SST_MEM_IRAM:
+ shift = sram_shift[index].iram_shift;
+ break;
+ default:
+ shift = 0;
+ }
+ } else
+ shift = 0;
+
bit = 1 << (block->index + shift);
return bit;
}
+/*dummy read a SRAM block.*/
+static void sst_mem_block_dummy_read(struct sst_mem_block *block)
+{
+ u32 size;
+ u8 tmp_buf[4];
+ struct sst_dsp *sst = block->dsp;
+
+ size = block->size > 4 ? 4 : block->size;
+ memcpy_fromio(tmp_buf, sst->addr.lpe + block->offset, size);
+}
+
/* enable 32kB memory block - locks held by caller */
static int hsw_block_enable(struct sst_mem_block *block)
{
@@ -378,6 +403,8 @@ static int hsw_block_enable(struct sst_mem_block *block)
/* wait 18 DSP clock ticks */
udelay(10);
+ /*add a dummy read before the SRAM block is written, otherwise the writing may miss bytes sometimes.*/
+ sst_mem_block_dummy_read(block);
return 0;
}
@@ -488,8 +515,9 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata)
}
}
- /* set default power gating mask */
- writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0);
+ /* set default power gating control, enable power gating control for all blocks. that is,
+ can't be accessed, please enable each block before accessing. */
+ writel(0xffffffff, sst->addr.pci_cfg + SST_VDRTCTL0);
return 0;
}
diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c
index 434236343ddf..b6291516dbbf 100644
--- a/sound/soc/intel/sst-haswell-ipc.c
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -183,7 +183,7 @@ struct sst_hsw_ipc_fw_ready {
u32 inbox_size;
u32 outbox_size;
u32 fw_info_size;
- u8 fw_info[1];
+ u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)];
} __attribute__((packed));
struct ipc_message {
@@ -457,9 +457,10 @@ static void ipc_tx_msgs(struct kthread_work *work)
return;
}
- /* if the DSP is busy we will TX messages after IRQ */
+ /* if the DSP is busy, we will TX messages after IRQ.
+ * also postpone if we are in the middle of procesing completion irq*/
ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX);
- if (ipcx & SST_IPCX_BUSY) {
+ if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) {
spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
return;
}
@@ -502,6 +503,7 @@ static int tx_wait_done(struct sst_hsw *hsw, struct ipc_message *msg,
ipc_shim_dbg(hsw, "message timeout");
trace_ipc_error("error message timeout for", msg->header);
+ list_del(&msg->list);
ret = -ETIMEDOUT;
} else {
@@ -569,6 +571,9 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header)
{
struct sst_hsw_ipc_fw_ready fw_ready;
u32 offset;
+ u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)];
+ char *tmp[5], *pinfo;
+ int i = 0;
offset = (header & 0x1FFFFFFF) << 3;
@@ -589,6 +594,19 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header)
fw_ready.inbox_offset, fw_ready.inbox_size);
dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n",
fw_ready.outbox_offset, fw_ready.outbox_size);
+ if (fw_ready.fw_info_size < sizeof(fw_ready.fw_info)) {
+ fw_ready.fw_info[fw_ready.fw_info_size] = 0;
+ dev_dbg(hsw->dev, " Firmware info: %s \n", fw_ready.fw_info);
+
+ /* log the FW version info got from the mailbox here. */
+ memcpy(fw_info, fw_ready.fw_info, fw_ready.fw_info_size);
+ pinfo = &fw_info[0];
+ for (i = 0; i < sizeof(tmp) / sizeof(char *); i++)
+ tmp[i] = strsep(&pinfo, " ");
+ dev_info(hsw->dev, "FW loaded, mailbox readback FW info: type %s, - "
+ "version: %s.%s, build %s, source commit id: %s\n",
+ tmp[0], tmp[1], tmp[2], tmp[3], tmp[4]);
+ }
}
static void hsw_notification_work(struct work_struct *work)
@@ -671,7 +689,9 @@ static void hsw_stream_update(struct sst_hsw *hsw, struct ipc_message *msg)
switch (stream_msg) {
case IPC_STR_STAGE_MESSAGE:
case IPC_STR_NOTIFICATION:
+ break;
case IPC_STR_RESET:
+ trace_ipc_notification("stream reset", stream->reply.stream_hw_id);
break;
case IPC_STR_PAUSE:
stream->running = false;
@@ -762,7 +782,8 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header)
}
/* update any stream states */
- hsw_stream_update(hsw, msg);
+ if (msg_get_global_type(header) == IPC_GLB_STREAM_MESSAGE)
+ hsw_stream_update(hsw, msg);
/* wake up and return the error if we have waiters on this message ? */
list_del(&msg->list);
@@ -1628,7 +1649,7 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw,
enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx)
{
u32 header, state_;
- int ret;
+ int ret, item;
header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE);
state_ = state;
@@ -1642,6 +1663,13 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw,
return ret;
}
+ for (item = 0; item < dx->entries_no; item++) {
+ dev_dbg(hsw->dev,
+ "Item[%d] offset[%x] - size[%x] - source[%x]\n",
+ item, dx->mem_info[item].offset,
+ dx->mem_info[item].size,
+ dx->mem_info[item].source);
+ }
dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n",
dx->entries_no, state);
@@ -1775,8 +1803,6 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata)
/* get the FW version */
sst_hsw_fw_get_version(hsw, &version);
- dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n",
- version.type, version.major, version.minor, version.build);
/* get the globalmixer */
ret = sst_hsw_mixer_get_info(hsw);
diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c
index 058efb17c568..61bf6da4bb02 100644
--- a/sound/soc/intel/sst-haswell-pcm.c
+++ b/sound/soc/intel/sst-haswell-pcm.c
@@ -80,7 +80,7 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = {
SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE |
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
SNDRV_PCM_FMTBIT_S32_LE,
.period_bytes_min = PAGE_SIZE,
.period_bytes_max = (HSW_PCM_PERIODS_MAX / HSW_PCM_PERIODS_MIN) * PAGE_SIZE,
@@ -400,7 +400,15 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream,
sst_hsw_stream_set_valid(hsw, pcm_data->stream, 16);
break;
case SNDRV_PCM_FORMAT_S24_LE:
- bits = SST_HSW_DEPTH_24BIT;
+ bits = SST_HSW_DEPTH_32BIT;
+ sst_hsw_stream_set_valid(hsw, pcm_data->stream, 24);
+ break;
+ case SNDRV_PCM_FORMAT_S8:
+ bits = SST_HSW_DEPTH_8BIT;
+ sst_hsw_stream_set_valid(hsw, pcm_data->stream, 8);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ bits = SST_HSW_DEPTH_32BIT;
sst_hsw_stream_set_valid(hsw, pcm_data->stream, 32);
break;
default:
@@ -685,8 +693,9 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd)
}
#define HSW_FORMATS \
- (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\
- SNDRV_PCM_FMTBIT_S32_LE)
+ (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S8)
static struct snd_soc_dai_driver hsw_dais[] = {
{
@@ -696,7 +705,7 @@ static struct snd_soc_dai_driver hsw_dais[] = {
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE,
},
},
{
@@ -727,8 +736,8 @@ static struct snd_soc_dai_driver hsw_dais[] = {
.stream_name = "Loopback Capture",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_192000,
- .formats = HSW_FORMATS,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE,
},
},
{
@@ -737,8 +746,8 @@ static struct snd_soc_dai_driver hsw_dais[] = {
.stream_name = "Analog Capture",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_192000,
- .formats = HSW_FORMATS,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE,
},
},
};
diff --git a/sound/soc/intel/sst-mfld-dsp.h b/sound/soc/intel/sst-mfld-dsp.h
index 8d482d76475a..4257263157cd 100644
--- a/sound/soc/intel/sst-mfld-dsp.h
+++ b/sound/soc/intel/sst-mfld-dsp.h
@@ -3,7 +3,7 @@
/*
* sst_mfld_dsp.h - Intel SST Driver for audio engine
*
- * Copyright (C) 2008-12 Intel Corporation
+ * Copyright (C) 2008-14 Intel Corporation
* Authors: Vinod Koul <vinod.koul@linux.intel.com>
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
@@ -19,6 +19,142 @@
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*/
+#define SST_MAX_BIN_BYTES 1024
+
+#define MAX_DBG_RW_BYTES 80
+#define MAX_NUM_SCATTER_BUFFERS 8
+#define MAX_LOOP_BACK_DWORDS 8
+/* IPC base address and mailbox, timestamp offsets */
+#define SST_MAILBOX_SIZE 0x0400
+#define SST_MAILBOX_SEND 0x0000
+#define SST_TIME_STAMP 0x1800
+#define SST_TIME_STAMP_MRFLD 0x800
+#define SST_RESERVED_OFFSET 0x1A00
+#define SST_SCU_LPE_MAILBOX 0x1000
+#define SST_LPE_SCU_MAILBOX 0x1400
+#define SST_SCU_LPE_LOG_BUF (SST_SCU_LPE_MAILBOX+16)
+#define PROCESS_MSG 0x80
+
+/* Message ID's for IPC messages */
+/* Bits B7: SST or IA/SC ; B6-B4: Msg Category; B3-B0: Msg Type */
+
+/* I2L Firmware/Codec Download msgs */
+#define IPC_IA_PREP_LIB_DNLD 0x01
+#define IPC_IA_LIB_DNLD_CMPLT 0x02
+#define IPC_IA_GET_FW_VERSION 0x04
+#define IPC_IA_GET_FW_BUILD_INF 0x05
+#define IPC_IA_GET_FW_INFO 0x06
+#define IPC_IA_GET_FW_CTXT 0x07
+#define IPC_IA_SET_FW_CTXT 0x08
+#define IPC_IA_PREPARE_SHUTDOWN 0x31
+/* I2L Codec Config/control msgs */
+#define IPC_PREP_D3 0x10
+#define IPC_IA_SET_CODEC_PARAMS 0x10
+#define IPC_IA_GET_CODEC_PARAMS 0x11
+#define IPC_IA_SET_PPP_PARAMS 0x12
+#define IPC_IA_GET_PPP_PARAMS 0x13
+#define IPC_SST_PERIOD_ELAPSED_MRFLD 0xA
+#define IPC_IA_ALG_PARAMS 0x1A
+#define IPC_IA_TUNING_PARAMS 0x1B
+#define IPC_IA_SET_RUNTIME_PARAMS 0x1C
+#define IPC_IA_SET_PARAMS 0x1
+#define IPC_IA_GET_PARAMS 0x2
+
+#define IPC_EFFECTS_CREATE 0xE
+#define IPC_EFFECTS_DESTROY 0xF
+
+/* I2L Stream config/control msgs */
+#define IPC_IA_ALLOC_STREAM_MRFLD 0x2
+#define IPC_IA_ALLOC_STREAM 0x20 /* Allocate a stream ID */
+#define IPC_IA_FREE_STREAM_MRFLD 0x03
+#define IPC_IA_FREE_STREAM 0x21 /* Free the stream ID */
+#define IPC_IA_SET_STREAM_PARAMS 0x22
+#define IPC_IA_SET_STREAM_PARAMS_MRFLD 0x12
+#define IPC_IA_GET_STREAM_PARAMS 0x23
+#define IPC_IA_PAUSE_STREAM 0x24
+#define IPC_IA_PAUSE_STREAM_MRFLD 0x4
+#define IPC_IA_RESUME_STREAM 0x25
+#define IPC_IA_RESUME_STREAM_MRFLD 0x5
+#define IPC_IA_DROP_STREAM 0x26
+#define IPC_IA_DROP_STREAM_MRFLD 0x07
+#define IPC_IA_DRAIN_STREAM 0x27 /* Short msg with str_id */
+#define IPC_IA_DRAIN_STREAM_MRFLD 0x8
+#define IPC_IA_CONTROL_ROUTING 0x29
+#define IPC_IA_VTSV_UPDATE_MODULES 0x20
+#define IPC_IA_VTSV_DETECTED 0x21
+
+#define IPC_IA_START_STREAM_MRFLD 0X06
+#define IPC_IA_START_STREAM 0x30 /* Short msg with str_id */
+
+#define IPC_IA_SET_GAIN_MRFLD 0x21
+/* Debug msgs */
+#define IPC_IA_DBG_MEM_READ 0x40
+#define IPC_IA_DBG_MEM_WRITE 0x41
+#define IPC_IA_DBG_LOOP_BACK 0x42
+#define IPC_IA_DBG_LOG_ENABLE 0x45
+#define IPC_IA_DBG_SET_PROBE_PARAMS 0x47
+
+/* L2I Firmware/Codec Download msgs */
+#define IPC_IA_FW_INIT_CMPLT 0x81
+#define IPC_IA_FW_INIT_CMPLT_MRFLD 0x01
+#define IPC_IA_FW_ASYNC_ERR_MRFLD 0x11
+
+/* L2I Codec Config/control msgs */
+#define IPC_SST_FRAGMENT_ELPASED 0x90 /* Request IA more data */
+
+#define IPC_SST_BUF_UNDER_RUN 0x92 /* PB Under run and stopped */
+#define IPC_SST_BUF_OVER_RUN 0x93 /* CAP Under run and stopped */
+#define IPC_SST_DRAIN_END 0x94 /* PB Drain complete and stopped */
+#define IPC_SST_CHNGE_SSP_PARAMS 0x95 /* PB SSP parameters changed */
+#define IPC_SST_STREAM_PROCESS_FATAL_ERR 0x96/* error in processing a stream */
+#define IPC_SST_PERIOD_ELAPSED 0x97 /* period elapsed */
+
+#define IPC_SST_ERROR_EVENT 0x99 /* Buffer over run occurred */
+/* L2S messages */
+#define IPC_SC_DDR_LINK_UP 0xC0
+#define IPC_SC_DDR_LINK_DOWN 0xC1
+#define IPC_SC_SET_LPECLK_REQ 0xC2
+#define IPC_SC_SSP_BIT_BANG 0xC3
+
+/* L2I Error reporting msgs */
+#define IPC_IA_MEM_ALLOC_FAIL 0xE0
+#define IPC_IA_PROC_ERR 0xE1 /* error in processing a
+ stream can be used by playback and
+ capture modules */
+
+/* L2I Debug msgs */
+#define IPC_IA_PRINT_STRING 0xF0
+
+/* Buffer under-run */
+#define IPC_IA_BUF_UNDER_RUN_MRFLD 0x0B
+
+/* Mrfld specific defines:
+ * For asynchronous messages(INIT_CMPLT, PERIOD_ELAPSED, ASYNC_ERROR)
+ * received from FW, the format is:
+ * - IPC High: pvt_id is set to zero. Always short message.
+ * - msg_id is in lower 16-bits of IPC low payload.
+ * - pipe_id is in higher 16-bits of IPC low payload for period_elapsed.
+ * - error id is in higher 16-bits of IPC low payload for async errors.
+ */
+#define SST_ASYNC_DRV_ID 0
+
+/* Command Response or Acknowledge message to any IPC message will have
+ * same message ID and stream ID information which is sent.
+ * There is no specific Ack message ID. The data field is used as response
+ * meaning.
+ */
+enum ackData {
+ IPC_ACK_SUCCESS = 0,
+ IPC_ACK_FAILURE,
+};
+
+enum ipc_ia_msg_id {
+ IPC_CMD = 1, /*!< Task Control message ID */
+ IPC_SET_PARAMS = 2,/*!< Task Set param message ID */
+ IPC_GET_PARAMS = 3, /*!< Task Get param message ID */
+ IPC_INVALID = 0xFF, /*!<Task Get param message ID */
+};
+
enum sst_codec_types {
/* AUDIO/MUSIC CODEC Type Definitions */
SST_CODEC_TYPE_UNKNOWN = 0,
@@ -35,14 +171,157 @@ enum stream_type {
SST_STREAM_TYPE_MUSIC = 1,
};
+enum sst_error_codes {
+ /* Error code,response to msgId: Description */
+ /* Common error codes */
+ SST_SUCCESS = 0, /* Success */
+ SST_ERR_INVALID_STREAM_ID = 1,
+ SST_ERR_INVALID_MSG_ID = 2,
+ SST_ERR_INVALID_STREAM_OP = 3,
+ SST_ERR_INVALID_PARAMS = 4,
+ SST_ERR_INVALID_CODEC = 5,
+ SST_ERR_INVALID_MEDIA_TYPE = 6,
+ SST_ERR_STREAM_ERR = 7,
+
+ SST_ERR_STREAM_IN_USE = 15,
+};
+
+struct ipc_dsp_hdr {
+ u16 mod_index_id:8; /*!< DSP Command ID specific to tasks */
+ u16 pipe_id:8; /*!< instance of the module in the pipeline */
+ u16 mod_id; /*!< Pipe_id */
+ u16 cmd_id; /*!< Module ID = lpe_algo_types_t */
+ u16 length; /*!< Length of the payload only */
+} __packed;
+
+union ipc_header_high {
+ struct {
+ u32 msg_id:8; /* Message ID - Max 256 Message Types */
+ u32 task_id:4; /* Task ID associated with this comand */
+ u32 drv_id:4; /* Identifier for the driver to track*/
+ u32 rsvd1:8; /* Reserved */
+ u32 result:4; /* Reserved */
+ u32 res_rqd:1; /* Response rqd */
+ u32 large:1; /* Large Message if large = 1 */
+ u32 done:1; /* bit 30 - Done bit */
+ u32 busy:1; /* bit 31 - busy bit*/
+ } part;
+ u32 full;
+} __packed;
+/* IPC header */
+union ipc_header_mrfld {
+ struct {
+ u32 header_low_payload;
+ union ipc_header_high header_high;
+ } p;
+ u64 full;
+} __packed;
+/* CAUTION NOTE: All IPC message body must be multiple of 32 bits.*/
+
+/* IPC Header */
+union ipc_header {
+ struct {
+ u32 msg_id:8; /* Message ID - Max 256 Message Types */
+ u32 str_id:5;
+ u32 large:1; /* Large Message if large = 1 */
+ u32 reserved:2; /* Reserved for future use */
+ u32 data:14; /* Ack/Info for msg, size of msg in Mailbox */
+ u32 done:1; /* bit 30 */
+ u32 busy:1; /* bit 31 */
+ } part;
+ u32 full;
+} __packed;
+
+/* Firmware build info */
+struct sst_fw_build_info {
+ unsigned char date[16]; /* Firmware build date */
+ unsigned char time[16]; /* Firmware build time */
+} __packed;
+
+/* Firmware Version info */
+struct snd_sst_fw_version {
+ u8 build; /* build number*/
+ u8 minor; /* minor number*/
+ u8 major; /* major number*/
+ u8 type; /* build type */
+};
+
+struct ipc_header_fw_init {
+ struct snd_sst_fw_version fw_version;/* Firmware version details */
+ struct sst_fw_build_info build_info;
+ u16 result; /* Fw init result */
+ u8 module_id; /* Module ID in case of error */
+ u8 debug_info; /* Debug info from Module ID in case of fail */
+} __packed;
+
+struct snd_sst_tstamp {
+ u64 ring_buffer_counter; /* PB/CP: Bytes copied from/to DDR. */
+ u64 hardware_counter; /* PB/CP: Bytes DMAed to/from SSP. */
+ u64 frames_decoded;
+ u64 bytes_decoded;
+ u64 bytes_copied;
+ u32 sampling_frequency;
+ u32 channel_peak[8];
+} __packed;
+
+/* Stream type params struture for Alloc stream */
+struct snd_sst_str_type {
+ u8 codec_type; /* Codec type */
+ u8 str_type; /* 1 = voice 2 = music */
+ u8 operation; /* Playback or Capture */
+ u8 protected_str; /* 0=Non DRM, 1=DRM */
+ u8 time_slots;
+ u8 reserved; /* Reserved */
+ u16 result; /* Result used for acknowledgment */
+} __packed;
+
+/* Library info structure */
+struct module_info {
+ u32 lib_version;
+ u32 lib_type;/*TBD- KLOCKWORK u8 lib_type;*/
+ u32 media_type;
+ u8 lib_name[12];
+ u32 lib_caps;
+ unsigned char b_date[16]; /* Lib build date */
+ unsigned char b_time[16]; /* Lib build time */
+} __packed;
+
+/* Library slot info */
+struct lib_slot_info {
+ u8 slot_num; /* 1 or 2 */
+ u8 reserved1;
+ u16 reserved2;
+ u32 iram_size; /* slot size in IRAM */
+ u32 dram_size; /* slot size in DRAM */
+ u32 iram_offset; /* starting offset of slot in IRAM */
+ u32 dram_offset; /* starting offset of slot in DRAM */
+} __packed;
+
+struct snd_ppp_mixer_params {
+ __u32 type; /*Type of the parameter */
+ __u32 size;
+ __u32 input_stream_bitmap; /*Input stream Bit Map*/
+} __packed;
+
+struct snd_sst_lib_download {
+ struct module_info lib_info; /* library info type, capabilities etc */
+ struct lib_slot_info slot_info; /* slot info to be downloaded */
+ u32 mod_entry_pt;
+};
+
+struct snd_sst_lib_download_info {
+ struct snd_sst_lib_download dload_lib;
+ u16 result; /* Result used for acknowledgment */
+ u8 pvt_id; /* Private ID */
+ u8 reserved; /* for alignment */
+};
struct snd_pcm_params {
u8 num_chan; /* 1=Mono, 2=Stereo */
u8 pcm_wd_sz; /* 16/24 - bit*/
- u32 reserved; /* Bitrate in bits per second */
- u32 sfreq; /* Sampling rate in Hz */
- u8 use_offload_path;
+ u8 use_offload_path; /* 0-PCM using period elpased & ALSA interfaces
+ 1-PCM stream via compressed interface */
u8 reserved2;
- u16 reserved3;
+ u32 sfreq; /* Sampling rate in Hz */
u8 channel_map[8];
} __packed;
@@ -76,6 +355,7 @@ struct snd_aac_params {
struct snd_wma_params {
u8 num_chan; /* 1=Mono, 2=Stereo */
u8 pcm_wd_sz; /* 16/24 - bit*/
+ u16 reserved1;
u32 brate; /* Use the hard coded value. */
u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */
u32 channel_mask; /* Channel Mask */
@@ -101,26 +381,153 @@ struct sst_address_info {
};
struct snd_sst_alloc_params_ext {
- struct sst_address_info ring_buf_info[8];
- u8 sg_count;
- u8 reserved;
- u16 reserved2;
- u32 frag_size; /*Number of samples after which period elapsed
+ __u16 sg_count;
+ __u16 reserved;
+ __u32 frag_size; /*Number of samples after which period elapsed
message is sent valid only if path = 0*/
-} __packed;
+ struct sst_address_info ring_buf_info[8];
+};
struct snd_sst_stream_params {
union snd_sst_codec_params uc;
} __packed;
struct snd_sst_params {
+ u32 result;
u32 stream_id;
u8 codec;
u8 ops;
u8 stream_type;
u8 device_type;
+ u8 task;
struct snd_sst_stream_params sparams;
struct snd_sst_alloc_params_ext aparams;
};
+struct snd_sst_alloc_mrfld {
+ u16 codec_type;
+ u8 operation;
+ u8 sg_count;
+ struct sst_address_info ring_buf_info[8];
+ u32 frag_size;
+ u32 ts;
+ struct snd_sst_stream_params codec_params;
+} __packed;
+
+/* Alloc stream params structure */
+struct snd_sst_alloc_params {
+ struct snd_sst_str_type str_type;
+ struct snd_sst_stream_params stream_params;
+ struct snd_sst_alloc_params_ext alloc_params;
+} __packed;
+
+/* Alloc stream response message */
+struct snd_sst_alloc_response {
+ struct snd_sst_str_type str_type; /* Stream type for allocation */
+ struct snd_sst_lib_download lib_dnld; /* Valid only for codec dnld */
+};
+
+/* Drop response */
+struct snd_sst_drop_response {
+ u32 result;
+ u32 bytes;
+};
+
+struct snd_sst_async_msg {
+ u32 msg_id; /* Async msg id */
+ u32 payload[0];
+};
+
+struct snd_sst_async_err_msg {
+ u32 fw_resp; /* Firmware Result */
+ u32 lib_resp; /*Library result */
+} __packed;
+
+struct snd_sst_vol {
+ u32 stream_id;
+ s32 volume;
+ u32 ramp_duration;
+ u32 ramp_type; /* Ramp type, default=0 */
+};
+
+/* Gain library parameters for mrfld
+ * based on DSP command spec v0.82
+ */
+struct snd_sst_gain_v2 {
+ u16 gain_cell_num; /* num of gain cells to modify*/
+ u8 cell_nbr_idx; /* instance index*/
+ u8 cell_path_idx; /* pipe-id */
+ u16 module_id; /*module id */
+ u16 left_cell_gain; /* left gain value in dB*/
+ u16 right_cell_gain; /* right gain value in dB*/
+ u16 gain_time_const; /* gain time constant*/
+} __packed;
+
+struct snd_sst_mute {
+ u32 stream_id;
+ u32 mute;
+};
+
+struct snd_sst_runtime_params {
+ u8 type;
+ u8 str_id;
+ u8 size;
+ u8 rsvd;
+ void *addr;
+} __packed;
+
+enum stream_param_type {
+ SST_SET_TIME_SLOT = 0,
+ SST_SET_CHANNEL_INFO = 1,
+ OTHERS = 2, /*reserved for future params*/
+};
+
+/* CSV Voice call routing structure */
+struct snd_sst_control_routing {
+ u8 control; /* 0=start, 1=Stop */
+ u8 reserved[3]; /* Reserved- for 32 bit alignment */
+};
+
+struct ipc_post {
+ struct list_head node;
+ union ipc_header header; /* driver specific */
+ bool is_large;
+ bool is_process_reply;
+ union ipc_header_mrfld mrfld_header;
+ char *mailbox_data;
+};
+
+struct snd_sst_ctxt_params {
+ u32 address; /* Physical Address in DDR where the context is stored */
+ u32 size; /* size of the context */
+};
+
+struct snd_sst_lpe_log_params {
+ u8 dbg_type;
+ u8 module_id;
+ u8 log_level;
+ u8 reserved;
+} __packed;
+
+enum snd_sst_bytes_type {
+ SND_SST_BYTES_SET = 0x1,
+ SND_SST_BYTES_GET = 0x2,
+};
+
+struct snd_sst_bytes_v2 {
+ u8 type;
+ u8 ipc_msg;
+ u8 block;
+ u8 task_id;
+ u8 pipe_id;
+ u8 rsvd;
+ u16 len;
+ char bytes[0];
+};
+
+#define MAX_VTSV_FILES 2
+struct snd_sst_vtsv_info {
+ struct sst_address_info vfiles[MAX_VTSV_FILES];
+} __packed;
+
#endif /* __SST_MFLD_DSP_H__ */
diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/sst-mfld-platform-compress.c
index 02abd19fce1d..29c059ca19e8 100644
--- a/sound/soc/intel/sst-mfld-platform-compress.c
+++ b/sound/soc/intel/sst-mfld-platform-compress.c
@@ -100,14 +100,19 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream,
int retval;
struct snd_sst_params str_params;
struct sst_compress_cb cb;
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct sst_data *ctx = snd_soc_platform_get_drvdata(platform);
stream = cstream->runtime->private_data;
/* construct fw structure for this*/
memset(&str_params, 0, sizeof(str_params));
- str_params.ops = STREAM_OPS_PLAYBACK;
- str_params.stream_type = SST_STREAM_TYPE_MUSIC;
- str_params.device_type = SND_SST_DEVICE_COMPRESS;
+ /* fill the device type and stream id to pass to SST driver */
+ retval = sst_fill_stream_params(cstream, ctx, &str_params, true);
+ pr_debug("compr_set_params: fill stream params ret_val = 0x%x\n", retval);
+ if (retval < 0)
+ return retval;
switch (params->codec.id) {
case SND_AUDIOCODEC_MP3: {
diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c
index 7c790f51d259..706212a6a68c 100644
--- a/sound/soc/intel/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/sst-mfld-platform-pcm.c
@@ -1,7 +1,7 @@
/*
* sst_mfld_platform.c - Intel MID Platform driver
*
- * Copyright (C) 2010-2013 Intel Corp
+ * Copyright (C) 2010-2014 Intel Corp
* Author: Vinod Koul <vinod.koul@intel.com>
* Author: Harsha Priya <priya.harsha@intel.com>
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
@@ -27,7 +27,9 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/compress_driver.h>
+#include <asm/platform_sst_audio.h>
#include "sst-mfld-platform.h"
+#include "sst-atom-controls.h"
struct sst_device *sst;
static DEFINE_MUTEX(sst_lock);
@@ -92,6 +94,13 @@ static struct snd_pcm_hardware sst_platform_pcm_hw = {
.fifo_size = SST_FIFO_SIZE,
};
+static struct sst_dev_stream_map dpcm_strm_map[] = {
+ {0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF}, /* Reserved, not in use */
+ {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA1_IN, SST_TASK_ID_MEDIA, 0},
+ {MERR_DPCM_COMPR, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA0_IN, SST_TASK_ID_MEDIA, 0},
+ {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0},
+};
+
/* MFLD - MSIC */
static struct snd_soc_dai_driver sst_platform_dai[] = {
{
@@ -143,58 +152,142 @@ static inline int sst_get_stream_status(struct sst_runtime_stream *stream)
return state;
}
+static void sst_fill_alloc_params(struct snd_pcm_substream *substream,
+ struct snd_sst_alloc_params_ext *alloc_param)
+{
+ unsigned int channels;
+ snd_pcm_uframes_t period_size;
+ ssize_t periodbytes;
+ ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
+ u32 buffer_addr = virt_to_phys(substream->dma_buffer.area);
+
+ channels = substream->runtime->channels;
+ period_size = substream->runtime->period_size;
+ periodbytes = samples_to_bytes(substream->runtime, period_size);
+ alloc_param->ring_buf_info[0].addr = buffer_addr;
+ alloc_param->ring_buf_info[0].size = buffer_bytes;
+ alloc_param->sg_count = 1;
+ alloc_param->reserved = 0;
+ alloc_param->frag_size = periodbytes * channels;
+
+}
static void sst_fill_pcm_params(struct snd_pcm_substream *substream,
- struct sst_pcm_params *param)
+ struct snd_sst_stream_params *param)
{
+ param->uc.pcm_params.num_chan = (u8) substream->runtime->channels;
+ param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits;
+ param->uc.pcm_params.sfreq = substream->runtime->rate;
+
+ /* PCM stream via ALSA interface */
+ param->uc.pcm_params.use_offload_path = 0;
+ param->uc.pcm_params.reserved2 = 0;
+ memset(param->uc.pcm_params.channel_map, 0, sizeof(u8));
- param->num_chan = (u8) substream->runtime->channels;
- param->pcm_wd_sz = substream->runtime->sample_bits;
- param->reserved = 0;
- param->sfreq = substream->runtime->rate;
- param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream);
- param->period_count = substream->runtime->period_size;
- param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area);
- pr_debug("period_cnt = %d\n", param->period_count);
- pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz);
}
-static int sst_platform_alloc_stream(struct snd_pcm_substream *substream)
+static int sst_get_stream_mapping(int dev, int sdev, int dir,
+ struct sst_dev_stream_map *map, int size)
+{
+ int i;
+
+ if (map == NULL)
+ return -EINVAL;
+
+
+ /* index 0 is not used in stream map */
+ for (i = 1; i < size; i++) {
+ if ((map[i].dev_num == dev) && (map[i].direction == dir))
+ return i;
+ }
+ return 0;
+}
+
+int sst_fill_stream_params(void *substream,
+ const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress)
+{
+ int map_size;
+ int index;
+ struct sst_dev_stream_map *map;
+ struct snd_pcm_substream *pstream = NULL;
+ struct snd_compr_stream *cstream = NULL;
+
+ map = ctx->pdata->pdev_strm_map;
+ map_size = ctx->pdata->strm_map_size;
+
+ if (is_compress == true)
+ cstream = (struct snd_compr_stream *)substream;
+ else
+ pstream = (struct snd_pcm_substream *)substream;
+
+ str_params->stream_type = SST_STREAM_TYPE_MUSIC;
+
+ /* For pcm streams */
+ if (pstream) {
+ index = sst_get_stream_mapping(pstream->pcm->device,
+ pstream->number, pstream->stream,
+ map, map_size);
+ if (index <= 0)
+ return -EINVAL;
+
+ str_params->stream_id = index;
+ str_params->device_type = map[index].device_id;
+ str_params->task = map[index].task_id;
+
+ str_params->ops = (u8)pstream->stream;
+ }
+
+ if (cstream) {
+ index = sst_get_stream_mapping(cstream->device->device,
+ 0, cstream->direction,
+ map, map_size);
+ if (index <= 0)
+ return -EINVAL;
+ str_params->stream_id = index;
+ str_params->device_type = map[index].device_id;
+ str_params->task = map[index].task_id;
+
+ str_params->ops = (u8)cstream->direction;
+ }
+ return 0;
+}
+
+static int sst_platform_alloc_stream(struct snd_pcm_substream *substream,
+ struct snd_soc_platform *platform)
{
struct sst_runtime_stream *stream =
substream->runtime->private_data;
- struct sst_pcm_params param = {0};
- struct sst_stream_params str_params = {0};
- int ret_val;
+ struct snd_sst_stream_params param = {{{0,},},};
+ struct snd_sst_params str_params = {0};
+ struct snd_sst_alloc_params_ext alloc_params = {0};
+ int ret_val = 0;
+ struct sst_data *ctx = snd_soc_platform_get_drvdata(platform);
/* set codec params and inform SST driver the same */
sst_fill_pcm_params(substream, &param);
+ sst_fill_alloc_params(substream, &alloc_params);
substream->runtime->dma_area = substream->dma_buffer.area;
str_params.sparams = param;
- str_params.codec = param.codec;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- str_params.ops = STREAM_OPS_PLAYBACK;
- str_params.device_type = substream->pcm->device + 1;
- pr_debug("Playbck stream,Device %d\n",
- substream->pcm->device);
- } else {
- str_params.ops = STREAM_OPS_CAPTURE;
- str_params.device_type = SND_SST_DEVICE_CAPTURE;
- pr_debug("Capture stream,Device %d\n",
- substream->pcm->device);
- }
- ret_val = stream->ops->open(&str_params);
- pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val);
+ str_params.aparams = alloc_params;
+ str_params.codec = SST_CODEC_TYPE_PCM;
+
+ /* fill the device type and stream id to pass to SST driver */
+ ret_val = sst_fill_stream_params(substream, ctx, &str_params, false);
if (ret_val < 0)
return ret_val;
- stream->stream_info.str_id = ret_val;
- pr_debug("str id : %d\n", stream->stream_info.str_id);
+ stream->stream_info.str_id = str_params.stream_id;
+
+ ret_val = stream->ops->open(&str_params);
+ if (ret_val <= 0)
+ return ret_val;
+
+
return ret_val;
}
-static void sst_period_elapsed(void *mad_substream)
+static void sst_period_elapsed(void *arg)
{
- struct snd_pcm_substream *substream = mad_substream;
+ struct snd_pcm_substream *substream = arg;
struct sst_runtime_stream *stream;
int status;
@@ -218,7 +311,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream)
pr_debug("setting buffer ptr param\n");
sst_set_stream_status(stream, SST_PLATFORM_INIT);
stream->stream_info.period_elapsed = sst_period_elapsed;
- stream->stream_info.mad_substream = substream;
+ stream->stream_info.arg = substream;
stream->stream_info.buffer_ptr = 0;
stream->stream_info.sfreq = substream->runtime->rate;
ret_val = stream->ops->device_control(
@@ -230,19 +323,12 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream)
}
/* end -- helper functions */
-static int sst_platform_open(struct snd_pcm_substream *substream)
+static int sst_media_open(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
+ int ret_val = 0;
struct snd_pcm_runtime *runtime = substream->runtime;
struct sst_runtime_stream *stream;
- int ret_val;
-
- pr_debug("sst_platform_open called\n");
-
- snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw);
- ret_val = snd_pcm_hw_constraint_integer(runtime,
- SNDRV_PCM_HW_PARAM_PERIODS);
- if (ret_val < 0)
- return ret_val;
stream = kzalloc(sizeof(*stream), GFP_KERNEL);
if (!stream)
@@ -251,50 +337,69 @@ static int sst_platform_open(struct snd_pcm_substream *substream)
/* get the sst ops */
mutex_lock(&sst_lock);
- if (!sst) {
+ if (!sst ||
+ !try_module_get(sst->dev->driver->owner)) {
pr_err("no device available to run\n");
- mutex_unlock(&sst_lock);
- kfree(stream);
- return -ENODEV;
- }
- if (!try_module_get(sst->dev->driver->owner)) {
- mutex_unlock(&sst_lock);
- kfree(stream);
- return -ENODEV;
+ ret_val = -ENODEV;
+ goto out_ops;
}
stream->ops = sst->ops;
mutex_unlock(&sst_lock);
stream->stream_info.str_id = 0;
- sst_set_stream_status(stream, SST_PLATFORM_INIT);
- stream->stream_info.mad_substream = substream;
+
+ stream->stream_info.arg = substream;
/* allocate memory for SST API set */
runtime->private_data = stream;
- return 0;
+ /* Make sure, that the period size is always even */
+ snd_pcm_hw_constraint_step(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIODS, 2);
+
+ return snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+out_ops:
+ kfree(stream);
+ mutex_unlock(&sst_lock);
+ return ret_val;
}
-static int sst_platform_close(struct snd_pcm_substream *substream)
+static void sst_media_close(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct sst_runtime_stream *stream;
int ret_val = 0, str_id;
- pr_debug("sst_platform_close called\n");
stream = substream->runtime->private_data;
str_id = stream->stream_info.str_id;
if (str_id)
ret_val = stream->ops->close(str_id);
module_put(sst->dev->driver->owner);
kfree(stream);
- return ret_val;
}
-static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream)
+static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform,
+ struct snd_pcm_substream *substream)
+{
+ struct sst_data *sst = snd_soc_platform_get_drvdata(platform);
+ struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map;
+ struct sst_runtime_stream *stream =
+ substream->runtime->private_data;
+ u32 str_id = stream->stream_info.str_id;
+ unsigned int pipe_id;
+ pipe_id = map[str_id].device_id;
+
+ pr_debug("%s: got pipe_id = %#x for str_id = %d\n",
+ __func__, pipe_id, str_id);
+ return pipe_id;
+}
+
+static int sst_media_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct sst_runtime_stream *stream;
int ret_val = 0, str_id;
- pr_debug("sst_platform_pcm_prepare called\n");
stream = substream->runtime->private_data;
str_id = stream->stream_info.str_id;
if (stream->stream_info.str_id) {
@@ -303,8 +408,8 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream)
return ret_val;
}
- ret_val = sst_platform_alloc_stream(substream);
- if (ret_val < 0)
+ ret_val = sst_platform_alloc_stream(substream, dai->platform);
+ if (ret_val <= 0)
return ret_val;
snprintf(substream->pcm->id, sizeof(substream->pcm->id),
"%d", stream->stream_info.str_id);
@@ -316,6 +421,41 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream)
return ret_val;
}
+static int sst_media_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
+ return 0;
+}
+
+static int sst_media_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static struct snd_soc_dai_ops sst_media_dai_ops = {
+ .startup = sst_media_open,
+ .shutdown = sst_media_close,
+ .prepare = sst_media_prepare,
+ .hw_params = sst_media_hw_params,
+ .hw_free = sst_media_hw_free,
+};
+
+static int sst_platform_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime;
+
+ if (substream->pcm->internal)
+ return 0;
+
+ runtime = substream->runtime;
+ runtime->hw = sst_platform_pcm_hw;
+ return 0;
+}
+
static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream,
int cmd)
{
@@ -331,7 +471,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream,
pr_debug("sst: Trigger Start\n");
str_cmd = SST_SND_START;
status = SST_PLATFORM_RUNNING;
- stream->stream_info.mad_substream = substream;
+ stream->stream_info.arg = substream;
break;
case SNDRV_PCM_TRIGGER_STOP:
pr_debug("sst: in stop\n");
@@ -377,32 +517,15 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer
pr_err("sst: error code = %d\n", ret_val);
return ret_val;
}
- return stream->stream_info.buffer_ptr;
-}
-
-static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
- memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
-
- return 0;
-}
-
-static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream)
-{
- return snd_pcm_lib_free_pages(substream);
+ substream->runtime->delay = str_info->pcm_delay;
+ return str_info->buffer_ptr;
}
static struct snd_pcm_ops sst_platform_ops = {
.open = sst_platform_open,
- .close = sst_platform_close,
.ioctl = snd_pcm_lib_ioctl,
- .prepare = sst_platform_pcm_prepare,
.trigger = sst_platform_pcm_trigger,
.pointer = sst_platform_pcm_pointer,
- .hw_params = sst_platform_pcm_hw_params,
- .hw_free = sst_platform_pcm_hw_free,
};
static void sst_pcm_free(struct snd_pcm *pcm)
@@ -413,15 +536,15 @@ static void sst_pcm_free(struct snd_pcm *pcm)
static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_soc_dai *dai = rtd->cpu_dai;
struct snd_pcm *pcm = rtd->pcm;
int retval = 0;
- pr_debug("sst_pcm_new called\n");
- if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream ||
- pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ if (dai->driver->playback.channels_min ||
+ dai->driver->capture.channels_min) {
retval = snd_pcm_lib_preallocate_pages_for_all(pcm,
SNDRV_DMA_TYPE_CONTINUOUS,
- snd_dma_continuous_data(GFP_KERNEL),
+ snd_dma_continuous_data(GFP_DMA),
SST_MIN_BUFFER, SST_MAX_BUFFER);
if (retval) {
pr_err("dma buffer allocationf fail\n");
@@ -445,10 +568,28 @@ static const struct snd_soc_component_driver sst_component = {
static int sst_platform_probe(struct platform_device *pdev)
{
+ struct sst_data *drv;
int ret;
+ struct sst_platform_data *pdata;
+
+ drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL);
+ if (drv == NULL) {
+ pr_err("kzalloc failed\n");
+ return -ENOMEM;
+ }
+
+ pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);
+ if (pdata == NULL) {
+ pr_err("kzalloc failed for pdata\n");
+ return -ENOMEM;
+ }
+
+ pdata->pdev_strm_map = dpcm_strm_map;
+ pdata->strm_map_size = ARRAY_SIZE(dpcm_strm_map);
+ drv->pdata = pdata;
+ mutex_init(&drv->lock);
+ dev_set_drvdata(&pdev->dev, drv);
- pr_debug("sst_platform_probe called\n");
- sst = NULL;
ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv);
if (ret) {
pr_err("registering soc platform failed\n");
diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h
index 6c5e7dc49e3c..6c6a42c08e24 100644
--- a/sound/soc/intel/sst-mfld-platform.h
+++ b/sound/soc/intel/sst-mfld-platform.h
@@ -39,9 +39,10 @@ extern struct sst_device *sst;
struct pcm_stream_info {
int str_id;
- void *mad_substream;
- void (*period_elapsed) (void *mad_substream);
+ void *arg;
+ void (*period_elapsed) (void *arg);
unsigned long long buffer_ptr;
+ unsigned long long pcm_delay;
int sfreq;
};
@@ -62,7 +63,9 @@ enum sst_controls {
SST_SND_BUFFER_POINTER = 0x05,
SST_SND_STREAM_INIT = 0x06,
SST_SND_START = 0x07,
- SST_MAX_CONTROLS = 0x07,
+ SST_SET_BYTE_STREAM = 0x100A,
+ SST_GET_BYTE_STREAM = 0x100B,
+ SST_MAX_CONTROLS = SST_GET_BYTE_STREAM,
};
enum sst_stream_ops {
@@ -124,8 +127,9 @@ struct compress_sst_ops {
};
struct sst_ops {
- int (*open) (struct sst_stream_params *str_param);
+ int (*open) (struct snd_sst_params *str_param);
int (*device_control) (int cmd, void *arg);
+ int (*set_generic_params)(enum sst_controls cmd, void *arg);
int (*close) (unsigned int str_id);
};
@@ -143,10 +147,27 @@ struct sst_device {
char *name;
struct device *dev;
struct sst_ops *ops;
+ struct platform_device *pdev;
struct compress_sst_ops *compr_ops;
};
+struct sst_data;
void sst_set_stream_status(struct sst_runtime_stream *stream, int state);
+int sst_fill_stream_params(void *substream, const struct sst_data *ctx,
+ struct snd_sst_params *str_params, bool is_compress);
+
+struct sst_algo_int_control_v2 {
+ struct soc_mixer_control mc;
+ u16 module_id; /* module identifieer */
+ u16 pipe_id; /* location info: pipe_id + instance_id */
+ u16 instance_id;
+ unsigned int value; /* Value received is stored here */
+};
+struct sst_data {
+ struct platform_device *pdev;
+ struct sst_platform_data *pdata;
+ struct mutex lock;
+};
int sst_register_dsp(struct sst_device *sst);
int sst_unregister_dsp(struct sst_device *sst);
#endif
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 06f4e8aa93ae..132bb83f8e99 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -1,6 +1,6 @@
config SND_KIRKWOOD_SOC
tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
- depends on ARCH_KIRKWOOD || ARCH_DOVE || ARCH_MVEBU || MACH_KIRKWOOD || COMPILE_TEST
+ depends on ARCH_DOVE || ARCH_MVEBU || 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
@@ -15,20 +15,3 @@ config SND_KIRKWOOD_SOC_ARMADA370_DB
Say Y if you want to add support for SoC audio on
the Armada 370 Development Board.
-config SND_KIRKWOOD_SOC_OPENRD
- tristate "SoC Audio support for Kirkwood Openrd Client"
- depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST)
- depends on I2C
- select SND_SOC_CS42L51
- help
- Say Y if you want to add support for SoC audio on
- Openrd Client.
-
-config SND_KIRKWOOD_SOC_T5325
- tristate "SoC Audio support for HP t5325"
- 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
- the HP t5325 thin client.
-
diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile
index 7c1d8fe09e6b..c36b03d8006c 100644
--- a/sound/soc/kirkwood/Makefile
+++ b/sound/soc/kirkwood/Makefile
@@ -2,10 +2,6 @@ snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o
obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o
-snd-soc-openrd-objs := kirkwood-openrd.o
-snd-soc-t5325-objs := kirkwood-t5325.o
snd-soc-armada-370-db-objs := armada-370-db.o
obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o
-obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o
-obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index aac22fccdcdc..4cf2245950d7 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -28,11 +28,12 @@ static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs)
}
static struct snd_pcm_hardware kirkwood_dma_snd_hw = {
- .info = (SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_PAUSE),
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
.buffer_bytes_max = KIRKWOOD_SND_MAX_BUFFER_BYTES,
.period_bytes_min = KIRKWOOD_SND_MIN_PERIOD_BYTES,
.period_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES,
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 9f842222e798..0704cd6d2314 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -212,7 +212,8 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream,
KIRKWOOD_PLAYCTL_SIZE_MASK);
priv->ctl_play |= ctl_play;
} else {
- priv->ctl_rec &= ~KIRKWOOD_RECCTL_SIZE_MASK;
+ priv->ctl_rec &= ~(KIRKWOOD_RECCTL_ENABLE_MASK |
+ KIRKWOOD_RECCTL_SIZE_MASK);
priv->ctl_rec |= ctl_rec;
}
@@ -221,14 +222,24 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream,
return 0;
}
+static unsigned kirkwood_i2s_play_mute(unsigned ctl)
+{
+ if (!(ctl & KIRKWOOD_PLAYCTL_I2S_EN))
+ ctl |= KIRKWOOD_PLAYCTL_I2S_MUTE;
+ if (!(ctl & KIRKWOOD_PLAYCTL_SPDIF_EN))
+ ctl |= KIRKWOOD_PLAYCTL_SPDIF_MUTE;
+ return ctl;
+}
+
static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
+ struct snd_pcm_runtime *runtime = substream->runtime;
struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai);
uint32_t ctl, value;
ctl = readl(priv->io + KIRKWOOD_PLAYCTL);
- if (ctl & KIRKWOOD_PLAYCTL_PAUSE) {
+ if ((ctl & KIRKWOOD_PLAYCTL_ENABLE_MASK) == 0) {
unsigned timeout = 5000;
/*
* The Armada510 spec says that if we enter pause mode, the
@@ -256,14 +267,16 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */
else
ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */
-
+ ctl = kirkwood_i2s_play_mute(ctl);
value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK;
writel(value, priv->io + KIRKWOOD_PLAYCTL);
/* enable interrupts */
- value = readl(priv->io + KIRKWOOD_INT_MASK);
- value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES;
- writel(value, priv->io + KIRKWOOD_INT_MASK);
+ if (!runtime->no_period_wakeup) {
+ value = readl(priv->io + KIRKWOOD_INT_MASK);
+ value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES;
+ writel(value, priv->io + KIRKWOOD_INT_MASK);
+ }
/* enable playback */
writel(ctl, priv->io + KIRKWOOD_PLAYCTL);
@@ -295,6 +308,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE |
KIRKWOOD_PLAYCTL_SPDIF_MUTE);
+ ctl = kirkwood_i2s_play_mute(ctl);
writel(ctl, priv->io + KIRKWOOD_PLAYCTL);
break;
@@ -322,8 +336,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream,
else
ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif */
- value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN |
- KIRKWOOD_RECCTL_SPDIF_EN);
+ value = ctl & ~KIRKWOOD_RECCTL_ENABLE_MASK;
writel(value, priv->io + KIRKWOOD_RECCTL);
/* enable interrupts */
@@ -347,7 +360,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream,
/* disable all records */
value = readl(priv->io + KIRKWOOD_RECCTL);
- value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN);
+ value &= ~KIRKWOOD_RECCTL_ENABLE_MASK;
writel(value, priv->io + KIRKWOOD_RECCTL);
break;
@@ -411,7 +424,7 @@ static int kirkwood_i2s_init(struct kirkwood_dma_data *priv)
writel(value, priv->io + KIRKWOOD_PLAYCTL);
value = readl(priv->io + KIRKWOOD_RECCTL);
- value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN);
+ value &= ~KIRKWOOD_RECCTL_ENABLE_MASK;
writel(value, priv->io + KIRKWOOD_RECCTL);
return 0;
diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c
deleted file mode 100644
index 65f2a5b9ec3b..000000000000
--- a/sound/soc/kirkwood/kirkwood-openrd.c
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * kirkwood-openrd.c
- *
- * (c) 2010 Arnaud Patard <apatard@mandriva.com>
- * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
- *
- * 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/module.h>
-#include <linux/moduleparam.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <sound/soc.h>
-#include <linux/platform_data/asoc-kirkwood.h>
-#include "../codecs/cs42l51.h"
-
-static int openrd_client_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;
- unsigned int freq;
-
- switch (params_rate(params)) {
- default:
- case 44100:
- freq = 11289600;
- break;
- case 48000:
- freq = 12288000;
- break;
- case 96000:
- freq = 24576000;
- break;
- }
-
- return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
-
-}
-
-static struct snd_soc_ops openrd_client_ops = {
- .hw_params = openrd_client_hw_params,
-};
-
-
-static struct snd_soc_dai_link openrd_client_dai[] = {
-{
- .name = "CS42L51",
- .stream_name = "CS42L51 HiFi",
- .cpu_dai_name = "i2s",
- .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,
- .ops = &openrd_client_ops,
-},
-};
-
-
-static struct snd_soc_card openrd_client = {
- .name = "OpenRD Client",
- .owner = THIS_MODULE,
- .dai_link = openrd_client_dai,
- .num_links = ARRAY_SIZE(openrd_client_dai),
-};
-
-static int openrd_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card = &openrd_client;
- int ret;
-
- card->dev = &pdev->dev;
-
- ret = snd_soc_register_card(card);
- if (ret)
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
- ret);
- return ret;
-}
-
-static int openrd_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
-static struct platform_driver openrd_driver = {
- .driver = {
- .name = "openrd-client-audio",
- .owner = THIS_MODULE,
- },
- .probe = openrd_probe,
- .remove = openrd_remove,
-};
-
-module_platform_driver(openrd_driver);
-
-/* Module information */
-MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
-MODULE_DESCRIPTION("ALSA SoC OpenRD Client");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:openrd-client-audio");
diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c
deleted file mode 100644
index 844b8415a011..000000000000
--- a/sound/soc/kirkwood/kirkwood-t5325.c
+++ /dev/null
@@ -1,116 +0,0 @@
-/*
- * kirkwood-t5325.c
- *
- * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
- *
- * 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/module.h>
-#include <linux/moduleparam.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <sound/soc.h>
-#include <linux/platform_data/asoc-kirkwood.h>
-#include "../codecs/alc5623.h"
-
-static int t5325_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;
- unsigned int freq;
-
- freq = params_rate(params) * 256;
-
- return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
-
-}
-
-static struct snd_soc_ops t5325_ops = {
- .hw_params = t5325_hw_params,
-};
-
-static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_SPK("Speaker", NULL),
- SND_SOC_DAPM_MIC("Mic Jack", NULL),
-};
-
-static const struct snd_soc_dapm_route t5325_route[] = {
- { "Headphone Jack", NULL, "HPL" },
- { "Headphone Jack", NULL, "HPR" },
-
- {"Speaker", NULL, "SPKOUT"},
- {"Speaker", NULL, "SPKOUTN"},
-
- { "MIC1", NULL, "Mic Jack" },
- { "MIC2", NULL, "Mic Jack" },
-};
-
-static struct snd_soc_dai_link t5325_dai[] = {
-{
- .name = "ALC5621",
- .stream_name = "ALC5621 HiFi",
- .cpu_dai_name = "i2s",
- .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,
- .ops = &t5325_ops,
-},
-};
-
-static struct snd_soc_card t5325 = {
- .name = "t5325",
- .owner = THIS_MODULE,
- .dai_link = t5325_dai,
- .num_links = ARRAY_SIZE(t5325_dai),
-
- .dapm_widgets = t5325_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(t5325_dapm_widgets),
- .dapm_routes = t5325_route,
- .num_dapm_routes = ARRAY_SIZE(t5325_route),
-};
-
-static int t5325_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card = &t5325;
- int ret;
-
- card->dev = &pdev->dev;
-
- ret = snd_soc_register_card(card);
- if (ret)
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
- ret);
- return ret;
-}
-
-static int t5325_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
-static struct platform_driver t5325_driver = {
- .driver = {
- .name = "t5325-audio",
- .owner = THIS_MODULE,
- },
- .probe = t5325_probe,
- .remove = t5325_remove,
-};
-
-module_platform_driver(t5325_driver);
-
-MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
-MODULE_DESCRIPTION("ALSA SoC t5325 audio client");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:t5325-audio");
diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h
index bf23afbba1d7..90e32a781424 100644
--- a/sound/soc/kirkwood/kirkwood.h
+++ b/sound/soc/kirkwood/kirkwood.h
@@ -38,6 +38,9 @@
#define KIRKWOOD_RECCTL_SIZE_24 (1<<0)
#define KIRKWOOD_RECCTL_SIZE_32 (0<<0)
+#define KIRKWOOD_RECCTL_ENABLE_MASK (KIRKWOOD_RECCTL_SPDIF_EN | \
+ KIRKWOOD_RECCTL_I2S_EN)
+
#define KIRKWOOD_REC_BUF_ADDR 0x1004
#define KIRKWOOD_REC_BUF_SIZE 0x1008
#define KIRKWOOD_REC_BYTE_COUNT 0x100C
@@ -121,9 +124,9 @@
/* Theses values come from the marvell alsa driver */
/* need to find where they come from */
-#define KIRKWOOD_SND_MIN_PERIODS 8
+#define KIRKWOOD_SND_MIN_PERIODS 2
#define KIRKWOOD_SND_MAX_PERIODS 16
-#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x800
+#define KIRKWOOD_SND_MIN_PERIOD_BYTES 256
#define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x8000
#define KIRKWOOD_SND_MAX_BUFFER_BYTES (KIRKWOOD_SND_MAX_PERIOD_BYTES \
* KIRKWOOD_SND_MAX_PERIODS)
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 0cc41f94de4e..8c9cc64a9dfb 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -301,7 +301,7 @@ static int cx81801_open(struct tty_struct *tty)
static void cx81801_close(struct tty_struct *tty)
{
struct snd_soc_codec *codec = tty->disc_data;
- struct snd_soc_dapm_context *dapm = &codec->card->dapm;
+ struct snd_soc_dapm_context *dapm = &codec->component.card->dapm;
del_timer_sync(&cx81801_timer);
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 6925d7141215..0f34e28a3d55 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -466,7 +466,7 @@ static int asoc_dmic_probe(struct platform_device *pdev)
mutex_init(&dmic->mutex);
- dmic->fclk = clk_get(dmic->dev, "fck");
+ dmic->fclk = devm_clk_get(dmic->dev, "fck");
if (IS_ERR(dmic->fclk)) {
dev_err(dmic->dev, "cant get fck\n");
return -ENODEV;
@@ -475,8 +475,7 @@ static int asoc_dmic_probe(struct platform_device *pdev)
res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma");
if (!res) {
dev_err(dmic->dev, "invalid dma memory resource\n");
- ret = -ENODEV;
- goto err_put_clk;
+ return -ENODEV;
}
dmic->dma_data.addr = res->start + OMAP_DMIC_DATA_REG;
@@ -484,34 +483,19 @@ static int asoc_dmic_probe(struct platform_device *pdev)
res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
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;
- }
+ 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);
+ ret = devm_snd_soc_register_component(&pdev->dev,
+ &omap_dmic_component,
+ &omap_dmic_dai, 1);
if (ret)
- goto err_put_clk;
+ return ret;
ret = omap_pcm_platform_register(&pdev->dev);
if (ret)
- goto err_put_clk;
-
- return 0;
-
-err_put_clk:
- clk_put(dmic->fclk);
- return ret;
-}
-
-static int asoc_dmic_remove(struct platform_device *pdev)
-{
- struct omap_dmic *dmic = platform_get_drvdata(pdev);
-
- snd_soc_unregister_component(&pdev->dev);
- clk_put(dmic->fclk);
+ return ret;
return 0;
}
@@ -529,7 +513,6 @@ static struct platform_driver asoc_dmic_driver = {
.of_match_table = omap_dmic_of_match,
},
.probe = asoc_dmic_probe,
- .remove = asoc_dmic_remove,
};
module_platform_driver(asoc_dmic_driver);
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index efe2cd699b77..bd3ef2a88be0 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -805,8 +805,9 @@ static int asoc_mcbsp_probe(struct platform_device *pdev)
if (ret)
return ret;
- ret = snd_soc_register_component(&pdev->dev, &omap_mcbsp_component,
- &omap_mcbsp_dai, 1);
+ ret = devm_snd_soc_register_component(&pdev->dev,
+ &omap_mcbsp_component,
+ &omap_mcbsp_dai, 1);
if (ret)
return ret;
@@ -817,8 +818,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev)
{
struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev);
- snd_soc_unregister_component(&pdev->dev);
-
if (mcbsp->pdata->ops && mcbsp->pdata->ops->free)
mcbsp->pdata->ops->free(mcbsp->id);
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 8d809f8509c8..f4b05bc23e4b 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -31,6 +31,7 @@
#include <sound/pcm_params.h>
#include <sound/dmaengine_pcm.h>
#include <sound/soc.h>
+#include <sound/omap-pcm.h>
#ifdef CONFIG_ARCH_OMAP1
#define pcm_omap1510() cpu_is_omap1510()
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 199a8b377553..0109f6c2334e 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -723,7 +723,8 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai)
ssp_handle = of_parse_phandle(dev->of_node, "port", 0);
if (!ssp_handle) {
dev_err(dev, "unable to get 'port' phandle\n");
- return -ENODEV;
+ ret = -ENODEV;
+ goto err_priv;
}
priv->ssp = pxa_ssp_request_of(ssp_handle, "SoC audio");
diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig
new file mode 100644
index 000000000000..c196a466eef6
--- /dev/null
+++ b/sound/soc/rockchip/Kconfig
@@ -0,0 +1,12 @@
+config SND_SOC_ROCKCHIP
+ tristate "ASoC support for Rockchip"
+ depends on COMPILE_TEST || ARCH_ROCKCHIP
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+ select SND_ROCKCHIP_I2S
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the Rockchip SoCs' Audio interfaces. You will also need to
+ select the audio interfaces to support below.
+
+config SND_ROCKCHIP_I2S
+ tristate
diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile
new file mode 100644
index 000000000000..1006418e1394
--- /dev/null
+++ b/sound/soc/rockchip/Makefile
@@ -0,0 +1,4 @@
+# ROCKCHIP Platform Support
+snd-soc-i2s-objs := rockchip_i2s.o
+
+obj-$(CONFIG_SND_ROCKCHIP_I2S) += snd-soc-i2s.o
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
new file mode 100644
index 000000000000..8d8e4b59049f
--- /dev/null
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -0,0 +1,529 @@
+/* sound/soc/rockchip/rockchip_i2s.c
+ *
+ * ALSA SoC Audio Layer - Rockchip I2S Controller driver
+ *
+ * Copyright (c) 2014 Rockchip Electronics Co. Ltd.
+ * Author: Jianqun <jay.xu@rock-chips.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/delay.h>
+#include <linux/of_gpio.h>
+#include <linux/clk.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <sound/pcm_params.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "rockchip_i2s.h"
+
+#define DRV_NAME "rockchip-i2s"
+
+struct rk_i2s_dev {
+ struct device *dev;
+
+ struct clk *hclk;
+ struct clk *mclk;
+
+ struct snd_dmaengine_dai_dma_data capture_dma_data;
+ struct snd_dmaengine_dai_dma_data playback_dma_data;
+
+ struct regmap *regmap;
+
+/*
+ * Used to indicate the tx/rx status.
+ * I2S controller hopes to start the tx and rx together,
+ * also to stop them when they are both try to stop.
+*/
+ bool tx_start;
+ bool rx_start;
+};
+
+static int i2s_runtime_suspend(struct device *dev)
+{
+ struct rk_i2s_dev *i2s = dev_get_drvdata(dev);
+
+ clk_disable_unprepare(i2s->mclk);
+
+ return 0;
+}
+
+static int i2s_runtime_resume(struct device *dev)
+{
+ struct rk_i2s_dev *i2s = dev_get_drvdata(dev);
+ int ret;
+
+ ret = clk_prepare_enable(i2s->mclk);
+ if (ret) {
+ dev_err(i2s->dev, "clock enable failed %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static inline struct rk_i2s_dev *to_info(struct snd_soc_dai *dai)
+{
+ return snd_soc_dai_get_drvdata(dai);
+}
+
+static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on)
+{
+ unsigned int val = 0;
+ int retry = 10;
+
+ if (on) {
+ regmap_update_bits(i2s->regmap, I2S_DMACR,
+ I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE);
+
+ regmap_update_bits(i2s->regmap, I2S_XFER,
+ I2S_XFER_TXS_START | I2S_XFER_RXS_START,
+ I2S_XFER_TXS_START | I2S_XFER_RXS_START);
+
+ i2s->tx_start = true;
+ } else {
+ i2s->tx_start = false;
+
+ regmap_update_bits(i2s->regmap, I2S_DMACR,
+ I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE);
+
+ if (!i2s->rx_start) {
+ regmap_update_bits(i2s->regmap, I2S_XFER,
+ I2S_XFER_TXS_START |
+ I2S_XFER_RXS_START,
+ I2S_XFER_TXS_STOP |
+ I2S_XFER_RXS_STOP);
+
+ regmap_update_bits(i2s->regmap, I2S_CLR,
+ I2S_CLR_TXC | I2S_CLR_RXC,
+ I2S_CLR_TXC | I2S_CLR_RXC);
+
+ regmap_read(i2s->regmap, I2S_CLR, &val);
+
+ /* Should wait for clear operation to finish */
+ while (val) {
+ regmap_read(i2s->regmap, I2S_CLR, &val);
+ retry--;
+ if (!retry)
+ dev_warn(i2s->dev, "fail to clear\n");
+ }
+ }
+ }
+}
+
+static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on)
+{
+ unsigned int val = 0;
+ int retry = 10;
+
+ if (on) {
+ regmap_update_bits(i2s->regmap, I2S_DMACR,
+ I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE);
+
+ regmap_update_bits(i2s->regmap, I2S_XFER,
+ I2S_XFER_TXS_START | I2S_XFER_RXS_START,
+ I2S_XFER_TXS_START | I2S_XFER_RXS_START);
+
+ i2s->rx_start = true;
+ } else {
+ i2s->rx_start = false;
+
+ regmap_update_bits(i2s->regmap, I2S_DMACR,
+ I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE);
+
+ if (!i2s->tx_start) {
+ regmap_update_bits(i2s->regmap, I2S_XFER,
+ I2S_XFER_TXS_START |
+ I2S_XFER_RXS_START,
+ I2S_XFER_TXS_STOP |
+ I2S_XFER_RXS_STOP);
+
+ regmap_update_bits(i2s->regmap, I2S_CLR,
+ I2S_CLR_TXC | I2S_CLR_RXC,
+ I2S_CLR_TXC | I2S_CLR_RXC);
+
+ regmap_read(i2s->regmap, I2S_CLR, &val);
+
+ /* Should wait for clear operation to finish */
+ while (val) {
+ regmap_read(i2s->regmap, I2S_CLR, &val);
+ retry--;
+ if (!retry)
+ dev_warn(i2s->dev, "fail to clear\n");
+ }
+ }
+ }
+}
+
+static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct rk_i2s_dev *i2s = to_info(cpu_dai);
+ unsigned int mask = 0, val = 0;
+
+ mask = I2S_CKR_MSS_SLAVE;
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ val = I2S_CKR_MSS_SLAVE;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ val = I2S_CKR_MSS_MASTER;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ regmap_update_bits(i2s->regmap, I2S_CKR, mask, val);
+
+ mask = I2S_TXCR_IBM_MASK;
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_RIGHT_J:
+ val = I2S_TXCR_IBM_RSJM;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ val = I2S_TXCR_IBM_LSJM;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ val = I2S_TXCR_IBM_NORMAL;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ regmap_update_bits(i2s->regmap, I2S_TXCR, mask, val);
+
+ mask = I2S_RXCR_IBM_MASK;
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_RIGHT_J:
+ val = I2S_RXCR_IBM_RSJM;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ val = I2S_RXCR_IBM_LSJM;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ val = I2S_RXCR_IBM_NORMAL;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ regmap_update_bits(i2s->regmap, I2S_RXCR, mask, val);
+
+ return 0;
+}
+
+static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct rk_i2s_dev *i2s = to_info(dai);
+ unsigned int val = 0;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ val |= I2S_TXCR_VDW(8);
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val |= I2S_TXCR_VDW(16);
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val |= I2S_TXCR_VDW(20);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val |= I2S_TXCR_VDW(24);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ regmap_update_bits(i2s->regmap, I2S_TXCR, I2S_TXCR_VDW_MASK, val);
+ regmap_update_bits(i2s->regmap, I2S_RXCR, I2S_RXCR_VDW_MASK, val);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ dai->playback_dma_data = &i2s->playback_dma_data;
+ regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK,
+ I2S_DMACR_TDL(1) | I2S_DMACR_TDE_ENABLE);
+ } else {
+ dai->capture_dma_data = &i2s->capture_dma_data;
+ regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK,
+ I2S_DMACR_RDL(1) | I2S_DMACR_RDE_ENABLE);
+ }
+
+ return 0;
+}
+
+static int rockchip_i2s_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct rk_i2s_dev *i2s = to_info(dai);
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ rockchip_snd_rxctrl(i2s, 1);
+ else
+ rockchip_snd_txctrl(i2s, 1);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ rockchip_snd_rxctrl(i2s, 0);
+ else
+ rockchip_snd_txctrl(i2s, 0);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int rockchip_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct rk_i2s_dev *i2s = to_info(cpu_dai);
+ int ret;
+
+ ret = clk_set_rate(i2s->mclk, freq);
+ if (ret)
+ dev_err(i2s->dev, "Fail to set mclk %d\n", ret);
+
+ return ret;
+}
+
+static const struct snd_soc_dai_ops rockchip_i2s_dai_ops = {
+ .hw_params = rockchip_i2s_hw_params,
+ .set_sysclk = rockchip_i2s_set_sysclk,
+ .set_fmt = rockchip_i2s_set_fmt,
+ .trigger = rockchip_i2s_trigger,
+};
+
+static struct snd_soc_dai_driver rockchip_i2s_dai = {
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = (SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_3LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = (SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_3LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ },
+ .ops = &rockchip_i2s_dai_ops,
+};
+
+static const struct snd_soc_component_driver rockchip_i2s_component = {
+ .name = DRV_NAME,
+};
+
+static bool rockchip_i2s_wr_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case I2S_TXCR:
+ case I2S_RXCR:
+ case I2S_CKR:
+ case I2S_DMACR:
+ case I2S_INTCR:
+ case I2S_XFER:
+ case I2S_CLR:
+ case I2S_TXDR:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case I2S_TXCR:
+ case I2S_RXCR:
+ case I2S_CKR:
+ case I2S_DMACR:
+ case I2S_INTCR:
+ case I2S_XFER:
+ case I2S_CLR:
+ case I2S_RXDR:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case I2S_FIFOLR:
+ case I2S_INTSR:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool rockchip_i2s_precious_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case I2S_FIFOLR:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const struct regmap_config rockchip_i2s_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = I2S_RXDR,
+ .writeable_reg = rockchip_i2s_wr_reg,
+ .readable_reg = rockchip_i2s_rd_reg,
+ .volatile_reg = rockchip_i2s_volatile_reg,
+ .precious_reg = rockchip_i2s_precious_reg,
+ .cache_type = REGCACHE_FLAT,
+};
+
+static int rockchip_i2s_probe(struct platform_device *pdev)
+{
+ struct rk_i2s_dev *i2s;
+ struct resource *res;
+ void __iomem *regs;
+ int ret;
+
+ i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL);
+ if (!i2s) {
+ dev_err(&pdev->dev, "Can't allocate rk_i2s_dev\n");
+ return -ENOMEM;
+ }
+
+ /* try to prepare related clocks */
+ i2s->hclk = devm_clk_get(&pdev->dev, "i2s_hclk");
+ if (IS_ERR(i2s->hclk)) {
+ dev_err(&pdev->dev, "Can't retrieve i2s bus clock\n");
+ return PTR_ERR(i2s->hclk);
+ }
+
+ i2s->mclk = devm_clk_get(&pdev->dev, "i2s_clk");
+ if (IS_ERR(i2s->mclk)) {
+ dev_err(&pdev->dev, "Can't retrieve i2s master clock\n");
+ return PTR_ERR(i2s->mclk);
+ }
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ regs = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs,
+ &rockchip_i2s_regmap_config);
+ if (IS_ERR(i2s->regmap)) {
+ dev_err(&pdev->dev,
+ "Failed to initialise managed register map\n");
+ return PTR_ERR(i2s->regmap);
+ }
+
+ i2s->playback_dma_data.addr = res->start + I2S_TXDR;
+ i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ i2s->playback_dma_data.maxburst = 16;
+
+ i2s->capture_dma_data.addr = res->start + I2S_RXDR;
+ i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ i2s->capture_dma_data.maxburst = 16;
+
+ i2s->dev = &pdev->dev;
+ dev_set_drvdata(&pdev->dev, i2s);
+
+ pm_runtime_enable(&pdev->dev);
+ if (!pm_runtime_enabled(&pdev->dev)) {
+ ret = i2s_runtime_resume(&pdev->dev);
+ if (ret)
+ goto err_pm_disable;
+ }
+
+ ret = devm_snd_soc_register_component(&pdev->dev,
+ &rockchip_i2s_component,
+ &rockchip_i2s_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "Could not register DAI\n");
+ goto err_suspend;
+ }
+
+ ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+ if (ret) {
+ dev_err(&pdev->dev, "Could not register PCM\n");
+ goto err_pcm_register;
+ }
+
+ return 0;
+
+err_pcm_register:
+ snd_dmaengine_pcm_unregister(&pdev->dev);
+err_suspend:
+ if (!pm_runtime_status_suspended(&pdev->dev))
+ i2s_runtime_suspend(&pdev->dev);
+err_pm_disable:
+ pm_runtime_disable(&pdev->dev);
+
+ return ret;
+}
+
+static int rockchip_i2s_remove(struct platform_device *pdev)
+{
+ struct rk_i2s_dev *i2s = dev_get_drvdata(&pdev->dev);
+
+ pm_runtime_disable(&pdev->dev);
+ if (!pm_runtime_status_suspended(&pdev->dev))
+ i2s_runtime_suspend(&pdev->dev);
+
+ clk_disable_unprepare(i2s->mclk);
+ clk_disable_unprepare(i2s->hclk);
+ snd_dmaengine_pcm_unregister(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
+
+ return 0;
+}
+
+static const struct of_device_id rockchip_i2s_match[] = {
+ { .compatible = "rockchip,rk3066-i2s", },
+ {},
+};
+
+static const struct dev_pm_ops rockchip_i2s_pm_ops = {
+ SET_RUNTIME_PM_OPS(i2s_runtime_suspend, i2s_runtime_resume,
+ NULL)
+};
+
+static struct platform_driver rockchip_i2s_driver = {
+ .probe = rockchip_i2s_probe,
+ .remove = rockchip_i2s_remove,
+ .driver = {
+ .name = DRV_NAME,
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(rockchip_i2s_match),
+ .pm = &rockchip_i2s_pm_ops,
+ },
+};
+module_platform_driver(rockchip_i2s_driver);
+
+MODULE_DESCRIPTION("ROCKCHIP IIS ASoC Interface");
+MODULE_AUTHOR("jianqun <jay.xu@rock-chips.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:" DRV_NAME);
+MODULE_DEVICE_TABLE(of, rockchip_i2s_match);
diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h
new file mode 100644
index 000000000000..89a5d8bc6ee7
--- /dev/null
+++ b/sound/soc/rockchip/rockchip_i2s.h
@@ -0,0 +1,223 @@
+/*
+ * sound/soc/rockchip/rockchip_i2s.h
+ *
+ * ALSA SoC Audio Layer - Rockchip I2S Controller driver
+ *
+ * Copyright (c) 2014 Rockchip Electronics Co. Ltd.
+ * Author: Jianqun xu <jay.xu@rock-chips.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 _ROCKCHIP_IIS_H
+#define _ROCKCHIP_IIS_H
+
+/*
+ * TXCR
+ * transmit operation control register
+*/
+#define I2S_TXCR_RCNT_SHIFT 17
+#define I2S_TXCR_RCNT_MASK (0x3f << I2S_TXCR_RCNT_SHIFT)
+#define I2S_TXCR_CSR_SHIFT 15
+#define I2S_TXCR_CSR(x) (x << I2S_TXCR_CSR_SHIFT)
+#define I2S_TXCR_CSR_MASK (3 << I2S_TXCR_CSR_SHIFT)
+#define I2S_TXCR_HWT BIT(14)
+#define I2S_TXCR_SJM_SHIFT 12
+#define I2S_TXCR_SJM_R (0 << I2S_TXCR_SJM_SHIFT)
+#define I2S_TXCR_SJM_L (1 << I2S_TXCR_SJM_SHIFT)
+#define I2S_TXCR_FBM_SHIFT 11
+#define I2S_TXCR_FBM_MSB (0 << I2S_TXCR_FBM_SHIFT)
+#define I2S_TXCR_FBM_LSB (1 << I2S_TXCR_FBM_SHIFT)
+#define I2S_TXCR_IBM_SHIFT 9
+#define I2S_TXCR_IBM_NORMAL (0 << I2S_TXCR_IBM_SHIFT)
+#define I2S_TXCR_IBM_LSJM (1 << I2S_TXCR_IBM_SHIFT)
+#define I2S_TXCR_IBM_RSJM (2 << I2S_TXCR_IBM_SHIFT)
+#define I2S_TXCR_IBM_MASK (3 << I2S_TXCR_IBM_SHIFT)
+#define I2S_TXCR_PBM_SHIFT 7
+#define I2S_TXCR_PBM_MODE(x) (x << I2S_TXCR_PBM_SHIFT)
+#define I2S_TXCR_PBM_MASK (3 << I2S_TXCR_PBM_SHIFT)
+#define I2S_TXCR_TFS_SHIFT 5
+#define I2S_TXCR_TFS_I2S (0 << I2S_TXCR_TFS_SHIFT)
+#define I2S_TXCR_TFS_PCM (1 << I2S_TXCR_TFS_SHIFT)
+#define I2S_TXCR_VDW_SHIFT 0
+#define I2S_TXCR_VDW(x) ((x - 1) << I2S_TXCR_VDW_SHIFT)
+#define I2S_TXCR_VDW_MASK (0x1f << I2S_TXCR_VDW_SHIFT)
+
+/*
+ * RXCR
+ * receive operation control register
+*/
+#define I2S_RXCR_HWT BIT(14)
+#define I2S_RXCR_SJM_SHIFT 12
+#define I2S_RXCR_SJM_R (0 << I2S_RXCR_SJM_SHIFT)
+#define I2S_RXCR_SJM_L (1 << I2S_RXCR_SJM_SHIFT)
+#define I2S_RXCR_FBM_SHIFT 11
+#define I2S_RXCR_FBM_MSB (0 << I2S_RXCR_FBM_SHIFT)
+#define I2S_RXCR_FBM_LSB (1 << I2S_RXCR_FBM_SHIFT)
+#define I2S_RXCR_IBM_SHIFT 9
+#define I2S_RXCR_IBM_NORMAL (0 << I2S_RXCR_IBM_SHIFT)
+#define I2S_RXCR_IBM_LSJM (1 << I2S_RXCR_IBM_SHIFT)
+#define I2S_RXCR_IBM_RSJM (2 << I2S_RXCR_IBM_SHIFT)
+#define I2S_RXCR_IBM_MASK (3 << I2S_RXCR_IBM_SHIFT)
+#define I2S_RXCR_PBM_SHIFT 7
+#define I2S_RXCR_PBM_MODE(x) (x << I2S_RXCR_PBM_SHIFT)
+#define I2S_RXCR_PBM_MASK (3 << I2S_RXCR_PBM_SHIFT)
+#define I2S_RXCR_TFS_SHIFT 5
+#define I2S_RXCR_TFS_I2S (0 << I2S_RXCR_TFS_SHIFT)
+#define I2S_RXCR_TFS_PCM (1 << I2S_RXCR_TFS_SHIFT)
+#define I2S_RXCR_VDW_SHIFT 0
+#define I2S_RXCR_VDW(x) ((x - 1) << I2S_RXCR_VDW_SHIFT)
+#define I2S_RXCR_VDW_MASK (0x1f << I2S_RXCR_VDW_SHIFT)
+
+/*
+ * CKR
+ * clock generation register
+*/
+#define I2S_CKR_MSS_SHIFT 27
+#define I2S_CKR_MSS_MASTER (0 << I2S_CKR_MSS_SHIFT)
+#define I2S_CKR_MSS_SLAVE (1 << I2S_CKR_MSS_SHIFT)
+#define I2S_CKR_MSS_MASK (1 << I2S_CKR_MSS_SHIFT)
+#define I2S_CKR_CKP_SHIFT 26
+#define I2S_CKR_CKP_NEG (0 << I2S_CKR_CKP_SHIFT)
+#define I2S_CKR_CKP_POS (1 << I2S_CKR_CKP_SHIFT)
+#define I2S_CKR_RLP_SHIFT 25
+#define I2S_CKR_RLP_NORMAL (0 << I2S_CKR_RLP_SHIFT)
+#define I2S_CKR_RLP_OPPSITE (1 << I2S_CKR_RLP_SHIFT)
+#define I2S_CKR_TLP_SHIFT 24
+#define I2S_CKR_TLP_NORMAL (0 << I2S_CKR_TLP_SHIFT)
+#define I2S_CKR_TLP_OPPSITE (1 << I2S_CKR_TLP_SHIFT)
+#define I2S_CKR_MDIV_SHIFT 16
+#define I2S_CKR_MDIV(x) ((x - 1) << I2S_CKR_MDIV_SHIFT)
+#define I2S_CKR_MDIV_MASK (0xff << I2S_CKR_MDIV_SHIFT)
+#define I2S_CKR_RSD_SHIFT 8
+#define I2S_CKR_RSD(x) ((x - 1) << I2S_CKR_RSD_SHIFT)
+#define I2S_CKR_RSD_MASK (0xff << I2S_CKR_RSD_SHIFT)
+#define I2S_CKR_TSD_SHIFT 0
+#define I2S_CKR_TSD(x) ((x - 1) << I2S_CKR_TSD_SHIFT)
+#define I2S_CKR_TSD_MASK (0xff << I2S_CKR_TSD_SHIFT)
+
+/*
+ * FIFOLR
+ * FIFO level register
+*/
+#define I2S_FIFOLR_RFL_SHIFT 24
+#define I2S_FIFOLR_RFL_MASK (0x3f << I2S_FIFOLR_RFL_SHIFT)
+#define I2S_FIFOLR_TFL3_SHIFT 18
+#define I2S_FIFOLR_TFL3_MASK (0x3f << I2S_FIFOLR_TFL3_SHIFT)
+#define I2S_FIFOLR_TFL2_SHIFT 12
+#define I2S_FIFOLR_TFL2_MASK (0x3f << I2S_FIFOLR_TFL2_SHIFT)
+#define I2S_FIFOLR_TFL1_SHIFT 6
+#define I2S_FIFOLR_TFL1_MASK (0x3f << I2S_FIFOLR_TFL1_SHIFT)
+#define I2S_FIFOLR_TFL0_SHIFT 0
+#define I2S_FIFOLR_TFL0_MASK (0x3f << I2S_FIFOLR_TFL0_SHIFT)
+
+/*
+ * DMACR
+ * DMA control register
+*/
+#define I2S_DMACR_RDE_SHIFT 24
+#define I2S_DMACR_RDE_DISABLE (0 << I2S_DMACR_RDE_SHIFT)
+#define I2S_DMACR_RDE_ENABLE (1 << I2S_DMACR_RDE_SHIFT)
+#define I2S_DMACR_RDL_SHIFT 16
+#define I2S_DMACR_RDL(x) ((x - 1) << I2S_DMACR_RDL_SHIFT)
+#define I2S_DMACR_RDL_MASK (0x1f << I2S_DMACR_RDL_SHIFT)
+#define I2S_DMACR_TDE_SHIFT 8
+#define I2S_DMACR_TDE_DISABLE (0 << I2S_DMACR_TDE_SHIFT)
+#define I2S_DMACR_TDE_ENABLE (1 << I2S_DMACR_TDE_SHIFT)
+#define I2S_DMACR_TDL_SHIFT 0
+#define I2S_DMACR_TDL(x) ((x - 1) << I2S_DMACR_TDL_SHIFT)
+#define I2S_DMACR_TDL_MASK (0x1f << I2S_DMACR_TDL_SHIFT)
+
+/*
+ * INTCR
+ * interrupt control register
+*/
+#define I2S_INTCR_RFT_SHIFT 20
+#define I2S_INTCR_RFT(x) ((x - 1) << I2S_INTCR_RFT_SHIFT)
+#define I2S_INTCR_RXOIC BIT(18)
+#define I2S_INTCR_RXOIE_SHIFT 17
+#define I2S_INTCR_RXOIE_DISABLE (0 << I2S_INTCR_RXOIE_SHIFT)
+#define I2S_INTCR_RXOIE_ENABLE (1 << I2S_INTCR_RXOIE_SHIFT)
+#define I2S_INTCR_RXFIE_SHIFT 16
+#define I2S_INTCR_RXFIE_DISABLE (0 << I2S_INTCR_RXFIE_SHIFT)
+#define I2S_INTCR_RXFIE_ENABLE (1 << I2S_INTCR_RXFIE_SHIFT)
+#define I2S_INTCR_TFT_SHIFT 4
+#define I2S_INTCR_TFT(x) ((x - 1) << I2S_INTCR_TFT_SHIFT)
+#define I2S_INTCR_TFT_MASK (0x1f << I2S_INTCR_TFT_SHIFT)
+#define I2S_INTCR_TXUIC BIT(2)
+#define I2S_INTCR_TXUIE_SHIFT 1
+#define I2S_INTCR_TXUIE_DISABLE (0 << I2S_INTCR_TXUIE_SHIFT)
+#define I2S_INTCR_TXUIE_ENABLE (1 << I2S_INTCR_TXUIE_SHIFT)
+
+/*
+ * INTSR
+ * interrupt status register
+*/
+#define I2S_INTSR_TXEIE_SHIFT 0
+#define I2S_INTSR_TXEIE_DISABLE (0 << I2S_INTSR_TXEIE_SHIFT)
+#define I2S_INTSR_TXEIE_ENABLE (1 << I2S_INTSR_TXEIE_SHIFT)
+#define I2S_INTSR_RXOI_SHIFT 17
+#define I2S_INTSR_RXOI_INA (0 << I2S_INTSR_RXOI_SHIFT)
+#define I2S_INTSR_RXOI_ACT (1 << I2S_INTSR_RXOI_SHIFT)
+#define I2S_INTSR_RXFI_SHIFT 16
+#define I2S_INTSR_RXFI_INA (0 << I2S_INTSR_RXFI_SHIFT)
+#define I2S_INTSR_RXFI_ACT (1 << I2S_INTSR_RXFI_SHIFT)
+#define I2S_INTSR_TXUI_SHIFT 1
+#define I2S_INTSR_TXUI_INA (0 << I2S_INTSR_TXUI_SHIFT)
+#define I2S_INTSR_TXUI_ACT (1 << I2S_INTSR_TXUI_SHIFT)
+#define I2S_INTSR_TXEI_SHIFT 0
+#define I2S_INTSR_TXEI_INA (0 << I2S_INTSR_TXEI_SHIFT)
+#define I2S_INTSR_TXEI_ACT (1 << I2S_INTSR_TXEI_SHIFT)
+
+/*
+ * XFER
+ * Transfer start register
+*/
+#define I2S_XFER_RXS_SHIFT 1
+#define I2S_XFER_RXS_STOP (0 << I2S_XFER_RXS_SHIFT)
+#define I2S_XFER_RXS_START (1 << I2S_XFER_RXS_SHIFT)
+#define I2S_XFER_TXS_SHIFT 0
+#define I2S_XFER_TXS_STOP (0 << I2S_XFER_TXS_SHIFT)
+#define I2S_XFER_TXS_START (1 << I2S_XFER_TXS_SHIFT)
+
+/*
+ * CLR
+ * clear SCLK domain logic register
+*/
+#define I2S_CLR_RXC BIT(1)
+#define I2S_CLR_TXC BIT(0)
+
+/*
+ * TXDR
+ * Transimt FIFO data register, write only.
+*/
+#define I2S_TXDR_MASK (0xff)
+
+/*
+ * RXDR
+ * Receive FIFO data register, write only.
+*/
+#define I2S_RXDR_MASK (0xff)
+
+/* Clock divider id */
+enum {
+ ROCKCHIP_DIV_MCLK = 0,
+ ROCKCHIP_DIV_BCLK,
+};
+
+/* I2S REGS */
+#define I2S_TXCR (0x0000)
+#define I2S_RXCR (0x0004)
+#define I2S_CKR (0x0008)
+#define I2S_FIFOLR (0x000c)
+#define I2S_DMACR (0x0010)
+#define I2S_INTCR (0x0014)
+#define I2S_INTSR (0x0018)
+#define I2S_XFER (0x001c)
+#define I2S_CLR (0x0020)
+#define I2S_TXDR (0x0024)
+#define I2S_RXDR (0x0028)
+
+#endif /* _ROCKCHIP_IIS_H */
diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig
index c74eb3d4a47c..f244a2566f20 100644
--- a/sound/soc/s6000/Kconfig
+++ b/sound/soc/s6000/Kconfig
@@ -1,17 +1,24 @@
config SND_S6000_SOC
tristate "SoC Audio for the Stretch s6000 family"
- depends on XTENSA_VARIANT_S6000
+ depends on XTENSA_VARIANT_S6000 || COMPILE_TEST
+ depends on HAS_IOMEM
+ select SND_S6000_SOC_PCM if XTENSA_VARIANT_S6000
help
Say Y or M if you want to add support for codecs attached to
s6000 family chips. You will also need to select the platform
to support below.
+config SND_S6000_SOC_PCM
+ tristate
+
config SND_S6000_SOC_I2S
tristate
config SND_S6000_SOC_S6IPCAM
- tristate "SoC Audio support for Stretch 6105 IP Camera"
- depends on SND_S6000_SOC && XTENSA_PLATFORM_S6105
+ bool "SoC Audio support for Stretch 6105 IP Camera"
+ depends on SND_S6000_SOC=y
+ depends on I2C=y
+ depends on XTENSA_PLATFORM_S6105 || COMPILE_TEST
select SND_S6000_SOC_I2S
select SND_SOC_TLV320AIC3X
help
diff --git a/sound/soc/s6000/Makefile b/sound/soc/s6000/Makefile
index 7a613612e010..0f0ae2a012aa 100644
--- a/sound/soc/s6000/Makefile
+++ b/sound/soc/s6000/Makefile
@@ -2,7 +2,7 @@
snd-soc-s6000-objs := s6000-pcm.o
snd-soc-s6000-i2s-objs := s6000-i2s.o
-obj-$(CONFIG_SND_S6000_SOC) += snd-soc-s6000.o
+obj-$(CONFIG_SND_S6000_SOC_PCM) += snd-soc-s6000.o
obj-$(CONFIG_SND_S6000_SOC_I2S) += snd-soc-s6000-i2s.o
# s6105 Machine Support
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
index 7eba7979b9af..1c8d01166e5b 100644
--- a/sound/soc/s6000/s6000-i2s.c
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -570,7 +570,7 @@ err_release_none:
return ret;
}
-static void s6000_i2s_remove(struct platform_device *pdev)
+static int s6000_i2s_remove(struct platform_device *pdev)
{
struct s6000_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
struct resource *region;
@@ -597,6 +597,8 @@ static void s6000_i2s_remove(struct platform_device *pdev)
iounmap(mmio);
region = platform_get_resource(pdev, IORESOURCE_IO, 0);
release_mem_region(region->start, resource_size(region));
+
+ return 0;
}
static struct platform_driver s6000_i2s_driver = {
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
index 0b21d1dc80c1..3510c01f8a6a 100644
--- a/sound/soc/s6000/s6105-ipcam.c
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -19,8 +19,6 @@
#include <sound/pcm.h>
#include <sound/soc.h>
-#include <variant/dmac.h>
-
#include "s6000-pcm.h"
#include "s6000-i2s.h"
@@ -135,22 +133,8 @@ static const struct snd_kcontrol_new audio_out_mux = {
/* Logic for a aic3x as connected on the s6105 ip camera ref design */
static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
struct snd_soc_card *card = rtd->card;
- /* not present */
- snd_soc_dapm_nc_pin(dapm, "MONO_LOUT");
- snd_soc_dapm_nc_pin(dapm, "LINE2L");
- snd_soc_dapm_nc_pin(dapm, "LINE2R");
-
- /* not connected */
- snd_soc_dapm_nc_pin(dapm, "MIC3L"); /* LINE2L on this chip */
- snd_soc_dapm_nc_pin(dapm, "MIC3R"); /* LINE2R on this chip */
- snd_soc_dapm_nc_pin(dapm, "LLOUT");
- snd_soc_dapm_nc_pin(dapm, "RLOUT");
- snd_soc_dapm_nc_pin(dapm, "HPRCOM");
-
/* must correspond to audio_out_mux.private_value initializer */
snd_soc_dapm_disable_pin(&card->dapm, "Audio Out Differential");
@@ -182,6 +166,7 @@ static struct snd_soc_card snd_soc_card_s6105 = {
.num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
+ .fully_routed = true,
};
static struct s6000_snd_platform_data s6105_snd_data __initdata = {
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 753b8c93ab51..55a38697443d 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -1,25 +1,16 @@
config SND_SOC_SAMSUNG
tristate "ASoC support for Samsung"
depends on PLAT_SAMSUNG
- select S3C2410_DMA if ARCH_S3C24XX
- select S3C64XX_PL080 if ARCH_S3C64XX
- select SND_S3C_DMA if !ARCH_S3C24XX
- select SND_S3C_DMA_LEGACY if ARCH_S3C24XX
- select SND_SOC_GENERIC_DMAENGINE_PCM if !ARCH_S3C24XX
+ depends on S3C64XX_PL080 || !ARCH_S3C64XX
+ depends on S3C24XX_DMAC || !ARCH_S3C24XX
+ select SND_SOC_GENERIC_DMAENGINE_PCM
help
Say Y or M if you want to add support for codecs attached to
the Samsung SoCs' Audio interfaces. You will also need to
select the audio interfaces to support below.
-config SND_S3C_DMA
- tristate
-
-config SND_S3C_DMA_LEGACY
- tristate
-
config SND_S3C24XX_I2S
tristate
- select S3C24XX_DMA
config SND_S3C_I2SV2_SOC
tristate
@@ -27,7 +18,6 @@ config SND_S3C_I2SV2_SOC
config SND_S3C2412_SOC_I2S
tristate
select SND_S3C_I2SV2_SOC
- select S3C2410_DMA
config SND_SAMSUNG_PCM
tristate
@@ -55,7 +45,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753
config SND_SOC_SAMSUNG_JIVE_WM8750
tristate "SoC I2S Audio support for Jive"
- depends on SND_SOC_SAMSUNG && MACH_JIVE
+ depends on SND_SOC_SAMSUNG && MACH_JIVE && I2C
select SND_SOC_WM8750
select SND_S3C2412_SOC_I2S
help
@@ -63,7 +53,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750
config SND_SOC_SAMSUNG_SMDK_WM8580
tristate "SoC I2S Audio support for WM8580 on SMDK"
- depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDK6440 || MACH_SMDK6450 || MACH_SMDKV210 || MACH_SMDKC110)
+ depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110)
depends on REGMAP_I2C
select SND_SOC_WM8580
select SND_SAMSUNG_I2S
@@ -83,7 +73,6 @@ config SND_SOC_SAMSUNG_SMDK_WM8994
config SND_SOC_SAMSUNG_SMDK2443_WM9710
tristate "SoC AC97 Audio support for SMDK2443 - WM9710"
depends on SND_SOC_SAMSUNG && MACH_SMDK2443
- select S3C2410_DMA
select AC97_BUS
select SND_SOC_AC97_CODEC
select SND_SAMSUNG_AC97
@@ -94,7 +83,6 @@ config SND_SOC_SAMSUNG_SMDK2443_WM9710
config SND_SOC_SAMSUNG_LN2440SBC_ALC650
tristate "SoC AC97 Audio support for LN2440SBC - ALC650"
depends on SND_SOC_SAMSUNG && ARCH_S3C24XX
- select S3C2410_DMA
select AC97_BUS
select SND_SOC_AC97_CODEC
select SND_SAMSUNG_AC97
@@ -154,7 +142,7 @@ config SND_SOC_SAMSUNG_SMDK_WM9713
config SND_SOC_SMARTQ
tristate "SoC I2S Audio support for SmartQ board"
- depends on SND_SOC_SAMSUNG && MACH_SMARTQ
+ depends on SND_SOC_SAMSUNG && MACH_SMARTQ && I2C
select SND_SAMSUNG_I2S
select SND_SOC_WM8750
@@ -178,7 +166,7 @@ config SND_SOC_SAMSUNG_SMDK_SPDIF
config SND_SOC_SMDK_WM8580_PCM
tristate "SoC PCM Audio support for WM8580 on SMDK"
- depends on SND_SOC_SAMSUNG && (MACH_SMDK6450 || MACH_SMDKV210 || MACH_SMDKC110)
+ depends on SND_SOC_SAMSUNG && (MACH_SMDKV210 || MACH_SMDKC110)
depends on REGMAP_I2C
select SND_SOC_WM8580
select SND_SAMSUNG_PCM
@@ -206,7 +194,7 @@ config SND_SOC_SPEYSIDE
config SND_SOC_TOBERMORY
tristate "Audio support for Wolfson Tobermory"
- depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && INPUT
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && INPUT && I2C
select SND_SAMSUNG_I2S
select SND_SOC_WM8962
@@ -222,7 +210,7 @@ config SND_SOC_BELLS
config SND_SOC_LOWLAND
tristate "Audio support for Wolfson Lowland"
- depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C
select SND_SAMSUNG_I2S
select SND_SOC_WM5100
select SND_SOC_WM9081
@@ -236,10 +224,18 @@ config SND_SOC_LITTLEMILL
config SND_SOC_SNOW
tristate "Audio support for Google Snow boards"
- depends on SND_SOC_SAMSUNG
+ depends on SND_SOC_SAMSUNG && I2C
select SND_SOC_MAX98090
select SND_SOC_MAX98095
select SND_SAMSUNG_I2S
help
Say Y if you want to add audio support for various Snow
boards based on Exynos5 series of SoCs.
+
+config SND_SOC_ODROIDX2
+ tristate "Audio support for Odroid-X2 and Odroid-U3"
+ depends on SND_SOC_SAMSUNG
+ select SND_SOC_MAX98090
+ select SND_SAMSUNG_I2S
+ help
+ Say Y here to enable audio support for the Odroid-X2/U3.
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 6d0212ba571c..91505ddaaf95 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -1,6 +1,5 @@
# S3c24XX Platform Support
snd-soc-s3c-dma-objs := dmaengine.o
-snd-soc-s3c-dma-legacy-objs := dma.o
snd-soc-idma-objs := idma.o
snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o
@@ -10,8 +9,7 @@ snd-soc-samsung-spdif-objs := spdif.o
snd-soc-pcm-objs := pcm.o
snd-soc-i2s-objs := i2s.o
-obj-$(CONFIG_SND_S3C_DMA) += snd-soc-s3c-dma.o
-obj-$(CONFIG_SND_S3C_DMA_LEGACY) += snd-soc-s3c-dma-legacy.o
+obj-$(CONFIG_SND_SOC_SAMSUNG) += snd-soc-s3c-dma.o
obj-$(CONFIG_SND_S3C24XX_I2S) += snd-soc-s3c24xx-i2s.o
obj-$(CONFIG_SND_SAMSUNG_AC97) += snd-soc-ac97.o
obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
@@ -46,6 +44,7 @@ snd-soc-tobermory-objs := tobermory.o
snd-soc-lowland-objs := lowland.o
snd-soc-littlemill-objs := littlemill.o
snd-soc-bells-objs := bells.o
+snd-soc-odroidx2-max98090-objs := odroidx2_max98090.o
obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o
obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
@@ -71,3 +70,4 @@ obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o
obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o
obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o
obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o
+obj-$(CONFIG_SND_SOC_ODROIDX2) += snd-soc-odroidx2-max98090.o
diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c
index 68d9303047e8..e1615113fd84 100644
--- a/sound/soc/samsung/ac97.c
+++ b/sound/soc/samsung/ac97.c
@@ -19,7 +19,6 @@
#include <sound/soc.h>
-#include <mach/dma.h>
#include "regs-ac97.h"
#include <linux/platform_data/asoc-s3c.h>
@@ -39,30 +38,15 @@ struct s3c_ac97_info {
};
static struct s3c_ac97_info s3c_ac97;
-static struct s3c_dma_client s3c_dma_client_out = {
- .name = "AC97 PCMOut"
-};
-
-static struct s3c_dma_client s3c_dma_client_in = {
- .name = "AC97 PCMIn"
-};
-
-static struct s3c_dma_client s3c_dma_client_micin = {
- .name = "AC97 MicIn"
-};
-
static struct s3c_dma_params s3c_ac97_pcm_out = {
- .client = &s3c_dma_client_out,
.dma_size = 4,
};
static struct s3c_dma_params s3c_ac97_pcm_in = {
- .client = &s3c_dma_client_in,
.dma_size = 4,
};
static struct s3c_dma_params s3c_ac97_mic_in = {
- .client = &s3c_dma_client_micin,
.dma_size = 4,
};
@@ -225,9 +209,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
u32 ac_glbctrl;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct s3c_dma_params *dma_data =
- snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL);
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
@@ -253,11 +234,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL);
- if (!dma_data->ops)
- dma_data->ops = samsung_dma_get_ops();
-
- dma_data->ops->started(dma_data->channel);
-
return 0;
}
@@ -265,9 +241,6 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
u32 ac_glbctrl;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct s3c_dma_params *dma_data =
- snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL);
ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK;
@@ -287,11 +260,6 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream,
writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL);
- if (!dma_data->ops)
- dma_data->ops = samsung_dma_get_ops();
-
- dma_data->ops->started(dma_data->channel);
-
return 0;
}
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
deleted file mode 100644
index d9dc7bcc0336..000000000000
--- a/sound/soc/samsung/dma.c
+++ /dev/null
@@ -1,454 +0,0 @@
-/*
- * dma.c -- ALSA Soc Audio Layer
- *
- * (c) 2006 Wolfson Microelectronics PLC.
- * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
- *
- * Copyright 2004-2005 Simtec Electronics
- * http://armlinux.simtec.co.uk/
- * Ben Dooks <ben@simtec.co.uk>
- *
- * 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/slab.h>
-#include <linux/dma-mapping.h>
-#include <linux/module.h>
-
-#include <sound/soc.h>
-#include <sound/pcm_params.h>
-
-#include <asm/dma.h>
-#include <mach/hardware.h>
-#include <mach/dma.h>
-
-#include "dma.h"
-
-#define ST_RUNNING (1<<0)
-#define ST_OPENED (1<<1)
-
-static const struct snd_pcm_hardware dma_hardware = {
- .info = SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_MMAP_VALID,
- .buffer_bytes_max = 128*1024,
- .period_bytes_min = PAGE_SIZE,
- .period_bytes_max = PAGE_SIZE*2,
- .periods_min = 2,
- .periods_max = 128,
- .fifo_size = 32,
-};
-
-struct runtime_data {
- spinlock_t lock;
- int state;
- unsigned int dma_loaded;
- unsigned int dma_period;
- dma_addr_t dma_start;
- dma_addr_t dma_pos;
- dma_addr_t dma_end;
- struct s3c_dma_params *params;
-};
-
-static void audio_buffdone(void *data);
-
-/* dma_enqueue
- *
- * place a dma buffer onto the queue for the dma system
- * to handle.
- */
-static void dma_enqueue(struct snd_pcm_substream *substream)
-{
- struct runtime_data *prtd = substream->runtime->private_data;
- dma_addr_t pos = prtd->dma_pos;
- unsigned int limit;
- struct samsung_dma_prep dma_info;
-
- pr_debug("Entered %s\n", __func__);
-
- limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period;
-
- pr_debug("%s: loaded %d, limit %d\n",
- __func__, prtd->dma_loaded, limit);
-
- dma_info.cap = (samsung_dma_has_circular() ? DMA_CYCLIC : DMA_SLAVE);
- dma_info.direction =
- (substream->stream == SNDRV_PCM_STREAM_PLAYBACK
- ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM);
- dma_info.fp = audio_buffdone;
- dma_info.fp_param = substream;
- 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);
-
- if ((pos + dma_info.period) > prtd->dma_end) {
- dma_info.period = prtd->dma_end - pos;
- pr_debug("%s: corrected dma len %ld\n",
- __func__, dma_info.period);
- }
-
- dma_info.buf = pos;
- prtd->params->ops->prepare(prtd->params->ch, &dma_info);
-
- prtd->dma_loaded++;
- pos += prtd->dma_period;
- if (pos >= prtd->dma_end)
- pos = prtd->dma_start;
- }
-
- prtd->dma_pos = pos;
-}
-
-static void audio_buffdone(void *data)
-{
- struct snd_pcm_substream *substream = data;
- struct runtime_data *prtd = substream->runtime->private_data;
-
- pr_debug("Entered %s\n", __func__);
-
- if (prtd->state & ST_RUNNING) {
- prtd->dma_pos += prtd->dma_period;
- if (prtd->dma_pos >= prtd->dma_end)
- prtd->dma_pos = prtd->dma_start;
-
- if (substream)
- snd_pcm_period_elapsed(substream);
-
- spin_lock(&prtd->lock);
- if (!samsung_dma_has_circular()) {
- prtd->dma_loaded--;
- dma_enqueue(substream);
- }
- spin_unlock(&prtd->lock);
- }
-}
-
-static int dma_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct runtime_data *prtd = runtime->private_data;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- unsigned long totbytes = params_buffer_bytes(params);
- struct s3c_dma_params *dma =
- snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
- struct samsung_dma_req req;
- struct samsung_dma_config config;
-
- pr_debug("Entered %s\n", __func__);
-
- /* return if this is a bufferless transfer e.g.
- * codec <--> BT codec or GSM modem -- lg FIXME */
- if (!dma)
- return 0;
-
- /* this may get called several times by oss emulation
- * with different params -HW */
- if (prtd->params == NULL) {
- /* prepare DMA */
- prtd->params = dma;
-
- pr_debug("params %p, client %p, channel %d\n", prtd->params,
- prtd->params->client, prtd->params->channel);
-
- prtd->params->ops = samsung_dma_get_ops();
-
- req.cap = (samsung_dma_has_circular() ?
- DMA_CYCLIC : DMA_SLAVE);
- req.client = prtd->params->client;
- config.direction =
- (substream->stream == SNDRV_PCM_STREAM_PLAYBACK
- ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM);
- config.width = prtd->params->dma_size;
- config.fifo = prtd->params->dma_addr;
- 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);
- }
-
- snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-
- runtime->dma_bytes = totbytes;
-
- spin_lock_irq(&prtd->lock);
- prtd->dma_loaded = 0;
- prtd->dma_period = params_period_bytes(params);
- prtd->dma_start = runtime->dma_addr;
- prtd->dma_pos = prtd->dma_start;
- prtd->dma_end = prtd->dma_start + totbytes;
- spin_unlock_irq(&prtd->lock);
-
- return 0;
-}
-
-static int dma_hw_free(struct snd_pcm_substream *substream)
-{
- struct runtime_data *prtd = substream->runtime->private_data;
-
- pr_debug("Entered %s\n", __func__);
-
- snd_pcm_set_runtime_buffer(substream, NULL);
-
- if (prtd->params) {
- prtd->params->ops->flush(prtd->params->ch);
- prtd->params->ops->release(prtd->params->ch,
- prtd->params->client);
- prtd->params = NULL;
- }
-
- return 0;
-}
-
-static int dma_prepare(struct snd_pcm_substream *substream)
-{
- struct runtime_data *prtd = substream->runtime->private_data;
- int ret = 0;
-
- pr_debug("Entered %s\n", __func__);
-
- /* return if this is a bufferless transfer e.g.
- * codec <--> BT codec or GSM modem -- lg FIXME */
- if (!prtd->params)
- return 0;
-
- /* flush the DMA channel */
- prtd->params->ops->flush(prtd->params->ch);
-
- prtd->dma_loaded = 0;
- prtd->dma_pos = prtd->dma_start;
-
- /* enqueue dma buffers */
- dma_enqueue(substream);
-
- return ret;
-}
-
-static int dma_trigger(struct snd_pcm_substream *substream, int cmd)
-{
- struct runtime_data *prtd = substream->runtime->private_data;
- int ret = 0;
-
- pr_debug("Entered %s\n", __func__);
-
- spin_lock(&prtd->lock);
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- prtd->state |= ST_RUNNING;
- prtd->params->ops->trigger(prtd->params->ch);
- break;
-
- case SNDRV_PCM_TRIGGER_STOP:
- prtd->state &= ~ST_RUNNING;
- prtd->params->ops->stop(prtd->params->ch);
- break;
-
- default:
- ret = -EINVAL;
- break;
- }
-
- spin_unlock(&prtd->lock);
-
- return ret;
-}
-
-static snd_pcm_uframes_t
-dma_pointer(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct runtime_data *prtd = runtime->private_data;
- unsigned long res;
-
- pr_debug("Entered %s\n", __func__);
-
- res = prtd->dma_pos - prtd->dma_start;
-
- pr_debug("Pointer offset: %lu\n", res);
-
- /* we seem to be getting the odd error from the pcm library due
- * to out-of-bounds pointers. this is maybe due to the dma engine
- * not having loaded the new values for the channel before being
- * called... (todo - fix )
- */
-
- if (res >= snd_pcm_lib_buffer_bytes(substream)) {
- if (res == snd_pcm_lib_buffer_bytes(substream))
- res = 0;
- }
-
- return bytes_to_frames(substream->runtime, res);
-}
-
-static int dma_open(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct runtime_data *prtd;
-
- pr_debug("Entered %s\n", __func__);
-
- snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
- snd_soc_set_runtime_hwparams(substream, &dma_hardware);
-
- prtd = kzalloc(sizeof(struct runtime_data), GFP_KERNEL);
- if (prtd == NULL)
- return -ENOMEM;
-
- spin_lock_init(&prtd->lock);
-
- runtime->private_data = prtd;
- return 0;
-}
-
-static int dma_close(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct runtime_data *prtd = runtime->private_data;
-
- pr_debug("Entered %s\n", __func__);
-
- if (!prtd)
- pr_debug("dma_close called with prtd == NULL\n");
-
- kfree(prtd);
-
- return 0;
-}
-
-static int dma_mmap(struct snd_pcm_substream *substream,
- struct vm_area_struct *vma)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
-
- pr_debug("Entered %s\n", __func__);
-
- return dma_mmap_writecombine(substream->pcm->card->dev, vma,
- runtime->dma_area,
- runtime->dma_addr,
- runtime->dma_bytes);
-}
-
-static struct snd_pcm_ops dma_ops = {
- .open = dma_open,
- .close = dma_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = dma_hw_params,
- .hw_free = dma_hw_free,
- .prepare = dma_prepare,
- .trigger = dma_trigger,
- .pointer = dma_pointer,
- .mmap = dma_mmap,
-};
-
-static int preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
-{
- struct snd_pcm_substream *substream = pcm->streams[stream].substream;
- struct snd_dma_buffer *buf = &substream->dma_buffer;
- size_t size = dma_hardware.buffer_bytes_max;
-
- pr_debug("Entered %s\n", __func__);
-
- buf->dev.type = SNDRV_DMA_TYPE_DEV;
- buf->dev.dev = pcm->card->dev;
- buf->private_data = NULL;
- buf->area = dma_alloc_writecombine(pcm->card->dev, size,
- &buf->addr, GFP_KERNEL);
- if (!buf->area)
- return -ENOMEM;
- buf->bytes = size;
- return 0;
-}
-
-static void dma_free_dma_buffers(struct snd_pcm *pcm)
-{
- struct snd_pcm_substream *substream;
- struct snd_dma_buffer *buf;
- int stream;
-
- pr_debug("Entered %s\n", __func__);
-
- for (stream = 0; stream < 2; stream++) {
- substream = pcm->streams[stream].substream;
- if (!substream)
- continue;
-
- buf = &substream->dma_buffer;
- if (!buf->area)
- continue;
-
- dma_free_writecombine(pcm->card->dev, buf->bytes,
- buf->area, buf->addr);
- buf->area = NULL;
- }
-}
-
-static int dma_new(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_card *card = rtd->card->snd_card;
- struct snd_pcm *pcm = rtd->pcm;
- int ret;
-
- pr_debug("Entered %s\n", __func__);
-
- ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
- if (ret)
- return ret;
-
- if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
- ret = preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_PLAYBACK);
- if (ret)
- goto out;
- }
-
- if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
- ret = preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_CAPTURE);
- if (ret)
- goto out;
- }
-out:
- return ret;
-}
-
-static struct snd_soc_platform_driver samsung_asoc_platform = {
- .ops = &dma_ops,
- .pcm_new = dma_new,
- .pcm_free = dma_free_dma_buffers,
-};
-
-void samsung_asoc_init_dma_data(struct snd_soc_dai *dai,
- struct s3c_dma_params *playback,
- struct s3c_dma_params *capture)
-{
- snd_soc_dai_init_dma_data(dai, playback, capture);
-}
-EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data);
-
-int samsung_asoc_dma_platform_register(struct device *dev)
-{
- return devm_snd_soc_register_platform(dev, &samsung_asoc_platform);
-}
-EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register);
-
-MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
-MODULE_DESCRIPTION("Samsung ASoC DMA Driver");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h
index 070ab0f09609..0e85dcfec023 100644
--- a/sound/soc/samsung/dma.h
+++ b/sound/soc/samsung/dma.h
@@ -14,17 +14,10 @@
#include <sound/dmaengine_pcm.h>
-struct s3c_dma_client {
- char *name;
-};
-
struct s3c_dma_params {
- struct s3c_dma_client *client; /* stream identifier */
int channel; /* Channel ID */
dma_addr_t dma_addr;
int dma_size; /* Size of the DMA transfer */
- unsigned ch;
- struct samsung_dma_ops *ops;
char *ch_name;
struct snd_dmaengine_dai_dma_data dma_data;
};
diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c
index a0e4e7948909..506f5bf6d082 100644
--- a/sound/soc/samsung/dmaengine.c
+++ b/sound/soc/samsung/dmaengine.c
@@ -17,6 +17,7 @@
#include <linux/module.h>
#include <linux/amba/pl08x.h>
+#include <linux/platform_data/dma-s3c24xx.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -29,6 +30,8 @@
#ifdef CONFIG_ARCH_S3C64XX
#define filter_fn pl08x_filter_id
+#elif defined(CONFIG_ARCH_S3C24XX)
+#define filter_fn s3c24xx_dma_filter
#else
#define filter_fn NULL
#endif
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 2ac76fa3e742..03eec22f0f46 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -68,6 +68,8 @@ struct i2s_dai {
#define DAI_OPENED (1 << 0) /* Dai is opened */
#define DAI_MANAGER (1 << 1) /* Dai is the manager */
unsigned mode;
+ /* CDCLK pin direction: 0 - input, 1 - output */
+ unsigned int cdclk_out:1;
/* Driver for this DAI */
struct snd_soc_dai_driver i2s_dai_drv;
/* DMA parameters */
@@ -737,6 +739,9 @@ static int i2s_startup(struct snd_pcm_substream *substream,
spin_unlock_irqrestore(&lock, flags);
+ if (!is_opened(other) && i2s->cdclk_out)
+ i2s_set_sysclk(dai, SAMSUNG_I2S_CDCLK,
+ 0, SND_SOC_CLOCK_OUT);
return 0;
}
@@ -752,9 +757,13 @@ static void i2s_shutdown(struct snd_pcm_substream *substream,
i2s->mode &= ~DAI_OPENED;
i2s->mode &= ~DAI_MANAGER;
- if (is_opened(other))
+ if (is_opened(other)) {
other->mode |= DAI_MANAGER;
-
+ } else {
+ u32 mod = readl(i2s->addr + I2SMOD);
+ i2s->cdclk_out = !(mod & MOD_CDCLKCON);
+ other->cdclk_out = i2s->cdclk_out;
+ }
/* Reset any constraint on RFS and BFS */
i2s->rfs = 0;
i2s->bfs = 0;
@@ -920,11 +929,9 @@ static int i2s_suspend(struct snd_soc_dai *dai)
{
struct i2s_dai *i2s = to_info(dai);
- if (dai->active) {
- i2s->suspend_i2smod = readl(i2s->addr + I2SMOD);
- i2s->suspend_i2scon = readl(i2s->addr + I2SCON);
- i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR);
- }
+ i2s->suspend_i2smod = readl(i2s->addr + I2SMOD);
+ i2s->suspend_i2scon = readl(i2s->addr + I2SCON);
+ i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR);
return 0;
}
@@ -933,11 +940,9 @@ static int i2s_resume(struct snd_soc_dai *dai)
{
struct i2s_dai *i2s = to_info(dai);
- if (dai->active) {
- writel(i2s->suspend_i2scon, i2s->addr + I2SCON);
- writel(i2s->suspend_i2smod, i2s->addr + I2SMOD);
- writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR);
- }
+ writel(i2s->suspend_i2scon, i2s->addr + I2SCON);
+ writel(i2s->suspend_i2smod, i2s->addr + I2SMOD);
+ writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR);
return 0;
}
@@ -1216,11 +1221,7 @@ static int samsung_i2s_probe(struct platform_device *pdev)
pri_dai->dma_playback.dma_addr = regs_base + I2STXD;
pri_dai->dma_capture.dma_addr = regs_base + I2SRXD;
- pri_dai->dma_playback.client =
- (struct s3c_dma_client *)&pri_dai->dma_playback;
pri_dai->dma_playback.ch_name = "tx";
- pri_dai->dma_capture.client =
- (struct s3c_dma_client *)&pri_dai->dma_capture;
pri_dai->dma_capture.ch_name = "rx";
pri_dai->dma_playback.dma_size = 4;
pri_dai->dma_capture.dma_size = 4;
@@ -1238,8 +1239,6 @@ static int samsung_i2s_probe(struct platform_device *pdev)
goto err;
}
sec_dai->dma_playback.dma_addr = regs_base + I2STXDS;
- sec_dai->dma_playback.client =
- (struct s3c_dma_client *)&sec_dai->dma_playback;
sec_dai->dma_playback.ch_name = "tx-sec";
if (!np) {
diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c
index 8cc5770abb39..db6cefa18017 100644
--- a/sound/soc/samsung/idma.c
+++ b/sound/soc/samsung/idma.c
@@ -261,10 +261,9 @@ static int idma_mmap(struct snd_pcm_substream *substream,
static irqreturn_t iis_irq(int irqno, void *dev_id)
{
struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id;
- u32 iiscon, iisahb, val, addr;
+ u32 iisahb, val, addr;
iisahb = readl(idma.regs + I2SAHB);
- iiscon = readl(idma.regs + I2SCON);
val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0;
diff --git a/sound/soc/samsung/odroidx2_max98090.c b/sound/soc/samsung/odroidx2_max98090.c
new file mode 100644
index 000000000000..278edf9e2a87
--- /dev/null
+++ b/sound/soc/samsung/odroidx2_max98090.c
@@ -0,0 +1,177 @@
+/*
+ * Copyright (C) 2014 Samsung Electronics Co., Ltd.
+ *
+ * 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/module.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include "i2s.h"
+
+struct odroidx2_drv_data {
+ const struct snd_soc_dapm_widget *dapm_widgets;
+ unsigned int num_dapm_widgets;
+};
+
+/* The I2S CDCLK output clock frequency for the MAX98090 codec */
+#define MAX98090_MCLK 19200000
+
+static int odroidx2_late_probe(struct snd_soc_card *card)
+{
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct snd_soc_dai *cpu_dai = card->rtd[0].cpu_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, MAX98090_MCLK,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* Set the cpu DAI configuration in order to use CDCLK */
+ return snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK,
+ 0, SND_SOC_CLOCK_OUT);
+}
+
+static const struct snd_soc_dapm_widget odroidx2_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_MIC("DMIC", NULL),
+};
+
+static const struct snd_soc_dapm_widget odroidu3_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Speakers", NULL),
+};
+
+static struct snd_soc_dai_link odroidx2_dai[] = {
+ {
+ .name = "MAX98090",
+ .stream_name = "MAX98090 PCM",
+ .codec_dai_name = "HiFi",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
+ }
+};
+
+static struct snd_soc_card odroidx2 = {
+ .owner = THIS_MODULE,
+ .dai_link = odroidx2_dai,
+ .num_links = ARRAY_SIZE(odroidx2_dai),
+ .fully_routed = true,
+ .late_probe = odroidx2_late_probe,
+};
+
+struct odroidx2_drv_data odroidx2_drvdata = {
+ .dapm_widgets = odroidx2_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(odroidx2_dapm_widgets),
+};
+
+struct odroidx2_drv_data odroidu3_drvdata = {
+ .dapm_widgets = odroidu3_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(odroidu3_dapm_widgets),
+};
+
+static const struct of_device_id odroidx2_audio_of_match[] = {
+ {
+ .compatible = "samsung,odroidx2-audio",
+ .data = &odroidx2_drvdata,
+ }, {
+ .compatible = "samsung,odroidu3-audio",
+ .data = &odroidu3_drvdata,
+ },
+ { },
+};
+MODULE_DEVICE_TABLE(of, odroidx2_audio_of_match);
+
+static int odroidx2_audio_probe(struct platform_device *pdev)
+{
+ struct device_node *snd_node = pdev->dev.of_node;
+ struct snd_soc_card *card = &odroidx2;
+ struct device_node *i2s_node, *codec_node;
+ struct odroidx2_drv_data *dd;
+ const struct of_device_id *of_id;
+ int ret;
+
+ of_id = of_match_node(odroidx2_audio_of_match, snd_node);
+ dd = (struct odroidx2_drv_data *)of_id->data;
+
+ card->num_dapm_widgets = dd->num_dapm_widgets;
+ card->dapm_widgets = dd->dapm_widgets;
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_of_parse_card_name(card, "samsung,model");
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_of_parse_audio_routing(card, "samsung,audio-routing");
+ if (ret < 0)
+ return ret;
+
+ codec_node = of_parse_phandle(snd_node, "samsung,audio-codec", 0);
+ if (!codec_node) {
+ dev_err(&pdev->dev,
+ "Failed parsing samsung,i2s-codec property\n");
+ return -EINVAL;
+ }
+
+ i2s_node = of_parse_phandle(snd_node, "samsung,i2s-controller", 0);
+ if (!i2s_node) {
+ dev_err(&pdev->dev,
+ "Failed parsing samsung,i2s-controller property\n");
+ ret = -EINVAL;
+ goto err_put_codec_n;
+ }
+
+ odroidx2_dai[0].codec_of_node = codec_node;
+ odroidx2_dai[0].cpu_of_node = i2s_node;
+ odroidx2_dai[0].platform_of_node = i2s_node;
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ goto err_put_i2s_n;
+ }
+ return 0;
+
+err_put_i2s_n:
+ of_node_put(i2s_node);
+err_put_codec_n:
+ of_node_put(codec_node);
+ return ret;
+}
+
+static int odroidx2_audio_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ of_node_put((struct device_node *)odroidx2_dai[0].cpu_of_node);
+ of_node_put((struct device_node *)odroidx2_dai[0].codec_of_node);
+
+ return 0;
+}
+
+static struct platform_driver odroidx2_audio_driver = {
+ .driver = {
+ .name = "odroidx2-audio",
+ .owner = THIS_MODULE,
+ .of_match_table = odroidx2_audio_of_match,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = odroidx2_audio_probe,
+ .remove = odroidx2_audio_remove,
+};
+module_platform_driver(odroidx2_audio_driver);
+
+MODULE_AUTHOR("Chen Zhen <zhen1.chen@samsung.com>");
+MODULE_AUTHOR("Sylwester Nawrocki <s.nawrocki@samsung.com>");
+MODULE_DESCRIPTION("ALSA SoC Odroid X2/U3 Audio Support");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index 4c5f97fe45c8..bac034b15a27 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -131,32 +131,20 @@ struct s3c_pcm_info {
struct s3c_dma_params *dma_capture;
};
-static struct s3c_dma_client s3c_pcm_dma_client_out = {
- .name = "PCM Stereo out"
-};
-
-static struct s3c_dma_client s3c_pcm_dma_client_in = {
- .name = "PCM Stereo in"
-};
-
static struct s3c_dma_params s3c_pcm_stereo_out[] = {
[0] = {
- .client = &s3c_pcm_dma_client_out,
.dma_size = 4,
},
[1] = {
- .client = &s3c_pcm_dma_client_out,
.dma_size = 4,
},
};
static struct s3c_dma_params s3c_pcm_stereo_in[] = {
[0] = {
- .client = &s3c_pcm_dma_client_in,
.dma_size = 4,
},
[1] = {
- .client = &s3c_pcm_dma_client_in,
.dma_size = 4,
},
};
diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c
index 0ff4bbe23af3..df65c5b494b1 100644
--- a/sound/soc/samsung/s3c-i2s-v2.c
+++ b/sound/soc/samsung/s3c-i2s-v2.c
@@ -22,8 +22,6 @@
#include <sound/soc.h>
#include <sound/pcm_params.h>
-#include <mach/dma.h>
-
#include "regs-i2s-v2.h"
#include "s3c-i2s-v2.h"
#include "dma.h"
@@ -392,8 +390,6 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
unsigned long irqs;
int ret = 0;
- struct s3c_dma_params *dma_data =
- snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
pr_debug("Entered %s\n", __func__);
@@ -424,13 +420,6 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
local_irq_restore(irqs);
- /*
- * Load the next buffer to DMA to meet the reqirement
- * of the auto reload mechanism of S3C24XX.
- * This call won't bother S3C64XX.
- */
- s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED);
-
break;
case SNDRV_PCM_TRIGGER_STOP:
@@ -644,12 +633,6 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai,
/* record our i2s structure for later use in the callbacks */
snd_soc_dai_set_drvdata(dai, i2s);
- i2s->regs = ioremap(base, 0x100);
- if (i2s->regs == NULL) {
- dev_err(dev, "cannot ioremap registers\n");
- return -ENXIO;
- }
-
i2s->iis_pclk = clk_get(dev, "iis");
if (IS_ERR(i2s->iis_pclk)) {
dev_err(dev, "failed to get iis_clock\n");
@@ -729,7 +712,7 @@ int s3c_i2sv2_register_component(struct device *dev, int id,
struct snd_soc_component_driver *cmp_drv,
struct snd_soc_dai_driver *dai_drv)
{
- struct snd_soc_dai_ops *ops = dai_drv->ops;
+ struct snd_soc_dai_ops *ops = (struct snd_soc_dai_ops *)dai_drv->ops;
ops->trigger = s3c2412_i2s_trigger;
if (!ops->hw_params)
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 08c059be9104..27b339c6580e 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -33,25 +33,15 @@
#include "regs-i2s-v2.h"
#include "s3c2412-i2s.h"
-static struct s3c_dma_client s3c2412_dma_client_out = {
- .name = "I2S PCM Stereo out"
-};
-
-static struct s3c_dma_client s3c2412_dma_client_in = {
- .name = "I2S PCM Stereo in"
-};
-
static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = {
- .client = &s3c2412_dma_client_out,
.channel = DMACH_I2S_OUT,
- .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD,
+ .ch_name = "tx",
.dma_size = 4,
};
static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = {
- .client = &s3c2412_dma_client_in,
.channel = DMACH_I2S_IN,
- .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD,
+ .ch_name = "rx",
.dma_size = 4,
};
@@ -63,6 +53,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai)
pr_debug("Entered %s\n", __func__);
+ samsung_asoc_init_dma_data(dai, &s3c2412_i2s_pcm_stereo_out,
+ &s3c2412_i2s_pcm_stereo_in);
+
ret = s3c_i2sv2_probe(dai, &s3c2412_i2s, S3C2410_PA_IIS);
if (ret)
return ret;
@@ -70,17 +63,16 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai)
s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in;
s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out;
- s3c2412_i2s.iis_cclk = clk_get(dai->dev, "i2sclk");
+ s3c2412_i2s.iis_cclk = devm_clk_get(dai->dev, "i2sclk");
if (IS_ERR(s3c2412_i2s.iis_cclk)) {
pr_err("failed to get i2sclk clock\n");
- iounmap(s3c2412_i2s.regs);
return PTR_ERR(s3c2412_i2s.iis_cclk);
}
/* Set MPLL as the source for IIS CLK */
clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll"));
- clk_enable(s3c2412_i2s.iis_cclk);
+ clk_prepare_enable(s3c2412_i2s.iis_cclk);
s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk;
@@ -93,9 +85,7 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai)
static int s3c2412_i2s_remove(struct snd_soc_dai *dai)
{
- clk_disable(s3c2412_i2s.iis_cclk);
- clk_put(s3c2412_i2s.iis_cclk);
- iounmap(s3c2412_i2s.regs);
+ clk_disable_unprepare(s3c2412_i2s.iis_cclk);
return 0;
}
@@ -105,18 +95,10 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai)
{
struct s3c_i2sv2_info *i2s = snd_soc_dai_get_drvdata(cpu_dai);
- struct s3c_dma_params *dma_data;
u32 iismod;
pr_debug("Entered %s\n", __func__);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- dma_data = i2s->dma_playback;
- else
- dma_data = i2s->dma_capture;
-
- snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
-
iismod = readl(i2s->regs + S3C2412_IISMOD);
pr_debug("%s: r: IISMOD: %x\n", __func__, iismod);
@@ -169,6 +151,15 @@ static const struct snd_soc_component_driver s3c2412_i2s_component = {
static int s3c2412_iis_dev_probe(struct platform_device *pdev)
{
int ret = 0;
+ struct resource *res;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ s3c2412_i2s.regs = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(s3c2412_i2s.regs))
+ return PTR_ERR(s3c2412_i2s.regs);
+
+ s3c2412_i2s_pcm_stereo_out.dma_addr = res->start + S3C2412_IISTXD;
+ s3c2412_i2s_pcm_stereo_in.dma_addr = res->start + S3C2412_IISRXD;
ret = s3c_i2sv2_register_component(&pdev->dev, -1,
&s3c2412_i2s_component,
diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c
index 9aba9fb7df0e..e87d9a2053b8 100644
--- a/sound/soc/samsung/s3c24xx-i2s.c
+++ b/sound/soc/samsung/s3c24xx-i2s.c
@@ -31,25 +31,15 @@
#include "dma.h"
#include "s3c24xx-i2s.h"
-static struct s3c_dma_client s3c24xx_dma_client_out = {
- .name = "I2S PCM Stereo out"
-};
-
-static struct s3c_dma_client s3c24xx_dma_client_in = {
- .name = "I2S PCM Stereo in"
-};
-
static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = {
- .client = &s3c24xx_dma_client_out,
.channel = DMACH_I2S_OUT,
- .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO,
+ .ch_name = "tx",
.dma_size = 2,
};
static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = {
- .client = &s3c24xx_dma_client_in,
.channel = DMACH_I2S_IN,
- .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO,
+ .ch_name = "rx",
.dma_size = 2,
};
@@ -231,18 +221,12 @@ static int s3c24xx_i2s_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 s3c_dma_params *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
u32 iismod;
pr_debug("Entered %s\n", __func__);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- dma_data = &s3c24xx_i2s_pcm_stereo_out;
- else
- dma_data = &s3c24xx_i2s_pcm_stereo_in;
-
- snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data);
+ dma_data = snd_soc_dai_get_dma_data(dai, substream);
/* Working copies of register */
iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
@@ -251,11 +235,11 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
switch (params_width(params)) {
case 8:
iismod &= ~S3C2410_IISMOD_16BIT;
- dma_data->dma_size = 1;
+ dma_data->addr_width = 1;
break;
case 16:
iismod |= S3C2410_IISMOD_16BIT;
- dma_data->dma_size = 2;
+ dma_data->addr_width = 2;
break;
default:
return -EINVAL;
@@ -270,8 +254,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
int ret = 0;
- struct s3c_dma_params *dma_data =
- snd_soc_dai_get_dma_data(dai, substream);
pr_debug("Entered %s\n", __func__);
@@ -290,7 +272,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
else
s3c24xx_snd_txctrl(1);
- s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
@@ -380,17 +361,15 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai)
{
pr_debug("Entered %s\n", __func__);
- s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100);
- if (s3c24xx_i2s.regs == NULL)
- return -ENXIO;
+ samsung_asoc_init_dma_data(dai, &s3c24xx_i2s_pcm_stereo_out,
+ &s3c24xx_i2s_pcm_stereo_in);
- s3c24xx_i2s.iis_clk = clk_get(dai->dev, "iis");
+ s3c24xx_i2s.iis_clk = devm_clk_get(dai->dev, "iis");
if (IS_ERR(s3c24xx_i2s.iis_clk)) {
pr_err("failed to get iis_clock\n");
- iounmap(s3c24xx_i2s.regs);
return PTR_ERR(s3c24xx_i2s.iis_clk);
}
- clk_enable(s3c24xx_i2s.iis_clk);
+ clk_prepare_enable(s3c24xx_i2s.iis_clk);
/* Configure the I2S pins (GPE0...GPE4) in correct mode */
s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2),
@@ -414,7 +393,7 @@ static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai)
s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
s3c24xx_i2s.iispsr = readl(s3c24xx_i2s.regs + S3C2410_IISPSR);
- clk_disable(s3c24xx_i2s.iis_clk);
+ clk_disable_unprepare(s3c24xx_i2s.iis_clk);
return 0;
}
@@ -422,7 +401,7 @@ static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai)
static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai)
{
pr_debug("Entered %s\n", __func__);
- clk_enable(s3c24xx_i2s.iis_clk);
+ clk_prepare_enable(s3c24xx_i2s.iis_clk);
writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON);
writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
@@ -474,6 +453,19 @@ static const struct snd_soc_component_driver s3c24xx_i2s_component = {
static int s3c24xx_iis_dev_probe(struct platform_device *pdev)
{
int ret = 0;
+ struct resource *res;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!res) {
+ dev_err(&pdev->dev, "Can't get IO resource.\n");
+ return -ENOENT;
+ }
+ s3c24xx_i2s.regs = devm_ioremap_resource(&pdev->dev, res);
+ if (s3c24xx_i2s.regs == NULL)
+ return -ENXIO;
+
+ s3c24xx_i2s_pcm_stereo_out.dma_addr = res->start + S3C2410_IISFIFO;
+ s3c24xx_i2s_pcm_stereo_in.dma_addr = res->start + S3C2410_IISFIFO;
ret = devm_snd_soc_register_component(&pdev->dev,
&s3c24xx_i2s_component, &s3c24xx_i2s_dai, 1);
diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c
index e119aaa91c28..63d079303561 100644
--- a/sound/soc/samsung/smdk_wm8580pcm.c
+++ b/sound/soc/samsung/smdk_wm8580pcm.c
@@ -25,7 +25,7 @@
* o '0' means 'OFF'
* o 'X' means 'Don't care'
*
- * SMDK6410, SMDK6440, SMDK6450 Base B/D: CFG1-0000, CFG2-1111
+ * SMDK6410 Base B/D: CFG1-0000, CFG2-1111
* SMDKC110, SMDKV210: CFGB11-100100, CFGB12-0000
*/
diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c
index 014c177840ba..0acf5d0eed53 100644
--- a/sound/soc/samsung/snow.c
+++ b/sound/soc/samsung/snow.c
@@ -92,6 +92,9 @@ static int snow_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
+ /* Update card-name if provided through DT, else use default name */
+ snd_soc_of_parse_card_name(card, "samsung,model");
+
ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
@@ -103,6 +106,7 @@ static int snow_probe(struct platform_device *pdev)
static const struct of_device_id snow_of_match[] = {
{ .compatible = "google,snow-audio-max98090", },
+ { .compatible = "google,snow-audio-max98091", },
{ .compatible = "google,snow-audio-max98095", },
{},
};
diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c
index d9ffc48fce5e..d7d2e208f486 100644
--- a/sound/soc/samsung/spdif.c
+++ b/sound/soc/samsung/spdif.c
@@ -93,10 +93,6 @@ struct samsung_spdif_info {
struct s3c_dma_params *dma_playback;
};
-static struct s3c_dma_client spdif_dma_client_out = {
- .name = "S/PDIF Stereo out",
-};
-
static struct s3c_dma_params spdif_stereo_out;
static struct samsung_spdif_info spdif_info;
@@ -435,7 +431,6 @@ static int spdif_probe(struct platform_device *pdev)
}
spdif_stereo_out.dma_size = 2;
- spdif_stereo_out.client = &spdif_dma_client_out;
spdif_stereo_out.dma_addr = mem_res->start + DATA_OUTBUF;
spdif_stereo_out.channel = dma_res->start;
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index b43fdf0d08af..80245b6eebd6 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -37,7 +37,7 @@ config SND_SOC_SH4_SIU
config SND_SOC_RCAR
tristate "R-Car series SRU/SCU/SSIU/SSI support"
select SND_SIMPLE_CARD
- select REGMAP
+ select REGMAP_MMIO
help
This option enables R-Car SUR/SCU/SSIU/SSI sound support
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 710a079a7377..c76344350e44 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -232,11 +232,7 @@ struct fsi_stream {
* these are for DMAEngine
*/
struct dma_chan *chan;
- struct work_struct work;
- dma_addr_t dma;
int dma_id;
- int loop_cnt;
- int additional_pos;
};
struct fsi_clk {
@@ -264,12 +260,12 @@ struct fsi_priv {
u32 fmt;
int chan_num:16;
- int clk_master:1;
- int clk_cpg:1;
- int spdif:1;
- int enable_stream:1;
- int bit_clk_inv:1;
- int lr_clk_inv:1;
+ unsigned int clk_master:1;
+ unsigned int clk_cpg:1;
+ unsigned int spdif:1;
+ unsigned int enable_stream:1;
+ unsigned int bit_clk_inv:1;
+ unsigned int lr_clk_inv:1;
};
struct fsi_stream_handler {
@@ -1042,6 +1038,26 @@ static int fsi_clk_set_rate_cpg(struct device *dev,
return ret;
}
+static void fsi_pointer_update(struct fsi_stream *io, int size)
+{
+ io->buff_sample_pos += size;
+
+ if (io->buff_sample_pos >=
+ io->period_samples * (io->period_pos + 1)) {
+ struct snd_pcm_substream *substream = io->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ io->period_pos++;
+
+ if (io->period_pos >= runtime->periods) {
+ io->buff_sample_pos = 0;
+ io->period_pos = 0;
+ }
+
+ snd_pcm_period_elapsed(substream);
+ }
+}
+
/*
* pio data transfer handler
*/
@@ -1108,31 +1124,11 @@ static int fsi_pio_transfer(struct fsi_priv *fsi, struct fsi_stream *io,
void (*run32)(struct fsi_priv *fsi, u8 *buf, int samples),
int samples)
{
- struct snd_pcm_runtime *runtime;
- struct snd_pcm_substream *substream;
u8 *buf;
- int over_period;
if (!fsi_stream_is_working(fsi, io))
return -EINVAL;
- over_period = 0;
- substream = io->substream;
- runtime = substream->runtime;
-
- /* FSI FIFO has limit.
- * So, this driver can not send periods data at a time
- */
- if (io->buff_sample_pos >=
- io->period_samples * (io->period_pos + 1)) {
-
- over_period = 1;
- io->period_pos = (io->period_pos + 1) % runtime->periods;
-
- if (0 == io->period_pos)
- io->buff_sample_pos = 0;
- }
-
buf = fsi_pio_get_area(fsi, io);
switch (io->sample_width) {
@@ -1146,11 +1142,7 @@ static int fsi_pio_transfer(struct fsi_priv *fsi, struct fsi_stream *io,
return -EINVAL;
}
- /* update buff_sample_pos */
- io->buff_sample_pos += samples;
-
- if (over_period)
- snd_pcm_period_elapsed(substream);
+ fsi_pointer_update(io, samples);
return 0;
}
@@ -1279,11 +1271,6 @@ static irqreturn_t fsi_interrupt(int irq, void *data)
*/
static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io)
{
- struct snd_pcm_runtime *runtime = io->substream->runtime;
- struct snd_soc_dai *dai = fsi_get_dai(io->substream);
- enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
- DMA_TO_DEVICE : DMA_FROM_DEVICE;
-
/*
* 24bit data : 24bit bus / package in back
* 16bit data : 16bit bus / stream mode
@@ -1291,107 +1278,48 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io)
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;
}
-static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io)
-{
- struct snd_soc_dai *dai = fsi_get_dai(io->substream);
- enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
- DMA_TO_DEVICE : DMA_FROM_DEVICE;
-
- dma_unmap_single(dai->dev, io->dma,
- snd_pcm_lib_buffer_bytes(io->substream), dir);
- return 0;
-}
-
-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;
-
- if (period >= runtime->periods)
- period = 0;
-
- return io->dma + samples_to_bytes(runtime, period * io->period_samples);
-}
-
static void fsi_dma_complete(void *data)
{
struct fsi_stream *io = (struct fsi_stream *)data;
struct fsi_priv *fsi = fsi_stream_to_priv(io);
- struct snd_pcm_runtime *runtime = io->substream->runtime;
- struct snd_soc_dai *dai = fsi_get_dai(io->substream);
- 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, 0),
- samples_to_bytes(runtime, io->period_samples), dir);
-
- io->buff_sample_pos += io->period_samples;
- io->period_pos++;
-
- if (io->period_pos >= runtime->periods) {
- io->period_pos = 0;
- io->buff_sample_pos = 0;
- }
+ fsi_pointer_update(io, io->period_samples);
fsi_count_fifo_err(fsi);
- fsi_stream_transfer(io);
-
- snd_pcm_period_elapsed(io->substream);
}
-static void fsi_dma_do_work(struct work_struct *work)
+static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io)
{
- struct fsi_stream *io = container_of(work, struct fsi_stream, work);
- struct fsi_priv *fsi = fsi_stream_to_priv(io);
- struct snd_soc_dai *dai;
+ struct snd_soc_dai *dai = fsi_get_dai(io->substream);
+ struct snd_pcm_substream *substream = io->substream;
struct dma_async_tx_descriptor *desc;
- struct snd_pcm_runtime *runtime;
- enum dma_data_direction dir;
int is_play = fsi_stream_is_play(fsi, io);
- int len, i;
- dma_addr_t buf;
-
- if (!fsi_stream_is_working(fsi, io))
- return;
-
- dai = fsi_get_dai(io->substream);
- runtime = io->substream->runtime;
- dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
- len = samples_to_bytes(runtime, io->period_samples);
-
- for (i = 0; i < io->loop_cnt; i++) {
- buf = fsi_dma_get_area(io, io->additional_pos);
-
- 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;
- }
+ enum dma_data_direction dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
+ int ret = -EIO;
+
+ desc = dmaengine_prep_dma_cyclic(io->chan,
+ substream->runtime->dma_addr,
+ snd_pcm_lib_buffer_bytes(substream),
+ snd_pcm_lib_period_bytes(substream),
+ dir,
+ DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+ if (!desc) {
+ dev_err(dai->dev, "dmaengine_prep_dma_cyclic() fail\n");
+ goto fsi_dma_transfer_err;
+ }
- dma_async_issue_pending(io->chan);
+ desc->callback = fsi_dma_complete;
+ desc->callback_param = io;
- io->additional_pos = 1;
+ if (dmaengine_submit(desc) < 0) {
+ dev_err(dai->dev, "tx_submit() fail\n");
+ goto fsi_dma_transfer_err;
}
- io->loop_cnt = 1;
+ dma_async_issue_pending(io->chan);
/*
* FIXME
@@ -1408,13 +1336,11 @@ static void fsi_dma_do_work(struct work_struct *work)
fsi_reg_write(fsi, DIFF_ST, 0);
}
}
-}
-static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io)
-{
- schedule_work(&io->work);
+ ret = 0;
- return 0;
+fsi_dma_transfer_err:
+ return ret;
}
static int fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io,
@@ -1475,15 +1401,11 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct dev
return fsi_stream_probe(fsi, dev);
}
- INIT_WORK(&io->work, fsi_dma_do_work);
-
return 0;
}
static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io)
{
- cancel_work_sync(&io->work);
-
fsi_stream_stop(fsi, io);
if (io->chan)
@@ -1495,7 +1417,6 @@ static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io)
static struct fsi_stream_handler fsi_dma_push_handler = {
.init = fsi_dma_init,
- .quit = fsi_dma_quit,
.probe = fsi_dma_probe,
.transfer = fsi_dma_transfer,
.remove = fsi_dma_remove,
@@ -1657,9 +1578,9 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
if (!ret)
ret = fsi_hw_startup(fsi, io, dai->dev);
if (!ret)
- ret = fsi_stream_transfer(io);
+ ret = fsi_stream_start(fsi, io);
if (!ret)
- fsi_stream_start(fsi, io);
+ ret = fsi_stream_transfer(io);
break;
case SNDRV_PCM_TRIGGER_STOP:
if (!ret)
@@ -1850,16 +1771,10 @@ static void fsi_pcm_free(struct snd_pcm *pcm)
static int fsi_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_pcm *pcm = rtd->pcm;
-
- /*
- * dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
- * in MMAP mode (i.e. aplay -M)
- */
return snd_pcm_lib_preallocate_pages_for_all(
- pcm,
- SNDRV_DMA_TYPE_CONTINUOUS,
- snd_dma_continuous_data(GFP_KERNEL),
+ rtd->pcm,
+ SNDRV_DMA_TYPE_DEV,
+ rtd->card->snd_card->dev,
PREALLOC_BUFFER, PREALLOC_BUFFER_MAX);
}
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 4e86265f625c..19f78963e8b9 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -138,6 +138,17 @@ char *rsnd_mod_name(struct rsnd_mod *mod)
return mod->ops->name;
}
+char *rsnd_mod_dma_name(struct rsnd_mod *mod)
+{
+ if (!mod || !mod->ops)
+ return "unknown";
+
+ if (!mod->ops->dma_name)
+ return mod->ops->name;
+
+ return mod->ops->dma_name(mod);
+}
+
void rsnd_mod_init(struct rsnd_priv *priv,
struct rsnd_mod *mod,
struct rsnd_mod_ops *ops,
@@ -153,26 +164,8 @@ void rsnd_mod_init(struct rsnd_priv *priv,
/*
* rsnd_dma functions
*/
-static void __rsnd_dma_start(struct rsnd_dma *dma);
-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->offset = 0;
- dma->submit_loop = 2;
- __rsnd_dma_start(dma);
-}
-
void rsnd_dma_stop(struct rsnd_dma *dma)
{
- dma->submit_loop = 0;
- cancel_work_sync(&dma->work);
dmaengine_terminate_all(dma->chan);
}
@@ -180,11 +173,7 @@ static void rsnd_dma_complete(void *data)
{
struct rsnd_dma *dma = (struct rsnd_dma *)data;
struct rsnd_mod *mod = rsnd_dma_to_mod(dma);
- struct rsnd_priv *priv = rsnd_mod_to_priv(rsnd_dma_to_mod(dma));
struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
- unsigned long flags;
-
- rsnd_lock(priv, flags);
/*
* Renesas sound Gen1 needs 1 DMAC,
@@ -197,57 +186,41 @@ static void rsnd_dma_complete(void *data)
* rsnd_dai_pointer_update() will be called twice,
* ant it will breaks io->byte_pos
*/
- if (dma->submit_loop)
- rsnd_dma_continue(dma);
-
- rsnd_unlock(priv, flags);
rsnd_dai_pointer_update(io, io->byte_per_period);
}
-static void __rsnd_dma_start(struct rsnd_dma *dma)
+void rsnd_dma_start(struct rsnd_dma *dma)
{
struct rsnd_mod *mod = rsnd_dma_to_mod(dma);
struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
- struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+ struct snd_pcm_substream *substream = io->substream;
struct device *dev = rsnd_priv_to_dev(priv);
struct dma_async_tx_descriptor *desc;
- dma_addr_t buf;
- size_t len = io->byte_per_period;
- int i;
- for (i = 0; i < dma->submit_loop; i++) {
-
- buf = runtime->dma_addr +
- rsnd_dai_pointer_offset(io, dma->offset + len);
- dma->offset = len;
-
- 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 = dmaengine_prep_dma_cyclic(dma->chan,
+ (dma->addr) ? dma->addr :
+ substream->runtime->dma_addr,
+ snd_pcm_lib_buffer_bytes(substream),
+ snd_pcm_lib_period_bytes(substream),
+ dma->dir,
+ DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
- desc->callback = rsnd_dma_complete;
- desc->callback_param = dma;
+ if (!desc) {
+ dev_err(dev, "dmaengine_prep_slave_sg() fail\n");
+ return;
+ }
- if (dmaengine_submit(desc) < 0) {
- dev_err(dev, "dmaengine_submit() fail\n");
- return;
- }
+ desc->callback = rsnd_dma_complete;
+ desc->callback_param = dma;
- dma_async_issue_pending(dma->chan);
+ if (dmaengine_submit(desc) < 0) {
+ dev_err(dev, "dmaengine_submit() fail\n");
+ return;
}
-}
-
-static void rsnd_dma_do_work(struct work_struct *work)
-{
- struct rsnd_dma *dma = container_of(work, struct rsnd_dma, work);
- __rsnd_dma_start(dma);
+ dma_async_issue_pending(dma->chan);
}
int rsnd_dma_available(struct rsnd_dma *dma)
@@ -261,14 +234,27 @@ static int _rsnd_dma_of_name(char *dma_name, struct rsnd_mod *mod)
{
if (mod)
return snprintf(dma_name, DMA_NAME_SIZE / 2, "%s%d",
- rsnd_mod_name(mod), rsnd_mod_id(mod));
+ rsnd_mod_dma_name(mod), rsnd_mod_id(mod));
else
return snprintf(dma_name, DMA_NAME_SIZE / 2, "mem");
}
-static void rsnd_dma_of_name(struct rsnd_dma *dma,
- int is_play, char *dma_name)
+static void rsnd_dma_of_name(struct rsnd_mod *mod_from,
+ struct rsnd_mod *mod_to,
+ char *dma_name)
+{
+ int index = 0;
+
+ index = _rsnd_dma_of_name(dma_name + index, mod_from);
+ *(dma_name + index++) = '_';
+ index = _rsnd_dma_of_name(dma_name + index, mod_to);
+}
+
+static void rsnd_dma_of_path(struct rsnd_dma *dma,
+ int is_play,
+ struct rsnd_mod **mod_from,
+ struct rsnd_mod **mod_to)
{
struct rsnd_mod *this = rsnd_dma_to_mod(dma);
struct rsnd_dai_stream *io = rsnd_mod_to_io(this);
@@ -276,7 +262,6 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma,
struct rsnd_mod *src = rsnd_io_to_mod_src(io);
struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io);
struct rsnd_mod *mod[MOD_MAX];
- struct rsnd_mod *src_mod, *dst_mod;
int i, index;
@@ -297,31 +282,34 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma,
for (i = 1; i < MOD_MAX; i++) {
if (!src) {
mod[i] = ssi;
- break;
} else if (!dvc) {
mod[i] = src;
src = NULL;
} else {
- mod[i] = dvc;
+ if ((!is_play) && (this == src))
+ this = dvc;
+
+ mod[i] = (is_play) ? src : dvc;
+ i++;
+ mod[i] = (is_play) ? dvc : src;
+ src = NULL;
dvc = NULL;
}
if (mod[i] == this)
index = i;
+
+ if (mod[i] == ssi)
+ break;
}
if (is_play) {
- src_mod = mod[index - 1];
- dst_mod = mod[index];
+ *mod_from = mod[index - 1];
+ *mod_to = mod[index];
} else {
- src_mod = mod[index];
- dst_mod = mod[index - 1];
+ *mod_from = mod[index];
+ *mod_to = mod[index - 1];
}
-
- index = 0;
- index = _rsnd_dma_of_name(dma_name + index, src_mod);
- *(dma_name + index++) = '_';
- index = _rsnd_dma_of_name(dma_name + index, dst_mod);
}
int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
@@ -329,6 +317,8 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
{
struct device *dev = rsnd_priv_to_dev(priv);
struct dma_slave_config cfg;
+ struct rsnd_mod *mod_from;
+ struct rsnd_mod *mod_to;
char dma_name[DMA_NAME_SIZE];
dma_cap_mask_t mask;
int ret;
@@ -341,13 +331,18 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
dma_cap_zero(mask);
dma_cap_set(DMA_SLAVE, mask);
- if (dev->of_node)
- rsnd_dma_of_name(dma, is_play, dma_name);
- else
- snprintf(dma_name, DMA_NAME_SIZE,
- is_play ? "tx" : "rx");
+ rsnd_dma_of_path(dma, is_play, &mod_from, &mod_to);
+ rsnd_dma_of_name(mod_from, mod_to, dma_name);
- dev_dbg(dev, "dma name : %s\n", dma_name);
+ cfg.slave_id = id;
+ cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM;
+ cfg.src_addr = rsnd_gen_dma_addr(priv, mod_from, is_play, 1);
+ cfg.dst_addr = rsnd_gen_dma_addr(priv, mod_to, is_play, 0);
+ cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+
+ dev_dbg(dev, "dma : %s %pad -> %pad\n",
+ dma_name, &cfg.src_addr, &cfg.dst_addr);
dma->chan = dma_request_slave_channel_compat(mask, shdma_chan_filter,
(void *)id, dev,
@@ -357,14 +352,12 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
return -EIO;
}
- rsnd_gen_dma_addr(priv, dma, &cfg, is_play, id);
-
ret = dmaengine_slave_config(dma->chan, &cfg);
if (ret < 0)
goto rsnd_dma_init_err;
- dma->dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
- INIT_WORK(&dma->work, rsnd_dma_do_work);
+ dma->addr = is_play ? cfg.src_addr : cfg.dst_addr;
+ dma->dir = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM;
return 0;
@@ -631,40 +624,41 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
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;
+ rdai->frm_clk_inv = 0;
break;
case SND_SOC_DAIFMT_LEFT_J:
rdai->sys_delay = 1;
rdai->data_alignment = 0;
+ rdai->frm_clk_inv = 1;
break;
case SND_SOC_DAIFMT_RIGHT_J:
rdai->sys_delay = 1;
rdai->data_alignment = 1;
+ rdai->frm_clk_inv = 1;
+ break;
+ }
+
+ /* set clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_IF:
+ rdai->bit_clk_inv = rdai->bit_clk_inv;
+ rdai->frm_clk_inv = !rdai->frm_clk_inv;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ rdai->bit_clk_inv = !rdai->bit_clk_inv;
+ rdai->frm_clk_inv = rdai->frm_clk_inv;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ rdai->bit_clk_inv = !rdai->bit_clk_inv;
+ rdai->frm_clk_inv = !rdai->frm_clk_inv;
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ default:
break;
}
@@ -734,12 +728,13 @@ static void rsnd_of_parse_dai(struct platform_device *pdev,
struct device_node *dai_node, *dai_np;
struct device_node *ssi_node, *ssi_np;
struct device_node *src_node, *src_np;
+ struct device_node *dvc_node, *dvc_np;
struct device_node *playback, *capture;
struct rsnd_dai_platform_info *dai_info;
struct rcar_snd_info *info = rsnd_priv_to_info(priv);
struct device *dev = &pdev->dev;
int nr, i;
- int dai_i, ssi_i, src_i;
+ int dai_i, ssi_i, src_i, dvc_i;
if (!of_data)
return;
@@ -765,6 +760,7 @@ static void rsnd_of_parse_dai(struct platform_device *pdev,
ssi_node = of_get_child_by_name(dev->of_node, "rcar_sound,ssi");
src_node = of_get_child_by_name(dev->of_node, "rcar_sound,src");
+ dvc_node = of_get_child_by_name(dev->of_node, "rcar_sound,dvc");
#define mod_parse(name) \
if (name##_node) { \
@@ -800,6 +796,7 @@ if (name##_node) { \
mod_parse(ssi);
mod_parse(src);
+ mod_parse(dvc);
if (playback)
of_node_put(playback);
@@ -948,19 +945,17 @@ static struct snd_pcm_ops rsnd_pcm_ops = {
static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
- struct rsnd_priv *priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
- struct rsnd_dai *rdai;
- int i, ret;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
+ int ret;
- for_each_rsnd_dai(rdai, priv, i) {
- ret = rsnd_dai_call(pcm_new, &rdai->playback, rdai, rtd);
- if (ret)
- return ret;
+ ret = rsnd_dai_call(pcm_new, &rdai->playback, rdai, rtd);
+ if (ret)
+ return ret;
- ret = rsnd_dai_call(pcm_new, &rdai->capture, rdai, rtd);
- if (ret)
- return ret;
- }
+ ret = rsnd_dai_call(pcm_new, &rdai->capture, rdai, rtd);
+ if (ret)
+ return ret;
return snd_pcm_lib_preallocate_pages_for_all(
rtd->pcm,
@@ -1047,11 +1042,11 @@ static int rsnd_probe(struct platform_device *pdev)
for_each_rsnd_dai(rdai, priv, i) {
ret = rsnd_dai_call(probe, &rdai->playback, rdai);
if (ret)
- return ret;
+ goto exit_snd_probe;
ret = rsnd_dai_call(probe, &rdai->capture, rdai);
if (ret)
- return ret;
+ goto exit_snd_probe;
}
/*
@@ -1079,6 +1074,11 @@ static int rsnd_probe(struct platform_device *pdev)
exit_snd_soc:
snd_soc_unregister_platform(dev);
+exit_snd_probe:
+ for_each_rsnd_dai(rdai, priv, i) {
+ rsnd_dai_call(remove, &rdai->playback, rdai);
+ rsnd_dai_call(remove, &rdai->capture, rdai);
+ }
return ret;
}
@@ -1087,21 +1087,16 @@ static int rsnd_remove(struct platform_device *pdev)
{
struct rsnd_priv *priv = dev_get_drvdata(&pdev->dev);
struct rsnd_dai *rdai;
- int ret, i;
+ int ret = 0, i;
pm_runtime_disable(&pdev->dev);
for_each_rsnd_dai(rdai, priv, i) {
- ret = rsnd_dai_call(remove, &rdai->playback, rdai);
- if (ret)
- return ret;
-
- ret = rsnd_dai_call(remove, &rdai->capture, rdai);
- if (ret)
- return ret;
+ ret |= rsnd_dai_call(remove, &rdai->playback, rdai);
+ ret |= rsnd_dai_call(remove, &rdai->capture, rdai);
}
- return 0;
+ return ret;
}
static struct platform_driver rsnd_driver = {
diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c
index ed0007006899..3f443930c2b1 100644
--- a/sound/soc/sh/rcar/dvc.c
+++ b/sound/soc/sh/rcar/dvc.c
@@ -20,7 +20,8 @@ struct rsnd_dvc {
struct rsnd_dvc_platform_info *info; /* rcar_snd.h */
struct rsnd_mod mod;
struct clk *clk;
- long volume[RSND_DVC_VOLUME_NUM];
+ u8 volume[RSND_DVC_VOLUME_NUM];
+ u8 mute[RSND_DVC_VOLUME_NUM];
};
#define rsnd_mod_to_dvc(_mod) \
@@ -37,13 +38,18 @@ static void rsnd_dvc_volume_update(struct rsnd_mod *mod)
struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod);
u32 max = (0x00800000 - 1);
u32 vol[RSND_DVC_VOLUME_NUM];
+ u32 mute = 0;
int i;
- for (i = 0; i < RSND_DVC_VOLUME_NUM; i++)
+ for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) {
vol[i] = max / RSND_DVC_VOLUME_MAX * dvc->volume[i];
+ mute |= (!!dvc->mute[i]) << i;
+ }
rsnd_mod_write(mod, DVC_VOL0R, vol[0]);
rsnd_mod_write(mod, DVC_VOL1R, vol[1]);
+
+ rsnd_mod_write(mod, DVC_ZCMCR, mute);
}
static int rsnd_dvc_probe_gen2(struct rsnd_mod *mod,
@@ -96,8 +102,8 @@ static int rsnd_dvc_init(struct rsnd_mod *dvc_mod,
rsnd_mod_write(dvc_mod, DVC_ADINR, rsnd_get_adinr(dvc_mod));
- /* enable Volume */
- rsnd_mod_write(dvc_mod, DVC_DVUCR, 0x100);
+ /* enable Volume / Mute */
+ rsnd_mod_write(dvc_mod, DVC_DVUCR, 0x101);
/* ch0/ch1 Volume */
rsnd_dvc_volume_update(dvc_mod);
@@ -140,10 +146,20 @@ static int rsnd_dvc_stop(struct rsnd_mod *mod,
static int rsnd_dvc_volume_info(struct snd_kcontrol *kctrl,
struct snd_ctl_elem_info *uinfo)
{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ struct rsnd_mod *mod = snd_kcontrol_chip(kctrl);
+ struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod);
+ u8 *val = (u8 *)kctrl->private_value;
+
uinfo->count = RSND_DVC_VOLUME_NUM;
uinfo->value.integer.min = 0;
- uinfo->value.integer.max = RSND_DVC_VOLUME_MAX;
+
+ if (val == dvc->volume) {
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->value.integer.max = RSND_DVC_VOLUME_MAX;
+ } else {
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->value.integer.max = 1;
+ }
return 0;
}
@@ -151,12 +167,11 @@ static int rsnd_dvc_volume_info(struct snd_kcontrol *kctrl,
static int rsnd_dvc_volume_get(struct snd_kcontrol *kctrl,
struct snd_ctl_elem_value *ucontrol)
{
- struct rsnd_mod *mod = snd_kcontrol_chip(kctrl);
- struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod);
+ u8 *val = (u8 *)kctrl->private_value;
int i;
for (i = 0; i < RSND_DVC_VOLUME_NUM; i++)
- ucontrol->value.integer.value[i] = dvc->volume[i];
+ ucontrol->value.integer.value[i] = val[i];
return 0;
}
@@ -165,51 +180,38 @@ static int rsnd_dvc_volume_put(struct snd_kcontrol *kctrl,
struct snd_ctl_elem_value *ucontrol)
{
struct rsnd_mod *mod = snd_kcontrol_chip(kctrl);
- struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod);
+ u8 *val = (u8 *)kctrl->private_value;
int i, change = 0;
for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) {
- if (ucontrol->value.integer.value[i] < 0 ||
- ucontrol->value.integer.value[i] > RSND_DVC_VOLUME_MAX)
- return -EINVAL;
-
- change |= (ucontrol->value.integer.value[i] != dvc->volume[i]);
+ change |= (ucontrol->value.integer.value[i] != val[i]);
+ val[i] = ucontrol->value.integer.value[i];
}
- if (change) {
- for (i = 0; i < RSND_DVC_VOLUME_NUM; i++)
- dvc->volume[i] = ucontrol->value.integer.value[i];
-
+ if (change)
rsnd_dvc_volume_update(mod);
- }
return change;
}
-static int rsnd_dvc_pcm_new(struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct snd_soc_pcm_runtime *rtd)
+static int __rsnd_dvc_pcm_new(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct snd_soc_pcm_runtime *rtd,
+ const unsigned char *name,
+ u8 *private)
{
- struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
- struct device *dev = rsnd_priv_to_dev(priv);
struct snd_card *card = rtd->card->snd_card;
struct snd_kcontrol *kctrl;
- static struct snd_kcontrol_new knew = {
+ struct snd_kcontrol_new knew = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Playback Volume",
+ .name = name,
.info = rsnd_dvc_volume_info,
.get = rsnd_dvc_volume_get,
.put = rsnd_dvc_volume_put,
+ .private_value = (unsigned long)private,
};
int ret;
- if (!rsnd_dai_is_play(rdai, io)) {
- dev_err(dev, "DVC%d is connected to Capture DAI\n",
- rsnd_mod_id(mod));
- return -EINVAL;
- }
-
kctrl = snd_ctl_new1(&knew, mod);
if (!kctrl)
return -ENOMEM;
@@ -221,6 +223,33 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod,
return 0;
}
+static int rsnd_dvc_pcm_new(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct snd_soc_pcm_runtime *rtd)
+{
+ struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
+ struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod);
+ int ret;
+
+ /* Volume */
+ ret = __rsnd_dvc_pcm_new(mod, rdai, rtd,
+ rsnd_dai_is_play(rdai, io) ?
+ "DVC Out Playback Volume" : "DVC In Capture Volume",
+ dvc->volume);
+ if (ret < 0)
+ return ret;
+
+ /* Mute */
+ ret = __rsnd_dvc_pcm_new(mod, rdai, rtd,
+ rsnd_dai_is_play(rdai, io) ?
+ "DVC Out Mute Switch" : "DVC In Mute Switch",
+ dvc->mute);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
static struct rsnd_mod_ops rsnd_dvc_ops = {
.name = DVC_NAME,
.probe = rsnd_dvc_probe_gen2,
@@ -239,6 +268,42 @@ struct rsnd_mod *rsnd_dvc_mod_get(struct rsnd_priv *priv, int id)
return &((struct rsnd_dvc *)(priv->dvc) + id)->mod;
}
+static void rsnd_of_parse_dvc(struct platform_device *pdev,
+ const struct rsnd_of_data *of_data,
+ struct rsnd_priv *priv)
+{
+ struct device_node *node;
+ struct rsnd_dvc_platform_info *dvc_info;
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+ struct device *dev = &pdev->dev;
+ int nr;
+
+ if (!of_data)
+ return;
+
+ node = of_get_child_by_name(dev->of_node, "rcar_sound,dvc");
+ if (!node)
+ return;
+
+ nr = of_get_child_count(node);
+ if (!nr)
+ goto rsnd_of_parse_dvc_end;
+
+ dvc_info = devm_kzalloc(dev,
+ sizeof(struct rsnd_dvc_platform_info) * nr,
+ GFP_KERNEL);
+ if (!dvc_info) {
+ dev_err(dev, "dvc info allocation error\n");
+ goto rsnd_of_parse_dvc_end;
+ }
+
+ info->dvc_info = dvc_info;
+ info->dvc_info_nr = nr;
+
+rsnd_of_parse_dvc_end:
+ of_node_put(node);
+}
+
int rsnd_dvc_probe(struct platform_device *pdev,
const struct rsnd_of_data *of_data,
struct rsnd_priv *priv)
@@ -250,6 +315,8 @@ int rsnd_dvc_probe(struct platform_device *pdev,
char name[RSND_DVC_NAME_SIZE];
int i, nr;
+ rsnd_of_parse_dvc(pdev, of_data, priv);
+
nr = info->dvc_info_nr;
if (!nr)
return 0;
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 1dd2b7d38c2c..3fdf3be7b99a 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -15,63 +15,35 @@ struct rsnd_gen {
struct rsnd_gen_ops *ops;
- struct regmap *regmap;
+ struct regmap *regmap[RSND_BASE_MAX];
struct regmap_field *regs[RSND_REG_MAX];
};
#define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen)
-#define RSND_REG_SET(gen, id, reg_id, offset, _id_offset, _id_size) \
- [id] = { \
- .reg = (unsigned int)gen->base[reg_id] + offset, \
- .lsb = 0, \
- .msb = 31, \
- .id_size = _id_size, \
- .id_offset = _id_offset, \
- }
-
-/*
- * basic function
- */
-static int rsnd_regmap_write32(void *context, const void *_data, size_t count)
-{
- struct rsnd_priv *priv = context;
- struct device *dev = rsnd_priv_to_dev(priv);
- u32 *data = (u32 *)_data;
- u32 val = data[1];
- void __iomem *reg = (void *)data[0];
-
- iowrite32(val, reg);
-
- dev_dbg(dev, "w %p : %08x\n", reg, val);
-
- return 0;
-}
-
-static int rsnd_regmap_read32(void *context,
- const void *_data, size_t reg_size,
- void *_val, size_t val_size)
-{
- struct rsnd_priv *priv = context;
- struct device *dev = rsnd_priv_to_dev(priv);
- u32 *data = (u32 *)_data;
- u32 *val = (u32 *)_val;
- void __iomem *reg = (void *)data[0];
-
- *val = ioread32(reg);
-
- dev_dbg(dev, "r %p : %08x\n", reg, *val);
+struct rsnd_regmap_field_conf {
+ int idx;
+ unsigned int reg_offset;
+ unsigned int id_offset;
+};
- return 0;
+#define RSND_REG_SET(id, offset, _id_offset) \
+{ \
+ .idx = id, \
+ .reg_offset = offset, \
+ .id_offset = _id_offset, \
}
+/* single address mapping */
+#define RSND_GEN_S_REG(id, offset) \
+ RSND_REG_SET(RSND_REG_##id, offset, 0)
-static struct regmap_bus rsnd_regmap_bus = {
- .write = rsnd_regmap_write32,
- .read = rsnd_regmap_read32,
- .reg_format_endian_default = REGMAP_ENDIAN_NATIVE,
- .val_format_endian_default = REGMAP_ENDIAN_NATIVE,
-};
+/* multi address mapping */
+#define RSND_GEN_M_REG(id, offset, _id_offset) \
+ RSND_REG_SET(RSND_REG_##id, offset, _id_offset)
+/*
+ * basic function
+ */
static int rsnd_is_accessible_reg(struct rsnd_priv *priv,
struct rsnd_gen *gen, enum rsnd_reg reg)
{
@@ -88,6 +60,7 @@ static int rsnd_is_accessible_reg(struct rsnd_priv *priv,
u32 rsnd_read(struct rsnd_priv *priv,
struct rsnd_mod *mod, enum rsnd_reg reg)
{
+ struct device *dev = rsnd_priv_to_dev(priv);
struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
u32 val;
@@ -96,6 +69,8 @@ u32 rsnd_read(struct rsnd_priv *priv,
regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val);
+ dev_dbg(dev, "r %s - 0x%04d : %08x\n", rsnd_mod_name(mod), reg, val);
+
return val;
}
@@ -103,17 +78,21 @@ void rsnd_write(struct rsnd_priv *priv,
struct rsnd_mod *mod,
enum rsnd_reg reg, u32 data)
{
+ struct device *dev = rsnd_priv_to_dev(priv);
struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
if (!rsnd_is_accessible_reg(priv, gen, reg))
return;
regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data);
+
+ dev_dbg(dev, "w %s - 0x%04d : %08x\n", rsnd_mod_name(mod), reg, data);
}
void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod,
enum rsnd_reg reg, u32 mask, u32 data)
{
+ struct device *dev = rsnd_priv_to_dev(priv);
struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
if (!rsnd_is_accessible_reg(priv, gen, reg))
@@ -121,35 +100,63 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod,
regmap_fields_update_bits(gen->regs[reg], rsnd_mod_id(mod),
mask, data);
+
+ dev_dbg(dev, "b %s - 0x%04d : %08x/%08x\n",
+ rsnd_mod_name(mod), reg, data, mask);
}
-static int rsnd_gen_regmap_init(struct rsnd_priv *priv,
- struct rsnd_gen *gen,
- struct reg_field *regf)
+#define rsnd_gen_regmap_init(priv, id_size, reg_id, conf) \
+ _rsnd_gen_regmap_init(priv, id_size, reg_id, conf, ARRAY_SIZE(conf))
+static int _rsnd_gen_regmap_init(struct rsnd_priv *priv,
+ int id_size,
+ int reg_id,
+ struct rsnd_regmap_field_conf *conf,
+ int conf_size)
{
- int i;
+ struct platform_device *pdev = rsnd_priv_to_pdev(priv);
+ struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
struct device *dev = rsnd_priv_to_dev(priv);
+ struct resource *res;
struct regmap_config regc;
+ struct regmap_field *regs;
+ struct regmap *regmap;
+ struct reg_field regf;
+ void __iomem *base;
+ int i;
memset(&regc, 0, sizeof(regc));
regc.reg_bits = 32;
regc.val_bits = 32;
+ regc.reg_stride = 4;
- gen->regmap = devm_regmap_init(dev, &rsnd_regmap_bus, priv, &regc);
- if (IS_ERR(gen->regmap)) {
- dev_err(dev, "regmap error %ld\n", PTR_ERR(gen->regmap));
- return PTR_ERR(gen->regmap);
- }
+ res = platform_get_resource(pdev, IORESOURCE_MEM, reg_id);
+ if (!res)
+ return -ENODEV;
- for (i = 0; i < RSND_REG_MAX; i++) {
- gen->regs[i] = NULL;
- if (!regf[i].reg)
- continue;
+ base = devm_ioremap_resource(dev, res);
+ if (IS_ERR(base))
+ return PTR_ERR(base);
- gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]);
- if (IS_ERR(gen->regs[i]))
- return PTR_ERR(gen->regs[i]);
+ regmap = devm_regmap_init_mmio(dev, base, &regc);
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+ gen->base[reg_id] = base;
+ gen->regmap[reg_id] = regmap;
+
+ for (i = 0; i < conf_size; i++) {
+
+ regf.reg = conf[i].reg_offset;
+ regf.id_offset = conf[i].id_offset;
+ regf.lsb = 0;
+ regf.msb = 31;
+ regf.id_size = id_size;
+
+ regs = devm_regmap_field_alloc(dev, regmap, regf);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ gen->regs[conf[i].idx] = regs;
}
return 0;
@@ -165,15 +172,19 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv,
*
* ex) R-Car H2 case
* mod / DMAC in / DMAC out / DMAC PP in / DMAC pp out
- * SSI : 0xec541000 / 0xec241008 / 0xec24100c / 0xec400000 / 0xec400000
+ * SSI : 0xec541000 / 0xec241008 / 0xec24100c
+ * SSIU: 0xec541000 / 0xec100000 / 0xec100000 / 0xec400000 / 0xec400000
* SCU : 0xec500000 / 0xec000000 / 0xec004000 / 0xec300000 / 0xec304000
- * CMD : 0xec500000 / 0xec008000 0xec308000
+ * CMD : 0xec500000 / / 0xec008000 0xec308000
*/
#define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8)
#define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc)
-#define RDMA_SSI_I_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i))
-#define RDMA_SSI_O_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i))
+#define RDMA_SSIU_I_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i))
+#define RDMA_SSIU_O_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i))
+
+#define RDMA_SSIU_I_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i))
+#define RDMA_SSIU_O_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i))
#define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i))
#define RDMA_SRC_O_N(addr, i) (addr ##_reg - 0x004fc000 + (0x400 * i))
@@ -184,14 +195,13 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv,
#define RDMA_CMD_O_N(addr, i) (addr ##_reg - 0x004f8000 + (0x400 * i))
#define RDMA_CMD_O_P(addr, i) (addr ##_reg - 0x001f8000 + (0x400 * i))
-void rsnd_gen_dma_addr(struct rsnd_priv *priv,
- struct rsnd_dma *dma,
- struct dma_slave_config *cfg,
- int is_play, int slave_id)
+static dma_addr_t
+rsnd_gen2_dma_addr(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ int is_play, int is_from)
{
struct platform_device *pdev = rsnd_priv_to_pdev(priv);
struct device *dev = rsnd_priv_to_dev(priv);
- struct rsnd_mod *mod = rsnd_dma_to_mod(dma);
struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
dma_addr_t ssi_reg = platform_get_resource(pdev,
IORESOURCE_MEM, RSND_GEN2_SSI)->start;
@@ -202,170 +212,152 @@ void rsnd_gen_dma_addr(struct rsnd_priv *priv,
int use_dvc = !!rsnd_io_to_mod_dvc(io);
int id = rsnd_mod_id(mod);
struct dma_addr {
- dma_addr_t src_addr;
- dma_addr_t dst_addr;
- } dma_addrs[2][2][3] = {
- { /* SRC */
- /* Capture */
- {{ 0, 0 },
- { RDMA_SRC_O_N(src, id), 0 },
- { RDMA_CMD_O_N(src, id), 0 }},
- /* Playback */
- {{ 0, 0, },
- { 0, RDMA_SRC_I_N(src, id) },
- { 0, RDMA_SRC_I_N(src, id) }}
- }, { /* SSI */
- /* Capture */
- {{ RDMA_SSI_O_N(ssi, id), 0 },
- { RDMA_SSI_O_P(ssi, id), RDMA_SRC_I_P(src, id) },
- { RDMA_SSI_O_P(ssi, id), RDMA_SRC_I_P(src, id) }},
- /* Playback */
- {{ 0, RDMA_SSI_I_N(ssi, id) },
- { RDMA_SRC_O_P(src, id), RDMA_SSI_I_P(ssi, id) },
- { RDMA_CMD_O_P(src, id), RDMA_SSI_I_P(ssi, id) }}
- }
+ dma_addr_t out_addr;
+ dma_addr_t in_addr;
+ } dma_addrs[3][2][3] = {
+ /* SRC */
+ {{{ 0, 0 },
+ /* Capture */
+ { RDMA_SRC_O_N(src, id), RDMA_SRC_I_P(src, id) },
+ { RDMA_CMD_O_N(src, id), RDMA_SRC_I_P(src, id) } },
+ /* Playback */
+ {{ 0, 0, },
+ { RDMA_SRC_O_P(src, id), RDMA_SRC_I_N(src, id) },
+ { RDMA_CMD_O_P(src, id), RDMA_SRC_I_N(src, id) } }
+ },
+ /* SSI */
+ /* Capture */
+ {{{ RDMA_SSI_O_N(ssi, id), 0 },
+ { RDMA_SSIU_O_P(ssi, id), 0 },
+ { RDMA_SSIU_O_P(ssi, id), 0 } },
+ /* Playback */
+ {{ 0, RDMA_SSI_I_N(ssi, id) },
+ { 0, RDMA_SSIU_I_P(ssi, id) },
+ { 0, RDMA_SSIU_I_P(ssi, id) } }
+ },
+ /* SSIU */
+ /* Capture */
+ {{{ RDMA_SSIU_O_N(ssi, id), 0 },
+ { RDMA_SSIU_O_P(ssi, id), 0 },
+ { RDMA_SSIU_O_P(ssi, id), 0 } },
+ /* Playback */
+ {{ 0, RDMA_SSIU_I_N(ssi, id) },
+ { 0, RDMA_SSIU_I_P(ssi, id) },
+ { 0, RDMA_SSIU_I_P(ssi, id) } } },
};
- cfg->slave_id = slave_id;
- cfg->src_addr = 0;
- cfg->dst_addr = 0;
- cfg->direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM;
+ /* it shouldn't happen */
+ if (use_dvc & !use_src)
+ dev_err(dev, "DVC is selected without SRC\n");
+
+ /* use SSIU or SSI ? */
+ if (is_ssi && (0 == strcmp(rsnd_mod_dma_name(mod), "ssiu")))
+ is_ssi++;
+
+ return (is_from) ?
+ dma_addrs[is_ssi][is_play][use_src + use_dvc].out_addr :
+ dma_addrs[is_ssi][is_play][use_src + use_dvc].in_addr;
+}
+dma_addr_t rsnd_gen_dma_addr(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ int is_play, int is_from)
+{
/*
* gen1 uses default DMA addr
*/
if (rsnd_is_gen1(priv))
- return;
-
- /* it shouldn't happen */
- if (use_dvc & !use_src) {
- dev_err(dev, "DVC is selected without SRC\n");
- return;
- }
+ return 0;
- cfg->src_addr = dma_addrs[is_ssi][is_play][use_src + use_dvc].src_addr;
- cfg->dst_addr = dma_addrs[is_ssi][is_play][use_src + use_dvc].dst_addr;
+ if (!mod)
+ return 0;
- dev_dbg(dev, "dma%d addr - src : %x / dst : %x\n",
- id, cfg->src_addr, cfg->dst_addr);
+ return rsnd_gen2_dma_addr(priv, mod, is_play, is_from);
}
/*
* Gen2
*/
-
-/* single address mapping */
-#define RSND_GEN2_S_REG(gen, reg, id, offset) \
- RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, 0, 10)
-
-/* multi address mapping */
-#define RSND_GEN2_M_REG(gen, reg, id, offset, _id_offset) \
- RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, _id_offset, 10)
-
-static int rsnd_gen2_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
-{
- struct reg_field regf[RSND_REG_MAX] = {
- RSND_GEN2_S_REG(gen, SSIU, SSI_MODE0, 0x800),
- RSND_GEN2_S_REG(gen, SSIU, SSI_MODE1, 0x804),
- /* FIXME: it needs SSI_MODE2/3 in the future */
- RSND_GEN2_M_REG(gen, SSIU, SSI_BUSIF_MODE, 0x0, 0x80),
- RSND_GEN2_M_REG(gen, SSIU, SSI_BUSIF_ADINR,0x4, 0x80),
- RSND_GEN2_M_REG(gen, SSIU, SSI_CTRL, 0x10, 0x80),
- RSND_GEN2_M_REG(gen, SSIU, INT_ENABLE, 0x18, 0x80),
-
- RSND_GEN2_M_REG(gen, SCU, SRC_BUSIF_MODE, 0x0, 0x20),
- RSND_GEN2_M_REG(gen, SCU, SRC_ROUTE_MODE0,0xc, 0x20),
- RSND_GEN2_M_REG(gen, SCU, SRC_CTRL, 0x10, 0x20),
- RSND_GEN2_M_REG(gen, SCU, CMD_ROUTE_SLCT, 0x18c, 0x20),
- RSND_GEN2_M_REG(gen, SCU, CMD_CTRL, 0x190, 0x20),
- RSND_GEN2_M_REG(gen, SCU, SRC_SWRSR, 0x200, 0x40),
- RSND_GEN2_M_REG(gen, SCU, SRC_SRCIR, 0x204, 0x40),
- RSND_GEN2_M_REG(gen, SCU, SRC_ADINR, 0x214, 0x40),
- RSND_GEN2_M_REG(gen, SCU, SRC_IFSCR, 0x21c, 0x40),
- RSND_GEN2_M_REG(gen, SCU, SRC_IFSVR, 0x220, 0x40),
- RSND_GEN2_M_REG(gen, SCU, SRC_SRCCR, 0x224, 0x40),
- RSND_GEN2_M_REG(gen, SCU, SRC_BSDSR, 0x22c, 0x40),
- RSND_GEN2_M_REG(gen, SCU, SRC_BSISR, 0x238, 0x40),
- RSND_GEN2_M_REG(gen, SCU, DVC_SWRSR, 0xe00, 0x100),
- RSND_GEN2_M_REG(gen, SCU, DVC_DVUIR, 0xe04, 0x100),
- RSND_GEN2_M_REG(gen, SCU, DVC_ADINR, 0xe08, 0x100),
- RSND_GEN2_M_REG(gen, SCU, DVC_DVUCR, 0xe10, 0x100),
- RSND_GEN2_M_REG(gen, SCU, DVC_ZCMCR, 0xe14, 0x100),
- RSND_GEN2_M_REG(gen, SCU, DVC_VOL0R, 0xe28, 0x100),
- RSND_GEN2_M_REG(gen, SCU, DVC_VOL1R, 0xe2c, 0x100),
- RSND_GEN2_M_REG(gen, SCU, DVC_DVUER, 0xe48, 0x100),
-
- RSND_GEN2_S_REG(gen, ADG, BRRA, 0x00),
- RSND_GEN2_S_REG(gen, ADG, BRRB, 0x04),
- RSND_GEN2_S_REG(gen, ADG, SSICKR, 0x08),
- RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c),
- RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10),
- RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL2, 0x14),
- RSND_GEN2_S_REG(gen, ADG, DIV_EN, 0x30),
- RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL0, 0x34),
- RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL1, 0x38),
- RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL2, 0x3c),
- RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL3, 0x40),
- RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL4, 0x44),
- RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL0, 0x48),
- RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL1, 0x4c),
- RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL2, 0x50),
- RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL3, 0x54),
- RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL4, 0x58),
- RSND_GEN2_S_REG(gen, ADG, CMDOUT_TIMSEL, 0x5c),
-
- RSND_GEN2_M_REG(gen, SSI, SSICR, 0x00, 0x40),
- RSND_GEN2_M_REG(gen, SSI, SSISR, 0x04, 0x40),
- RSND_GEN2_M_REG(gen, SSI, SSITDR, 0x08, 0x40),
- RSND_GEN2_M_REG(gen, SSI, SSIRDR, 0x0c, 0x40),
- RSND_GEN2_M_REG(gen, SSI, SSIWSR, 0x20, 0x40),
- };
-
- return rsnd_gen_regmap_init(priv, gen, regf);
-}
-
static int rsnd_gen2_probe(struct platform_device *pdev,
struct rsnd_priv *priv)
{
struct device *dev = rsnd_priv_to_dev(priv);
- struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
- struct resource *scu_res;
- struct resource *adg_res;
- struct resource *ssiu_res;
- struct resource *ssi_res;
- int ret;
-
- /*
- * map address
- */
- scu_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SCU);
- adg_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_ADG);
- ssiu_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SSIU);
- ssi_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SSI);
-
- gen->base[RSND_GEN2_SCU] = devm_ioremap_resource(dev, scu_res);
- gen->base[RSND_GEN2_ADG] = devm_ioremap_resource(dev, adg_res);
- gen->base[RSND_GEN2_SSIU] = devm_ioremap_resource(dev, ssiu_res);
- gen->base[RSND_GEN2_SSI] = devm_ioremap_resource(dev, ssi_res);
- if (IS_ERR(gen->base[RSND_GEN2_SCU]) ||
- IS_ERR(gen->base[RSND_GEN2_ADG]) ||
- IS_ERR(gen->base[RSND_GEN2_SSIU]) ||
- IS_ERR(gen->base[RSND_GEN2_SSI]))
- return -ENODEV;
-
- ret = rsnd_gen2_regmap_init(priv, gen);
- if (ret < 0)
- return ret;
-
- dev_dbg(dev, "Gen2 device probed\n");
- dev_dbg(dev, "SCU : %pap => %p\n", &scu_res->start,
- gen->base[RSND_GEN2_SCU]);
- dev_dbg(dev, "ADG : %pap => %p\n", &adg_res->start,
- gen->base[RSND_GEN2_ADG]);
- dev_dbg(dev, "SSIU : %pap => %p\n", &ssiu_res->start,
- gen->base[RSND_GEN2_SSIU]);
- dev_dbg(dev, "SSI : %pap => %p\n", &ssi_res->start,
- gen->base[RSND_GEN2_SSI]);
+ struct rsnd_regmap_field_conf conf_ssiu[] = {
+ RSND_GEN_S_REG(SSI_MODE0, 0x800),
+ RSND_GEN_S_REG(SSI_MODE1, 0x804),
+ /* FIXME: it needs SSI_MODE2/3 in the future */
+ RSND_GEN_M_REG(SSI_BUSIF_MODE, 0x0, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF_ADINR, 0x4, 0x80),
+ RSND_GEN_M_REG(BUSIF_DALIGN, 0x8, 0x80),
+ RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80),
+ RSND_GEN_M_REG(INT_ENABLE, 0x18, 0x80),
+ };
+ struct rsnd_regmap_field_conf conf_scu[] = {
+ RSND_GEN_M_REG(SRC_BUSIF_MODE, 0x0, 0x20),
+ RSND_GEN_M_REG(SRC_ROUTE_MODE0, 0xc, 0x20),
+ RSND_GEN_M_REG(SRC_CTRL, 0x10, 0x20),
+ RSND_GEN_M_REG(CMD_ROUTE_SLCT, 0x18c, 0x20),
+ RSND_GEN_M_REG(CMD_CTRL, 0x190, 0x20),
+ RSND_GEN_M_REG(SRC_SWRSR, 0x200, 0x40),
+ RSND_GEN_M_REG(SRC_SRCIR, 0x204, 0x40),
+ RSND_GEN_M_REG(SRC_ADINR, 0x214, 0x40),
+ RSND_GEN_M_REG(SRC_IFSCR, 0x21c, 0x40),
+ RSND_GEN_M_REG(SRC_IFSVR, 0x220, 0x40),
+ RSND_GEN_M_REG(SRC_SRCCR, 0x224, 0x40),
+ RSND_GEN_M_REG(SRC_BSDSR, 0x22c, 0x40),
+ RSND_GEN_M_REG(SRC_BSISR, 0x238, 0x40),
+ RSND_GEN_M_REG(DVC_SWRSR, 0xe00, 0x100),
+ RSND_GEN_M_REG(DVC_DVUIR, 0xe04, 0x100),
+ RSND_GEN_M_REG(DVC_ADINR, 0xe08, 0x100),
+ RSND_GEN_M_REG(DVC_DVUCR, 0xe10, 0x100),
+ RSND_GEN_M_REG(DVC_ZCMCR, 0xe14, 0x100),
+ RSND_GEN_M_REG(DVC_VOL0R, 0xe28, 0x100),
+ RSND_GEN_M_REG(DVC_VOL1R, 0xe2c, 0x100),
+ RSND_GEN_M_REG(DVC_DVUER, 0xe48, 0x100),
+ };
+ struct rsnd_regmap_field_conf conf_adg[] = {
+ RSND_GEN_S_REG(BRRA, 0x00),
+ RSND_GEN_S_REG(BRRB, 0x04),
+ RSND_GEN_S_REG(SSICKR, 0x08),
+ RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c),
+ RSND_GEN_S_REG(AUDIO_CLK_SEL1, 0x10),
+ RSND_GEN_S_REG(AUDIO_CLK_SEL2, 0x14),
+ RSND_GEN_S_REG(DIV_EN, 0x30),
+ RSND_GEN_S_REG(SRCIN_TIMSEL0, 0x34),
+ RSND_GEN_S_REG(SRCIN_TIMSEL1, 0x38),
+ RSND_GEN_S_REG(SRCIN_TIMSEL2, 0x3c),
+ RSND_GEN_S_REG(SRCIN_TIMSEL3, 0x40),
+ RSND_GEN_S_REG(SRCIN_TIMSEL4, 0x44),
+ RSND_GEN_S_REG(SRCOUT_TIMSEL0, 0x48),
+ RSND_GEN_S_REG(SRCOUT_TIMSEL1, 0x4c),
+ RSND_GEN_S_REG(SRCOUT_TIMSEL2, 0x50),
+ RSND_GEN_S_REG(SRCOUT_TIMSEL3, 0x54),
+ RSND_GEN_S_REG(SRCOUT_TIMSEL4, 0x58),
+ RSND_GEN_S_REG(CMDOUT_TIMSEL, 0x5c),
+ };
+ struct rsnd_regmap_field_conf conf_ssi[] = {
+ RSND_GEN_M_REG(SSICR, 0x00, 0x40),
+ RSND_GEN_M_REG(SSISR, 0x04, 0x40),
+ RSND_GEN_M_REG(SSITDR, 0x08, 0x40),
+ RSND_GEN_M_REG(SSIRDR, 0x0c, 0x40),
+ RSND_GEN_M_REG(SSIWSR, 0x20, 0x40),
+ };
+ int ret_ssiu;
+ int ret_scu;
+ int ret_adg;
+ int ret_ssi;
+
+ ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSIU, conf_ssiu);
+ ret_scu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SCU, conf_scu);
+ ret_adg = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_ADG, conf_adg);
+ ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSI, conf_ssi);
+ if (ret_ssiu < 0 ||
+ ret_scu < 0 ||
+ ret_adg < 0 ||
+ ret_ssi < 0)
+ return ret_ssiu | ret_scu | ret_adg | ret_ssi;
+
+ dev_dbg(dev, "Gen2 is probed\n");
return 0;
}
@@ -374,92 +366,60 @@ static int rsnd_gen2_probe(struct platform_device *pdev,
* Gen1
*/
-/* single address mapping */
-#define RSND_GEN1_S_REG(gen, reg, id, offset) \
- RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, 0, 9)
-
-/* multi address mapping */
-#define RSND_GEN1_M_REG(gen, reg, id, offset, _id_offset) \
- RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, _id_offset, 9)
-
-static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
-{
- struct reg_field regf[RSND_REG_MAX] = {
- RSND_GEN1_S_REG(gen, SRU, SRC_ROUTE_SEL, 0x00),
- RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL0, 0x08),
- RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL1, 0x0c),
- RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL2, 0x10),
- RSND_GEN1_S_REG(gen, SRU, SRC_ROUTE_CTRL, 0xc0),
- RSND_GEN1_S_REG(gen, SRU, SSI_MODE0, 0xD0),
- RSND_GEN1_S_REG(gen, SRU, SSI_MODE1, 0xD4),
- RSND_GEN1_M_REG(gen, SRU, SRC_BUSIF_MODE, 0x20, 0x4),
- RSND_GEN1_M_REG(gen, SRU, SRC_ROUTE_MODE0,0x50, 0x8),
- RSND_GEN1_M_REG(gen, SRU, SRC_SWRSR, 0x200, 0x40),
- RSND_GEN1_M_REG(gen, SRU, SRC_SRCIR, 0x204, 0x40),
- RSND_GEN1_M_REG(gen, SRU, SRC_ADINR, 0x214, 0x40),
- RSND_GEN1_M_REG(gen, SRU, SRC_IFSCR, 0x21c, 0x40),
- RSND_GEN1_M_REG(gen, SRU, SRC_IFSVR, 0x220, 0x40),
- RSND_GEN1_M_REG(gen, SRU, SRC_SRCCR, 0x224, 0x40),
- RSND_GEN1_M_REG(gen, SRU, SRC_MNFSR, 0x228, 0x40),
-
- RSND_GEN1_S_REG(gen, ADG, BRRA, 0x00),
- RSND_GEN1_S_REG(gen, ADG, BRRB, 0x04),
- RSND_GEN1_S_REG(gen, ADG, SSICKR, 0x08),
- RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c),
- RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10),
- RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL3, 0x18),
- RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL4, 0x1c),
- RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL5, 0x20),
-
- RSND_GEN1_M_REG(gen, SSI, SSICR, 0x00, 0x40),
- RSND_GEN1_M_REG(gen, SSI, SSISR, 0x04, 0x40),
- RSND_GEN1_M_REG(gen, SSI, SSITDR, 0x08, 0x40),
- RSND_GEN1_M_REG(gen, SSI, SSIRDR, 0x0c, 0x40),
- RSND_GEN1_M_REG(gen, SSI, SSIWSR, 0x20, 0x40),
- };
-
- return rsnd_gen_regmap_init(priv, gen, regf);
-}
-
static int rsnd_gen1_probe(struct platform_device *pdev,
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;
- int ret;
-
- /*
- * 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;
+ struct rsnd_regmap_field_conf conf_sru[] = {
+ RSND_GEN_S_REG(SRC_ROUTE_SEL, 0x00),
+ RSND_GEN_S_REG(SRC_TMG_SEL0, 0x08),
+ RSND_GEN_S_REG(SRC_TMG_SEL1, 0x0c),
+ RSND_GEN_S_REG(SRC_TMG_SEL2, 0x10),
+ RSND_GEN_S_REG(SRC_ROUTE_CTRL, 0xc0),
+ RSND_GEN_S_REG(SSI_MODE0, 0xD0),
+ RSND_GEN_S_REG(SSI_MODE1, 0xD4),
+ RSND_GEN_M_REG(SRC_BUSIF_MODE, 0x20, 0x4),
+ RSND_GEN_M_REG(SRC_ROUTE_MODE0, 0x50, 0x8),
+ RSND_GEN_M_REG(SRC_SWRSR, 0x200, 0x40),
+ RSND_GEN_M_REG(SRC_SRCIR, 0x204, 0x40),
+ RSND_GEN_M_REG(SRC_ADINR, 0x214, 0x40),
+ RSND_GEN_M_REG(SRC_IFSCR, 0x21c, 0x40),
+ RSND_GEN_M_REG(SRC_IFSVR, 0x220, 0x40),
+ RSND_GEN_M_REG(SRC_SRCCR, 0x224, 0x40),
+ RSND_GEN_M_REG(SRC_MNFSR, 0x228, 0x40),
+ };
+ struct rsnd_regmap_field_conf conf_adg[] = {
+ RSND_GEN_S_REG(BRRA, 0x00),
+ RSND_GEN_S_REG(BRRB, 0x04),
+ RSND_GEN_S_REG(SSICKR, 0x08),
+ RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c),
+ RSND_GEN_S_REG(AUDIO_CLK_SEL1, 0x10),
+ RSND_GEN_S_REG(AUDIO_CLK_SEL3, 0x18),
+ RSND_GEN_S_REG(AUDIO_CLK_SEL4, 0x1c),
+ RSND_GEN_S_REG(AUDIO_CLK_SEL5, 0x20),
+ };
+ struct rsnd_regmap_field_conf conf_ssi[] = {
+ RSND_GEN_M_REG(SSICR, 0x00, 0x40),
+ RSND_GEN_M_REG(SSISR, 0x04, 0x40),
+ RSND_GEN_M_REG(SSITDR, 0x08, 0x40),
+ RSND_GEN_M_REG(SSIRDR, 0x0c, 0x40),
+ RSND_GEN_M_REG(SSIWSR, 0x20, 0x40),
+ };
+ int ret_sru;
+ int ret_adg;
+ int ret_ssi;
- ret = rsnd_gen1_regmap_init(priv, gen);
- if (ret < 0)
- return ret;
+ ret_sru = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SRU, conf_sru);
+ ret_adg = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_ADG, conf_adg);
+ ret_ssi = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SSI, conf_ssi);
+ if (ret_sru < 0 ||
+ ret_adg < 0 ||
+ ret_ssi < 0)
+ return ret_sru | ret_adg | ret_ssi;
- dev_dbg(dev, "Gen1 device probed\n");
- dev_dbg(dev, "SRU : %pap => %p\n", &sru_res->start,
- gen->base[RSND_GEN1_SRU]);
- dev_dbg(dev, "ADG : %pap => %p\n", &adg_res->start,
- gen->base[RSND_GEN1_ADG]);
- dev_dbg(dev, "SSI : %pap => %p\n", &ssi_res->start,
- gen->base[RSND_GEN1_SSI]);
+ dev_dbg(dev, "Gen1 is probed\n");
return 0;
-
}
/*
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 39d98af5ee05..d119adf97c9c 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -90,6 +90,7 @@ enum rsnd_reg {
RSND_REG_SHARE19,
RSND_REG_SHARE20,
RSND_REG_SHARE21,
+ RSND_REG_SHARE22,
RSND_REG_MAX,
};
@@ -127,6 +128,7 @@ enum rsnd_reg {
#define RSND_REG_AUDIO_CLK_SEL2 RSND_REG_SHARE19
#define RSND_REG_CMD_CTRL RSND_REG_SHARE20
#define RSND_REG_CMDOUT_TIMSEL RSND_REG_SHARE21
+#define RSND_REG_BUSIF_DALIGN RSND_REG_SHARE22
struct rsnd_of_data;
struct rsnd_priv;
@@ -156,12 +158,9 @@ u32 rsnd_get_adinr(struct rsnd_mod *mod);
*/
struct rsnd_dma {
struct sh_dmae_slave slave;
- struct work_struct work;
struct dma_chan *chan;
- enum dma_data_direction dir;
-
- int submit_loop;
- int offset; /* it cares A/B plane */
+ enum dma_transfer_direction dir;
+ dma_addr_t addr;
};
void rsnd_dma_start(struct rsnd_dma *dma);
@@ -185,6 +184,7 @@ enum rsnd_mod_type {
struct rsnd_mod_ops {
char *name;
+ char* (*dma_name)(struct rsnd_mod *mod);
int (*probe)(struct rsnd_mod *mod,
struct rsnd_dai *rdai);
int (*remove)(struct rsnd_mod *mod,
@@ -224,6 +224,7 @@ void rsnd_mod_init(struct rsnd_priv *priv,
enum rsnd_mod_type type,
int id);
char *rsnd_mod_name(struct rsnd_mod *mod);
+char *rsnd_mod_dma_name(struct rsnd_mod *mod);
/*
* R-Car sound DAI
@@ -281,10 +282,9 @@ int rsnd_gen_probe(struct platform_device *pdev,
void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv,
struct rsnd_mod *mod,
enum rsnd_reg reg);
-void rsnd_gen_dma_addr(struct rsnd_priv *priv,
- struct rsnd_dma *dma,
- struct dma_slave_config *cfg,
- int is_play, int slave_id);
+dma_addr_t rsnd_gen_dma_addr(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ int is_play, int is_from);
#define rsnd_is_gen1(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN1)
#define rsnd_is_gen2(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN2)
@@ -391,8 +391,12 @@ struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id);
unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv,
struct rsnd_dai_stream *io,
struct snd_pcm_runtime *runtime);
-int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod,
- struct rsnd_dai *rdai);
+int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod,
+ struct rsnd_dai *rdai,
+ int use_busif);
+int rsnd_src_ssiu_stop(struct rsnd_mod *ssi_mod,
+ struct rsnd_dai *rdai,
+ int use_busif);
int rsnd_src_enable_ssi_irq(struct rsnd_mod *ssi_mod,
struct rsnd_dai *rdai);
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 200eda019bc7..9183e0145503 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -106,18 +106,19 @@ struct rsnd_src {
/*
* Gen1/Gen2 common functions
*/
-int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod,
- struct rsnd_dai *rdai)
+int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod,
+ struct rsnd_dai *rdai,
+ int use_busif)
{
struct rsnd_dai_stream *io = rsnd_mod_to_io(ssi_mod);
- struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io);
+ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
int ssi_id = rsnd_mod_id(ssi_mod);
/*
* SSI_MODE0
*/
rsnd_mod_bset(ssi_mod, SSI_MODE0, (1 << ssi_id),
- src_mod ? 0 : (1 << ssi_id));
+ !use_busif << ssi_id);
/*
* SSI_MODE1
@@ -143,6 +144,46 @@ int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod,
0x2 << shift : 0x1 << shift);
}
+ /*
+ * DMA settings for SSIU
+ */
+ if (use_busif) {
+ u32 val = 0x76543210;
+ u32 mask = ~0;
+
+ rsnd_mod_write(ssi_mod, SSI_BUSIF_ADINR,
+ rsnd_get_adinr(ssi_mod));
+ rsnd_mod_write(ssi_mod, SSI_BUSIF_MODE, 1);
+ rsnd_mod_write(ssi_mod, SSI_CTRL, 0x1);
+
+ mask <<= runtime->channels * 4;
+ val = val & mask;
+
+ switch (runtime->sample_bits) {
+ case 16:
+ val |= 0x67452301 & ~mask;
+ break;
+ case 32:
+ val |= 0x76543210 & ~mask;
+ break;
+ }
+ rsnd_mod_write(ssi_mod, BUSIF_DALIGN, val);
+
+ }
+
+ return 0;
+}
+
+int rsnd_src_ssiu_stop(struct rsnd_mod *ssi_mod,
+ struct rsnd_dai *rdai,
+ int use_busif)
+{
+ /*
+ * DMA settings for SSIU
+ */
+ if (use_busif)
+ rsnd_mod_write(ssi_mod, SSI_CTRL, 0);
+
return 0;
}
@@ -461,18 +502,45 @@ static struct rsnd_mod_ops rsnd_src_gen1_ops = {
static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod,
struct rsnd_dai *rdai)
{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
+ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+ uint ratio;
int ret;
+ /* 6 - 1/6 are very enough ratio for SRC_BSDSR */
+ if (!rsnd_src_convert_rate(src))
+ ratio = 0;
+ else if (rsnd_src_convert_rate(src) > runtime->rate)
+ ratio = 100 * rsnd_src_convert_rate(src) / runtime->rate;
+ else
+ ratio = 100 * runtime->rate / rsnd_src_convert_rate(src);
+
+ if (ratio > 600) {
+ dev_err(dev, "FSO/FSI ratio error\n");
+ return -EINVAL;
+ }
+
ret = rsnd_src_set_convert_rate(mod, rdai);
if (ret < 0)
return ret;
- rsnd_mod_write(mod, SSI_BUSIF_ADINR, rsnd_get_adinr(mod));
- rsnd_mod_write(mod, SSI_BUSIF_MODE, 1);
-
rsnd_mod_write(mod, SRC_SRCCR, 0x00011110);
- rsnd_mod_write(mod, SRC_BSDSR, 0x01800000);
+ switch (rsnd_mod_id(mod)) {
+ case 5:
+ case 6:
+ case 7:
+ case 8:
+ rsnd_mod_write(mod, SRC_BSDSR, 0x02400000);
+ break;
+ default:
+ rsnd_mod_write(mod, SRC_BSDSR, 0x01800000);
+ break;
+ }
+
rsnd_mod_write(mod, SRC_BSISR, 0x00100060);
return 0;
@@ -554,7 +622,6 @@ static int rsnd_src_start_gen2(struct rsnd_mod *mod,
rsnd_dma_start(rsnd_mod_to_dma(&src->mod));
- rsnd_mod_write(mod, SSI_CTRL, 0x1);
rsnd_mod_write(mod, SRC_CTRL, val);
return rsnd_src_start(mod, rdai);
@@ -565,7 +632,6 @@ static int rsnd_src_stop_gen2(struct rsnd_mod *mod,
{
struct rsnd_src *src = rsnd_mod_to_src(mod);
- rsnd_mod_write(mod, SSI_CTRL, 0);
rsnd_mod_write(mod, SRC_CTRL, 0);
rsnd_dma_stop(rsnd_mod_to_dma(&src->mod));
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 2df723df5d19..34e84009162b 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -90,6 +90,20 @@ struct rsnd_ssi {
#define rsnd_ssi_mode_flags(p) ((p)->info->flags)
#define rsnd_ssi_dai_id(ssi) ((ssi)->info->dai_id)
+static int rsnd_ssi_use_busif(struct rsnd_mod *mod)
+{
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+ struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
+ int use_busif = 0;
+
+ if (!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_NO_BUSIF))
+ use_busif = 1;
+ if (rsnd_io_to_mod_src(io))
+ use_busif = 1;
+
+ return use_busif;
+}
+
static void rsnd_ssi_status_check(struct rsnd_mod *mod,
u32 bit)
{
@@ -289,8 +303,6 @@ static int rsnd_ssi_init(struct rsnd_mod *mod,
ssi->cr_own = cr;
ssi->err = -1; /* ignore 1st error */
- rsnd_src_ssi_mode_init(mod, rdai);
-
return 0;
}
@@ -389,6 +401,8 @@ static int rsnd_ssi_pio_start(struct rsnd_mod *mod,
/* enable PIO IRQ */
ssi->cr_etc = UIEN | OIEN | DIEN;
+ rsnd_src_ssiu_start(mod, rdai, 0);
+
rsnd_src_enable_ssi_irq(mod, rdai);
rsnd_ssi_hw_start(ssi, rdai, io);
@@ -405,6 +419,8 @@ static int rsnd_ssi_pio_stop(struct rsnd_mod *mod,
rsnd_ssi_hw_stop(ssi, rdai);
+ rsnd_src_ssiu_stop(mod, rdai, 0);
+
return 0;
}
@@ -457,6 +473,8 @@ static int rsnd_ssi_dma_start(struct rsnd_mod *mod,
/* enable DMA transfer */
ssi->cr_etc = DMEN;
+ rsnd_src_ssiu_start(mod, rdai, rsnd_ssi_use_busif(mod));
+
rsnd_dma_start(dma);
rsnd_ssi_hw_start(ssi, ssi->rdai, io);
@@ -482,11 +500,19 @@ static int rsnd_ssi_dma_stop(struct rsnd_mod *mod,
rsnd_dma_stop(dma);
+ rsnd_src_ssiu_stop(mod, rdai, 1);
+
return 0;
}
+static char *rsnd_ssi_dma_name(struct rsnd_mod *mod)
+{
+ return rsnd_ssi_use_busif(mod) ? "ssiu" : SSI_NAME;
+}
+
static struct rsnd_mod_ops rsnd_ssi_dma_ops = {
.name = SSI_NAME,
+ .dma_name = rsnd_ssi_dma_name,
.probe = rsnd_ssi_dma_probe,
.remove = rsnd_ssi_dma_remove,
.init = rsnd_ssi_init,
@@ -595,6 +621,9 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev,
*/
ssi_info->dma_id = of_get_property(np, "pio-transfer", NULL) ?
0 : 1;
+
+ if (of_get_property(np, "no-busif", NULL))
+ ssi_info->flags |= RSND_SSI_NO_BUSIF;
}
rsnd_of_parse_ssi_end:
diff --git a/sound/soc/sirf/Kconfig b/sound/soc/sirf/Kconfig
index 89e89429b04a..840058dcad09 100644
--- a/sound/soc/sirf/Kconfig
+++ b/sound/soc/sirf/Kconfig
@@ -12,3 +12,9 @@ config SND_SOC_SIRF_AUDIO
config SND_SOC_SIRF_AUDIO_PORT
select REGMAP_MMIO
tristate
+
+config SND_SOC_SIRF_USP
+ tristate "SoC Audio (I2S protocol) for SiRF SoC USP interface"
+ depends on SND_SOC_SIRF
+ select REGMAP_MMIO
+ tristate
diff --git a/sound/soc/sirf/Makefile b/sound/soc/sirf/Makefile
index 913b93231d4e..dd917f20f12f 100644
--- a/sound/soc/sirf/Makefile
+++ b/sound/soc/sirf/Makefile
@@ -1,5 +1,7 @@
snd-soc-sirf-audio-objs := sirf-audio.o
snd-soc-sirf-audio-port-objs := sirf-audio-port.o
+snd-soc-sirf-usp-objs := sirf-usp.o
obj-$(CONFIG_SND_SOC_SIRF_AUDIO) += snd-soc-sirf-audio.o
obj-$(CONFIG_SND_SOC_SIRF_AUDIO_PORT) += snd-soc-sirf-audio-port.o
+obj-$(CONFIG_SND_SOC_SIRF_USP) += snd-soc-sirf-usp.o
diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c
new file mode 100644
index 000000000000..3a730374e259
--- /dev/null
+++ b/sound/soc/sirf/sirf-usp.c
@@ -0,0 +1,415 @@
+/*
+ * SiRF USP in I2S/DSP mode
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+#include <linux/module.h>
+#include <linux/io.h>
+#include <linux/of.h>
+#include <linux/clk.h>
+#include <linux/pm_runtime.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "sirf-usp.h"
+
+struct sirf_usp {
+ struct regmap *regmap;
+ struct clk *clk;
+ u32 mode1_reg;
+ u32 mode2_reg;
+ int daifmt_format;
+ struct snd_dmaengine_dai_dma_data playback_dma_data;
+ struct snd_dmaengine_dai_dma_data capture_dma_data;
+};
+
+static void sirf_usp_tx_enable(struct sirf_usp *usp)
+{
+ regmap_update_bits(usp->regmap, USP_TX_FIFO_OP,
+ USP_TX_FIFO_RESET, USP_TX_FIFO_RESET);
+ regmap_write(usp->regmap, USP_TX_FIFO_OP, 0);
+
+ regmap_update_bits(usp->regmap, USP_TX_FIFO_OP,
+ USP_TX_FIFO_START, USP_TX_FIFO_START);
+
+ regmap_update_bits(usp->regmap, USP_TX_RX_ENABLE,
+ USP_TX_ENA, USP_TX_ENA);
+}
+
+static void sirf_usp_tx_disable(struct sirf_usp *usp)
+{
+ regmap_update_bits(usp->regmap, USP_TX_RX_ENABLE,
+ USP_TX_ENA, ~USP_TX_ENA);
+ /* FIFO stop */
+ regmap_write(usp->regmap, USP_TX_FIFO_OP, 0);
+}
+
+static void sirf_usp_rx_enable(struct sirf_usp *usp)
+{
+ regmap_update_bits(usp->regmap, USP_RX_FIFO_OP,
+ USP_RX_FIFO_RESET, USP_RX_FIFO_RESET);
+ regmap_write(usp->regmap, USP_RX_FIFO_OP, 0);
+
+ regmap_update_bits(usp->regmap, USP_RX_FIFO_OP,
+ USP_RX_FIFO_START, USP_RX_FIFO_START);
+
+ regmap_update_bits(usp->regmap, USP_TX_RX_ENABLE,
+ USP_RX_ENA, USP_RX_ENA);
+}
+
+static void sirf_usp_rx_disable(struct sirf_usp *usp)
+{
+ regmap_update_bits(usp->regmap, USP_TX_RX_ENABLE,
+ USP_RX_ENA, ~USP_RX_ENA);
+ /* FIFO stop */
+ regmap_write(usp->regmap, USP_RX_FIFO_OP, 0);
+}
+
+static int sirf_usp_pcm_dai_probe(struct snd_soc_dai *dai)
+{
+ struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai);
+ snd_soc_dai_init_dma_data(dai, &usp->playback_dma_data,
+ &usp->capture_dma_data);
+ return 0;
+}
+
+static int sirf_usp_pcm_set_dai_fmt(struct snd_soc_dai *dai,
+ unsigned int fmt)
+{
+ struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai);
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+ default:
+ dev_err(dai->dev, "Only CBM and CFM supported\n");
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_DSP_A:
+ usp->daifmt_format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+ break;
+ default:
+ dev_err(dai->dev, "Only I2S and DSP_A format supported\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static void sirf_usp_i2s_init(struct sirf_usp *usp)
+{
+ /* Configure RISC mode */
+ regmap_update_bits(usp->regmap, USP_RISC_DSP_MODE,
+ USP_RISC_DSP_SEL, ~USP_RISC_DSP_SEL);
+
+ /*
+ * Configure DMA IO Length register
+ * Set no limit, USP can receive data continuously until it is diabled
+ */
+ regmap_write(usp->regmap, USP_TX_DMA_IO_LEN, 0);
+ regmap_write(usp->regmap, USP_RX_DMA_IO_LEN, 0);
+
+ /* Configure Mode2 register */
+ regmap_write(usp->regmap, USP_MODE2, (1 << USP_RXD_DELAY_LEN_OFFSET) |
+ (0 << USP_TXD_DELAY_LEN_OFFSET) |
+ USP_TFS_CLK_SLAVE_MODE | USP_RFS_CLK_SLAVE_MODE);
+
+ /* Configure Mode1 register */
+ regmap_write(usp->regmap, USP_MODE1,
+ USP_SYNC_MODE | USP_EN | USP_TXD_ACT_EDGE_FALLING |
+ USP_RFS_ACT_LEVEL_LOGIC1 | USP_TFS_ACT_LEVEL_LOGIC1 |
+ USP_TX_UFLOW_REPEAT_ZERO | USP_CLOCK_MODE_SLAVE);
+
+ /* Configure RX DMA IO Control register */
+ regmap_write(usp->regmap, USP_RX_DMA_IO_CTRL, 0);
+
+ /* Congiure RX FIFO Control register */
+ regmap_write(usp->regmap, USP_RX_FIFO_CTRL,
+ (USP_RX_FIFO_THRESHOLD << USP_RX_FIFO_THD_OFFSET) |
+ (USP_TX_RX_FIFO_WIDTH_DWORD << USP_RX_FIFO_WIDTH_OFFSET));
+
+ /* Congiure RX FIFO Level Check register */
+ regmap_write(usp->regmap, USP_RX_FIFO_LEVEL_CHK,
+ RX_FIFO_SC(0x04) | RX_FIFO_LC(0x0E) | RX_FIFO_HC(0x1B));
+
+ /* Configure TX DMA IO Control register*/
+ regmap_write(usp->regmap, USP_TX_DMA_IO_CTRL, 0);
+
+ /* Configure TX FIFO Control register */
+ regmap_write(usp->regmap, USP_TX_FIFO_CTRL,
+ (USP_TX_FIFO_THRESHOLD << USP_TX_FIFO_THD_OFFSET) |
+ (USP_TX_RX_FIFO_WIDTH_DWORD << USP_TX_FIFO_WIDTH_OFFSET));
+ /* Congiure TX FIFO Level Check register */
+ regmap_write(usp->regmap, USP_TX_FIFO_LEVEL_CHK,
+ TX_FIFO_SC(0x1B) | TX_FIFO_LC(0x0E) | TX_FIFO_HC(0x04));
+}
+
+static int sirf_usp_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai);
+ u32 data_len, frame_len, shifter_len;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ data_len = 16;
+ frame_len = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ data_len = 24;
+ frame_len = 32;
+ break;
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ data_len = 24;
+ frame_len = 24;
+ break;
+ default:
+ dev_err(dai->dev, "Format unsupported\n");
+ return -EINVAL;
+ }
+
+ shifter_len = data_len;
+
+ switch (usp->daifmt_format) {
+ case SND_SOC_DAIFMT_I2S:
+ regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL,
+ USP_I2S_SYNC_CHG, USP_I2S_SYNC_CHG);
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL,
+ USP_I2S_SYNC_CHG, 0);
+ frame_len = data_len * params_channels(params);
+ data_len = frame_len;
+ break;
+ default:
+ dev_err(dai->dev, "Only support I2S and DSP_A mode\n");
+ return -EINVAL;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ regmap_update_bits(usp->regmap, USP_TX_FRAME_CTRL,
+ USP_TXC_DATA_LEN_MASK | USP_TXC_FRAME_LEN_MASK
+ | USP_TXC_SHIFTER_LEN_MASK | USP_TXC_SLAVE_CLK_SAMPLE,
+ ((data_len - 1) << USP_TXC_DATA_LEN_OFFSET)
+ | ((frame_len - 1) << USP_TXC_FRAME_LEN_OFFSET)
+ | ((shifter_len - 1) << USP_TXC_SHIFTER_LEN_OFFSET)
+ | USP_TXC_SLAVE_CLK_SAMPLE);
+ else
+ regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL,
+ USP_RXC_DATA_LEN_MASK | USP_RXC_FRAME_LEN_MASK
+ | USP_RXC_SHIFTER_LEN_MASK | USP_SINGLE_SYNC_MODE,
+ ((data_len - 1) << USP_RXC_DATA_LEN_OFFSET)
+ | ((frame_len - 1) << USP_RXC_FRAME_LEN_OFFSET)
+ | ((shifter_len - 1) << USP_RXC_SHIFTER_LEN_OFFSET)
+ | USP_SINGLE_SYNC_MODE);
+
+ return 0;
+}
+
+static int sirf_usp_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ sirf_usp_tx_enable(usp);
+ else
+ sirf_usp_rx_enable(usp);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ sirf_usp_tx_disable(usp);
+ else
+ sirf_usp_rx_disable(usp);
+ break;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops sirf_usp_pcm_dai_ops = {
+ .trigger = sirf_usp_pcm_trigger,
+ .set_fmt = sirf_usp_pcm_set_dai_fmt,
+ .hw_params = sirf_usp_pcm_hw_params,
+};
+
+static struct snd_soc_dai_driver sirf_usp_pcm_dai = {
+ .probe = sirf_usp_pcm_dai_probe,
+ .name = "sirf-usp-pcm",
+ .id = 0,
+ .playback = {
+ .stream_name = "SiRF USP PCM Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE,
+ },
+ .capture = {
+ .stream_name = "SiRF USP PCM Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE,
+ },
+ .ops = &sirf_usp_pcm_dai_ops,
+};
+
+static int sirf_usp_pcm_runtime_suspend(struct device *dev)
+{
+ struct sirf_usp *usp = dev_get_drvdata(dev);
+ clk_disable_unprepare(usp->clk);
+ return 0;
+}
+
+static int sirf_usp_pcm_runtime_resume(struct device *dev)
+{
+ struct sirf_usp *usp = dev_get_drvdata(dev);
+ int ret;
+ ret = clk_prepare_enable(usp->clk);
+ if (ret) {
+ dev_err(dev, "clk_enable failed: %d\n", ret);
+ return ret;
+ }
+ sirf_usp_i2s_init(usp);
+ return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+static int sirf_usp_pcm_suspend(struct device *dev)
+{
+ struct sirf_usp *usp = dev_get_drvdata(dev);
+
+ if (!pm_runtime_status_suspended(dev)) {
+ regmap_read(usp->regmap, USP_MODE1, &usp->mode1_reg);
+ regmap_read(usp->regmap, USP_MODE2, &usp->mode2_reg);
+ sirf_usp_pcm_runtime_suspend(dev);
+ }
+ return 0;
+}
+
+static int sirf_usp_pcm_resume(struct device *dev)
+{
+ struct sirf_usp *usp = dev_get_drvdata(dev);
+ int ret;
+
+ if (!pm_runtime_status_suspended(dev)) {
+ ret = sirf_usp_pcm_runtime_resume(dev);
+ if (ret)
+ return ret;
+ regmap_write(usp->regmap, USP_MODE1, usp->mode1_reg);
+ regmap_write(usp->regmap, USP_MODE2, usp->mode2_reg);
+ }
+ return 0;
+}
+#endif
+
+static const struct snd_soc_component_driver sirf_usp_component = {
+ .name = "sirf-usp",
+};
+
+static const struct regmap_config sirf_usp_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = USP_RX_FIFO_DATA,
+ .cache_type = REGCACHE_NONE,
+};
+
+static int sirf_usp_pcm_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct sirf_usp *usp;
+ void __iomem *base;
+ struct resource *mem_res;
+
+ usp = devm_kzalloc(&pdev->dev, sizeof(struct sirf_usp),
+ GFP_KERNEL);
+ if (!usp)
+ return -ENOMEM;
+
+ platform_set_drvdata(pdev, usp);
+
+ mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ base = devm_ioremap(&pdev->dev, mem_res->start,
+ resource_size(mem_res));
+ if (base == NULL)
+ return -ENOMEM;
+ usp->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+ &sirf_usp_regmap_config);
+ if (IS_ERR(usp->regmap))
+ return PTR_ERR(usp->regmap);
+
+ usp->clk = devm_clk_get(&pdev->dev, NULL);
+ if (IS_ERR(usp->clk)) {
+ dev_err(&pdev->dev, "Get clock failed.\n");
+ return PTR_ERR(usp->clk);
+ }
+
+ pm_runtime_enable(&pdev->dev);
+ if (!pm_runtime_enabled(&pdev->dev)) {
+ ret = sirf_usp_pcm_runtime_resume(&pdev->dev);
+ if (ret)
+ return ret;
+ }
+
+ ret = devm_snd_soc_register_component(&pdev->dev, &sirf_usp_component,
+ &sirf_usp_pcm_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "Register Audio SoC dai failed.\n");
+ return ret;
+ }
+ return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+}
+
+static int sirf_usp_pcm_remove(struct platform_device *pdev)
+{
+ if (!pm_runtime_enabled(&pdev->dev))
+ sirf_usp_pcm_runtime_suspend(&pdev->dev);
+ else
+ pm_runtime_disable(&pdev->dev);
+ return 0;
+}
+
+static const struct of_device_id sirf_usp_pcm_of_match[] = {
+ { .compatible = "sirf,prima2-usp-pcm", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, sirf_usp_pcm_of_match);
+
+static const struct dev_pm_ops sirf_usp_pcm_pm_ops = {
+ SET_RUNTIME_PM_OPS(sirf_usp_pcm_runtime_suspend,
+ sirf_usp_pcm_runtime_resume, NULL)
+ SET_SYSTEM_SLEEP_PM_OPS(sirf_usp_pcm_suspend, sirf_usp_pcm_resume)
+};
+
+static struct platform_driver sirf_usp_pcm_driver = {
+ .driver = {
+ .name = "sirf-usp-pcm",
+ .owner = THIS_MODULE,
+ .of_match_table = sirf_usp_pcm_of_match,
+ .pm = &sirf_usp_pcm_pm_ops,
+ },
+ .probe = sirf_usp_pcm_probe,
+ .remove = sirf_usp_pcm_remove,
+};
+
+module_platform_driver(sirf_usp_pcm_driver);
+
+MODULE_DESCRIPTION("SiRF SoC USP PCM bus driver");
+MODULE_AUTHOR("RongJun Ying <Rongjun.Ying@csr.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/sirf/sirf-usp.h b/sound/soc/sirf/sirf-usp.h
new file mode 100644
index 000000000000..bf0201cb15bc
--- /dev/null
+++ b/sound/soc/sirf/sirf-usp.h
@@ -0,0 +1,293 @@
+/*
+ * arch/arm/mach-prima2/include/mach/sirfsoc_usp.h
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#ifndef _SIRF_USP_H
+#define _SIRF_USP_H
+
+/* USP Registers */
+#define USP_MODE1 0x00
+#define USP_MODE2 0x04
+#define USP_TX_FRAME_CTRL 0x08
+#define USP_RX_FRAME_CTRL 0x0C
+#define USP_TX_RX_ENABLE 0x10
+#define USP_INT_ENABLE 0x14
+#define USP_INT_STATUS 0x18
+#define USP_PIN_IO_DATA 0x1C
+#define USP_RISC_DSP_MODE 0x20
+#define USP_AYSNC_PARAM_REG 0x24
+#define USP_IRDA_X_MODE_DIV 0x28
+#define USP_SM_CFG 0x2C
+#define USP_TX_DMA_IO_CTRL 0x100
+#define USP_TX_DMA_IO_LEN 0x104
+#define USP_TX_FIFO_CTRL 0x108
+#define USP_TX_FIFO_LEVEL_CHK 0x10C
+#define USP_TX_FIFO_OP 0x110
+#define USP_TX_FIFO_STATUS 0x114
+#define USP_TX_FIFO_DATA 0x118
+#define USP_RX_DMA_IO_CTRL 0x120
+#define USP_RX_DMA_IO_LEN 0x124
+#define USP_RX_FIFO_CTRL 0x128
+#define USP_RX_FIFO_LEVEL_CHK 0x12C
+#define USP_RX_FIFO_OP 0x130
+#define USP_RX_FIFO_STATUS 0x134
+#define USP_RX_FIFO_DATA 0x138
+
+/* USP MODE register-1 */
+#define USP_SYNC_MODE 0x00000001
+#define USP_CLOCK_MODE_SLAVE 0x00000002
+#define USP_LOOP_BACK_EN 0x00000004
+#define USP_HPSIR_EN 0x00000008
+#define USP_ENDIAN_CTRL_LSBF 0x00000010
+#define USP_EN 0x00000020
+#define USP_RXD_ACT_EDGE_FALLING 0x00000040
+#define USP_TXD_ACT_EDGE_FALLING 0x00000080
+#define USP_RFS_ACT_LEVEL_LOGIC1 0x00000100
+#define USP_TFS_ACT_LEVEL_LOGIC1 0x00000200
+#define USP_SCLK_IDLE_MODE_TOGGLE 0x00000400
+#define USP_SCLK_IDLE_LEVEL_LOGIC1 0x00000800
+#define USP_SCLK_PIN_MODE_IO 0x00001000
+#define USP_RFS_PIN_MODE_IO 0x00002000
+#define USP_TFS_PIN_MODE_IO 0x00004000
+#define USP_RXD_PIN_MODE_IO 0x00008000
+#define USP_TXD_PIN_MODE_IO 0x00010000
+#define USP_SCLK_IO_MODE_INPUT 0x00020000
+#define USP_RFS_IO_MODE_INPUT 0x00040000
+#define USP_TFS_IO_MODE_INPUT 0x00080000
+#define USP_RXD_IO_MODE_INPUT 0x00100000
+#define USP_TXD_IO_MODE_INPUT 0x00200000
+#define USP_IRDA_WIDTH_DIV_MASK 0x3FC00000
+#define USP_IRDA_WIDTH_DIV_OFFSET 0
+#define USP_IRDA_IDLE_LEVEL_HIGH 0x40000000
+#define USP_TX_UFLOW_REPEAT_ZERO 0x80000000
+#define USP_TX_ENDIAN_MODE 0x00000020
+#define USP_RX_ENDIAN_MODE 0x00000020
+
+/* USP Mode Register-2 */
+#define USP_RXD_DELAY_LEN_MASK 0x000000FF
+#define USP_RXD_DELAY_LEN_OFFSET 0
+
+#define USP_TXD_DELAY_LEN_MASK 0x0000FF00
+#define USP_TXD_DELAY_LEN_OFFSET 8
+
+#define USP_ENA_CTRL_MODE 0x00010000
+#define USP_FRAME_CTRL_MODE 0x00020000
+#define USP_TFS_SOURCE_MODE 0x00040000
+#define USP_TFS_MS_MODE 0x00080000
+#define USP_CLK_DIVISOR_MASK 0x7FE00000
+#define USP_CLK_DIVISOR_OFFSET 21
+
+#define USP_TFS_CLK_SLAVE_MODE (1<<20)
+#define USP_RFS_CLK_SLAVE_MODE (1<<19)
+
+#define USP_IRDA_DATA_WIDTH 0x80000000
+
+/* USP Transmit Frame Control Register */
+
+#define USP_TXC_DATA_LEN_MASK 0x000000FF
+#define USP_TXC_DATA_LEN_OFFSET 0
+
+#define USP_TXC_SYNC_LEN_MASK 0x0000FF00
+#define USP_TXC_SYNC_LEN_OFFSET 8
+
+#define USP_TXC_FRAME_LEN_MASK 0x00FF0000
+#define USP_TXC_FRAME_LEN_OFFSET 16
+
+#define USP_TXC_SHIFTER_LEN_MASK 0x1F000000
+#define USP_TXC_SHIFTER_LEN_OFFSET 24
+
+#define USP_TXC_SLAVE_CLK_SAMPLE 0x20000000
+
+#define USP_TXC_CLK_DIVISOR_MASK 0xC0000000
+#define USP_TXC_CLK_DIVISOR_OFFSET 30
+
+/* USP Receive Frame Control Register */
+
+#define USP_RXC_DATA_LEN_MASK 0x000000FF
+#define USP_RXC_DATA_LEN_OFFSET 0
+
+#define USP_RXC_FRAME_LEN_MASK 0x0000FF00
+#define USP_RXC_FRAME_LEN_OFFSET 8
+
+#define USP_RXC_SHIFTER_LEN_MASK 0x001F0000
+#define USP_RXC_SHIFTER_LEN_OFFSET 16
+
+#define USP_START_EDGE_MODE 0x00800000
+#define USP_I2S_SYNC_CHG 0x00200000
+
+#define USP_RXC_CLK_DIVISOR_MASK 0x0F000000
+#define USP_RXC_CLK_DIVISOR_OFFSET 24
+#define USP_SINGLE_SYNC_MODE 0x00400000
+
+/* Tx - RX Enable Register */
+
+#define USP_RX_ENA 0x00000001
+#define USP_TX_ENA 0x00000002
+
+/* USP Interrupt Enable and status Register */
+#define USP_RX_DONE_INT 0x00000001
+#define USP_TX_DONE_INT 0x00000002
+#define USP_RX_OFLOW_INT 0x00000004
+#define USP_TX_UFLOW_INT 0x00000008
+#define USP_RX_IO_DMA_INT 0x00000010
+#define USP_TX_IO_DMA_INT 0x00000020
+#define USP_RXFIFO_FULL_INT 0x00000040
+#define USP_TXFIFO_EMPTY_INT 0x00000080
+#define USP_RXFIFO_THD_INT 0x00000100
+#define USP_TXFIFO_THD_INT 0x00000200
+#define USP_UART_FRM_ERR_INT 0x00000400
+#define USP_RX_TIMEOUT_INT 0x00000800
+#define USP_TX_ALLOUT_INT 0x00001000
+#define USP_RXD_BREAK_INT 0x00008000
+
+/* All possible TX interruots */
+#define USP_TX_INTERRUPT (USP_TX_DONE_INT|USP_TX_UFLOW_INT|\
+ USP_TX_IO_DMA_INT|\
+ USP_TXFIFO_EMPTY_INT|\
+ USP_TXFIFO_THD_INT)
+/* All possible RX interruots */
+#define USP_RX_INTERRUPT (USP_RX_DONE_INT|USP_RX_OFLOW_INT|\
+ USP_RX_IO_DMA_INT|\
+ USP_RXFIFO_FULL_INT|\
+ USP_RXFIFO_THD_INT|\
+ USP_RXFIFO_THD_INT|USP_RX_TIMEOUT_INT)
+
+#define USP_INT_ALL 0x1FFF
+
+/* USP Pin I/O Data Register */
+
+#define USP_RFS_PIN_VALUE_MASK 0x00000001
+#define USP_TFS_PIN_VALUE_MASK 0x00000002
+#define USP_RXD_PIN_VALUE_MASK 0x00000004
+#define USP_TXD_PIN_VALUE_MASK 0x00000008
+#define USP_SCLK_PIN_VALUE_MASK 0x00000010
+
+/* USP RISC/DSP Mode Register */
+#define USP_RISC_DSP_SEL 0x00000001
+
+/* USP ASYNC PARAMETER Register*/
+
+#define USP_ASYNC_TIMEOUT_MASK 0x0000FFFF
+#define USP_ASYNC_TIMEOUT_OFFSET 0
+#define USP_ASYNC_TIMEOUT(x) (((x)&USP_ASYNC_TIMEOUT_MASK) \
+ <<USP_ASYNC_TIMEOUT_OFFSET)
+
+#define USP_ASYNC_DIV2_MASK 0x003F0000
+#define USP_ASYNC_DIV2_OFFSET 16
+
+/* USP TX DMA I/O MODE Register */
+#define USP_TX_MODE_IO 0x00000001
+
+/* USP TX DMA I/O Length Register */
+#define USP_TX_DATA_LEN_MASK 0xFFFFFFFF
+#define USP_TX_DATA_LEN_OFFSET 0
+
+/* USP TX FIFO Control Register */
+#define USP_TX_FIFO_WIDTH_MASK 0x00000003
+#define USP_TX_FIFO_WIDTH_OFFSET 0
+
+#define USP_TX_FIFO_THD_MASK 0x000001FC
+#define USP_TX_FIFO_THD_OFFSET 2
+
+/* USP TX FIFO Level Check Register */
+#define USP_TX_FIFO_LEVEL_CHECK_MASK 0x1F
+#define USP_TX_FIFO_SC_OFFSET 0
+#define USP_TX_FIFO_LC_OFFSET 10
+#define USP_TX_FIFO_HC_OFFSET 20
+
+#define TX_FIFO_SC(x) (((x) & USP_TX_FIFO_LEVEL_CHECK_MASK) \
+ << USP_TX_FIFO_SC_OFFSET)
+#define TX_FIFO_LC(x) (((x) & USP_TX_FIFO_LEVEL_CHECK_MASK) \
+ << USP_TX_FIFO_LC_OFFSET)
+#define TX_FIFO_HC(x) (((x) & USP_TX_FIFO_LEVEL_CHECK_MASK) \
+ << USP_TX_FIFO_HC_OFFSET)
+
+/* USP TX FIFO Operation Register */
+#define USP_TX_FIFO_RESET 0x00000001
+#define USP_TX_FIFO_START 0x00000002
+
+/* USP TX FIFO Status Register */
+#define USP_TX_FIFO_LEVEL_MASK 0x0000007F
+#define USP_TX_FIFO_LEVEL_OFFSET 0
+
+#define USP_TX_FIFO_FULL 0x00000080
+#define USP_TX_FIFO_EMPTY 0x00000100
+
+/* USP TX FIFO Data Register */
+#define USP_TX_FIFO_DATA_MASK 0xFFFFFFFF
+#define USP_TX_FIFO_DATA_OFFSET 0
+
+/* USP RX DMA I/O MODE Register */
+#define USP_RX_MODE_IO 0x00000001
+#define USP_RX_DMA_FLUSH 0x00000004
+
+/* USP RX DMA I/O Length Register */
+#define USP_RX_DATA_LEN_MASK 0xFFFFFFFF
+#define USP_RX_DATA_LEN_OFFSET 0
+
+/* USP RX FIFO Control Register */
+#define USP_RX_FIFO_WIDTH_MASK 0x00000003
+#define USP_RX_FIFO_WIDTH_OFFSET 0
+
+#define USP_RX_FIFO_THD_MASK 0x000001FC
+#define USP_RX_FIFO_THD_OFFSET 2
+
+/* USP RX FIFO Level Check Register */
+
+#define USP_RX_FIFO_LEVEL_CHECK_MASK 0x1F
+#define USP_RX_FIFO_SC_OFFSET 0
+#define USP_RX_FIFO_LC_OFFSET 10
+#define USP_RX_FIFO_HC_OFFSET 20
+
+#define RX_FIFO_SC(x) (((x) & USP_RX_FIFO_LEVEL_CHECK_MASK) \
+ << USP_RX_FIFO_SC_OFFSET)
+#define RX_FIFO_LC(x) (((x) & USP_RX_FIFO_LEVEL_CHECK_MASK) \
+ << USP_RX_FIFO_LC_OFFSET)
+#define RX_FIFO_HC(x) (((x) & USP_RX_FIFO_LEVEL_CHECK_MASK) \
+ << USP_RX_FIFO_HC_OFFSET)
+
+/* USP RX FIFO Operation Register */
+#define USP_RX_FIFO_RESET 0x00000001
+#define USP_RX_FIFO_START 0x00000002
+
+/* USP RX FIFO Status Register */
+
+#define USP_RX_FIFO_LEVEL_MASK 0x0000007F
+#define USP_RX_FIFO_LEVEL_OFFSET 0
+
+#define USP_RX_FIFO_FULL 0x00000080
+#define USP_RX_FIFO_EMPTY 0x00000100
+
+/* USP RX FIFO Data Register */
+
+#define USP_RX_FIFO_DATA_MASK 0xFFFFFFFF
+#define USP_RX_FIFO_DATA_OFFSET 0
+
+/*
+ * When rx thd irq occur, sender just disable tx empty irq,
+ * Remaining data in tx fifo wil also be sent out.
+ */
+#define USP_FIFO_SIZE 128
+#define USP_TX_FIFO_THRESHOLD (USP_FIFO_SIZE/2)
+#define USP_RX_FIFO_THRESHOLD (USP_FIFO_SIZE/2)
+
+/* FIFO_WIDTH for the USP_TX_FIFO_CTRL and USP_RX_FIFO_CTRL registers */
+#define USP_FIFO_WIDTH_BYTE 0x00
+#define USP_FIFO_WIDTH_WORD 0x01
+#define USP_FIFO_WIDTH_DWORD 0x02
+
+#define USP_ASYNC_DIV2 16
+
+#define USP_PLUGOUT_RETRY_CNT 2
+
+#define USP_TX_RX_FIFO_WIDTH_DWORD 2
+
+#define SIRF_USP_DIV_MCLK 0
+
+#define SIRF_USP_I2S_TFS_SYNC 0
+#define SIRF_USP_I2S_RFS_SYNC 1
+#endif
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 00e70b6c7da2..a9f82b5aba9d 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -78,7 +78,7 @@ int snd_soc_cache_init(struct snd_soc_codec *codec)
mutex_init(&codec->cache_rw_mutex);
dev_dbg(codec->dev, "ASoC: Initializing cache for %s codec\n",
- codec->name);
+ codec->component.name);
if (codec_drv->reg_cache_default)
codec->reg_cache = kmemdup(codec_drv->reg_cache_default,
@@ -98,8 +98,7 @@ int snd_soc_cache_init(struct snd_soc_codec *codec)
int snd_soc_cache_exit(struct snd_soc_codec *codec)
{
dev_dbg(codec->dev, "ASoC: Destroying cache for %s codec\n",
- codec->name);
-
+ codec->component.name);
kfree(codec->reg_cache);
codec->reg_cache = NULL;
return 0;
@@ -192,7 +191,7 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec)
return 0;
dev_dbg(codec->dev, "ASoC: Syncing cache for %s codec\n",
- codec->name);
+ codec->component.name);
trace_snd_soc_cache_sync(codec, name, "start");
ret = snd_soc_flat_cache_sync(codec);
if (!ret)
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 10f7f1da2aca..27c06acce205 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -37,7 +37,8 @@ static int soc_compr_open(struct snd_compr_stream *cstream)
if (platform->driver->compr_ops && platform->driver->compr_ops->open) {
ret = platform->driver->compr_ops->open(cstream);
if (ret < 0) {
- pr_err("compress asoc: can't open platform %s\n", platform->name);
+ pr_err("compress asoc: can't open platform %s\n",
+ platform->component.name);
goto out;
}
}
@@ -84,7 +85,8 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
if (platform->driver->compr_ops && platform->driver->compr_ops->open) {
ret = platform->driver->compr_ops->open(cstream);
if (ret < 0) {
- pr_err("compress asoc: can't open platform %s\n", platform->name);
+ pr_err("compress asoc: can't open platform %s\n",
+ platform->component.name);
goto out;
}
}
@@ -627,6 +629,11 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
char new_name[64];
int ret = 0, direction = 0;
+ if (rtd->num_codecs > 1) {
+ dev_err(rtd->card->dev, "Multicodec not supported for compressed stream\n");
+ return -EINVAL;
+ }
+
/* check client and interface hw capabilities */
snprintf(new_name, sizeof(new_name), "%s %s-%d",
rtd->dai_link->stream_name, codec_dai->name, num);
@@ -680,7 +687,7 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
ret = snd_compress_new(rtd->card->snd_card, num, direction, compr);
if (ret < 0) {
pr_err("compress asoc: can't create compress for codec %s\n",
- codec->name);
+ codec->component.name);
goto compr_err;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b87d7d882e6d..d4bfd4a9076f 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -270,12 +270,33 @@ static const struct file_operations codec_reg_fops = {
.llseek = default_llseek,
};
+static struct dentry *soc_debugfs_create_dir(struct dentry *parent,
+ const char *fmt, ...)
+{
+ struct dentry *de;
+ va_list ap;
+ char *s;
+
+ va_start(ap, fmt);
+ s = kvasprintf(GFP_KERNEL, fmt, ap);
+ va_end(ap);
+
+ if (!s)
+ return NULL;
+
+ de = debugfs_create_dir(s, parent);
+ kfree(s);
+
+ return de;
+}
+
static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
{
- struct dentry *debugfs_card_root = codec->card->debugfs_card_root;
+ struct dentry *debugfs_card_root = codec->component.card->debugfs_card_root;
- codec->debugfs_codec_root = debugfs_create_dir(codec->name,
- debugfs_card_root);
+ codec->debugfs_codec_root = soc_debugfs_create_dir(debugfs_card_root,
+ "codec:%s",
+ codec->component.name);
if (!codec->debugfs_codec_root) {
dev_warn(codec->dev,
"ASoC: Failed to create codec debugfs directory\n");
@@ -304,17 +325,18 @@ static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
static void soc_init_platform_debugfs(struct snd_soc_platform *platform)
{
- struct dentry *debugfs_card_root = platform->card->debugfs_card_root;
+ struct dentry *debugfs_card_root = platform->component.card->debugfs_card_root;
- platform->debugfs_platform_root = debugfs_create_dir(platform->name,
- debugfs_card_root);
+ platform->debugfs_platform_root = soc_debugfs_create_dir(debugfs_card_root,
+ "platform:%s",
+ platform->component.name);
if (!platform->debugfs_platform_root) {
dev_warn(platform->dev,
"ASoC: Failed to create platform debugfs directory\n");
return;
}
- snd_soc_dapm_debugfs_init(&platform->dapm,
+ snd_soc_dapm_debugfs_init(&platform->component.dapm,
platform->debugfs_platform_root);
}
@@ -335,7 +357,7 @@ static ssize_t codec_list_read_file(struct file *file, char __user *user_buf,
list_for_each_entry(codec, &codec_list, list) {
len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n",
- codec->name);
+ codec->component.name);
if (len >= 0)
ret += len;
if (ret > PAGE_SIZE) {
@@ -406,7 +428,7 @@ static ssize_t platform_list_read_file(struct file *file,
list_for_each_entry(platform, &platform_list, list) {
len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n",
- platform->name);
+ platform->component.name);
if (len >= 0)
ret += len;
if (ret > PAGE_SIZE) {
@@ -524,11 +546,12 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
int err;
codec->ac97->dev.bus = &ac97_bus_type;
- codec->ac97->dev.parent = codec->card->dev;
+ codec->ac97->dev.parent = codec->component.card->dev;
codec->ac97->dev.release = soc_ac97_device_release;
dev_set_name(&codec->ac97->dev, "%d-%d:%s",
- codec->card->snd_card->number, 0, codec->name);
+ codec->component.card->snd_card->number, 0,
+ codec->component.name);
err = device_register(&codec->ac97->dev);
if (err < 0) {
dev_err(codec->dev, "ASoC: Can't register ac97 bus\n");
@@ -554,7 +577,7 @@ int snd_soc_suspend(struct device *dev)
{
struct snd_soc_card *card = dev_get_drvdata(dev);
struct snd_soc_codec *codec;
- int i;
+ int i, j;
/* If the initialization of this soc device failed, there is no codec
* associated with it. Just bail out in this case.
@@ -574,14 +597,17 @@ int snd_soc_suspend(struct device *dev)
/* mute any active DACs */
for (i = 0; i < card->num_rtd; i++) {
- struct snd_soc_dai *dai = card->rtd[i].codec_dai;
- struct snd_soc_dai_driver *drv = dai->driver;
if (card->rtd[i].dai_link->ignore_suspend)
continue;
- if (drv->ops->digital_mute && dai->playback_active)
- drv->ops->digital_mute(dai, 1);
+ for (j = 0; j < card->rtd[i].num_codecs; j++) {
+ struct snd_soc_dai *dai = card->rtd[i].codec_dais[j];
+ struct snd_soc_dai_driver *drv = dai->driver;
+
+ if (drv->ops->digital_mute && dai->playback_active)
+ drv->ops->digital_mute(dai, 1);
+ }
}
/* suspend all pcms */
@@ -612,8 +638,12 @@ int snd_soc_suspend(struct device *dev)
/* close any waiting streams and save state */
for (i = 0; i < card->num_rtd; i++) {
+ struct snd_soc_dai **codec_dais = card->rtd[i].codec_dais;
flush_delayed_work(&card->rtd[i].delayed_work);
- card->rtd[i].codec->dapm.suspend_bias_level = card->rtd[i].codec->dapm.bias_level;
+ for (j = 0; j < card->rtd[i].num_codecs; j++) {
+ codec_dais[j]->codec->dapm.suspend_bias_level =
+ codec_dais[j]->codec->dapm.bias_level;
+ }
}
for (i = 0; i < card->num_rtd; i++) {
@@ -697,7 +727,7 @@ static void soc_resume_deferred(struct work_struct *work)
struct snd_soc_card *card =
container_of(work, struct snd_soc_card, deferred_resume_work);
struct snd_soc_codec *codec;
- int i;
+ int i, j;
/* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
* so userspace apps are blocked from touching us
@@ -758,14 +788,17 @@ static void soc_resume_deferred(struct work_struct *work)
/* unmute any active DACs */
for (i = 0; i < card->num_rtd; i++) {
- struct snd_soc_dai *dai = card->rtd[i].codec_dai;
- struct snd_soc_dai_driver *drv = dai->driver;
if (card->rtd[i].dai_link->ignore_suspend)
continue;
- if (drv->ops->digital_mute && dai->playback_active)
- drv->ops->digital_mute(dai, 0);
+ for (j = 0; j < card->rtd[i].num_codecs; j++) {
+ struct snd_soc_dai *dai = card->rtd[i].codec_dais[j];
+ struct snd_soc_dai_driver *drv = dai->driver;
+
+ if (drv->ops->digital_mute && dai->playback_active)
+ drv->ops->digital_mute(dai, 0);
+ }
}
for (i = 0; i < card->num_rtd; i++) {
@@ -810,12 +843,19 @@ int snd_soc_resume(struct device *dev)
/* activate pins from sleep state */
for (i = 0; i < card->num_rtd; i++) {
- struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai;
- struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai;
+ struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
+ struct snd_soc_dai **codec_dais = rtd->codec_dais;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int j;
+
if (cpu_dai->active)
pinctrl_pm_select_default_state(cpu_dai->dev);
- if (codec_dai->active)
- pinctrl_pm_select_default_state(codec_dai->dev);
+
+ for (j = 0; j < rtd->num_codecs; j++) {
+ struct snd_soc_dai *codec_dai = codec_dais[j];
+ if (codec_dai->active)
+ pinctrl_pm_select_default_state(codec_dai->dev);
+ }
}
/* AC97 devices might have other drivers hanging off them so
@@ -847,8 +887,9 @@ EXPORT_SYMBOL_GPL(snd_soc_resume);
static const struct snd_soc_dai_ops null_dai_ops = {
};
-static struct snd_soc_codec *soc_find_codec(const struct device_node *codec_of_node,
- const char *codec_name)
+static struct snd_soc_codec *soc_find_codec(
+ const struct device_node *codec_of_node,
+ const char *codec_name)
{
struct snd_soc_codec *codec;
@@ -857,7 +898,7 @@ static struct snd_soc_codec *soc_find_codec(const struct device_node *codec_of_n
if (codec->dev->of_node != codec_of_node)
continue;
} else {
- if (strcmp(codec->name, codec_name))
+ if (strcmp(codec->component.name, codec_name))
continue;
}
@@ -886,9 +927,12 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
struct snd_soc_dai_link *dai_link = &card->dai_link[num];
struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
struct snd_soc_component *component;
+ struct snd_soc_dai_link_component *codecs = dai_link->codecs;
+ struct snd_soc_dai **codec_dais = rtd->codec_dais;
struct snd_soc_platform *platform;
struct snd_soc_dai *cpu_dai;
const char *platform_name;
+ int i;
dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num);
@@ -915,24 +959,30 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
return -EPROBE_DEFER;
}
- /* Find CODEC from registered list */
- rtd->codec = soc_find_codec(dai_link->codec_of_node,
- dai_link->codec_name);
- if (!rtd->codec) {
- dev_err(card->dev, "ASoC: CODEC %s not registered\n",
- dai_link->codec_name);
- return -EPROBE_DEFER;
- }
+ rtd->num_codecs = dai_link->num_codecs;
- /* Find CODEC DAI from registered list */
- rtd->codec_dai = soc_find_codec_dai(rtd->codec,
- dai_link->codec_dai_name);
- if (!rtd->codec_dai) {
- dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n",
- dai_link->codec_dai_name);
- return -EPROBE_DEFER;
+ /* Find CODEC from registered CODECs */
+ for (i = 0; i < rtd->num_codecs; i++) {
+ struct snd_soc_codec *codec;
+ codec = soc_find_codec(codecs[i].of_node, codecs[i].name);
+ if (!codec) {
+ dev_err(card->dev, "ASoC: CODEC %s not registered\n",
+ codecs[i].name);
+ return -EPROBE_DEFER;
+ }
+
+ codec_dais[i] = soc_find_codec_dai(codec, codecs[i].dai_name);
+ if (!codec_dais[i]) {
+ dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n",
+ codecs[i].dai_name);
+ return -EPROBE_DEFER;
+ }
}
+ /* Single codec links expect codec and codec_dai in runtime data */
+ rtd->codec_dai = codec_dais[0];
+ rtd->codec = rtd->codec_dai->codec;
+
/* if there's no platform we match on the empty platform */
platform_name = dai_link->platform_name;
if (!platform_name && !dai_link->platform_of_node)
@@ -945,7 +995,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
dai_link->platform_of_node)
continue;
} else {
- if (strcmp(platform->name, platform_name))
+ if (strcmp(platform->component.name, platform_name))
continue;
}
@@ -974,11 +1024,10 @@ static int soc_remove_platform(struct snd_soc_platform *platform)
}
/* Make sure all DAPM widgets are freed */
- snd_soc_dapm_free(&platform->dapm);
+ snd_soc_dapm_free(&platform->component.dapm);
soc_cleanup_platform_debugfs(platform);
platform->probed = 0;
- list_del(&platform->card_list);
module_put(platform->dev->driver->owner);
return 0;
@@ -1023,8 +1072,8 @@ static void soc_remove_codec_dai(struct snd_soc_dai *codec_dai, int order)
static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order)
{
struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
- struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai;
- int err;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int i, err;
/* unregister the rtd device */
if (rtd->dev_registered) {
@@ -1035,7 +1084,8 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order)
}
/* remove the CODEC DAI */
- soc_remove_codec_dai(codec_dai, order);
+ for (i = 0; i < rtd->num_codecs; i++)
+ soc_remove_codec_dai(rtd->codec_dais[i], order);
/* remove the cpu_dai */
if (cpu_dai && cpu_dai->probed &&
@@ -1048,11 +1098,8 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order)
cpu_dai->name, err);
}
cpu_dai->probed = 0;
-
- if (!cpu_dai->codec) {
- snd_soc_dapm_free(&cpu_dai->dapm);
+ if (!cpu_dai->codec)
module_put(cpu_dai->dev->driver->owner);
- }
}
}
@@ -1061,9 +1108,9 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num,
{
struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_codec *codec;
+ int i;
/* remove the platform */
if (platform && platform->probed &&
@@ -1072,8 +1119,8 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num,
}
/* remove the CODEC-side CODEC */
- if (codec_dai) {
- codec = codec_dai->codec;
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec = rtd->codec_dais[i]->codec;
if (codec && codec->probed &&
codec->driver->remove_order == order)
soc_remove_codec(codec);
@@ -1108,7 +1155,7 @@ static void soc_remove_dai_links(struct snd_soc_card *card)
}
static void soc_set_name_prefix(struct snd_soc_card *card,
- struct snd_soc_codec *codec)
+ struct snd_soc_component *component)
{
int i;
@@ -1117,11 +1164,11 @@ static void soc_set_name_prefix(struct snd_soc_card *card,
for (i = 0; i < card->num_configs; i++) {
struct snd_soc_codec_conf *map = &card->codec_conf[i];
- if (map->of_node && codec->dev->of_node != map->of_node)
+ if (map->of_node && component->dev->of_node != map->of_node)
continue;
- if (map->dev_name && strcmp(codec->name, map->dev_name))
+ if (map->dev_name && strcmp(component->name, map->dev_name))
continue;
- codec->name_prefix = map->name_prefix;
+ component->name_prefix = map->name_prefix;
break;
}
}
@@ -1133,9 +1180,9 @@ static int soc_probe_codec(struct snd_soc_card *card,
const struct snd_soc_codec_driver *driver = codec->driver;
struct snd_soc_dai *dai;
- codec->card = card;
+ codec->component.card = card;
codec->dapm.card = card;
- soc_set_name_prefix(card, codec);
+ soc_set_name_prefix(card, &codec->component);
if (!try_module_get(codec->dev->driver->owner))
return -ENODEV;
@@ -1177,7 +1224,7 @@ static int soc_probe_codec(struct snd_soc_card *card,
WARN(codec->dapm.idle_bias_off &&
codec->dapm.bias_level != SND_SOC_BIAS_OFF,
"codec %s can not start from non-off bias with idle_bias_off==1\n",
- codec->name);
+ codec->component.name);
}
if (driver->controls)
@@ -1209,8 +1256,8 @@ static int soc_probe_platform(struct snd_soc_card *card,
struct snd_soc_component *component;
struct snd_soc_dai *dai;
- platform->card = card;
- platform->dapm.card = card;
+ platform->component.card = card;
+ platform->component.dapm.card = card;
if (!try_module_get(platform->dev->driver->owner))
return -ENODEV;
@@ -1218,7 +1265,7 @@ static int soc_probe_platform(struct snd_soc_card *card,
soc_init_platform_debugfs(platform);
if (driver->dapm_widgets)
- snd_soc_dapm_new_controls(&platform->dapm,
+ snd_soc_dapm_new_controls(&platform->component.dapm,
driver->dapm_widgets, driver->num_dapm_widgets);
/* Create DAPM widgets for each DAI stream */
@@ -1226,10 +1273,11 @@ static int soc_probe_platform(struct snd_soc_card *card,
if (component->dev != platform->dev)
continue;
list_for_each_entry(dai, &component->dai_list, list)
- snd_soc_dapm_new_dai_widgets(&platform->dapm, dai);
+ snd_soc_dapm_new_dai_widgets(&platform->component.dapm,
+ dai);
}
- platform->dapm.idle_bias_off = 1;
+ platform->component.dapm.idle_bias_off = 1;
if (driver->probe) {
ret = driver->probe(platform);
@@ -1244,13 +1292,12 @@ static int soc_probe_platform(struct snd_soc_card *card,
snd_soc_add_platform_controls(platform, driver->controls,
driver->num_controls);
if (driver->dapm_routes)
- snd_soc_dapm_add_routes(&platform->dapm, driver->dapm_routes,
- driver->num_dapm_routes);
+ snd_soc_dapm_add_routes(&platform->component.dapm,
+ driver->dapm_routes, driver->num_dapm_routes);
/* mark platform as probed and add to card platform list */
platform->probed = 1;
- list_add(&platform->card_list, &card->platform_dev_list);
- list_add(&platform->dapm.list, &card->dapm_list);
+ list_add(&platform->component.dapm.list, &card->dapm_list);
return 0;
@@ -1266,83 +1313,17 @@ static void rtd_release(struct device *dev)
kfree(dev);
}
-static int soc_aux_dev_init(struct snd_soc_card *card,
- struct snd_soc_codec *codec,
- int num)
-{
- struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num];
- struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num];
- int ret;
-
- rtd->card = card;
-
- /* do machine specific initialization */
- if (aux_dev->init) {
- ret = aux_dev->init(&codec->dapm);
- if (ret < 0)
- return ret;
- }
-
- rtd->codec = codec;
-
- return 0;
-}
-
-static int soc_dai_link_init(struct snd_soc_card *card,
- struct snd_soc_codec *codec,
- int num)
+static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
+ const char *name)
{
- struct snd_soc_dai_link *dai_link = &card->dai_link[num];
- struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
- int ret;
-
- rtd->card = card;
-
- /* do machine specific initialization */
- if (dai_link->init) {
- ret = dai_link->init(rtd);
- if (ret < 0)
- return ret;
- }
-
- rtd->codec = codec;
-
- return 0;
-}
-
-static int soc_post_component_init(struct snd_soc_card *card,
- struct snd_soc_codec *codec,
- int num, int dailess)
-{
- struct snd_soc_dai_link *dai_link = NULL;
- struct snd_soc_aux_dev *aux_dev = NULL;
- struct snd_soc_pcm_runtime *rtd;
- const char *name;
int ret = 0;
- if (!dailess) {
- dai_link = &card->dai_link[num];
- rtd = &card->rtd[num];
- name = dai_link->name;
- ret = soc_dai_link_init(card, codec, num);
- } else {
- aux_dev = &card->aux_dev[num];
- rtd = &card->rtd_aux[num];
- name = aux_dev->name;
- ret = soc_aux_dev_init(card, codec, num);
- }
-
- if (ret < 0) {
- dev_err(card->dev, "ASoC: failed to init %s: %d\n", name, ret);
- return ret;
- }
-
/* register the rtd device */
rtd->dev = kzalloc(sizeof(struct device), GFP_KERNEL);
if (!rtd->dev)
return -ENOMEM;
device_initialize(rtd->dev);
- rtd->dev->parent = card->dev;
+ rtd->dev->parent = rtd->card->dev;
rtd->dev->release = rtd_release;
rtd->dev->init_name = name;
dev_set_drvdata(rtd->dev, rtd);
@@ -1355,7 +1336,7 @@ static int soc_post_component_init(struct snd_soc_card *card,
if (ret < 0) {
/* calling put_device() here to free the rtd->dev */
put_device(rtd->dev);
- dev_err(card->dev,
+ dev_err(rtd->card->dev,
"ASoC: failed to register runtime device: %d\n", ret);
return ret;
}
@@ -1364,26 +1345,15 @@ static int soc_post_component_init(struct snd_soc_card *card,
/* add DAPM sysfs entries for this codec */
ret = snd_soc_dapm_sys_add(rtd->dev);
if (ret < 0)
- dev_err(codec->dev,
+ dev_err(rtd->dev,
"ASoC: failed to add codec dapm sysfs entries: %d\n", ret);
/* add codec sysfs entries */
ret = device_create_file(rtd->dev, &dev_attr_codec_reg);
if (ret < 0)
- dev_err(codec->dev,
+ dev_err(rtd->dev,
"ASoC: failed to add codec sysfs files: %d\n", ret);
-#ifdef CONFIG_DEBUG_FS
- /* add DPCM sysfs entries */
- if (!dailess && !dai_link->dynamic)
- goto out;
-
- ret = soc_dpcm_debugfs_add(rtd);
- if (ret < 0)
- dev_err(rtd->dev, "ASoC: failed to add dpcm sysfs entries: %d\n", ret);
-
-out:
-#endif
return 0;
}
@@ -1392,9 +1362,8 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num,
{
struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_platform *platform = rtd->platform;
- int ret;
+ int i, ret;
/* probe the CPU-side component, if it is a CODEC */
if (cpu_dai->codec &&
@@ -1405,12 +1374,14 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num,
return ret;
}
- /* probe the CODEC-side component */
- if (!codec_dai->codec->probed &&
- codec_dai->codec->driver->probe_order == order) {
- ret = soc_probe_codec(card, codec_dai->codec);
- if (ret < 0)
- return ret;
+ /* probe the CODEC-side components */
+ for (i = 0; i < rtd->num_codecs; i++) {
+ if (!rtd->codec_dais[i]->codec->probed &&
+ rtd->codec_dais[i]->codec->driver->probe_order == order) {
+ ret = soc_probe_codec(card, rtd->codec_dais[i]->codec);
+ if (ret < 0)
+ return ret;
+ }
}
/* probe the platform */
@@ -1450,12 +1421,16 @@ static int soc_probe_codec_dai(struct snd_soc_card *card,
static int soc_link_dai_widgets(struct snd_soc_card *card,
struct snd_soc_dai_link *dai_link,
- struct snd_soc_dai *cpu_dai,
- struct snd_soc_dai *codec_dai)
+ struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dapm_widget *play_w, *capture_w;
int ret;
+ if (rtd->num_codecs > 1)
+ dev_warn(card->dev, "ASoC: Multiple codecs not supported yet\n");
+
/* link the DAI widgets */
play_w = codec_dai->playback_widget;
capture_w = cpu_dai->capture_widget;
@@ -1488,19 +1463,18 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order)
{
struct snd_soc_dai_link *dai_link = &card->dai_link[num];
struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
- struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- int ret;
+ int i, ret;
dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n",
card->name, num, order);
/* config components */
cpu_dai->platform = platform;
- codec_dai->card = card;
cpu_dai->card = card;
+ for (i = 0; i < rtd->num_codecs; i++)
+ rtd->codec_dais[i]->card = card;
/* set default power off timeout */
rtd->pmdown_time = pmdown_time;
@@ -1509,11 +1483,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order)
if (!cpu_dai->probed &&
cpu_dai->driver->probe_order == order) {
if (!cpu_dai->codec) {
- cpu_dai->dapm.card = card;
if (!try_module_get(cpu_dai->dev->driver->owner))
return -ENODEV;
-
- list_add(&cpu_dai->dapm.list, &card->dapm_list);
}
if (cpu_dai->driver->probe) {
@@ -1530,18 +1501,43 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order)
}
/* probe the CODEC DAI */
- ret = soc_probe_codec_dai(card, codec_dai, order);
- if (ret)
- return ret;
+ for (i = 0; i < rtd->num_codecs; i++) {
+ ret = soc_probe_codec_dai(card, rtd->codec_dais[i], order);
+ if (ret)
+ return ret;
+ }
/* complete DAI probe during last probe */
if (order != SND_SOC_COMP_ORDER_LAST)
return 0;
- ret = soc_post_component_init(card, codec, num, 0);
+ /* do machine specific initialization */
+ if (dai_link->init) {
+ ret = dai_link->init(rtd);
+ if (ret < 0) {
+ dev_err(card->dev, "ASoC: failed to init %s: %d\n",
+ dai_link->name, ret);
+ return ret;
+ }
+ }
+
+ ret = soc_post_component_init(rtd, dai_link->name);
if (ret)
return ret;
+#ifdef CONFIG_DEBUG_FS
+ /* add DPCM sysfs entries */
+ if (dai_link->dynamic) {
+ ret = soc_dpcm_debugfs_add(rtd);
+ if (ret < 0) {
+ dev_err(rtd->dev,
+ "ASoC: failed to add dpcm sysfs entries: %d\n",
+ ret);
+ return ret;
+ }
+ }
+#endif
+
ret = device_create_file(rtd->dev, &dev_attr_pmdown_time);
if (ret < 0)
dev_warn(rtd->dev, "ASoC: failed to add pmdown_time sysfs: %d\n",
@@ -1570,16 +1566,18 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order)
codec2codec_close_delayed_work);
/* link the DAI widgets */
- ret = soc_link_dai_widgets(card, dai_link,
- cpu_dai, codec_dai);
+ ret = soc_link_dai_widgets(card, dai_link, rtd);
if (ret)
return ret;
}
}
/* add platform data for AC97 devices */
- if (rtd->codec_dai->driver->ac97_control)
- snd_ac97_dev_add_pdata(codec->ac97, rtd->cpu_dai->ac97_pdata);
+ for (i = 0; i < rtd->num_codecs; i++) {
+ if (rtd->codec_dais[i]->driver->ac97_control)
+ snd_ac97_dev_add_pdata(rtd->codec_dais[i]->codec->ac97,
+ rtd->cpu_dai->ac97_pdata);
+ }
return 0;
}
@@ -1617,11 +1615,6 @@ static int soc_register_ac97_codec(struct snd_soc_codec *codec,
return 0;
}
-static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd)
-{
- return soc_register_ac97_codec(rtd->codec, rtd->codec_dai);
-}
-
static void soc_unregister_ac97_codec(struct snd_soc_codec *codec)
{
if (codec->ac97_registered) {
@@ -1630,74 +1623,77 @@ static void soc_unregister_ac97_codec(struct snd_soc_codec *codec)
}
}
-static void soc_unregister_ac97_dai_link(struct snd_soc_pcm_runtime *rtd)
+static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd)
{
- soc_unregister_ac97_codec(rtd->codec);
-}
-#endif
+ int i, ret;
-static struct snd_soc_codec *soc_find_matching_codec(struct snd_soc_card *card,
- int num)
-{
- struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num];
- struct snd_soc_codec *codec;
+ for (i = 0; i < rtd->num_codecs; i++) {
+ struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
- /* find CODEC from registered CODECs */
- list_for_each_entry(codec, &codec_list, list) {
- if (aux_dev->codec_of_node &&
- (codec->dev->of_node != aux_dev->codec_of_node))
- continue;
- if (aux_dev->codec_name && strcmp(codec->name, aux_dev->codec_name))
- continue;
- return codec;
+ ret = soc_register_ac97_codec(codec_dai->codec, codec_dai);
+ if (ret) {
+ while (--i >= 0)
+ soc_unregister_ac97_codec(codec_dai->codec);
+ return ret;
+ }
}
- return NULL;
+ return 0;
}
-static int soc_check_aux_dev(struct snd_soc_card *card, int num)
+static void soc_unregister_ac97_dai_link(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num];
- const char *codecname = aux_dev->codec_name;
- struct snd_soc_codec *codec = soc_find_matching_codec(card, num);
-
- if (codec)
- return 0;
- if (aux_dev->codec_of_node)
- codecname = of_node_full_name(aux_dev->codec_of_node);
+ int i;
- dev_err(card->dev, "ASoC: %s not registered\n", codecname);
- return -EPROBE_DEFER;
+ for (i = 0; i < rtd->num_codecs; i++)
+ soc_unregister_ac97_codec(rtd->codec_dais[i]->codec);
}
+#endif
-static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
+static int soc_bind_aux_dev(struct snd_soc_card *card, int num)
{
+ struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num];
struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num];
const char *codecname = aux_dev->codec_name;
- int ret = -ENODEV;
- struct snd_soc_codec *codec = soc_find_matching_codec(card, num);
- if (!codec) {
+ rtd->codec = soc_find_codec(aux_dev->codec_of_node, codecname);
+ if (!rtd->codec) {
if (aux_dev->codec_of_node)
codecname = of_node_full_name(aux_dev->codec_of_node);
- /* codec not found */
- dev_err(card->dev, "ASoC: codec %s not found", codecname);
+ dev_err(card->dev, "ASoC: %s not registered\n", codecname);
return -EPROBE_DEFER;
}
- if (codec->probed) {
- dev_err(codec->dev, "ASoC: codec already probed");
+ return 0;
+}
+
+static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
+{
+ struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num];
+ struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num];
+ int ret;
+
+ if (rtd->codec->probed) {
+ dev_err(rtd->codec->dev, "ASoC: codec already probed\n");
return -EBUSY;
}
- ret = soc_probe_codec(card, codec);
+ ret = soc_probe_codec(card, rtd->codec);
if (ret < 0)
return ret;
- ret = soc_post_component_init(card, codec, num, 1);
+ /* do machine specific initialization */
+ if (aux_dev->init) {
+ ret = aux_dev->init(&rtd->codec->dapm);
+ if (ret < 0) {
+ dev_err(card->dev, "ASoC: failed to init %s: %d\n",
+ aux_dev->name, ret);
+ return ret;
+ }
+ }
- return ret;
+ return soc_post_component_init(rtd, aux_dev->name);
}
static void soc_remove_aux_dev(struct snd_soc_card *card, int num)
@@ -1749,9 +1745,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
goto base_error;
}
- /* check aux_devs too */
+ /* bind aux_devs too */
for (i = 0; i < card->num_aux_devs; i++) {
- ret = soc_check_aux_dev(card, i);
+ ret = soc_bind_aux_dev(card, i);
if (ret != 0)
goto base_error;
}
@@ -1849,16 +1845,23 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
card->num_dapm_routes);
for (i = 0; i < card->num_links; i++) {
+ struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
dai_link = &card->dai_link[i];
dai_fmt = dai_link->dai_fmt;
if (dai_fmt) {
- ret = snd_soc_dai_set_fmt(card->rtd[i].codec_dai,
- dai_fmt);
- if (ret != 0 && ret != -ENOTSUPP)
- dev_warn(card->rtd[i].codec_dai->dev,
- "ASoC: Failed to set DAI format: %d\n",
- ret);
+ struct snd_soc_dai **codec_dais = rtd->codec_dais;
+ int j;
+
+ for (j = 0; j < rtd->num_codecs; j++) {
+ struct snd_soc_dai *codec_dai = codec_dais[j];
+
+ ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt);
+ if (ret != 0 && ret != -ENOTSUPP)
+ dev_warn(codec_dai->dev,
+ "ASoC: Failed to set DAI format: %d\n",
+ ret);
+ }
}
/* If this is a regular CPU link there will be a platform */
@@ -1927,8 +1930,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
}
if (card->fully_routed)
- list_for_each_entry(codec, &card->codec_dev_list, card_list)
- snd_soc_dapm_auto_nc_codec_pins(codec);
+ snd_soc_dapm_auto_nc_pins(card);
snd_soc_dapm_new_widgets(card);
@@ -2058,10 +2060,15 @@ int snd_soc_poweroff(struct device *dev)
/* deactivate pins to sleep state */
for (i = 0; i < card->num_rtd; i++) {
- struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai;
- struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai;
- pinctrl_pm_select_sleep_state(codec_dai->dev);
+ struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int j;
+
pinctrl_pm_select_sleep_state(cpu_dai->dev);
+ for (j = 0; j < rtd->num_codecs; j++) {
+ struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+ pinctrl_pm_select_sleep_state(codec_dai->dev);
+ }
}
return 0;
@@ -2387,6 +2394,25 @@ struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card,
EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol);
/**
+ * snd_soc_add_component_controls - Add an array of controls to a component.
+ *
+ * @component: Component to add controls to
+ * @controls: Array of controls to add
+ * @num_controls: Number of elements in the array
+ *
+ * Return: 0 for success, else error.
+ */
+int snd_soc_add_component_controls(struct snd_soc_component *component,
+ const struct snd_kcontrol_new *controls, unsigned int num_controls)
+{
+ struct snd_card *card = component->card->snd_card;
+
+ return snd_soc_add_controls(card, component->dev, controls,
+ num_controls, component->name_prefix, component);
+}
+EXPORT_SYMBOL_GPL(snd_soc_add_component_controls);
+
+/**
* snd_soc_add_codec_controls - add an array of controls to a codec.
* Convenience function to add a list of controls. Many codecs were
* duplicating this code.
@@ -2398,12 +2424,10 @@ EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol);
* Return 0 for success, else error.
*/
int snd_soc_add_codec_controls(struct snd_soc_codec *codec,
- const struct snd_kcontrol_new *controls, int num_controls)
+ const struct snd_kcontrol_new *controls, unsigned int num_controls)
{
- struct snd_card *card = codec->card->snd_card;
-
- return snd_soc_add_controls(card, codec->dev, controls, num_controls,
- codec->name_prefix, &codec->component);
+ return snd_soc_add_component_controls(&codec->component, controls,
+ num_controls);
}
EXPORT_SYMBOL_GPL(snd_soc_add_codec_controls);
@@ -2418,12 +2442,10 @@ EXPORT_SYMBOL_GPL(snd_soc_add_codec_controls);
* Return 0 for success, else error.
*/
int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
- const struct snd_kcontrol_new *controls, int num_controls)
+ const struct snd_kcontrol_new *controls, unsigned int num_controls)
{
- struct snd_card *card = platform->card->snd_card;
-
- return snd_soc_add_controls(card, platform->dev, controls, num_controls,
- NULL, &platform->component);
+ return snd_soc_add_component_controls(&platform->component, controls,
+ num_controls);
}
EXPORT_SYMBOL_GPL(snd_soc_add_platform_controls);
@@ -3095,7 +3117,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range);
int snd_soc_limit_volume(struct snd_soc_codec *codec,
const char *name, int max)
{
- struct snd_card *card = codec->card->snd_card;
+ struct snd_card *card = codec->component.card->snd_card;
struct snd_kcontrol *kctl;
struct soc_mixer_control *mc;
int found = 0;
@@ -3267,6 +3289,27 @@ int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol,
}
EXPORT_SYMBOL_GPL(snd_soc_bytes_info_ext);
+int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *tlv)
+{
+ struct soc_bytes_ext *params = (void *)kcontrol->private_value;
+ unsigned int count = size < params->max ? size : params->max;
+ int ret = -ENXIO;
+
+ switch (op_flag) {
+ case SNDRV_CTL_TLV_OP_READ:
+ if (params->get)
+ ret = params->get(tlv, count);
+ break;
+ case SNDRV_CTL_TLV_OP_WRITE:
+ if (params->put)
+ ret = params->put(tlv, count);
+ break;
+ }
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_bytes_tlv_callback);
+
/**
* snd_soc_info_xr_sx - signed multi register info callback
* @kcontrol: mreg control
@@ -3641,6 +3684,9 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
else
snd_soc_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask);
+ dai->tx_mask = tx_mask;
+ dai->rx_mask = rx_mask;
+
if (dai->driver && dai->driver->ops->set_tdm_slot)
return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask,
slots, slot_width);
@@ -3713,6 +3759,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
}
EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
+static int snd_soc_init_multicodec(struct snd_soc_card *card,
+ struct snd_soc_dai_link *dai_link)
+{
+ /* Legacy codec/codec_dai link is a single entry in multicodec */
+ if (dai_link->codec_name || dai_link->codec_of_node ||
+ dai_link->codec_dai_name) {
+ dai_link->num_codecs = 1;
+
+ dai_link->codecs = devm_kzalloc(card->dev,
+ sizeof(struct snd_soc_dai_link_component),
+ GFP_KERNEL);
+ if (!dai_link->codecs)
+ return -ENOMEM;
+
+ dai_link->codecs[0].name = dai_link->codec_name;
+ dai_link->codecs[0].of_node = dai_link->codec_of_node;
+ dai_link->codecs[0].dai_name = dai_link->codec_dai_name;
+ }
+
+ if (!dai_link->codecs) {
+ dev_err(card->dev, "ASoC: DAI link has no CODECs\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
/**
* snd_soc_register_card - Register a card with the ASoC core
*
@@ -3721,7 +3794,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
*/
int snd_soc_register_card(struct snd_soc_card *card)
{
- int i, ret;
+ int i, j, ret;
if (!card->name || !card->dev)
return -EINVAL;
@@ -3729,22 +3802,29 @@ int snd_soc_register_card(struct snd_soc_card *card)
for (i = 0; i < card->num_links; i++) {
struct snd_soc_dai_link *link = &card->dai_link[i];
- /*
- * Codec must be specified by 1 of name or OF node,
- * not both or neither.
- */
- if (!!link->codec_name == !!link->codec_of_node) {
- dev_err(card->dev,
- "ASoC: Neither/both codec name/of_node are set for %s\n",
- link->name);
- return -EINVAL;
+ ret = snd_soc_init_multicodec(card, link);
+ if (ret) {
+ dev_err(card->dev, "ASoC: failed to init multicodec\n");
+ return ret;
}
- /* Codec DAI name must be specified */
- if (!link->codec_dai_name) {
- dev_err(card->dev,
- "ASoC: codec_dai_name not set for %s\n",
- link->name);
- return -EINVAL;
+
+ for (j = 0; j < link->num_codecs; j++) {
+ /*
+ * Codec must be specified by 1 of name or OF node,
+ * not both or neither.
+ */
+ if (!!link->codecs[j].name ==
+ !!link->codecs[j].of_node) {
+ dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n",
+ link->name);
+ return -EINVAL;
+ }
+ /* Codec DAI name must be specified */
+ if (!link->codecs[j].dai_name) {
+ dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n",
+ link->name);
+ return -EINVAL;
+ }
}
/*
@@ -3797,8 +3877,19 @@ int snd_soc_register_card(struct snd_soc_card *card)
card->num_rtd = 0;
card->rtd_aux = &card->rtd[card->num_links];
- for (i = 0; i < card->num_links; i++)
+ for (i = 0; i < card->num_links; i++) {
+ card->rtd[i].card = card;
card->rtd[i].dai_link = &card->dai_link[i];
+ card->rtd[i].codec_dais = devm_kzalloc(card->dev,
+ sizeof(struct snd_soc_dai *) *
+ (card->rtd[i].dai_link->num_codecs),
+ GFP_KERNEL);
+ if (card->rtd[i].codec_dais == NULL)
+ return -ENOMEM;
+ }
+
+ for (i = 0; i < card->num_aux_devs; i++)
+ card->rtd_aux[i].card = card;
INIT_LIST_HEAD(&card->dapm_dirty);
card->instantiated = 0;
@@ -3811,10 +3902,16 @@ int snd_soc_register_card(struct snd_soc_card *card)
/* deactivate pins to sleep state */
for (i = 0; i < card->num_rtd; i++) {
- struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai;
- struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai;
- if (!codec_dai->active)
- pinctrl_pm_select_sleep_state(codec_dai->dev);
+ struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int j;
+
+ for (j = 0; j < rtd->num_codecs; j++) {
+ struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+ if (!codec_dai->active)
+ pinctrl_pm_select_sleep_state(codec_dai->dev);
+ }
+
if (!cpu_dai->active)
pinctrl_pm_select_sleep_state(cpu_dai->dev);
}
@@ -3921,16 +4018,14 @@ static void snd_soc_unregister_dais(struct snd_soc_component *component)
* snd_soc_register_dais - Register a DAI with the ASoC core
*
* @component: The component the DAIs are registered for
- * @codec: The CODEC that the DAIs are registered for, NULL if the component is
- * not a CODEC.
* @dai_drv: DAI driver to use for the DAIs
* @count: Number of DAIs
* @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the
* parent's name.
*/
static int snd_soc_register_dais(struct snd_soc_component *component,
- struct snd_soc_codec *codec, struct snd_soc_dai_driver *dai_drv,
- size_t count, bool legacy_dai_naming)
+ struct snd_soc_dai_driver *dai_drv, size_t count,
+ bool legacy_dai_naming)
{
struct device *dev = component->dev;
struct snd_soc_dai *dai;
@@ -3939,6 +4034,9 @@ static int snd_soc_register_dais(struct snd_soc_component *component,
dev_dbg(dev, "ASoC: dai register %s #%Zu\n", dev_name(dev), count);
+ component->dai_drv = dai_drv;
+ component->num_dai = count;
+
for (i = 0; i < count; i++) {
dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL);
@@ -3971,16 +4069,11 @@ static int snd_soc_register_dais(struct snd_soc_component *component,
}
dai->component = component;
- dai->codec = codec;
dai->dev = dev;
dai->driver = &dai_drv[i];
- dai->dapm.dev = dev;
if (!dai->driver->ops)
dai->driver->ops = &null_dai_ops;
- if (!dai->codec)
- dai->dapm.idle_bias_off = 1;
-
list_add(&dai->list, &component->dai_list);
dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name);
@@ -3994,60 +4087,82 @@ err:
return ret;
}
-/**
- * snd_soc_register_component - Register a component with the ASoC core
- *
- */
-static int
-__snd_soc_register_component(struct device *dev,
- struct snd_soc_component *cmpnt,
- const struct snd_soc_component_driver *cmpnt_drv,
- struct snd_soc_codec *codec,
- struct snd_soc_dai_driver *dai_drv,
- int num_dai, bool allow_single_dai)
+static void snd_soc_component_seq_notifier(struct snd_soc_dapm_context *dapm,
+ enum snd_soc_dapm_type type, int subseq)
{
- int ret;
+ struct snd_soc_component *component = dapm->component;
- dev_dbg(dev, "component register %s\n", dev_name(dev));
+ component->driver->seq_notifier(component, type, subseq);
+}
- if (!cmpnt) {
- dev_err(dev, "ASoC: Failed to connecting component\n");
- return -ENOMEM;
- }
+static int snd_soc_component_stream_event(struct snd_soc_dapm_context *dapm,
+ int event)
+{
+ struct snd_soc_component *component = dapm->component;
- mutex_init(&cmpnt->io_mutex);
+ return component->driver->stream_event(component, event);
+}
+
+static int snd_soc_component_initialize(struct snd_soc_component *component,
+ const struct snd_soc_component_driver *driver, struct device *dev)
+{
+ struct snd_soc_dapm_context *dapm;
- cmpnt->name = fmt_single_name(dev, &cmpnt->id);
- if (!cmpnt->name) {
- dev_err(dev, "ASoC: Failed to simplifying name\n");
+ component->name = fmt_single_name(dev, &component->id);
+ if (!component->name) {
+ dev_err(dev, "ASoC: Failed to allocate name\n");
return -ENOMEM;
}
- cmpnt->dev = dev;
- cmpnt->driver = cmpnt_drv;
- cmpnt->dai_drv = dai_drv;
- cmpnt->num_dai = num_dai;
- INIT_LIST_HEAD(&cmpnt->dai_list);
+ component->dev = dev;
+ component->driver = driver;
- ret = snd_soc_register_dais(cmpnt, codec, dai_drv, num_dai,
- allow_single_dai);
- if (ret < 0) {
- dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret);
- goto error_component_name;
- }
+ if (!component->dapm_ptr)
+ component->dapm_ptr = &component->dapm;
+
+ dapm = component->dapm_ptr;
+ dapm->dev = dev;
+ dapm->component = component;
+ dapm->bias_level = SND_SOC_BIAS_OFF;
+ if (driver->seq_notifier)
+ dapm->seq_notifier = snd_soc_component_seq_notifier;
+ if (driver->stream_event)
+ dapm->stream_event = snd_soc_component_stream_event;
+
+ INIT_LIST_HEAD(&component->dai_list);
+ mutex_init(&component->io_mutex);
+ return 0;
+}
+
+static void snd_soc_component_add_unlocked(struct snd_soc_component *component)
+{
+ list_add(&component->list, &component_list);
+}
+
+static void snd_soc_component_add(struct snd_soc_component *component)
+{
mutex_lock(&client_mutex);
- list_add(&cmpnt->list, &component_list);
+ snd_soc_component_add_unlocked(component);
mutex_unlock(&client_mutex);
+}
- dev_dbg(cmpnt->dev, "ASoC: Registered component '%s'\n", cmpnt->name);
-
- return ret;
+static void snd_soc_component_cleanup(struct snd_soc_component *component)
+{
+ snd_soc_unregister_dais(component);
+ kfree(component->name);
+}
-error_component_name:
- kfree(cmpnt->name);
+static void snd_soc_component_del_unlocked(struct snd_soc_component *component)
+{
+ list_del(&component->list);
+}
- return ret;
+static void snd_soc_component_del(struct snd_soc_component *component)
+{
+ mutex_lock(&client_mutex);
+ snd_soc_component_del_unlocked(component);
+ mutex_unlock(&client_mutex);
}
int snd_soc_register_component(struct device *dev,
@@ -4056,32 +4171,38 @@ int snd_soc_register_component(struct device *dev,
int num_dai)
{
struct snd_soc_component *cmpnt;
+ int ret;
- cmpnt = devm_kzalloc(dev, sizeof(*cmpnt), GFP_KERNEL);
+ cmpnt = kzalloc(sizeof(*cmpnt), GFP_KERNEL);
if (!cmpnt) {
dev_err(dev, "ASoC: Failed to allocate memory\n");
return -ENOMEM;
}
+ ret = snd_soc_component_initialize(cmpnt, cmpnt_drv, dev);
+ if (ret)
+ goto err_free;
+
cmpnt->ignore_pmdown_time = true;
cmpnt->registered_as_component = true;
- return __snd_soc_register_component(dev, cmpnt, cmpnt_drv, NULL,
- dai_drv, num_dai, true);
-}
-EXPORT_SYMBOL_GPL(snd_soc_register_component);
+ ret = snd_soc_register_dais(cmpnt, dai_drv, num_dai, true);
+ if (ret < 0) {
+ dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret);
+ goto err_cleanup;
+ }
-static void __snd_soc_unregister_component(struct snd_soc_component *cmpnt)
-{
- snd_soc_unregister_dais(cmpnt);
+ snd_soc_component_add(cmpnt);
- mutex_lock(&client_mutex);
- list_del(&cmpnt->list);
- mutex_unlock(&client_mutex);
+ return 0;
- dev_dbg(cmpnt->dev, "ASoC: Unregistered component '%s'\n", cmpnt->name);
- kfree(cmpnt->name);
+err_cleanup:
+ snd_soc_component_cleanup(cmpnt);
+err_free:
+ kfree(cmpnt);
+ return ret;
}
+EXPORT_SYMBOL_GPL(snd_soc_register_component);
/**
* snd_soc_unregister_component - Unregister a component from the ASoC core
@@ -4098,7 +4219,9 @@ void snd_soc_unregister_component(struct device *dev)
return;
found:
- __snd_soc_unregister_component(cmpnt);
+ snd_soc_component_del(cmpnt);
+ snd_soc_component_cleanup(cmpnt);
+ kfree(cmpnt);
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_component);
@@ -4131,37 +4254,25 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform,
{
int ret;
- /* create platform component name */
- platform->name = fmt_single_name(dev, &platform->id);
- if (platform->name == NULL)
- return -ENOMEM;
+ ret = snd_soc_component_initialize(&platform->component,
+ &platform_drv->component_driver, dev);
+ if (ret)
+ return ret;
platform->dev = dev;
platform->driver = platform_drv;
- platform->dapm.dev = dev;
- platform->dapm.platform = platform;
- platform->dapm.component = &platform->component;
- platform->dapm.stream_event = platform_drv->stream_event;
if (platform_drv->write)
platform->component.write = snd_soc_platform_drv_write;
if (platform_drv->read)
platform->component.read = snd_soc_platform_drv_read;
- /* register component */
- ret = __snd_soc_register_component(dev, &platform->component,
- &platform_drv->component_driver,
- NULL, NULL, 0, false);
- if (ret < 0) {
- dev_err(platform->component.dev,
- "ASoC: Failed to register component: %d\n", ret);
- return ret;
- }
-
mutex_lock(&client_mutex);
+ snd_soc_component_add_unlocked(&platform->component);
list_add(&platform->list, &platform_list);
mutex_unlock(&client_mutex);
- dev_dbg(dev, "ASoC: Registered platform '%s'\n", platform->name);
+ dev_dbg(dev, "ASoC: Registered platform '%s'\n",
+ platform->component.name);
return 0;
}
@@ -4198,15 +4309,16 @@ EXPORT_SYMBOL_GPL(snd_soc_register_platform);
*/
void snd_soc_remove_platform(struct snd_soc_platform *platform)
{
- __snd_soc_unregister_component(&platform->component);
mutex_lock(&client_mutex);
list_del(&platform->list);
+ snd_soc_component_del_unlocked(&platform->component);
mutex_unlock(&client_mutex);
+ snd_soc_component_cleanup(&platform->component);
+
dev_dbg(platform->dev, "ASoC: Unregistered platform '%s'\n",
- platform->name);
- kfree(platform->name);
+ platform->component.name);
}
EXPORT_SYMBOL_GPL(snd_soc_remove_platform);
@@ -4292,6 +4404,14 @@ static int snd_soc_codec_drv_read(struct snd_soc_component *component,
return 0;
}
+static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
+
+ return codec->driver->set_bias_level(codec, level);
+}
+
/**
* snd_soc_register_codec - Register a codec with the ASoC core
*
@@ -4303,6 +4423,7 @@ int snd_soc_register_codec(struct device *dev,
int num_dai)
{
struct snd_soc_codec *codec;
+ struct snd_soc_dai *dai;
struct regmap *regmap;
int ret, i;
@@ -4312,24 +4433,23 @@ int snd_soc_register_codec(struct device *dev,
if (codec == NULL)
return -ENOMEM;
- /* create CODEC component name */
- codec->name = fmt_single_name(dev, &codec->id);
- if (codec->name == NULL) {
- ret = -ENOMEM;
- goto fail_codec;
- }
+ codec->component.dapm_ptr = &codec->dapm;
+
+ ret = snd_soc_component_initialize(&codec->component,
+ &codec_drv->component_driver, dev);
+ if (ret)
+ goto err_free;
if (codec_drv->write)
codec->component.write = snd_soc_codec_drv_write;
if (codec_drv->read)
codec->component.read = snd_soc_codec_drv_read;
codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time;
- codec->dapm.bias_level = SND_SOC_BIAS_OFF;
- codec->dapm.dev = dev;
codec->dapm.codec = codec;
- codec->dapm.component = &codec->component;
- codec->dapm.seq_notifier = codec_drv->seq_notifier;
- codec->dapm.stream_event = codec_drv->stream_event;
+ if (codec_drv->seq_notifier)
+ codec->dapm.seq_notifier = codec_drv->seq_notifier;
+ if (codec_drv->set_bias_level)
+ codec->dapm.set_bias_level = snd_soc_codec_set_bias_level;
codec->dev = dev;
codec->driver = codec_drv;
codec->component.val_bytes = codec_drv->reg_word_size;
@@ -4348,7 +4468,7 @@ int snd_soc_register_codec(struct device *dev,
dev_err(codec->dev,
"Failed to set cache I/O:%d\n",
ret);
- return ret;
+ goto err_cleanup;
}
}
}
@@ -4358,29 +4478,27 @@ int snd_soc_register_codec(struct device *dev,
fixup_codec_formats(&dai_drv[i].capture);
}
- mutex_lock(&client_mutex);
- list_add(&codec->list, &codec_list);
- mutex_unlock(&client_mutex);
-
- /* register component */
- ret = __snd_soc_register_component(dev, &codec->component,
- &codec_drv->component_driver,
- codec, dai_drv, num_dai, false);
+ ret = snd_soc_register_dais(&codec->component, dai_drv, num_dai, false);
if (ret < 0) {
- dev_err(codec->dev, "ASoC: Failed to regster component: %d\n", ret);
- goto fail_codec_name;
+ dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret);
+ goto err_cleanup;
}
- dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name);
- return 0;
+ list_for_each_entry(dai, &codec->component.dai_list, list)
+ dai->codec = codec;
-fail_codec_name:
mutex_lock(&client_mutex);
- list_del(&codec->list);
+ snd_soc_component_add_unlocked(&codec->component);
+ list_add(&codec->list, &codec_list);
mutex_unlock(&client_mutex);
- kfree(codec->name);
-fail_codec:
+ dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n",
+ codec->component.name);
+ return 0;
+
+err_cleanup:
+ snd_soc_component_cleanup(&codec->component);
+err_free:
kfree(codec);
return ret;
}
@@ -4402,16 +4520,17 @@ void snd_soc_unregister_codec(struct device *dev)
return;
found:
- __snd_soc_unregister_component(&codec->component);
mutex_lock(&client_mutex);
list_del(&codec->list);
+ snd_soc_component_del_unlocked(&codec->component);
mutex_unlock(&client_mutex);
- dev_dbg(codec->dev, "ASoC: Unregistered codec '%s'\n", codec->name);
+ dev_dbg(codec->dev, "ASoC: Unregistered codec '%s'\n",
+ codec->component.name);
+ snd_soc_component_cleanup(&codec->component);
snd_soc_cache_exit(codec);
- kfree(codec->name);
kfree(codec);
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_codec);
@@ -4420,9 +4539,16 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_codec);
int snd_soc_of_parse_card_name(struct snd_soc_card *card,
const char *propname)
{
- struct device_node *np = card->dev->of_node;
+ struct device_node *np;
int ret;
+ if (!card->dev) {
+ pr_err("card->dev is not set before calling %s\n", __func__);
+ return -EINVAL;
+ }
+
+ np = card->dev->of_node;
+
ret = of_property_read_string_index(np, propname, 0, &card->name);
/*
* EINVAL means the property does not exist. This is fine providing
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index cdc837ed144d..8348352dc2c6 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -350,12 +350,27 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol,
}
/**
+ * snd_soc_dapm_kcontrol_dapm() - Returns the dapm context associated to a
+ * kcontrol
+ * @kcontrol: The kcontrol
+ *
+ * Note: This function must only be used on kcontrols that are known to have
+ * been registered for a CODEC. Otherwise the behaviour is undefined.
+ */
+struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm(
+ struct snd_kcontrol *kcontrol)
+{
+ return dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->dapm;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_dapm);
+
+/**
* snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol
* @kcontrol: The kcontrol
*/
struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol)
{
- return dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->codec;
+ return snd_soc_dapm_to_codec(snd_soc_dapm_kcontrol_dapm(kcontrol));
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_codec);
@@ -375,23 +390,38 @@ static void dapm_reset(struct snd_soc_card *card)
}
}
-static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg,
+static const char *soc_dapm_prefix(struct snd_soc_dapm_context *dapm)
+{
+ if (!dapm->component)
+ return NULL;
+ return dapm->component->name_prefix;
+}
+
+static int soc_dapm_read(struct snd_soc_dapm_context *dapm, int reg,
unsigned int *value)
{
- if (!w->dapm->component)
+ if (!dapm->component)
return -EIO;
- return snd_soc_component_read(w->dapm->component, reg, value);
+ return snd_soc_component_read(dapm->component, reg, value);
}
-static int soc_widget_update_bits(struct snd_soc_dapm_widget *w,
+static int soc_dapm_update_bits(struct snd_soc_dapm_context *dapm,
int reg, unsigned int mask, unsigned int value)
{
- if (!w->dapm->component)
+ if (!dapm->component)
return -EIO;
- return snd_soc_component_update_bits_async(w->dapm->component, reg,
+ return snd_soc_component_update_bits_async(dapm->component, reg,
mask, value);
}
+static int soc_dapm_test_bits(struct snd_soc_dapm_context *dapm,
+ int reg, unsigned int mask, unsigned int value)
+{
+ if (!dapm->component)
+ return -EIO;
+ return snd_soc_component_test_bits(dapm->component, reg, mask, value);
+}
+
static void soc_dapm_async_complete(struct snd_soc_dapm_context *dapm)
{
if (dapm->component)
@@ -420,15 +450,10 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
if (ret != 0)
goto out;
- if (dapm->codec) {
- if (dapm->codec->driver->set_bias_level)
- ret = dapm->codec->driver->set_bias_level(dapm->codec,
- level);
- else
- dapm->bias_level = level;
- } else if (!card || dapm != &card->dapm) {
+ if (dapm->set_bias_level)
+ ret = dapm->set_bias_level(dapm, level);
+ else if (!card || dapm != &card->dapm)
dapm->bias_level = level;
- }
if (ret != 0)
goto out;
@@ -452,7 +477,7 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm,
int i;
if (e->reg != SND_SOC_NOPM) {
- soc_widget_read(dest, e->reg, &val);
+ soc_dapm_read(dapm, e->reg, &val);
val = (val >> e->shift_l) & e->mask;
item = snd_soc_enum_val_to_item(e, val);
} else {
@@ -496,7 +521,7 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_widget *w,
unsigned int val;
if (reg != SND_SOC_NOPM) {
- soc_widget_read(w, reg, &val);
+ soc_dapm_read(w->dapm, reg, &val);
val = (val >> shift) & mask;
if (invert)
val = max - val;
@@ -570,11 +595,7 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
const char *name;
int ret;
- if (dapm->codec)
- prefix = dapm->codec->name_prefix;
- else
- prefix = NULL;
-
+ prefix = soc_dapm_prefix(dapm);
if (prefix)
prefix_len = strlen(prefix) + 1;
else
@@ -1308,16 +1329,18 @@ static void dapm_seq_check_event(struct snd_soc_card *card,
static void dapm_seq_run_coalesced(struct snd_soc_card *card,
struct list_head *pending)
{
+ struct snd_soc_dapm_context *dapm;
struct snd_soc_dapm_widget *w;
int reg;
unsigned int value = 0;
unsigned int mask = 0;
- reg = list_first_entry(pending, struct snd_soc_dapm_widget,
- power_list)->reg;
+ w = list_first_entry(pending, struct snd_soc_dapm_widget, power_list);
+ reg = w->reg;
+ dapm = w->dapm;
list_for_each_entry(w, pending, power_list) {
- WARN_ON(reg != w->reg);
+ WARN_ON(reg != w->reg || dapm != w->dapm);
w->power = w->new_power;
mask |= w->mask << w->shift;
@@ -1326,7 +1349,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card,
else
value |= w->off_val << w->shift;
- pop_dbg(w->dapm->dev, card->pop_time,
+ pop_dbg(dapm->dev, card->pop_time,
"pop test : Queue %s: reg=0x%x, 0x%x/0x%x\n",
w->name, reg, value, mask);
@@ -1339,14 +1362,12 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card,
/* Any widget will do, they should all be updating the
* same register.
*/
- w = list_first_entry(pending, struct snd_soc_dapm_widget,
- power_list);
- pop_dbg(w->dapm->dev, card->pop_time,
+ pop_dbg(dapm->dev, card->pop_time,
"pop test : Applying 0x%x/0x%x to %x in %dms\n",
value, mask, reg, card->pop_time);
pop_wait(card->pop_time);
- soc_widget_update_bits(w, reg, mask, value);
+ soc_dapm_update_bits(dapm, reg, mask, value);
}
list_for_each_entry(w, pending, power_list) {
@@ -1492,7 +1513,8 @@ static void dapm_widget_update(struct snd_soc_card *card)
if (!w)
return;
- ret = soc_widget_update_bits(w, update->reg, update->mask, update->val);
+ ret = soc_dapm_update_bits(w->dapm, update->reg, update->mask,
+ update->val);
if (ret < 0)
dev_err(w->dapm->dev, "ASoC: %s DAPM update failed: %d\n",
w->name, ret);
@@ -2062,17 +2084,13 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm,
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power);
-/* show dapm widget status in sys fs */
-static ssize_t dapm_widget_show(struct device *dev,
- struct device_attribute *attr, char *buf)
+static ssize_t dapm_widget_show_codec(struct snd_soc_codec *codec, char *buf)
{
- struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
- struct snd_soc_codec *codec =rtd->codec;
struct snd_soc_dapm_widget *w;
int count = 0;
char *state = "not set";
- list_for_each_entry(w, &codec->card->widgets, list) {
+ list_for_each_entry(w, &codec->component.card->widgets, list) {
if (w->dapm != &codec->dapm)
continue;
@@ -2120,6 +2138,21 @@ static ssize_t dapm_widget_show(struct device *dev,
return count;
}
+/* show dapm widget status in sys fs */
+static ssize_t dapm_widget_show(struct device *dev,
+ struct device_attribute *attr, char *buf)
+{
+ struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
+ int i, count = 0;
+
+ for (i = 0; i < rtd->num_codecs; i++) {
+ struct snd_soc_codec *codec = rtd->codec_dais[i]->codec;
+ count += dapm_widget_show_codec(codec, buf + count);
+ }
+
+ return count;
+}
+
static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL);
int snd_soc_dapm_sys_add(struct device *dev)
@@ -2371,14 +2404,16 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
const char *source;
char prefixed_sink[80];
char prefixed_source[80];
+ const char *prefix;
int ret;
- if (dapm->codec && dapm->codec->name_prefix) {
+ prefix = soc_dapm_prefix(dapm);
+ if (prefix) {
snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s",
- dapm->codec->name_prefix, route->sink);
+ prefix, route->sink);
sink = prefixed_sink;
snprintf(prefixed_source, sizeof(prefixed_source), "%s %s",
- dapm->codec->name_prefix, route->source);
+ prefix, route->source);
source = prefixed_source;
} else {
sink = route->sink;
@@ -2439,6 +2474,7 @@ static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm,
const char *source;
char prefixed_sink[80];
char prefixed_source[80];
+ const char *prefix;
if (route->control) {
dev_err(dapm->dev,
@@ -2446,12 +2482,13 @@ static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm,
return -EINVAL;
}
- if (dapm->codec && dapm->codec->name_prefix) {
+ prefix = soc_dapm_prefix(dapm);
+ if (prefix) {
snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s",
- dapm->codec->name_prefix, route->sink);
+ prefix, route->sink);
sink = prefixed_sink;
snprintf(prefixed_source, sizeof(prefixed_source), "%s %s",
- dapm->codec->name_prefix, route->source);
+ prefix, route->source);
source = prefixed_source;
} else {
sink = route->sink;
@@ -2670,7 +2707,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card)
/* Read the initial power state from the device */
if (w->reg >= 0) {
- soc_widget_read(w, w->reg, &val);
+ soc_dapm_read(w->dapm, w->reg, &val);
val = val >> w->shift;
val &= w->mask;
if (val == w->on_val)
@@ -2701,8 +2738,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets);
int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
- struct snd_soc_card *card = codec->card;
+ struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct snd_soc_card *card = dapm->card;
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
int reg = mc->reg;
@@ -2711,17 +2748,20 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
unsigned int val;
+ int ret = 0;
if (snd_soc_volsw_is_stereo(mc))
- dev_warn(codec->dapm.dev,
+ dev_warn(dapm->dev,
"ASoC: Control '%s' is stereo, which is not supported\n",
kcontrol->id.name);
mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
- if (dapm_kcontrol_is_powered(kcontrol) && reg != SND_SOC_NOPM)
- val = (snd_soc_read(codec, reg) >> shift) & mask;
- else
+ if (dapm_kcontrol_is_powered(kcontrol) && reg != SND_SOC_NOPM) {
+ ret = soc_dapm_read(dapm, reg, &val);
+ val = (val >> shift) & mask;
+ } else {
val = dapm_kcontrol_get_value(kcontrol);
+ }
mutex_unlock(&card->dapm_mutex);
if (invert)
@@ -2729,7 +2769,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
else
ucontrol->value.integer.value[0] = val;
- return 0;
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw);
@@ -2745,8 +2785,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw);
int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
- struct snd_soc_card *card = codec->card;
+ struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct snd_soc_card *card = dapm->card;
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
int reg = mc->reg;
@@ -2760,7 +2800,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
int ret = 0;
if (snd_soc_volsw_is_stereo(mc))
- dev_warn(codec->dapm.dev,
+ dev_warn(dapm->dev,
"ASoC: Control '%s' is stereo, which is not supported\n",
kcontrol->id.name);
@@ -2778,7 +2818,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
mask = mask << shift;
val = val << shift;
- reg_change = snd_soc_test_bits(codec, reg, mask, val);
+ reg_change = soc_dapm_test_bits(dapm, reg, mask, val);
}
if (change || reg_change) {
@@ -2817,12 +2857,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw);
int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
unsigned int reg_val, val;
+ int ret = 0;
if (e->reg != SND_SOC_NOPM)
- reg_val = snd_soc_read(codec, e->reg);
+ ret = soc_dapm_read(dapm, e->reg, &reg_val);
else
reg_val = dapm_kcontrol_get_value(kcontrol);
@@ -2834,7 +2875,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
ucontrol->value.enumerated.item[1] = val;
}
- return 0;
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);
@@ -2850,8 +2891,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);
int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
- struct snd_soc_card *card = codec->card;
+ struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct snd_soc_card *card = dapm->card;
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
unsigned int *item = ucontrol->value.enumerated.item;
unsigned int val, change;
@@ -2874,7 +2915,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
if (e->reg != SND_SOC_NOPM)
- change = snd_soc_test_bits(codec, e->reg, mask, val);
+ change = soc_dapm_test_bits(dapm, e->reg, mask, val);
else
change = dapm_kcontrol_set_value(kcontrol, val);
@@ -2971,6 +3012,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_widget *widget)
{
struct snd_soc_dapm_widget *w;
+ const char *prefix;
int ret;
if ((w = dapm_cnew_widget(widget)) == NULL)
@@ -3011,9 +3053,9 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
break;
}
- if (dapm->codec && dapm->codec->name_prefix)
- w->name = kasprintf(GFP_KERNEL, "%s %s",
- dapm->codec->name_prefix, widget->name);
+ prefix = soc_dapm_prefix(dapm);
+ if (prefix)
+ w->name = kasprintf(GFP_KERNEL, "%s %s", prefix, widget->name);
else
w->name = kasprintf(GFP_KERNEL, "%s", widget->name);
@@ -3066,7 +3108,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
w->dapm = dapm;
w->codec = dapm->codec;
- w->platform = dapm->platform;
INIT_LIST_HEAD(&w->sources);
INIT_LIST_HEAD(&w->sinks);
INIT_LIST_HEAD(&w->list);
@@ -3173,27 +3214,15 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
- if (source->driver->ops && source->driver->ops->hw_params) {
- substream.stream = SNDRV_PCM_STREAM_CAPTURE;
- ret = source->driver->ops->hw_params(&substream,
- params, source);
- if (ret != 0) {
- dev_err(source->dev,
- "ASoC: hw_params() failed: %d\n", ret);
- goto out;
- }
- }
+ substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+ ret = soc_dai_hw_params(&substream, params, source);
+ if (ret < 0)
+ goto out;
- if (sink->driver->ops && sink->driver->ops->hw_params) {
- substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
- ret = sink->driver->ops->hw_params(&substream, params,
- sink);
- if (ret != 0) {
- dev_err(sink->dev,
- "ASoC: hw_params() failed: %d\n", ret);
- goto out;
- }
- }
+ substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+ ret = soc_dai_hw_params(&substream, params, sink);
+ if (ret < 0)
+ goto out;
break;
case SND_SOC_DAPM_POST_PMU:
@@ -3365,25 +3394,15 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
return 0;
}
-void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
+static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
+ struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_pcm_runtime *rtd = card->rtd;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dapm_widget *sink, *source;
- struct snd_soc_dai *cpu_dai, *codec_dai;
int i;
- /* for each BE DAI link... */
- for (i = 0; i < card->num_rtd; i++) {
- rtd = &card->rtd[i];
- cpu_dai = rtd->cpu_dai;
- codec_dai = rtd->codec_dai;
-
- /*
- * dynamic FE links have no fixed DAI mapping.
- * CODEC<->CODEC links have no direct connection.
- */
- if (rtd->dai_link->dynamic || rtd->dai_link->params)
- continue;
+ for (i = 0; i < rtd->num_codecs; i++) {
+ struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
/* there is no point in connecting BE DAI links with dummies */
if (snd_soc_dai_is_dummy(codec_dai) ||
@@ -3395,8 +3414,8 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
source = cpu_dai->playback_widget;
sink = codec_dai->playback_widget;
dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
- cpu_dai->codec->name, source->name,
- codec_dai->platform->name, sink->name);
+ cpu_dai->component->name, source->name,
+ codec_dai->component->name, sink->name);
snd_soc_dapm_add_path(&card->dapm, source, sink,
NULL, NULL);
@@ -3407,8 +3426,8 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
source = codec_dai->capture_widget;
sink = cpu_dai->capture_widget;
dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
- codec_dai->codec->name, source->name,
- cpu_dai->platform->name, sink->name);
+ codec_dai->component->name, source->name,
+ cpu_dai->component->name, sink->name);
snd_soc_dapm_add_path(&card->dapm, source, sink,
NULL, NULL);
@@ -3445,11 +3464,34 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream,
}
}
+void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
+{
+ struct snd_soc_pcm_runtime *rtd = card->rtd;
+ int i;
+
+ /* for each BE DAI link... */
+ for (i = 0; i < card->num_rtd; i++) {
+ rtd = &card->rtd[i];
+
+ /*
+ * dynamic FE links have no fixed DAI mapping.
+ * CODEC<->CODEC links have no direct connection.
+ */
+ if (rtd->dai_link->dynamic || rtd->dai_link->params)
+ continue;
+
+ dapm_connect_dai_link_widgets(card, rtd);
+ }
+}
+
static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
int event)
{
+ int i;
+
soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event);
- soc_dapm_dai_stream_event(rtd->codec_dai, stream, event);
+ for (i = 0; i < rtd->num_codecs; i++)
+ soc_dapm_dai_stream_event(rtd->codec_dais[i], stream, event);
dapm_power_widgets(rtd->card, event);
}
@@ -3758,36 +3800,31 @@ static bool snd_soc_dapm_widget_in_card_paths(struct snd_soc_card *card,
}
/**
- * snd_soc_dapm_auto_nc_codec_pins - call snd_soc_dapm_nc_pin for unused pins
- * @codec: The codec whose pins should be processed
+ * snd_soc_dapm_auto_nc_pins - call snd_soc_dapm_nc_pin for unused pins
+ * @card: The card whose pins should be processed
*
- * Automatically call snd_soc_dapm_nc_pin() for any external pins in the codec
- * which are unused. Pins are used if they are connected externally to the
- * codec, whether that be to some other device, or a loop-back connection to
- * the codec itself.
+ * Automatically call snd_soc_dapm_nc_pin() for any external pins in the card
+ * which are unused. Pins are used if they are connected externally to a
+ * component, whether that be to some other device, or a loop-back connection to
+ * the component itself.
*/
-void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec)
+void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card)
{
- struct snd_soc_card *card = codec->card;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
struct snd_soc_dapm_widget *w;
- dev_dbg(codec->dev, "ASoC: Auto NC: DAPMs: card:%p codec:%p\n",
- &card->dapm, &codec->dapm);
+ dev_dbg(card->dev, "ASoC: Auto NC: DAPMs: card:%p\n", &card->dapm);
list_for_each_entry(w, &card->widgets, list) {
- if (w->dapm != dapm)
- continue;
switch (w->id) {
case snd_soc_dapm_input:
case snd_soc_dapm_output:
case snd_soc_dapm_micbias:
- dev_dbg(codec->dev, "ASoC: Auto NC: Checking widget %s\n",
+ dev_dbg(card->dev, "ASoC: Auto NC: Checking widget %s\n",
w->name);
if (!snd_soc_dapm_widget_in_card_paths(card, w)) {
- dev_dbg(codec->dev,
+ dev_dbg(card->dev,
"... Not in map; disabling\n");
- snd_soc_dapm_nc_pin(dapm, w->name);
+ snd_soc_dapm_nc_pin(w->dapm, w->name);
}
break;
default:
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 5bace124ef43..6307f85e871b 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -119,7 +119,10 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea
struct snd_dmaengine_dai_dma_data *dma_data;
struct dma_slave_caps dma_caps;
struct snd_pcm_hardware hw;
- int ret;
+ u32 addr_widths = BIT(DMA_SLAVE_BUSWIDTH_1_BYTE) |
+ BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) |
+ BIT(DMA_SLAVE_BUSWIDTH_4_BYTES);
+ int i, ret;
if (pcm->config && pcm->config->pcm_hardware)
return snd_soc_set_runtime_hwparams(substream,
@@ -146,6 +149,38 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea
hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME;
if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT)
hw.info |= SNDRV_PCM_INFO_BATCH;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ addr_widths = dma_caps.dstn_addr_widths;
+ else
+ addr_widths = dma_caps.src_addr_widths;
+ }
+
+ /*
+ * Prepare formats mask for valid/allowed sample types. If the dma does
+ * not have support for the given physical word size, it needs to be
+ * masked out so user space can not use the format which produces
+ * corrupted audio.
+ * In case the dma driver does not implement the slave_caps the default
+ * assumption is that it supports 1, 2 and 4 bytes widths.
+ */
+ for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+ int bits = snd_pcm_format_physical_width(i);
+
+ /* Enable only samples with DMA supported physical widths */
+ switch (bits) {
+ case 8:
+ case 16:
+ case 24:
+ case 32:
+ case 64:
+ if (addr_widths & (1 << (bits / 8)))
+ hw.formats |= (1LL << i);
+ break;
+ default:
+ /* Unsupported types */
+ break;
+ }
}
return snd_soc_set_runtime_hwparams(substream, &hw);
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index d0d98810af91..ab47fea997a3 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -43,7 +43,7 @@ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type,
INIT_LIST_HEAD(&jack->jack_zones);
BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier);
- return snd_jack_new(codec->card->snd_card, id, type, &jack->jack);
+ return snd_jack_new(codec->component.card->snd_card, id, type, &jack->jack);
}
EXPORT_SYMBOL_GPL(snd_soc_jack_new);
@@ -260,7 +260,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio)
static irqreturn_t gpio_handler(int irq, void *data)
{
struct snd_soc_jack_gpio *gpio = data;
- struct device *dev = gpio->jack->codec->card->dev;
+ struct device *dev = gpio->jack->codec->component.card->dev;
trace_snd_soc_jack_irq(gpio->name);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 54d18f22a33e..731fdb5b5f9b 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -7,7 +7,7 @@
* Copyright (C) 2010 Texas Instruments Inc.
*
* Authors: Liam Girdwood <lrg@ti.com>
- * Mark Brown <broonie@opensource.wolfsonmicro.com>
+ * 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 as published by the
@@ -47,22 +47,26 @@
void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
{
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int i;
lockdep_assert_held(&rtd->pcm_mutex);
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
cpu_dai->playback_active++;
- codec_dai->playback_active++;
+ for (i = 0; i < rtd->num_codecs; i++)
+ rtd->codec_dais[i]->playback_active++;
} else {
cpu_dai->capture_active++;
- codec_dai->capture_active++;
+ for (i = 0; i < rtd->num_codecs; i++)
+ rtd->codec_dais[i]->capture_active++;
}
cpu_dai->active++;
- codec_dai->active++;
cpu_dai->component->active++;
- codec_dai->component->active++;
+ for (i = 0; i < rtd->num_codecs; i++) {
+ rtd->codec_dais[i]->active++;
+ rtd->codec_dais[i]->component->active++;
+ }
}
/**
@@ -78,22 +82,26 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream)
{
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int i;
lockdep_assert_held(&rtd->pcm_mutex);
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
cpu_dai->playback_active--;
- codec_dai->playback_active--;
+ for (i = 0; i < rtd->num_codecs; i++)
+ rtd->codec_dais[i]->playback_active--;
} else {
cpu_dai->capture_active--;
- codec_dai->capture_active--;
+ for (i = 0; i < rtd->num_codecs; i++)
+ rtd->codec_dais[i]->capture_active--;
}
cpu_dai->active--;
- codec_dai->active--;
cpu_dai->component->active--;
- codec_dai->component->active--;
+ for (i = 0; i < rtd->num_codecs; i++) {
+ rtd->codec_dais[i]->component->active--;
+ rtd->codec_dais[i]->active--;
+ }
}
/**
@@ -107,11 +115,16 @@ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream)
*/
bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd)
{
+ int i;
+ bool ignore = true;
+
if (!rtd->pmdown_time || rtd->dai_link->ignore_pmdown_time)
return true;
- return rtd->cpu_dai->component->ignore_pmdown_time &&
- rtd->codec_dai->component->ignore_pmdown_time;
+ for (i = 0; i < rtd->num_codecs; i++)
+ ignore &= rtd->codec_dais[i]->component->ignore_pmdown_time;
+
+ return rtd->cpu_dai->component->ignore_pmdown_time && ignore;
}
/**
@@ -222,8 +235,7 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
{
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 rate, channels, sample_bits, symmetry;
+ unsigned int rate, channels, sample_bits, symmetry, i;
rate = params_rate(params);
channels = params_channels(params);
@@ -231,8 +243,11 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
/* reject unmatched parameters when applying symmetry */
symmetry = cpu_dai->driver->symmetric_rates ||
- codec_dai->driver->symmetric_rates ||
rtd->dai_link->symmetric_rates;
+
+ for (i = 0; i < rtd->num_codecs; i++)
+ symmetry |= rtd->codec_dais[i]->driver->symmetric_rates;
+
if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) {
dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n",
cpu_dai->rate, rate);
@@ -240,8 +255,11 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
}
symmetry = cpu_dai->driver->symmetric_channels ||
- codec_dai->driver->symmetric_channels ||
rtd->dai_link->symmetric_channels;
+
+ for (i = 0; i < rtd->num_codecs; i++)
+ symmetry |= rtd->codec_dais[i]->driver->symmetric_channels;
+
if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) {
dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n",
cpu_dai->channels, channels);
@@ -249,8 +267,11 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
}
symmetry = cpu_dai->driver->symmetric_samplebits ||
- codec_dai->driver->symmetric_samplebits ||
rtd->dai_link->symmetric_samplebits;
+
+ for (i = 0; i < rtd->num_codecs; i++)
+ symmetry |= rtd->codec_dais[i]->driver->symmetric_samplebits;
+
if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) {
dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n",
cpu_dai->sample_bits, sample_bits);
@@ -264,15 +285,20 @@ static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver;
- struct snd_soc_dai_driver *codec_driver = rtd->codec_dai->driver;
struct snd_soc_dai_link *link = rtd->dai_link;
+ unsigned int symmetry, i;
- return cpu_driver->symmetric_rates || codec_driver->symmetric_rates ||
- link->symmetric_rates || cpu_driver->symmetric_channels ||
- codec_driver->symmetric_channels || link->symmetric_channels ||
- cpu_driver->symmetric_samplebits ||
- codec_driver->symmetric_samplebits ||
- link->symmetric_samplebits;
+ symmetry = cpu_driver->symmetric_rates || link->symmetric_rates ||
+ cpu_driver->symmetric_channels || link->symmetric_channels ||
+ cpu_driver->symmetric_samplebits || link->symmetric_samplebits;
+
+ for (i = 0; i < rtd->num_codecs; i++)
+ symmetry = symmetry ||
+ rtd->codec_dais[i]->driver->symmetric_rates ||
+ rtd->codec_dais[i]->driver->symmetric_channels ||
+ rtd->codec_dais[i]->driver->symmetric_samplebits;
+
+ return symmetry;
}
/*
@@ -284,15 +310,10 @@ static int sample_sizes[] = {
24, 32,
};
-static void soc_pcm_apply_msb(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
+static void soc_pcm_set_msb(struct snd_pcm_substream *substream, int bits)
{
- int ret, i, bits;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- bits = dai->driver->playback.sig_bits;
- else
- bits = dai->driver->capture.sig_bits;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int ret, i;
if (!bits)
return;
@@ -304,38 +325,105 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream,
ret = snd_pcm_hw_constraint_msbits(substream->runtime, 0,
sample_sizes[i], bits);
if (ret != 0)
- dev_warn(dai->dev,
+ dev_warn(rtd->dev,
"ASoC: Failed to set MSB %d/%d: %d\n",
bits, sample_sizes[i], ret);
}
}
-static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
- struct snd_soc_pcm_stream *codec_stream,
- struct snd_soc_pcm_stream *cpu_stream)
+static void soc_pcm_apply_msb(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai;
+ int i;
+ unsigned int bits = 0, cpu_bits;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec_dai = rtd->codec_dais[i];
+ if (codec_dai->driver->playback.sig_bits == 0) {
+ bits = 0;
+ break;
+ }
+ bits = max(codec_dai->driver->playback.sig_bits, bits);
+ }
+ cpu_bits = cpu_dai->driver->playback.sig_bits;
+ } else {
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec_dai = rtd->codec_dais[i];
+ if (codec_dai->driver->playback.sig_bits == 0) {
+ bits = 0;
+ break;
+ }
+ bits = max(codec_dai->driver->capture.sig_bits, bits);
+ }
+ cpu_bits = cpu_dai->driver->capture.sig_bits;
+ }
+
+ soc_pcm_set_msb(substream, bits);
+ soc_pcm_set_msb(substream, cpu_bits);
+}
+
+static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
{
+ struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_pcm_hardware *hw = &runtime->hw;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai_driver *cpu_dai_drv = rtd->cpu_dai->driver;
+ struct snd_soc_dai_driver *codec_dai_drv;
+ struct snd_soc_pcm_stream *codec_stream;
+ struct snd_soc_pcm_stream *cpu_stream;
+ unsigned int chan_min = 0, chan_max = UINT_MAX;
+ unsigned int rate_min = 0, rate_max = UINT_MAX;
+ unsigned int rates = UINT_MAX;
+ u64 formats = ULLONG_MAX;
+ int i;
- hw->channels_min = max(codec_stream->channels_min,
- cpu_stream->channels_min);
- hw->channels_max = min(codec_stream->channels_max,
- cpu_stream->channels_max);
- if (hw->formats)
- hw->formats &= codec_stream->formats & cpu_stream->formats;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_stream = &cpu_dai_drv->playback;
else
- hw->formats = codec_stream->formats & cpu_stream->formats;
- hw->rates = snd_pcm_rate_mask_intersect(codec_stream->rates,
- cpu_stream->rates);
+ cpu_stream = &cpu_dai_drv->capture;
- hw->rate_min = 0;
- hw->rate_max = UINT_MAX;
+ /* first calculate min/max only for CODECs in the DAI link */
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec_dai_drv = rtd->codec_dais[i]->driver;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ codec_stream = &codec_dai_drv->playback;
+ else
+ codec_stream = &codec_dai_drv->capture;
+ chan_min = max(chan_min, codec_stream->channels_min);
+ chan_max = min(chan_max, codec_stream->channels_max);
+ rate_min = max(rate_min, codec_stream->rate_min);
+ rate_max = min_not_zero(rate_max, codec_stream->rate_max);
+ formats &= codec_stream->formats;
+ rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates);
+ }
+
+ /*
+ * chan min/max cannot be enforced if there are multiple CODEC DAIs
+ * connected to a single CPU DAI, use CPU DAI's directly and let
+ * channel allocation be fixed up later
+ */
+ if (rtd->num_codecs > 1) {
+ chan_min = cpu_stream->channels_min;
+ chan_max = cpu_stream->channels_max;
+ }
+
+ hw->channels_min = max(chan_min, cpu_stream->channels_min);
+ hw->channels_max = min(chan_max, cpu_stream->channels_max);
+ if (hw->formats)
+ hw->formats &= formats & cpu_stream->formats;
+ else
+ hw->formats = formats & cpu_stream->formats;
+ hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_stream->rates);
snd_pcm_limit_hw_rates(runtime);
hw->rate_min = max(hw->rate_min, cpu_stream->rate_min);
- hw->rate_min = max(hw->rate_min, codec_stream->rate_min);
+ hw->rate_min = max(hw->rate_min, rate_min);
hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max);
- hw->rate_max = min_not_zero(hw->rate_max, codec_stream->rate_max);
+ hw->rate_max = min_not_zero(hw->rate_max, rate_max);
}
/*
@@ -349,15 +437,16 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
- struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver;
- int ret = 0;
+ struct snd_soc_dai *codec_dai;
+ const char *codec_dai_name = "multicodec";
+ int i, ret = 0;
pinctrl_pm_select_default_state(cpu_dai->dev);
- pinctrl_pm_select_default_state(codec_dai->dev);
+ for (i = 0; i < rtd->num_codecs; i++)
+ pinctrl_pm_select_default_state(rtd->codec_dais[i]->dev);
pm_runtime_get_sync(cpu_dai->dev);
- pm_runtime_get_sync(codec_dai->dev);
+ for (i = 0; i < rtd->num_codecs; i++)
+ pm_runtime_get_sync(rtd->codec_dais[i]->dev);
pm_runtime_get_sync(platform->dev);
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
@@ -376,18 +465,28 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
ret = platform->driver->ops->open(substream);
if (ret < 0) {
dev_err(platform->dev, "ASoC: can't open platform"
- " %s: %d\n", platform->name, ret);
+ " %s: %d\n", platform->component.name, ret);
goto platform_err;
}
}
- if (codec_dai->driver->ops && codec_dai->driver->ops->startup) {
- ret = codec_dai->driver->ops->startup(substream, codec_dai);
- if (ret < 0) {
- dev_err(codec_dai->dev, "ASoC: can't open codec"
- " %s: %d\n", codec_dai->name, ret);
- goto codec_dai_err;
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec_dai = rtd->codec_dais[i];
+ if (codec_dai->driver->ops && codec_dai->driver->ops->startup) {
+ ret = codec_dai->driver->ops->startup(substream,
+ codec_dai);
+ if (ret < 0) {
+ dev_err(codec_dai->dev,
+ "ASoC: can't open codec %s: %d\n",
+ codec_dai->name, ret);
+ goto codec_dai_err;
+ }
}
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ codec_dai->tx_mask = 0;
+ else
+ codec_dai->rx_mask = 0;
}
if (rtd->dai_link->ops && rtd->dai_link->ops->startup) {
@@ -404,13 +503,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
goto dynamic;
/* Check that the codec and cpu DAIs are compatible */
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->playback,
- &cpu_dai_drv->playback);
- } else {
- soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->capture,
- &cpu_dai_drv->capture);
- }
+ soc_pcm_init_runtime_hw(substream);
+
+ if (rtd->num_codecs == 1)
+ codec_dai_name = rtd->codec_dai->name;
if (soc_pcm_has_symmetry(substream))
runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX;
@@ -418,23 +514,22 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
ret = -EINVAL;
if (!runtime->hw.rates) {
printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n",
- codec_dai->name, cpu_dai->name);
+ codec_dai_name, cpu_dai->name);
goto config_err;
}
if (!runtime->hw.formats) {
printk(KERN_ERR "ASoC: %s <-> %s No matching formats\n",
- codec_dai->name, cpu_dai->name);
+ codec_dai_name, cpu_dai->name);
goto config_err;
}
if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
runtime->hw.channels_min > runtime->hw.channels_max) {
printk(KERN_ERR "ASoC: %s <-> %s No matching channels\n",
- codec_dai->name, cpu_dai->name);
+ codec_dai_name, cpu_dai->name);
goto config_err;
}
- soc_pcm_apply_msb(substream, codec_dai);
- soc_pcm_apply_msb(substream, cpu_dai);
+ soc_pcm_apply_msb(substream);
/* Symmetry only applies if we've already got an active stream. */
if (cpu_dai->active) {
@@ -443,14 +538,17 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
goto config_err;
}
- if (codec_dai->active) {
- ret = soc_pcm_apply_symmetry(substream, codec_dai);
- if (ret != 0)
- goto config_err;
+ for (i = 0; i < rtd->num_codecs; i++) {
+ if (rtd->codec_dais[i]->active) {
+ ret = soc_pcm_apply_symmetry(substream,
+ rtd->codec_dais[i]);
+ if (ret != 0)
+ goto config_err;
+ }
}
pr_debug("ASoC: %s <-> %s info:\n",
- codec_dai->name, cpu_dai->name);
+ codec_dai_name, cpu_dai->name);
pr_debug("ASoC: rate mask 0x%x\n", runtime->hw.rates);
pr_debug("ASoC: min ch %d max ch %d\n", runtime->hw.channels_min,
runtime->hw.channels_max);
@@ -469,10 +567,15 @@ config_err:
rtd->dai_link->ops->shutdown(substream);
machine_err:
- if (codec_dai->driver->ops->shutdown)
- codec_dai->driver->ops->shutdown(substream, codec_dai);
+ i = rtd->num_codecs;
codec_dai_err:
+ while (--i >= 0) {
+ codec_dai = rtd->codec_dais[i];
+ if (codec_dai->driver->ops->shutdown)
+ codec_dai->driver->ops->shutdown(substream, codec_dai);
+ }
+
if (platform->driver->ops && platform->driver->ops->close)
platform->driver->ops->close(substream);
@@ -483,10 +586,13 @@ out:
mutex_unlock(&rtd->pcm_mutex);
pm_runtime_put(platform->dev);
- pm_runtime_put(codec_dai->dev);
+ for (i = 0; i < rtd->num_codecs; i++)
+ pm_runtime_put(rtd->codec_dais[i]->dev);
pm_runtime_put(cpu_dai->dev);
- if (!codec_dai->active)
- pinctrl_pm_select_sleep_state(codec_dai->dev);
+ for (i = 0; i < rtd->num_codecs; i++) {
+ if (!rtd->codec_dais[i]->active)
+ pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev);
+ }
if (!cpu_dai->active)
pinctrl_pm_select_sleep_state(cpu_dai->dev);
@@ -502,7 +608,7 @@ static void close_delayed_work(struct work_struct *work)
{
struct snd_soc_pcm_runtime *rtd =
container_of(work, struct snd_soc_pcm_runtime, delayed_work.work);
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dais[0];
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
@@ -531,7 +637,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai;
+ int i;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
@@ -541,14 +648,20 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
if (!cpu_dai->active)
cpu_dai->rate = 0;
- if (!codec_dai->active)
- codec_dai->rate = 0;
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec_dai = rtd->codec_dais[i];
+ if (!codec_dai->active)
+ codec_dai->rate = 0;
+ }
if (cpu_dai->driver->ops->shutdown)
cpu_dai->driver->ops->shutdown(substream, cpu_dai);
- if (codec_dai->driver->ops->shutdown)
- codec_dai->driver->ops->shutdown(substream, codec_dai);
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec_dai = rtd->codec_dais[i];
+ if (codec_dai->driver->ops->shutdown)
+ codec_dai->driver->ops->shutdown(substream, codec_dai);
+ }
if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown)
rtd->dai_link->ops->shutdown(substream);
@@ -578,10 +691,13 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
mutex_unlock(&rtd->pcm_mutex);
pm_runtime_put(platform->dev);
- pm_runtime_put(codec_dai->dev);
+ for (i = 0; i < rtd->num_codecs; i++)
+ pm_runtime_put(rtd->codec_dais[i]->dev);
pm_runtime_put(cpu_dai->dev);
- if (!codec_dai->active)
- pinctrl_pm_select_sleep_state(codec_dai->dev);
+ for (i = 0; i < rtd->num_codecs; i++) {
+ if (!rtd->codec_dais[i]->active)
+ pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev);
+ }
if (!cpu_dai->active)
pinctrl_pm_select_sleep_state(cpu_dai->dev);
@@ -598,8 +714,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret = 0;
+ struct snd_soc_dai *codec_dai;
+ int i, ret = 0;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
@@ -621,12 +737,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
}
}
- if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) {
- ret = codec_dai->driver->ops->prepare(substream, codec_dai);
- if (ret < 0) {
- dev_err(codec_dai->dev, "ASoC: DAI prepare error: %d\n",
- ret);
- goto out;
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec_dai = rtd->codec_dais[i];
+ if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) {
+ ret = codec_dai->driver->ops->prepare(substream,
+ codec_dai);
+ if (ret < 0) {
+ dev_err(codec_dai->dev,
+ "ASoC: DAI prepare error: %d\n", ret);
+ goto out;
+ }
}
}
@@ -649,13 +769,44 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
snd_soc_dapm_stream_event(rtd, substream->stream,
SND_SOC_DAPM_STREAM_START);
- snd_soc_dai_digital_mute(codec_dai, 0, substream->stream);
+ for (i = 0; i < rtd->num_codecs; i++)
+ snd_soc_dai_digital_mute(rtd->codec_dais[i], 0,
+ substream->stream);
out:
mutex_unlock(&rtd->pcm_mutex);
return ret;
}
+static void soc_pcm_codec_params_fixup(struct snd_pcm_hw_params *params,
+ unsigned int mask)
+{
+ struct snd_interval *interval;
+ int channels = hweight_long(mask);
+
+ interval = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ interval->min = channels;
+ interval->max = channels;
+}
+
+int soc_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ int ret;
+
+ if (dai->driver->ops && dai->driver->ops->hw_params) {
+ ret = dai->driver->ops->hw_params(substream, params, dai);
+ if (ret < 0) {
+ dev_err(dai->dev, "ASoC: can't set %s hw params: %d\n",
+ dai->name, ret);
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
/*
* Called by ALSA when the hardware params are set by application. This
* function can also be called multiple times and can allocate buffers
@@ -667,8 +818,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret = 0;
+ int i, ret = 0;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
@@ -685,29 +835,40 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
}
}
- if (codec_dai->driver->ops && codec_dai->driver->ops->hw_params) {
- ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai);
- if (ret < 0) {
- dev_err(codec_dai->dev, "ASoC: can't set %s hw params:"
- " %d\n", codec_dai->name, ret);
+ for (i = 0; i < rtd->num_codecs; i++) {
+ struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+ struct snd_pcm_hw_params codec_params;
+
+ /* copy params for each codec */
+ codec_params = *params;
+
+ /* fixup params based on TDM slot masks */
+ if (codec_dai->tx_mask)
+ soc_pcm_codec_params_fixup(&codec_params,
+ codec_dai->tx_mask);
+ if (codec_dai->rx_mask)
+ soc_pcm_codec_params_fixup(&codec_params,
+ codec_dai->rx_mask);
+
+ ret = soc_dai_hw_params(substream, &codec_params, codec_dai);
+ if(ret < 0)
goto codec_err;
- }
- }
- if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_params) {
- ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai);
- if (ret < 0) {
- dev_err(cpu_dai->dev, "ASoC: %s hw params failed: %d\n",
- cpu_dai->name, ret);
- goto interface_err;
- }
+ codec_dai->rate = params_rate(&codec_params);
+ codec_dai->channels = params_channels(&codec_params);
+ codec_dai->sample_bits = snd_pcm_format_physical_width(
+ params_format(&codec_params));
}
+ ret = soc_dai_hw_params(substream, params, cpu_dai);
+ if (ret < 0)
+ goto interface_err;
+
if (platform->driver->ops && platform->driver->ops->hw_params) {
ret = platform->driver->ops->hw_params(substream, params);
if (ret < 0) {
dev_err(platform->dev, "ASoC: %s hw params failed: %d\n",
- platform->name, ret);
+ platform->component.name, ret);
goto platform_err;
}
}
@@ -718,11 +879,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
cpu_dai->sample_bits =
snd_pcm_format_physical_width(params_format(params));
- codec_dai->rate = params_rate(params);
- codec_dai->channels = params_channels(params);
- codec_dai->sample_bits =
- snd_pcm_format_physical_width(params_format(params));
-
out:
mutex_unlock(&rtd->pcm_mutex);
return ret;
@@ -732,10 +888,16 @@ platform_err:
cpu_dai->driver->ops->hw_free(substream, cpu_dai);
interface_err:
- if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free)
- codec_dai->driver->ops->hw_free(substream, codec_dai);
+ i = rtd->num_codecs;
codec_err:
+ while (--i >= 0) {
+ struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+ if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free)
+ codec_dai->driver->ops->hw_free(substream, codec_dai);
+ codec_dai->rate = 0;
+ }
+
if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free)
rtd->dai_link->ops->hw_free(substream);
@@ -751,8 +913,9 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai;
bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ int i;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
@@ -763,16 +926,22 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
cpu_dai->sample_bits = 0;
}
- if (codec_dai->active == 1) {
- codec_dai->rate = 0;
- codec_dai->channels = 0;
- codec_dai->sample_bits = 0;
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec_dai = rtd->codec_dais[i];
+ if (codec_dai->active == 1) {
+ codec_dai->rate = 0;
+ codec_dai->channels = 0;
+ codec_dai->sample_bits = 0;
+ }
}
/* apply codec digital mute */
- if ((playback && codec_dai->playback_active == 1) ||
- (!playback && codec_dai->capture_active == 1))
- snd_soc_dai_digital_mute(codec_dai, 1, substream->stream);
+ for (i = 0; i < rtd->num_codecs; i++) {
+ if ((playback && rtd->codec_dais[i]->playback_active == 1) ||
+ (!playback && rtd->codec_dais[i]->capture_active == 1))
+ snd_soc_dai_digital_mute(rtd->codec_dais[i], 1,
+ substream->stream);
+ }
/* free any machine hw params */
if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free)
@@ -783,8 +952,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
platform->driver->ops->hw_free(substream);
/* now free hw params for the DAIs */
- if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free)
- codec_dai->driver->ops->hw_free(substream, codec_dai);
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec_dai = rtd->codec_dais[i];
+ if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free)
+ codec_dai->driver->ops->hw_free(substream, codec_dai);
+ }
if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free)
cpu_dai->driver->ops->hw_free(substream, cpu_dai);
@@ -798,13 +970,17 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret;
-
- if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) {
- ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai);
- if (ret < 0)
- return ret;
+ struct snd_soc_dai *codec_dai;
+ int i, ret;
+
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec_dai = rtd->codec_dais[i];
+ if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) {
+ ret = codec_dai->driver->ops->trigger(substream,
+ cmd, codec_dai);
+ if (ret < 0)
+ return ret;
+ }
}
if (platform->driver->ops && platform->driver->ops->trigger) {
@@ -834,14 +1010,18 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret;
-
- if (codec_dai->driver->ops &&
- codec_dai->driver->ops->bespoke_trigger) {
- ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai);
- if (ret < 0)
- return ret;
+ struct snd_soc_dai *codec_dai;
+ int i, ret;
+
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec_dai = rtd->codec_dais[i];
+ if (codec_dai->driver->ops &&
+ codec_dai->driver->ops->bespoke_trigger) {
+ ret = codec_dai->driver->ops->bespoke_trigger(substream,
+ cmd, codec_dai);
+ if (ret < 0)
+ return ret;
+ }
}
if (platform->driver->bespoke_trigger) {
@@ -867,10 +1047,12 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai;
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_uframes_t offset = 0;
snd_pcm_sframes_t delay = 0;
+ snd_pcm_sframes_t codec_delay = 0;
+ int i;
if (platform->driver->ops && platform->driver->ops->pointer)
offset = platform->driver->ops->pointer(substream);
@@ -878,11 +1060,21 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
if (cpu_dai->driver->ops && cpu_dai->driver->ops->delay)
delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
- if (codec_dai->driver->ops && codec_dai->driver->ops->delay)
- delay += codec_dai->driver->ops->delay(substream, codec_dai);
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec_dai = rtd->codec_dais[i];
+ if (codec_dai->driver->ops && codec_dai->driver->ops->delay)
+ codec_delay = max(codec_delay,
+ codec_dai->driver->ops->delay(substream,
+ codec_dai));
+ }
+ delay += codec_delay;
+ /*
+ * None of the existing platform drivers implement delay(), so
+ * for now the codec_dai of first multicodec entry is used
+ */
if (platform->driver->delay)
- delay += platform->driver->delay(substream, codec_dai);
+ delay += platform->driver->delay(substream, rtd->codec_dais[0]);
runtime->delay = delay;
@@ -985,7 +1177,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
struct snd_soc_dapm_widget *widget, int stream)
{
struct snd_soc_pcm_runtime *be;
- int i;
+ int i, j;
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
for (i = 0; i < card->num_links; i++) {
@@ -994,9 +1186,14 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
if (!be->dai_link->no_pcm)
continue;
- if (be->cpu_dai->playback_widget == widget ||
- be->codec_dai->playback_widget == widget)
+ if (be->cpu_dai->playback_widget == widget)
return be;
+
+ for (j = 0; j < be->num_codecs; j++) {
+ struct snd_soc_dai *dai = be->codec_dais[j];
+ if (dai->playback_widget == widget)
+ return be;
+ }
}
} else {
@@ -1006,9 +1203,14 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
if (!be->dai_link->no_pcm)
continue;
- if (be->cpu_dai->capture_widget == widget ||
- be->codec_dai->capture_widget == widget)
+ if (be->cpu_dai->capture_widget == widget)
return be;
+
+ for (j = 0; j < be->num_codecs; j++) {
+ struct snd_soc_dai *dai = be->codec_dais[j];
+ if (dai->capture_widget == widget)
+ return be;
+ }
}
}
@@ -1071,6 +1273,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
/* Destroy any old FE <--> BE connections */
list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ unsigned int i;
/* is there a valid CPU DAI widget for this BE */
widget = dai_get_widget(dpcm->be->cpu_dai, stream);
@@ -1080,11 +1283,14 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
continue;
/* is there a valid CODEC DAI widget for this BE */
- widget = dai_get_widget(dpcm->be->codec_dai, stream);
+ for (i = 0; i < dpcm->be->num_codecs; i++) {
+ struct snd_soc_dai *dai = dpcm->be->codec_dais[i];
+ widget = dai_get_widget(dai, stream);
- /* prune the BE if it's no longer in our active list */
- if (widget && widget_in_list(list, widget))
- continue;
+ /* prune the BE if it's no longer in our active list */
+ if (widget && widget_in_list(list, widget))
+ continue;
+ }
dev_dbg(fe->dev, "ASoC: pruning %s BE %s for %s\n",
stream ? "capture" : "playback",
@@ -2069,6 +2275,7 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
}
+ dpcm_path_put(&list);
capture:
/* skip if FE doesn't have capture capability */
if (!fe->cpu_dai->driver->capture.channels_min)
@@ -2113,16 +2320,22 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
list_for_each_entry(dpcm, clients, list_be) {
struct snd_soc_pcm_runtime *be = dpcm->be;
- struct snd_soc_dai *dai = be->codec_dai;
- struct snd_soc_dai_driver *drv = dai->driver;
+ int i;
if (be->dai_link->ignore_suspend)
continue;
- dev_dbg(be->dev, "ASoC: BE digital mute %s\n", be->dai_link->name);
+ for (i = 0; i < be->num_codecs; i++) {
+ struct snd_soc_dai *dai = be->codec_dais[i];
+ struct snd_soc_dai_driver *drv = dai->driver;
+
+ dev_dbg(be->dev, "ASoC: BE digital mute %s\n",
+ be->dai_link->name);
- if (drv->ops && drv->ops->digital_mute && dai->playback_active)
- drv->ops->digital_mute(dai, mute);
+ if (drv->ops && drv->ops->digital_mute &&
+ dai->playback_active)
+ drv->ops->digital_mute(dai, mute);
+ }
}
return 0;
@@ -2187,22 +2400,27 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
{
struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_pcm *pcm;
char new_name[64];
int ret = 0, playback = 0, capture = 0;
+ int i;
if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) {
playback = rtd->dai_link->dpcm_playback;
capture = rtd->dai_link->dpcm_capture;
} else {
- if (codec_dai->driver->playback.channels_min &&
- cpu_dai->driver->playback.channels_min)
- playback = 1;
- if (codec_dai->driver->capture.channels_min &&
- cpu_dai->driver->capture.channels_min)
- capture = 1;
+ for (i = 0; i < rtd->num_codecs; i++) {
+ codec_dai = rtd->codec_dais[i];
+ if (codec_dai->driver->playback.channels_min)
+ playback = 1;
+ if (codec_dai->driver->capture.channels_min)
+ capture = 1;
+ }
+
+ capture = capture && cpu_dai->driver->capture.channels_min;
+ playback = playback && cpu_dai->driver->playback.channels_min;
}
if (rtd->dai_link->playback_only) {
@@ -2228,7 +2446,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
rtd->dai_link->stream_name);
else
snprintf(new_name, sizeof(new_name), "%s %s-%d",
- rtd->dai_link->stream_name, codec_dai->name, num);
+ rtd->dai_link->stream_name,
+ (rtd->num_codecs > 1) ?
+ "multicodec" : rtd->codec_dai->name, num);
ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback,
capture, &pcm);
@@ -2301,8 +2521,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
pcm->private_free = platform->driver->pcm_free;
out:
- dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", codec_dai->name,
- cpu_dai->name);
+ dev_info(rtd->card->dev, "%s <-> %s mapping ok\n",
+ (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name,
+ cpu_dai->name);
return ret;
}
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 02734bd4f09b..a83aff09dce2 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -41,8 +41,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = codec_dai->codec;
- struct snd_soc_card *card = codec->card;
+ struct snd_soc_card *card = rtd->card;
struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card);
int srate, mclk;
int err;
@@ -105,7 +104,7 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_codec *codec = codec_dai->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card);
+ struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card);
snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET,
&tegra_alc5632_hs_jack);
diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c
index ce73e1f62c4b..b86cd9936ef1 100644
--- a/sound/soc/tegra/tegra_max98090.c
+++ b/sound/soc/tegra/tegra_max98090.c
@@ -49,8 +49,7 @@ static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = codec_dai->codec;
- struct snd_soc_card *card = codec->card;
+ struct snd_soc_card *card = rtd->card;
struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card);
int srate, mclk;
int err;
@@ -127,7 +126,7 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_codec *codec = codec_dai->codec;
- struct tegra_max98090 *machine = snd_soc_card_get_drvdata(codec->card);
+ struct tegra_max98090 *machine = snd_soc_card_get_drvdata(rtd->card);
if (gpio_is_valid(machine->gpio_hp_det)) {
snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE,
diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c
index 4feb16a99e02..a6898831fb9f 100644
--- a/sound/soc/tegra/tegra_rt5640.c
+++ b/sound/soc/tegra/tegra_rt5640.c
@@ -51,8 +51,7 @@ static int tegra_rt5640_asoc_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = codec_dai->codec;
- struct snd_soc_card *card = codec->card;
+ struct snd_soc_card *card = rtd->card;
struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card);
int srate, mclk;
int err;
@@ -110,7 +109,7 @@ static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_codec *codec = codec_dai->codec;
- struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(codec->card);
+ struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(rtd->card);
snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE,
&tegra_rt5640_hp_jack);
diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c
index 8e774d1a243c..769e28f6642e 100644
--- a/sound/soc/tegra/tegra_wm8753.c
+++ b/sound/soc/tegra/tegra_wm8753.c
@@ -55,8 +55,7 @@ static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = codec_dai->codec;
- struct snd_soc_card *card = codec->card;
+ struct snd_soc_card *card = rtd->card;
struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card);
int srate, mclk;
int err;
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 0939661df60b..86e05e938585 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -60,8 +60,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = codec_dai->codec;
- struct snd_soc_card *card = codec->card;
+ struct snd_soc_card *card = rtd->card;
struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
int srate, mclk;
int err;
@@ -173,7 +172,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_codec *codec = codec_dai->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- struct snd_soc_card *card = codec->card;
+ struct snd_soc_card *card = rtd->card;
struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
if (gpio_is_valid(machine->gpio_hp_det)) {
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 734bfcd21148..589d2d9b553a 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -50,8 +50,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = codec_dai->codec;
- struct snd_soc_card *card = codec->card;
+ struct snd_soc_card *card = rtd->card;
struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card);
int srate, mclk;
int err;
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index be1b1aa96b7e..b2c3d0d5dca3 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -2534,12 +2534,10 @@ static int snd_dbri_create(struct snd_card *card,
dbri->op = op;
dbri->irq = irq;
- dbri->dma = dma_alloc_coherent(&op->dev,
- sizeof(struct dbri_dma),
- &dbri->dma_dvma, GFP_ATOMIC);
+ dbri->dma = dma_zalloc_coherent(&op->dev, sizeof(struct dbri_dma),
+ &dbri->dma_dvma, GFP_ATOMIC);
if (!dbri->dma)
return -ENOMEM;
- memset((void *)dbri->dma, 0, sizeof(struct dbri_dma));
dprintk(D_GEN, "DMA Cmd Block 0x%p (0x%08x)\n",
dbri->dma, dbri->dma_dvma);
diff --git a/sound/usb/card.c b/sound/usb/card.c
index a09e5f3519e3..7ecd0e8a5c51 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -680,6 +680,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
struct snd_usb_audio *chip = usb_get_intfdata(intf);
struct snd_usb_stream *as;
struct usb_mixer_interface *mixer;
+ struct list_head *p;
if (chip == (void *)-1L)
return 0;
@@ -692,6 +693,9 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
as->substream[0].need_setup_ep =
as->substream[1].need_setup_ep = true;
}
+ list_for_each(p, &chip->midi_list) {
+ snd_usbmidi_suspend(p);
+ }
}
} else {
/*
@@ -713,6 +717,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
{
struct snd_usb_audio *chip = usb_get_intfdata(intf);
struct usb_mixer_interface *mixer;
+ struct list_head *p;
int err = 0;
if (chip == (void *)-1L)
@@ -731,6 +736,10 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
goto err_out;
}
+ list_for_each(p, &chip->midi_list) {
+ snd_usbmidi_resume(p);
+ }
+
if (!chip->autosuspended)
snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
chip->autosuspended = 0;
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 9da74d2e8eee..7b166c2be0f7 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -102,8 +102,8 @@ struct usb_protocol_ops {
void (*input)(struct snd_usb_midi_in_endpoint*, uint8_t*, int);
void (*output)(struct snd_usb_midi_out_endpoint *ep, struct urb *urb);
void (*output_packet)(struct urb*, uint8_t, uint8_t, uint8_t, uint8_t);
- void (*init_out_endpoint)(struct snd_usb_midi_out_endpoint*);
- void (*finish_out_endpoint)(struct snd_usb_midi_out_endpoint*);
+ void (*init_out_endpoint)(struct snd_usb_midi_out_endpoint *);
+ void (*finish_out_endpoint)(struct snd_usb_midi_out_endpoint *);
};
struct snd_usb_midi {
@@ -112,7 +112,7 @@ struct snd_usb_midi {
struct usb_interface *iface;
const struct snd_usb_audio_quirk *quirk;
struct snd_rawmidi *rmidi;
- struct usb_protocol_ops* usb_protocol_ops;
+ struct usb_protocol_ops *usb_protocol_ops;
struct list_head list;
struct timer_list error_timer;
spinlock_t disc_lock;
@@ -134,7 +134,7 @@ struct snd_usb_midi {
};
struct snd_usb_midi_out_endpoint {
- struct snd_usb_midi* umidi;
+ struct snd_usb_midi *umidi;
struct out_urb_context {
struct urb *urb;
struct snd_usb_midi_out_endpoint *ep;
@@ -147,7 +147,7 @@ struct snd_usb_midi_out_endpoint {
spinlock_t buffer_lock;
struct usbmidi_out_port {
- struct snd_usb_midi_out_endpoint* ep;
+ struct snd_usb_midi_out_endpoint *ep;
struct snd_rawmidi_substream *substream;
int active;
uint8_t cable; /* cable number << 4 */
@@ -167,8 +167,8 @@ struct snd_usb_midi_out_endpoint {
};
struct snd_usb_midi_in_endpoint {
- struct snd_usb_midi* umidi;
- struct urb* urbs[INPUT_URBS];
+ struct snd_usb_midi *umidi;
+ struct urb *urbs[INPUT_URBS];
struct usbmidi_in_port {
struct snd_rawmidi_substream *substream;
u8 running_status_length;
@@ -178,7 +178,7 @@ struct snd_usb_midi_in_endpoint {
int current_port;
};
-static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep);
+static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint *ep);
static const uint8_t snd_usbmidi_cin_length[] = {
0, 0, 2, 3, 3, 1, 2, 3, 3, 3, 3, 3, 2, 2, 3, 1
@@ -187,7 +187,7 @@ static const uint8_t snd_usbmidi_cin_length[] = {
/*
* Submits the URB, with error handling.
*/
-static int snd_usbmidi_submit_urb(struct urb* urb, gfp_t flags)
+static int snd_usbmidi_submit_urb(struct urb *urb, gfp_t flags)
{
int err = usb_submit_urb(urb, flags);
if (err < 0 && err != -ENODEV)
@@ -221,10 +221,10 @@ static int snd_usbmidi_urb_error(const struct urb *urb)
/*
* Receives a chunk of MIDI data.
*/
-static void snd_usbmidi_input_data(struct snd_usb_midi_in_endpoint* ep, int portidx,
- uint8_t* data, int length)
+static void snd_usbmidi_input_data(struct snd_usb_midi_in_endpoint *ep,
+ int portidx, uint8_t *data, int length)
{
- struct usbmidi_in_port* port = &ep->ports[portidx];
+ struct usbmidi_in_port *port = &ep->ports[portidx];
if (!port->substream) {
dev_dbg(&ep->umidi->dev->dev, "unexpected port %d!\n", portidx);
@@ -250,9 +250,9 @@ static void dump_urb(const char *type, const u8 *data, int length)
/*
* Processes the data read from the device.
*/
-static void snd_usbmidi_in_urb_complete(struct urb* urb)
+static void snd_usbmidi_in_urb_complete(struct urb *urb)
{
- struct snd_usb_midi_in_endpoint* ep = urb->context;
+ struct snd_usb_midi_in_endpoint *ep = urb->context;
if (urb->status == 0) {
dump_urb("received", urb->transfer_buffer, urb->actual_length);
@@ -274,10 +274,10 @@ static void snd_usbmidi_in_urb_complete(struct urb* urb)
snd_usbmidi_submit_urb(urb, GFP_ATOMIC);
}
-static void snd_usbmidi_out_urb_complete(struct urb* urb)
+static void snd_usbmidi_out_urb_complete(struct urb *urb)
{
struct out_urb_context *context = urb->context;
- struct snd_usb_midi_out_endpoint* ep = context->ep;
+ struct snd_usb_midi_out_endpoint *ep = context->ep;
unsigned int urb_index;
spin_lock(&ep->buffer_lock);
@@ -304,10 +304,10 @@ static void snd_usbmidi_out_urb_complete(struct urb* urb)
* This is called when some data should be transferred to the device
* (from one or more substreams).
*/
-static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep)
+static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint *ep)
{
unsigned int urb_index;
- struct urb* urb;
+ struct urb *urb;
unsigned long flags;
spin_lock_irqsave(&ep->buffer_lock, flags);
@@ -343,7 +343,8 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep)
static void snd_usbmidi_out_tasklet(unsigned long data)
{
- struct snd_usb_midi_out_endpoint* ep = (struct snd_usb_midi_out_endpoint *) data;
+ struct snd_usb_midi_out_endpoint *ep =
+ (struct snd_usb_midi_out_endpoint *) data;
snd_usbmidi_do_output(ep);
}
@@ -375,7 +376,7 @@ static void snd_usbmidi_error_timer(unsigned long data)
}
/* helper function to send static data that may not DMA-able */
-static int send_bulk_static_data(struct snd_usb_midi_out_endpoint* ep,
+static int send_bulk_static_data(struct snd_usb_midi_out_endpoint *ep,
const void *data, int len)
{
int err = 0;
@@ -396,8 +397,8 @@ static int send_bulk_static_data(struct snd_usb_midi_out_endpoint* ep,
* fourth byte in each packet, and uses length instead of CIN.
*/
-static void snd_usbmidi_standard_input(struct snd_usb_midi_in_endpoint* ep,
- uint8_t* buffer, int buffer_length)
+static void snd_usbmidi_standard_input(struct snd_usb_midi_in_endpoint *ep,
+ uint8_t *buffer, int buffer_length)
{
int i;
@@ -405,12 +406,13 @@ static void snd_usbmidi_standard_input(struct snd_usb_midi_in_endpoint* ep,
if (buffer[i] != 0) {
int cable = buffer[i] >> 4;
int length = snd_usbmidi_cin_length[buffer[i] & 0x0f];
- snd_usbmidi_input_data(ep, cable, &buffer[i + 1], length);
+ snd_usbmidi_input_data(ep, cable, &buffer[i + 1],
+ length);
}
}
-static void snd_usbmidi_midiman_input(struct snd_usb_midi_in_endpoint* ep,
- uint8_t* buffer, int buffer_length)
+static void snd_usbmidi_midiman_input(struct snd_usb_midi_in_endpoint *ep,
+ uint8_t *buffer, int buffer_length)
{
int i;
@@ -427,8 +429,8 @@ static void snd_usbmidi_midiman_input(struct snd_usb_midi_in_endpoint* ep,
* the data bytes but not the status byte and that is marked with CIN 4.
*/
static void snd_usbmidi_maudio_broken_running_status_input(
- struct snd_usb_midi_in_endpoint* ep,
- uint8_t* buffer, int buffer_length)
+ struct snd_usb_midi_in_endpoint *ep,
+ uint8_t *buffer, int buffer_length)
{
int i;
@@ -458,7 +460,8 @@ static void snd_usbmidi_maudio_broken_running_status_input(
* doesn't use this format.)
*/
port->running_status_length = 0;
- snd_usbmidi_input_data(ep, cable, &buffer[i + 1], length);
+ snd_usbmidi_input_data(ep, cable, &buffer[i + 1],
+ length);
}
}
@@ -479,11 +482,13 @@ static void snd_usbmidi_cme_input(struct snd_usb_midi_in_endpoint *ep,
/*
* Adds one USB MIDI packet to the output buffer.
*/
-static void snd_usbmidi_output_standard_packet(struct urb* urb, uint8_t p0,
- uint8_t p1, uint8_t p2, uint8_t p3)
+static void snd_usbmidi_output_standard_packet(struct urb *urb, uint8_t p0,
+ uint8_t p1, uint8_t p2,
+ uint8_t p3)
{
- uint8_t* buf = (uint8_t*)urb->transfer_buffer + urb->transfer_buffer_length;
+ uint8_t *buf =
+ (uint8_t *)urb->transfer_buffer + urb->transfer_buffer_length;
buf[0] = p0;
buf[1] = p1;
buf[2] = p2;
@@ -494,11 +499,13 @@ static void snd_usbmidi_output_standard_packet(struct urb* urb, uint8_t p0,
/*
* Adds one Midiman packet to the output buffer.
*/
-static void snd_usbmidi_output_midiman_packet(struct urb* urb, uint8_t p0,
- uint8_t p1, uint8_t p2, uint8_t p3)
+static void snd_usbmidi_output_midiman_packet(struct urb *urb, uint8_t p0,
+ uint8_t p1, uint8_t p2,
+ uint8_t p3)
{
- uint8_t* buf = (uint8_t*)urb->transfer_buffer + urb->transfer_buffer_length;
+ uint8_t *buf =
+ (uint8_t *)urb->transfer_buffer + urb->transfer_buffer_length;
buf[0] = p1;
buf[1] = p2;
buf[2] = p3;
@@ -509,8 +516,8 @@ static void snd_usbmidi_output_midiman_packet(struct urb* urb, uint8_t p0,
/*
* Converts MIDI commands to USB MIDI packets.
*/
-static void snd_usbmidi_transmit_byte(struct usbmidi_out_port* port,
- uint8_t b, struct urb* urb)
+static void snd_usbmidi_transmit_byte(struct usbmidi_out_port *port,
+ uint8_t b, struct urb *urb)
{
uint8_t p0 = port->cable;
void (*output_packet)(struct urb*, uint8_t, uint8_t, uint8_t, uint8_t) =
@@ -547,10 +554,12 @@ static void snd_usbmidi_transmit_byte(struct usbmidi_out_port* port,
output_packet(urb, p0 | 0x05, 0xf7, 0, 0);
break;
case STATE_SYSEX_1:
- output_packet(urb, p0 | 0x06, port->data[0], 0xf7, 0);
+ output_packet(urb, p0 | 0x06, port->data[0],
+ 0xf7, 0);
break;
case STATE_SYSEX_2:
- output_packet(urb, p0 | 0x07, port->data[0], port->data[1], 0xf7);
+ output_packet(urb, p0 | 0x07, port->data[0],
+ port->data[1], 0xf7);
break;
}
port->state = STATE_UNKNOWN;
@@ -596,21 +605,22 @@ static void snd_usbmidi_transmit_byte(struct usbmidi_out_port* port,
port->state = STATE_SYSEX_2;
break;
case STATE_SYSEX_2:
- output_packet(urb, p0 | 0x04, port->data[0], port->data[1], b);
+ output_packet(urb, p0 | 0x04, port->data[0],
+ port->data[1], b);
port->state = STATE_SYSEX_0;
break;
}
}
}
-static void snd_usbmidi_standard_output(struct snd_usb_midi_out_endpoint* ep,
+static void snd_usbmidi_standard_output(struct snd_usb_midi_out_endpoint *ep,
struct urb *urb)
{
int p;
/* FIXME: lower-numbered ports can starve higher-numbered ports */
for (p = 0; p < 0x10; ++p) {
- struct usbmidi_out_port* port = &ep->ports[p];
+ struct usbmidi_out_port *port = &ep->ports[p];
if (!port->active)
continue;
while (urb->transfer_buffer_length + 3 < ep->max_transfer) {
@@ -753,18 +763,18 @@ static struct usb_protocol_ops snd_usbmidi_akai_ops = {
* at the third byte.
*/
-static void snd_usbmidi_novation_input(struct snd_usb_midi_in_endpoint* ep,
- uint8_t* buffer, int buffer_length)
+static void snd_usbmidi_novation_input(struct snd_usb_midi_in_endpoint *ep,
+ uint8_t *buffer, int buffer_length)
{
if (buffer_length < 2 || !buffer[0] || buffer_length < buffer[0] + 1)
return;
snd_usbmidi_input_data(ep, 0, &buffer[2], buffer[0] - 1);
}
-static void snd_usbmidi_novation_output(struct snd_usb_midi_out_endpoint* ep,
+static void snd_usbmidi_novation_output(struct snd_usb_midi_out_endpoint *ep,
struct urb *urb)
{
- uint8_t* transfer_buffer;
+ uint8_t *transfer_buffer;
int count;
if (!ep->ports[0].active)
@@ -791,13 +801,13 @@ static struct usb_protocol_ops snd_usbmidi_novation_ops = {
* "raw" protocol: just move raw MIDI bytes from/to the endpoint
*/
-static void snd_usbmidi_raw_input(struct snd_usb_midi_in_endpoint* ep,
- uint8_t* buffer, int buffer_length)
+static void snd_usbmidi_raw_input(struct snd_usb_midi_in_endpoint *ep,
+ uint8_t *buffer, int buffer_length)
{
snd_usbmidi_input_data(ep, 0, buffer, buffer_length);
}
-static void snd_usbmidi_raw_output(struct snd_usb_midi_out_endpoint* ep,
+static void snd_usbmidi_raw_output(struct snd_usb_midi_out_endpoint *ep,
struct urb *urb)
{
int count;
@@ -823,8 +833,8 @@ static struct usb_protocol_ops snd_usbmidi_raw_ops = {
* FTDI protocol: raw MIDI bytes, but input packets have two modem status bytes.
*/
-static void snd_usbmidi_ftdi_input(struct snd_usb_midi_in_endpoint* ep,
- uint8_t* buffer, int buffer_length)
+static void snd_usbmidi_ftdi_input(struct snd_usb_midi_in_endpoint *ep,
+ uint8_t *buffer, int buffer_length)
{
if (buffer_length > 2)
snd_usbmidi_input_data(ep, 0, buffer + 2, buffer_length - 2);
@@ -883,7 +893,7 @@ static struct usb_protocol_ops snd_usbmidi_122l_ops = {
* Emagic USB MIDI protocol: raw MIDI with "F5 xx" port switching.
*/
-static void snd_usbmidi_emagic_init_out(struct snd_usb_midi_out_endpoint* ep)
+static void snd_usbmidi_emagic_init_out(struct snd_usb_midi_out_endpoint *ep)
{
static const u8 init_data[] = {
/* initialization magic: "get version" */
@@ -900,7 +910,7 @@ static void snd_usbmidi_emagic_init_out(struct snd_usb_midi_out_endpoint* ep)
send_bulk_static_data(ep, init_data, sizeof(init_data));
}
-static void snd_usbmidi_emagic_finish_out(struct snd_usb_midi_out_endpoint* ep)
+static void snd_usbmidi_emagic_finish_out(struct snd_usb_midi_out_endpoint *ep)
{
static const u8 finish_data[] = {
/* switch to patch mode with last preset */
@@ -916,8 +926,8 @@ static void snd_usbmidi_emagic_finish_out(struct snd_usb_midi_out_endpoint* ep)
send_bulk_static_data(ep, finish_data, sizeof(finish_data));
}
-static void snd_usbmidi_emagic_input(struct snd_usb_midi_in_endpoint* ep,
- uint8_t* buffer, int buffer_length)
+static void snd_usbmidi_emagic_input(struct snd_usb_midi_in_endpoint *ep,
+ uint8_t *buffer, int buffer_length)
{
int i;
@@ -960,18 +970,18 @@ static void snd_usbmidi_emagic_input(struct snd_usb_midi_in_endpoint* ep,
}
}
-static void snd_usbmidi_emagic_output(struct snd_usb_midi_out_endpoint* ep,
+static void snd_usbmidi_emagic_output(struct snd_usb_midi_out_endpoint *ep,
struct urb *urb)
{
int port0 = ep->current_port;
- uint8_t* buf = urb->transfer_buffer;
+ uint8_t *buf = urb->transfer_buffer;
int buf_free = ep->max_transfer;
int length, i;
for (i = 0; i < 0x10; ++i) {
/* round-robin, starting at the last current port */
int portnum = (port0 + i) & 15;
- struct usbmidi_out_port* port = &ep->ports[portnum];
+ struct usbmidi_out_port *port = &ep->ports[portnum];
if (!port->active)
continue;
@@ -1015,7 +1025,7 @@ static struct usb_protocol_ops snd_usbmidi_emagic_ops = {
};
-static void update_roland_altsetting(struct snd_usb_midi* umidi)
+static void update_roland_altsetting(struct snd_usb_midi *umidi)
{
struct usb_interface *intf;
struct usb_host_interface *hostif;
@@ -1037,7 +1047,7 @@ static void update_roland_altsetting(struct snd_usb_midi* umidi)
static int substream_open(struct snd_rawmidi_substream *substream, int dir,
int open)
{
- struct snd_usb_midi* umidi = substream->rmidi->private_data;
+ struct snd_usb_midi *umidi = substream->rmidi->private_data;
struct snd_kcontrol *ctl;
down_read(&umidi->disc_rwsem);
@@ -1051,7 +1061,8 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir,
if (!umidi->opened[0] && !umidi->opened[1]) {
if (umidi->roland_load_ctl) {
ctl = umidi->roland_load_ctl;
- ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ ctl->vd[0].access |=
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE;
snd_ctl_notify(umidi->card,
SNDRV_CTL_EVENT_MASK_INFO, &ctl->id);
update_roland_altsetting(umidi);
@@ -1067,7 +1078,8 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir,
if (!umidi->opened[0] && !umidi->opened[1]) {
if (umidi->roland_load_ctl) {
ctl = umidi->roland_load_ctl;
- ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ ctl->vd[0].access &=
+ ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
snd_ctl_notify(umidi->card,
SNDRV_CTL_EVENT_MASK_INFO, &ctl->id);
}
@@ -1080,8 +1092,8 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir,
static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream)
{
- struct snd_usb_midi* umidi = substream->rmidi->private_data;
- struct usbmidi_out_port* port = NULL;
+ struct snd_usb_midi *umidi = substream->rmidi->private_data;
+ struct usbmidi_out_port *port = NULL;
int i, j;
for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i)
@@ -1106,9 +1118,11 @@ static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream)
return substream_open(substream, 0, 0);
}
-static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream,
+ int up)
{
- struct usbmidi_out_port* port = (struct usbmidi_out_port*)substream->runtime->private_data;
+ struct usbmidi_out_port *port =
+ (struct usbmidi_out_port *)substream->runtime->private_data;
port->active = up;
if (up) {
@@ -1125,7 +1139,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream,
static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream)
{
- struct usbmidi_out_port* port = substream->runtime->private_data;
+ struct usbmidi_out_port *port = substream->runtime->private_data;
struct snd_usb_midi_out_endpoint *ep = port->ep;
unsigned int drain_urbs;
DEFINE_WAIT(wait);
@@ -1164,9 +1178,10 @@ static int snd_usbmidi_input_close(struct snd_rawmidi_substream *substream)
return substream_open(substream, 1, 0);
}
-static void snd_usbmidi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+static void snd_usbmidi_input_trigger(struct snd_rawmidi_substream *substream,
+ int up)
{
- struct snd_usb_midi* umidi = substream->rmidi->private_data;
+ struct snd_usb_midi *umidi = substream->rmidi->private_data;
if (up)
set_bit(substream->number, &umidi->input_triggered);
@@ -1199,7 +1214,7 @@ static void free_urb_and_buffer(struct snd_usb_midi *umidi, struct urb *urb,
* Frees an input endpoint.
* May be called when ep hasn't been initialized completely.
*/
-static void snd_usbmidi_in_endpoint_delete(struct snd_usb_midi_in_endpoint* ep)
+static void snd_usbmidi_in_endpoint_delete(struct snd_usb_midi_in_endpoint *ep)
{
unsigned int i;
@@ -1213,12 +1228,12 @@ static void snd_usbmidi_in_endpoint_delete(struct snd_usb_midi_in_endpoint* ep)
/*
* Creates an input endpoint.
*/
-static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi,
- struct snd_usb_midi_endpoint_info* ep_info,
- struct snd_usb_midi_endpoint* rep)
+static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi *umidi,
+ struct snd_usb_midi_endpoint_info *ep_info,
+ struct snd_usb_midi_endpoint *rep)
{
- struct snd_usb_midi_in_endpoint* ep;
- void* buffer;
+ struct snd_usb_midi_in_endpoint *ep;
+ void *buffer;
unsigned int pipe;
int length;
unsigned int i;
@@ -1289,14 +1304,14 @@ static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint *ep
/*
* Creates an output endpoint, and initializes output ports.
*/
-static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi,
- struct snd_usb_midi_endpoint_info* ep_info,
- struct snd_usb_midi_endpoint* rep)
+static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi,
+ struct snd_usb_midi_endpoint_info *ep_info,
+ struct snd_usb_midi_endpoint *rep)
{
- struct snd_usb_midi_out_endpoint* ep;
+ struct snd_usb_midi_out_endpoint *ep;
unsigned int i;
unsigned int pipe;
- void* buffer;
+ void *buffer;
rep->out = NULL;
ep = kzalloc(sizeof(*ep), GFP_KERNEL);
@@ -1381,12 +1396,12 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi,
/*
* Frees everything.
*/
-static void snd_usbmidi_free(struct snd_usb_midi* umidi)
+static void snd_usbmidi_free(struct snd_usb_midi *umidi)
{
int i;
for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
- struct snd_usb_midi_endpoint* ep = &umidi->endpoints[i];
+ struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i];
if (ep->out)
snd_usbmidi_out_endpoint_delete(ep->out);
if (ep->in)
@@ -1399,9 +1414,9 @@ static void snd_usbmidi_free(struct snd_usb_midi* umidi)
/*
* Unlinks all URBs (must be done before the usb_device is deleted).
*/
-void snd_usbmidi_disconnect(struct list_head* p)
+void snd_usbmidi_disconnect(struct list_head *p)
{
- struct snd_usb_midi* umidi;
+ struct snd_usb_midi *umidi;
unsigned int i, j;
umidi = list_entry(p, struct snd_usb_midi, list);
@@ -1417,7 +1432,7 @@ void snd_usbmidi_disconnect(struct list_head* p)
up_write(&umidi->disc_rwsem);
for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
- struct snd_usb_midi_endpoint* ep = &umidi->endpoints[i];
+ struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i];
if (ep->out)
tasklet_kill(&ep->out->tasklet);
if (ep->out) {
@@ -1448,16 +1463,18 @@ EXPORT_SYMBOL(snd_usbmidi_disconnect);
static void snd_usbmidi_rawmidi_free(struct snd_rawmidi *rmidi)
{
- struct snd_usb_midi* umidi = rmidi->private_data;
+ struct snd_usb_midi *umidi = rmidi->private_data;
snd_usbmidi_free(umidi);
}
-static struct snd_rawmidi_substream *snd_usbmidi_find_substream(struct snd_usb_midi* umidi,
- int stream, int number)
+static struct snd_rawmidi_substream *snd_usbmidi_find_substream(struct snd_usb_midi *umidi,
+ int stream,
+ int number)
{
struct snd_rawmidi_substream *substream;
- list_for_each_entry(substream, &umidi->rmidi->streams[stream].substreams, list) {
+ list_for_each_entry(substream, &umidi->rmidi->streams[stream].substreams,
+ list) {
if (substream->number == number)
return substream;
}
@@ -1633,7 +1650,7 @@ static struct port_info {
SNDRV_SEQ_PORT_TYPE_SYNTHESIZER),
};
-static struct port_info *find_port_info(struct snd_usb_midi* umidi, int number)
+static struct port_info *find_port_info(struct snd_usb_midi *umidi, int number)
{
int i;
@@ -1659,16 +1676,18 @@ static void snd_usbmidi_get_port_info(struct snd_rawmidi *rmidi, int number,
}
}
-static void snd_usbmidi_init_substream(struct snd_usb_midi* umidi,
+static void snd_usbmidi_init_substream(struct snd_usb_midi *umidi,
int stream, int number,
- struct snd_rawmidi_substream ** rsubstream)
+ struct snd_rawmidi_substream **rsubstream)
{
struct port_info *port_info;
const char *name_format;
- struct snd_rawmidi_substream *substream = snd_usbmidi_find_substream(umidi, stream, number);
+ struct snd_rawmidi_substream *substream =
+ snd_usbmidi_find_substream(umidi, stream, number);
if (!substream) {
- dev_err(&umidi->dev->dev, "substream %d:%d not found\n", stream, number);
+ dev_err(&umidi->dev->dev, "substream %d:%d not found\n", stream,
+ number);
return;
}
@@ -1684,21 +1703,23 @@ static void snd_usbmidi_init_substream(struct snd_usb_midi* umidi,
/*
* Creates the endpoints and their ports.
*/
-static int snd_usbmidi_create_endpoints(struct snd_usb_midi* umidi,
- struct snd_usb_midi_endpoint_info* endpoints)
+static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi,
+ struct snd_usb_midi_endpoint_info *endpoints)
{
int i, j, err;
int out_ports = 0, in_ports = 0;
for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
if (endpoints[i].out_cables) {
- err = snd_usbmidi_out_endpoint_create(umidi, &endpoints[i],
+ err = snd_usbmidi_out_endpoint_create(umidi,
+ &endpoints[i],
&umidi->endpoints[i]);
if (err < 0)
return err;
}
if (endpoints[i].in_cables) {
- err = snd_usbmidi_in_endpoint_create(umidi, &endpoints[i],
+ err = snd_usbmidi_in_endpoint_create(umidi,
+ &endpoints[i],
&umidi->endpoints[i]);
if (err < 0)
return err;
@@ -1706,12 +1727,16 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi* umidi,
for (j = 0; j < 0x10; ++j) {
if (endpoints[i].out_cables & (1 << j)) {
- snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_OUTPUT, out_ports,
+ snd_usbmidi_init_substream(umidi,
+ SNDRV_RAWMIDI_STREAM_OUTPUT,
+ out_ports,
&umidi->endpoints[i].out->ports[j].substream);
++out_ports;
}
if (endpoints[i].in_cables & (1 << j)) {
- snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_INPUT, in_ports,
+ snd_usbmidi_init_substream(umidi,
+ SNDRV_RAWMIDI_STREAM_INPUT,
+ in_ports,
&umidi->endpoints[i].in->ports[j].substream);
++in_ports;
}
@@ -1725,16 +1750,16 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi* umidi,
/*
* Returns MIDIStreaming device capabilities.
*/
-static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi,
- struct snd_usb_midi_endpoint_info* endpoints)
+static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi,
+ struct snd_usb_midi_endpoint_info *endpoints)
{
- struct usb_interface* intf;
+ struct usb_interface *intf;
struct usb_host_interface *hostif;
- struct usb_interface_descriptor* intfd;
- struct usb_ms_header_descriptor* ms_header;
+ struct usb_interface_descriptor *intfd;
+ struct usb_ms_header_descriptor *ms_header;
struct usb_host_endpoint *hostep;
- struct usb_endpoint_descriptor* ep;
- struct usb_ms_endpoint_descriptor* ms_ep;
+ struct usb_endpoint_descriptor *ep;
+ struct usb_ms_endpoint_descriptor *ms_ep;
int i, epidx;
intf = umidi->iface;
@@ -1742,7 +1767,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi,
return -ENXIO;
hostif = &intf->altsetting[0];
intfd = get_iface_desc(hostif);
- ms_header = (struct usb_ms_header_descriptor*)hostif->extra;
+ ms_header = (struct usb_ms_header_descriptor *)hostif->extra;
if (hostif->extralen >= 7 &&
ms_header->bLength >= 7 &&
ms_header->bDescriptorType == USB_DT_CS_INTERFACE &&
@@ -1759,7 +1784,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi,
ep = get_ep_desc(hostep);
if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep))
continue;
- ms_ep = (struct usb_ms_endpoint_descriptor*)hostep->extra;
+ ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra;
if (hostep->extralen < 4 ||
ms_ep->bLength < 4 ||
ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT ||
@@ -1783,9 +1808,10 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi,
* ESI MIDI Mate that try to use them anyway.
*/
endpoints[epidx].out_interval = 1;
- endpoints[epidx].out_cables = (1 << ms_ep->bNumEmbMIDIJack) - 1;
+ endpoints[epidx].out_cables =
+ (1 << ms_ep->bNumEmbMIDIJack) - 1;
dev_dbg(&umidi->dev->dev, "EP %02X: %d jack(s)\n",
- ep->bEndpointAddress, ms_ep->bNumEmbMIDIJack);
+ ep->bEndpointAddress, ms_ep->bNumEmbMIDIJack);
} else {
if (endpoints[epidx].in_ep) {
if (++epidx >= MIDI_MAX_ENDPOINTS) {
@@ -1799,9 +1825,10 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi,
endpoints[epidx].in_interval = ep->bInterval;
else if (snd_usb_get_speed(umidi->dev) == USB_SPEED_LOW)
endpoints[epidx].in_interval = 1;
- endpoints[epidx].in_cables = (1 << ms_ep->bNumEmbMIDIJack) - 1;
+ endpoints[epidx].in_cables =
+ (1 << ms_ep->bNumEmbMIDIJack) - 1;
dev_dbg(&umidi->dev->dev, "EP %02X: %d jack(s)\n",
- ep->bEndpointAddress, ms_ep->bNumEmbMIDIJack);
+ ep->bEndpointAddress, ms_ep->bNumEmbMIDIJack);
}
}
return 0;
@@ -1825,7 +1852,7 @@ static int roland_load_get(struct snd_kcontrol *kcontrol,
static int roland_load_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *value)
{
- struct snd_usb_midi* umidi = kcontrol->private_data;
+ struct snd_usb_midi *umidi = kcontrol->private_data;
int changed;
if (value->value.enumerated.item[0] > 1)
@@ -1851,11 +1878,11 @@ static struct snd_kcontrol_new roland_load_ctl = {
* On Roland devices, use the second alternate setting to be able to use
* the interrupt input endpoint.
*/
-static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi)
+static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi *umidi)
{
- struct usb_interface* intf;
+ struct usb_interface *intf;
struct usb_host_interface *hostif;
- struct usb_interface_descriptor* intfd;
+ struct usb_interface_descriptor *intfd;
intf = umidi->iface;
if (!intf || intf->num_altsetting != 2)
@@ -1864,8 +1891,10 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi)
hostif = &intf->altsetting[1];
intfd = get_iface_desc(hostif);
if (intfd->bNumEndpoints != 2 ||
- (get_endpoint(hostif, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK ||
- (get_endpoint(hostif, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT)
+ (get_endpoint(hostif, 0)->bmAttributes &
+ USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK ||
+ (get_endpoint(hostif, 1)->bmAttributes &
+ USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT)
return;
dev_dbg(&umidi->dev->dev, "switching to altsetting %d with int ep\n",
@@ -1881,14 +1910,14 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi)
/*
* Try to find any usable endpoints in the interface.
*/
-static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi,
- struct snd_usb_midi_endpoint_info* endpoint,
+static int snd_usbmidi_detect_endpoints(struct snd_usb_midi *umidi,
+ struct snd_usb_midi_endpoint_info *endpoint,
int max_endpoints)
{
- struct usb_interface* intf;
+ struct usb_interface *intf;
struct usb_host_interface *hostif;
- struct usb_interface_descriptor* intfd;
- struct usb_endpoint_descriptor* epd;
+ struct usb_interface_descriptor *intfd;
+ struct usb_endpoint_descriptor *epd;
int i, out_eps = 0, in_eps = 0;
if (USB_ID_VENDOR(umidi->usb_id) == 0x0582)
@@ -1929,8 +1958,8 @@ static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi,
/*
* Detects the endpoints for one-port-per-endpoint protocols.
*/
-static int snd_usbmidi_detect_per_port_endpoints(struct snd_usb_midi* umidi,
- struct snd_usb_midi_endpoint_info* endpoints)
+static int snd_usbmidi_detect_per_port_endpoints(struct snd_usb_midi *umidi,
+ struct snd_usb_midi_endpoint_info *endpoints)
{
int err, i;
@@ -1947,13 +1976,13 @@ static int snd_usbmidi_detect_per_port_endpoints(struct snd_usb_midi* umidi,
/*
* Detects the endpoints and ports of Yamaha devices.
*/
-static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi,
- struct snd_usb_midi_endpoint_info* endpoint)
+static int snd_usbmidi_detect_yamaha(struct snd_usb_midi *umidi,
+ struct snd_usb_midi_endpoint_info *endpoint)
{
- struct usb_interface* intf;
+ struct usb_interface *intf;
struct usb_host_interface *hostif;
- struct usb_interface_descriptor* intfd;
- uint8_t* cs_desc;
+ struct usb_interface_descriptor *intfd;
+ uint8_t *cs_desc;
intf = umidi->iface;
if (!intf)
@@ -1972,9 +2001,11 @@ static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi,
cs_desc += cs_desc[0]) {
if (cs_desc[1] == USB_DT_CS_INTERFACE) {
if (cs_desc[2] == UAC_MIDI_IN_JACK)
- endpoint->in_cables = (endpoint->in_cables << 1) | 1;
+ endpoint->in_cables =
+ (endpoint->in_cables << 1) | 1;
else if (cs_desc[2] == UAC_MIDI_OUT_JACK)
- endpoint->out_cables = (endpoint->out_cables << 1) | 1;
+ endpoint->out_cables =
+ (endpoint->out_cables << 1) | 1;
}
}
if (!endpoint->in_cables && !endpoint->out_cables)
@@ -1986,12 +2017,12 @@ static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi,
/*
* Detects the endpoints and ports of Roland devices.
*/
-static int snd_usbmidi_detect_roland(struct snd_usb_midi* umidi,
- struct snd_usb_midi_endpoint_info* endpoint)
+static int snd_usbmidi_detect_roland(struct snd_usb_midi *umidi,
+ struct snd_usb_midi_endpoint_info *endpoint)
{
- struct usb_interface* intf;
+ struct usb_interface *intf;
struct usb_host_interface *hostif;
- u8* cs_desc;
+ u8 *cs_desc;
intf = umidi->iface;
if (!intf)
@@ -2024,14 +2055,14 @@ static int snd_usbmidi_detect_roland(struct snd_usb_midi* umidi,
/*
* Creates the endpoints and their ports for Midiman devices.
*/
-static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi,
- struct snd_usb_midi_endpoint_info* endpoint)
+static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi *umidi,
+ struct snd_usb_midi_endpoint_info *endpoint)
{
struct snd_usb_midi_endpoint_info ep_info;
- struct usb_interface* intf;
+ struct usb_interface *intf;
struct usb_host_interface *hostif;
- struct usb_interface_descriptor* intfd;
- struct usb_endpoint_descriptor* epd;
+ struct usb_interface_descriptor *intfd;
+ struct usb_endpoint_descriptor *epd;
int cable, err;
intf = umidi->iface;
@@ -2068,39 +2099,50 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi,
epd = get_endpoint(hostif, 4);
if (!usb_endpoint_dir_out(epd) ||
!usb_endpoint_xfer_bulk(epd)) {
- dev_dbg(&umidi->dev->dev, "endpoint[4] isn't bulk output\n");
+ dev_dbg(&umidi->dev->dev,
+ "endpoint[4] isn't bulk output\n");
return -ENXIO;
}
}
- ep_info.out_ep = get_endpoint(hostif, 2)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+ ep_info.out_ep = get_endpoint(hostif, 2)->bEndpointAddress &
+ USB_ENDPOINT_NUMBER_MASK;
ep_info.out_interval = 0;
ep_info.out_cables = endpoint->out_cables & 0x5555;
- err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]);
+ err = snd_usbmidi_out_endpoint_create(umidi, &ep_info,
+ &umidi->endpoints[0]);
if (err < 0)
return err;
- ep_info.in_ep = get_endpoint(hostif, 0)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+ ep_info.in_ep = get_endpoint(hostif, 0)->bEndpointAddress &
+ USB_ENDPOINT_NUMBER_MASK;
ep_info.in_interval = get_endpoint(hostif, 0)->bInterval;
ep_info.in_cables = endpoint->in_cables;
- err = snd_usbmidi_in_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]);
+ err = snd_usbmidi_in_endpoint_create(umidi, &ep_info,
+ &umidi->endpoints[0]);
if (err < 0)
return err;
if (endpoint->out_cables > 0x0001) {
- ep_info.out_ep = get_endpoint(hostif, 4)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+ ep_info.out_ep = get_endpoint(hostif, 4)->bEndpointAddress &
+ USB_ENDPOINT_NUMBER_MASK;
ep_info.out_cables = endpoint->out_cables & 0xaaaa;
- err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[1]);
+ err = snd_usbmidi_out_endpoint_create(umidi, &ep_info,
+ &umidi->endpoints[1]);
if (err < 0)
return err;
}
for (cable = 0; cable < 0x10; ++cable) {
if (endpoint->out_cables & (1 << cable))
- snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_OUTPUT, cable,
+ snd_usbmidi_init_substream(umidi,
+ SNDRV_RAWMIDI_STREAM_OUTPUT,
+ cable,
&umidi->endpoints[cable & 1].out->ports[cable].substream);
if (endpoint->in_cables & (1 << cable))
- snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_INPUT, cable,
+ snd_usbmidi_init_substream(umidi,
+ SNDRV_RAWMIDI_STREAM_INPUT,
+ cable,
&umidi->endpoints[0].in->ports[cable].substream);
}
return 0;
@@ -2110,7 +2152,7 @@ static struct snd_rawmidi_global_ops snd_usbmidi_ops = {
.get_port_info = snd_usbmidi_get_port_info,
};
-static int snd_usbmidi_create_rawmidi(struct snd_usb_midi* umidi,
+static int snd_usbmidi_create_rawmidi(struct snd_usb_midi *umidi,
int out_ports, int in_ports)
{
struct snd_rawmidi *rmidi;
@@ -2128,8 +2170,10 @@ static int snd_usbmidi_create_rawmidi(struct snd_usb_midi* umidi,
rmidi->ops = &snd_usbmidi_ops;
rmidi->private_data = umidi;
rmidi->private_free = snd_usbmidi_rawmidi_free;
- snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_usbmidi_output_ops);
- snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_usbmidi_input_ops);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &snd_usbmidi_output_ops);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ &snd_usbmidi_input_ops);
umidi->rmidi = rmidi;
return 0;
@@ -2138,16 +2182,16 @@ static int snd_usbmidi_create_rawmidi(struct snd_usb_midi* umidi,
/*
* Temporarily stop input.
*/
-void snd_usbmidi_input_stop(struct list_head* p)
+void snd_usbmidi_input_stop(struct list_head *p)
{
- struct snd_usb_midi* umidi;
+ struct snd_usb_midi *umidi;
unsigned int i, j;
umidi = list_entry(p, struct snd_usb_midi, list);
if (!umidi->input_running)
return;
for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
- struct snd_usb_midi_endpoint* ep = &umidi->endpoints[i];
+ struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i];
if (ep->in)
for (j = 0; j < INPUT_URBS; ++j)
usb_kill_urb(ep->in->urbs[j]);
@@ -2156,14 +2200,14 @@ void snd_usbmidi_input_stop(struct list_head* p)
}
EXPORT_SYMBOL(snd_usbmidi_input_stop);
-static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint* ep)
+static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint *ep)
{
unsigned int i;
if (!ep)
return;
for (i = 0; i < INPUT_URBS; ++i) {
- struct urb* urb = ep->urbs[i];
+ struct urb *urb = ep->urbs[i];
urb->dev = ep->umidi->dev;
snd_usbmidi_submit_urb(urb, GFP_KERNEL);
}
@@ -2172,9 +2216,9 @@ static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint* ep)
/*
* Resume input after a call to snd_usbmidi_input_stop().
*/
-void snd_usbmidi_input_start(struct list_head* p)
+void snd_usbmidi_input_start(struct list_head *p)
{
- struct snd_usb_midi* umidi;
+ struct snd_usb_midi *umidi;
int i;
umidi = list_entry(p, struct snd_usb_midi, list);
@@ -2187,14 +2231,42 @@ void snd_usbmidi_input_start(struct list_head* p)
EXPORT_SYMBOL(snd_usbmidi_input_start);
/*
+ * Prepare for suspend. Typically called from the USB suspend callback.
+ */
+void snd_usbmidi_suspend(struct list_head *p)
+{
+ struct snd_usb_midi *umidi;
+
+ umidi = list_entry(p, struct snd_usb_midi, list);
+ mutex_lock(&umidi->mutex);
+ snd_usbmidi_input_stop(p);
+ mutex_unlock(&umidi->mutex);
+}
+EXPORT_SYMBOL(snd_usbmidi_suspend);
+
+/*
+ * Resume. Typically called from the USB resume callback.
+ */
+void snd_usbmidi_resume(struct list_head *p)
+{
+ struct snd_usb_midi *umidi;
+
+ umidi = list_entry(p, struct snd_usb_midi, list);
+ mutex_lock(&umidi->mutex);
+ snd_usbmidi_input_start(p);
+ mutex_unlock(&umidi->mutex);
+}
+EXPORT_SYMBOL(snd_usbmidi_resume);
+
+/*
* Creates and registers everything needed for a MIDI streaming interface.
*/
int snd_usbmidi_create(struct snd_card *card,
- struct usb_interface* iface,
+ struct usb_interface *iface,
struct list_head *midi_list,
- const struct snd_usb_audio_quirk* quirk)
+ const struct snd_usb_audio_quirk *quirk)
{
- struct snd_usb_midi* umidi;
+ struct snd_usb_midi *umidi;
struct snd_usb_midi_endpoint_info endpoints[MIDI_MAX_ENDPOINTS];
int out_ports, in_ports;
int i, err;
@@ -2292,7 +2364,8 @@ int snd_usbmidi_create(struct snd_card *card,
err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
break;
default:
- dev_err(&umidi->dev->dev, "invalid quirk type %d\n", quirk->type);
+ dev_err(&umidi->dev->dev, "invalid quirk type %d\n",
+ quirk->type);
err = -ENXIO;
break;
}
diff --git a/sound/usb/midi.h b/sound/usb/midi.h
index 2fca80b744c0..ad8a3211f8e7 100644
--- a/sound/usb/midi.h
+++ b/sound/usb/midi.h
@@ -43,8 +43,10 @@ int snd_usbmidi_create(struct snd_card *card,
struct usb_interface *iface,
struct list_head *midi_list,
const struct snd_usb_audio_quirk *quirk);
-void snd_usbmidi_input_stop(struct list_head* p);
-void snd_usbmidi_input_start(struct list_head* p);
+void snd_usbmidi_input_stop(struct list_head *p);
+void snd_usbmidi_input_start(struct list_head *p);
void snd_usbmidi_disconnect(struct list_head *p);
+void snd_usbmidi_suspend(struct list_head *p);
+void snd_usbmidi_resume(struct list_head *p);
#endif /* __USBMIDI_H */
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 0b728d886f0d..2e4a9dbc51fa 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1340,12 +1340,11 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
*/
if (range > 384) {
usb_audio_warn(state->chip,
- "Warning! Unlikely big volume range (=%u), "
- "cval->res is probably wrong.",
+ "Warning! Unlikely big volume range (=%u), cval->res is probably wrong.",
range);
- usb_audio_warn(state->chip, "[%d] FU [%s] ch = %d, "
- "val = %d/%d/%d", cval->id,
- kctl->id.name, cval->channels,
+ usb_audio_warn(state->chip,
+ "[%d] FU [%s] ch = %d, val = %d/%d/%d",
+ cval->id, kctl->id.name, cval->channels,
cval->min, cval->max, cval->res);
}
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 7c57f2268dd7..19a921eb75f1 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -670,7 +670,7 @@ static int snd_usb_gamecon780_boot_quirk(struct usb_device *dev)
/* set the initial volume and don't change; other values are either
* too loud or silent due to firmware bug (bko#65251)
*/
- u8 buf[2] = { 0x74, 0xdc };
+ u8 buf[2] = { 0x74, 0xe3 };
return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
UAC_FU_VOLUME << 8, 9 << 8, buf, 2);