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authorLinus Torvalds <torvalds@linux-foundation.org>2018-10-25 09:00:15 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2018-10-25 09:00:15 -0700
commit3acbd2de6bc3af215c6ed7732dfc097d1e238503 (patch)
tree5152e90a4d2d586dd6ad1cf0b8f28c4de2e46e66
parentd49f8a52b15bf35db778035340d8a673149f9f93 (diff)
parentde7d83da84bdf0b5ec50b3b09249e608c0e4b81d (diff)
Merge tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "There have been little changes in ALSA core stuff, but ASoC core still kept rolling for the continued restructuring. The rest are lots of small driver-specific changes and some minor API updates. Here are highlights: General: - Appropriate fall-through annotations everywhere - Some code cleanup in memalloc code, handling non-cacahed pages more commonly in the helper - Deployment of SNDRV_PCM_INFO_SYNC_APPLPTR flag consistently Drivers: - More HD-audio CA0132 codec improvement for supporting other Creative boards - Plumbing legacy HD-audio codecs as ASoC BE on Intel SST; this will give move support of existing HD-audio devices with DSP - A few device-specific HD-audio quirks as usual - New quirk for RME CC devices and correction for B&W PX for USB-audio - FireWire: code refactoring including devres usages ASoC Core: - Continued componentization works; it's almost done! - A bunch of new for_each_foo macros - Cleanups and fixes in DAPM code ASoC Drivers: - MCLK support for several different devices, including CS42L51, STM32 SAI, and MAX98373 - Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and TI PCM3060" * tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (299 commits) ASoC: stm32: sai: fix master clock naming ASoC: stm32: add clock dependency for sai ALSA: hda/ca0132 - Actually fix microphone issue ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks ASoC: wm2000: Remove wm2000_read helper function ASoC: cs42l51: fix mclk support ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer ASoC: cs42l51: add mclk support ASoC: stm32: sai: set sai as mclk clock provider ASoC: dt-bindings: add mclk support to cs42l51 ASoC: dt-bindings: add mclk provider support to stm32 sai ASoC: soc-core: fix trivial checkpatch issues ASoC: dapm: Add support for hw_free on CODEC to CODEC links ASoC: Intel: kbl_da7219_max98927: minor white space clean up ALSA: i2c/cs8427: Fix int to char conversion ALSA: doc: Brush up the old writing-an-alsa-driver ASoC: rsnd: tidyup SSICR::SWSP for TDM ASoC: rsnd: enable TDM settings for SSI parent ASoC: pcm3168a: add hw constraint for capture channel ...
-rw-r--r--Documentation/devicetree/bindings/sound/adi,adau1977.txt54
-rw-r--r--Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt24
-rw-r--r--Documentation/devicetree/bindings/sound/cs42l51.txt17
-rw-r--r--Documentation/devicetree/bindings/sound/maxim,max98088.txt23
-rw-r--r--Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt23
-rw-r--r--Documentation/devicetree/bindings/sound/nau8822.txt16
-rw-r--r--Documentation/devicetree/bindings/sound/pcm3060.txt17
-rw-r--r--Documentation/devicetree/bindings/sound/qcom,q6afe.txt18
-rw-r--r--Documentation/devicetree/bindings/sound/renesas,rsnd.txt5
-rw-r--r--Documentation/devicetree/bindings/sound/st,sta32x.txt9
-rw-r--r--Documentation/devicetree/bindings/sound/st,stm32-sai.txt7
-rw-r--r--Documentation/devicetree/bindings/sound/sun4i-i2s.txt2
-rw-r--r--Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt12
-rw-r--r--Documentation/devicetree/bindings/sound/ts3a227e.txt2
-rw-r--r--Documentation/devicetree/bindings/sound/wm8782.txt17
-rw-r--r--Documentation/devicetree/bindings/trivial-devices.txt1
-rw-r--r--Documentation/devicetree/bindings/vendor-prefixes.txt1
-rw-r--r--Documentation/sound/hd-audio/models.rst2
-rw-r--r--Documentation/sound/kernel-api/writing-an-alsa-driver.rst307
-rw-r--r--MAINTAINERS7
-rw-r--r--include/sound/hda_codec.h (renamed from sound/pci/hda/hda_codec.h)0
-rw-r--r--include/sound/memalloc.h3
-rw-r--r--include/sound/rawmidi.h1
-rw-r--r--include/sound/simple_card_utils.h27
-rw-r--r--include/sound/soc-acpi-intel-match.h6
-rw-r--r--include/sound/soc-dapm.h9
-rw-r--r--include/sound/soc-dpcm.h10
-rw-r--r--include/sound/soc.h45
-rw-r--r--include/uapi/sound/asound.h2
-rw-r--r--sound/aoa/soundbus/i2sbus/core.c15
-rw-r--r--sound/arm/Kconfig1
-rw-r--r--sound/core/memalloc.c41
-rw-r--r--sound/core/oss/pcm_plugin.c4
-rw-r--r--sound/core/pcm_lib.c21
-rw-r--r--sound/core/rawmidi.c22
-rw-r--r--sound/core/seq/oss/seq_oss_timer.c2
-rw-r--r--sound/core/seq/seq_system.c22
-rw-r--r--sound/core/seq/seq_virmidi.c4
-rw-r--r--sound/core/sgbuf.c15
-rw-r--r--sound/firewire/Kconfig2
-rw-r--r--sound/firewire/amdtp-stream.c78
-rw-r--r--sound/firewire/bebob/bebob.c58
-rw-r--r--sound/firewire/bebob/bebob_maudio.c5
-rw-r--r--sound/firewire/dice/dice.c41
-rw-r--r--sound/firewire/digi00x/digi00x.c35
-rw-r--r--sound/firewire/fireface/ff.c36
-rw-r--r--sound/firewire/fireworks/fireworks.c69
-rw-r--r--sound/firewire/isight.c18
-rw-r--r--sound/firewire/motu/motu.c47
-rw-r--r--sound/firewire/oxfw/oxfw-scs1x.c5
-rw-r--r--sound/firewire/oxfw/oxfw-spkr.c5
-rw-r--r--sound/firewire/oxfw/oxfw-stream.c13
-rw-r--r--sound/firewire/oxfw/oxfw.c63
-rw-r--r--sound/firewire/tascam/tascam.c40
-rw-r--r--sound/hda/ext/hdac_ext_controller.c22
-rw-r--r--sound/i2c/cs8427.c2
-rw-r--r--sound/isa/opti9xx/opti92x-ad1848.c6
-rw-r--r--sound/isa/sb/sb8_main.c10
-rw-r--r--sound/mips/hal2.c13
-rw-r--r--sound/pci/asihpi/hpios.c2
-rw-r--r--sound/pci/atiixp.c6
-rw-r--r--sound/pci/au88x0/au88x0_core.c6
-rw-r--r--sound/pci/cs46xx/cs46xx_lib.c6
-rw-r--r--sound/pci/emu10k1/emupcm.c3
-rw-r--r--sound/pci/hda/hda_auto_parser.c2
-rw-r--r--sound/pci/hda/hda_beep.h2
-rw-r--r--sound/pci/hda/hda_bind.c14
-rw-r--r--sound/pci/hda/hda_codec.c2
-rw-r--r--sound/pci/hda/hda_controller.c36
-rw-r--r--sound/pci/hda/hda_controller.h20
-rw-r--r--sound/pci/hda/hda_eld.c2
-rw-r--r--sound/pci/hda/hda_generic.c2
-rw-r--r--sound/pci/hda/hda_hwdep.c2
-rw-r--r--sound/pci/hda/hda_intel.c112
-rw-r--r--sound/pci/hda/hda_jack.c2
-rw-r--r--sound/pci/hda/hda_proc.c2
-rw-r--r--sound/pci/hda/hda_sysfs.c2
-rw-r--r--sound/pci/hda/hda_tegra.c20
-rw-r--r--sound/pci/hda/patch_analog.c2
-rw-r--r--sound/pci/hda/patch_ca0110.c2
-rw-r--r--sound/pci/hda/patch_ca0132.c1617
-rw-r--r--sound/pci/hda/patch_cirrus.c2
-rw-r--r--sound/pci/hda/patch_cmedia.c2
-rw-r--r--sound/pci/hda/patch_conexant.c3
-rw-r--r--sound/pci/hda/patch_hdmi.c2
-rw-r--r--sound/pci/hda/patch_realtek.c29
-rw-r--r--sound/pci/hda/patch_si3054.c2
-rw-r--r--sound/pci/hda/patch_sigmatel.c22
-rw-r--r--sound/pci/hda/patch_via.c2
-rw-r--r--sound/pci/intel8x0.c97
-rw-r--r--sound/pci/intel8x0m.c20
-rw-r--r--sound/pci/rme32.c22
-rw-r--r--sound/pci/rme9652/hdspm.c2
-rw-r--r--sound/soc/amd/acp-da7219-max98357a.c77
-rw-r--r--sound/soc/amd/acp-pcm-dma.c30
-rw-r--r--sound/soc/amd/acp.h3
-rw-r--r--sound/soc/atmel/Kconfig12
-rw-r--r--sound/soc/atmel/Makefile2
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c13
-rw-r--r--sound/soc/atmel/mikroe-proto.c165
-rw-r--r--sound/soc/atmel/tse850-pcm5142.c78
-rw-r--r--sound/soc/bcm/cygnus-ssp.c13
-rw-r--r--sound/soc/codecs/Kconfig36
-rw-r--r--sound/soc/codecs/Makefile10
-rw-r--r--sound/soc/codecs/adau1761.c3
-rw-r--r--sound/soc/codecs/adau17x1.c86
-rw-r--r--sound/soc/codecs/adau17x1.h4
-rw-r--r--sound/soc/codecs/cs4265.c12
-rw-r--r--sound/soc/codecs/cs42l51.c21
-rw-r--r--sound/soc/codecs/dmic.c1
-rw-r--r--sound/soc/codecs/es8328.c4
-rw-r--r--sound/soc/codecs/hdac_hda.c483
-rw-r--r--sound/soc/codecs/hdac_hda.h24
-rw-r--r--sound/soc/codecs/hdac_hdmi.c11
-rw-r--r--sound/soc/codecs/max98088.c36
-rw-r--r--sound/soc/codecs/max98373.c47
-rw-r--r--sound/soc/codecs/nau8822.c1136
-rw-r--r--sound/soc/codecs/nau8822.h204
-rw-r--r--sound/soc/codecs/pcm186x.c3
-rw-r--r--sound/soc/codecs/pcm3060-i2c.c60
-rw-r--r--sound/soc/codecs/pcm3060-spi.c59
-rw-r--r--sound/soc/codecs/pcm3060.c295
-rw-r--r--sound/soc/codecs/pcm3060.h88
-rw-r--r--sound/soc/codecs/pcm3168a.c82
-rw-r--r--sound/soc/codecs/rt274.c2
-rw-r--r--sound/soc/codecs/rt5514-spi.c14
-rw-r--r--sound/soc/codecs/rt5651.c1
-rw-r--r--sound/soc/codecs/rt5663.c7
-rw-r--r--sound/soc/codecs/rt5668.c10
-rw-r--r--sound/soc/codecs/rt5670.c12
-rw-r--r--sound/soc/codecs/rt5677-spi.c1
-rw-r--r--sound/soc/codecs/rt5682.c86
-rw-r--r--sound/soc/codecs/rt5682.h14
-rw-r--r--sound/soc/codecs/sgtl5000.c2
-rw-r--r--sound/soc/codecs/sta32x.c30
-rw-r--r--sound/soc/codecs/tas5720.c103
-rw-r--r--sound/soc/codecs/tas6424.c58
-rw-r--r--sound/soc/codecs/tas6424.h10
-rw-r--r--sound/soc/codecs/tlv320aic31xx.c85
-rw-r--r--sound/soc/codecs/tlv320aic31xx.h23
-rw-r--r--sound/soc/codecs/tscs454.c2
-rw-r--r--sound/soc/codecs/wm2000.c54
-rw-r--r--sound/soc/codecs/wm8782.c63
-rw-r--r--sound/soc/codecs/wm8904.c1
-rw-r--r--sound/soc/codecs/wm8974.c1
-rw-r--r--sound/soc/codecs/wm9712.c3
-rw-r--r--sound/soc/codecs/wm_adsp.c26
-rw-r--r--sound/soc/davinci/davinci-mcasp.c37
-rw-r--r--sound/soc/fsl/fsl_asrc_dma.c2
-rw-r--r--sound/soc/fsl/fsl_esai.c2
-rw-r--r--sound/soc/fsl/fsl_utils.c4
-rw-r--r--sound/soc/fsl/pcm030-audio-fabric.c5
-rw-r--r--sound/soc/generic/audio-graph-card.c21
-rw-r--r--sound/soc/generic/audio-graph-scu-card.c55
-rw-r--r--sound/soc/generic/simple-card-utils.c53
-rw-r--r--sound/soc/generic/simple-card.c30
-rw-r--r--sound/soc/generic/simple-scu-card.c54
-rw-r--r--sound/soc/hisilicon/hi6210-i2s.c4
-rw-r--r--sound/soc/intel/atom/sst-mfld-platform-pcm.c4
-rw-r--r--sound/soc/intel/boards/Kconfig22
-rw-r--r--sound/soc/intel/boards/Makefile4
-rw-r--r--sound/soc/intel/boards/broadwell.c4
-rw-r--r--sound/soc/intel/boards/bytcr_rt5640.c4
-rw-r--r--sound/soc/intel/boards/bytcr_rt5651.c4
-rw-r--r--sound/soc/intel/boards/cht_bsw_rt5672.c9
-rw-r--r--sound/soc/intel/boards/kbl_da7219_max98927.c983
-rw-r--r--sound/soc/intel/boards/kbl_rt5663_max98927.c5
-rw-r--r--sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c5
-rw-r--r--sound/soc/intel/boards/skl_hda_dsp_common.c127
-rw-r--r--sound/soc/intel/boards/skl_hda_dsp_common.h38
-rw-r--r--sound/soc/intel/boards/skl_hda_dsp_generic.c183
-rw-r--r--sound/soc/intel/common/Makefile3
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-byt-match.c7
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-hda-match.c40
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-kbl-match.c13
-rw-r--r--sound/soc/intel/common/sst-firmware.c2
-rw-r--r--sound/soc/intel/skylake/skl-pcm.c71
-rw-r--r--sound/soc/intel/skylake/skl-topology.c4
-rw-r--r--sound/soc/intel/skylake/skl.c96
-rw-r--r--sound/soc/intel/skylake/skl.h12
-rw-r--r--sound/soc/mediatek/mt2701/mt2701-cs42448.c13
-rw-r--r--sound/soc/mediatek/mt2701/mt2701-wm8960.c14
-rw-r--r--sound/soc/mediatek/mt6797/mt6797-mt6351.c14
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-max98090.c13
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c12
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c12
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-rt5650.c12
-rw-r--r--sound/soc/meson/Kconfig13
-rw-r--r--sound/soc/meson/Makefile2
-rw-r--r--sound/soc/meson/axg-card.c16
-rw-r--r--sound/soc/meson/axg-fifo.c2
-rw-r--r--sound/soc/meson/axg-pdm.c654
-rw-r--r--sound/soc/meson/axg-tdm-interface.c50
-rw-r--r--sound/soc/nuc900/nuc900-ac97.c4
-rw-r--r--sound/soc/omap/omap-hdmi-audio.c4
-rw-r--r--sound/soc/pxa/Kconfig13
-rw-r--r--sound/soc/pxa/pxa-ssp.c6
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c48
-rw-r--r--sound/soc/qcom/apq8096.c7
-rw-r--r--sound/soc/qcom/qdsp6/q6adm.c17
-rw-r--r--sound/soc/qcom/qdsp6/q6asm-dai.c8
-rw-r--r--sound/soc/qcom/qdsp6/q6asm.c1
-rw-r--r--sound/soc/qcom/qdsp6/q6core.c9
-rw-r--r--sound/soc/qcom/sdm845.c7
-rw-r--r--sound/soc/rockchip/rk3288_hdmi_analog.c1
-rw-r--r--sound/soc/rockchip/rockchip_pcm.c3
-rw-r--r--sound/soc/samsung/tm2_wm5110.c13
-rw-r--r--sound/soc/sh/hac.c3
-rw-r--r--sound/soc/sh/rcar/adg.c4
-rw-r--r--sound/soc/sh/rcar/core.c124
-rw-r--r--sound/soc/sh/rcar/ctu.c2
-rw-r--r--sound/soc/sh/rcar/dma.c109
-rw-r--r--sound/soc/sh/rcar/gen.c33
-rw-r--r--sound/soc/sh/rcar/rsnd.h63
-rw-r--r--sound/soc/sh/rcar/src.c2
-rw-r--r--sound/soc/sh/rcar/ssi.c112
-rw-r--r--sound/soc/sh/rcar/ssiu.c92
-rw-r--r--sound/soc/soc-compress.c4
-rw-r--r--sound/soc/soc-core.c582
-rw-r--r--sound/soc/soc-dapm.c437
-rw-r--r--sound/soc/soc-ops.c4
-rw-r--r--sound/soc/soc-pcm.c253
-rw-r--r--sound/soc/soc-topology.c15
-rw-r--r--sound/soc/soc-utils.c4
-rw-r--r--sound/soc/stm/Kconfig1
-rw-r--r--sound/soc/stm/stm32_sai.c2
-rw-r--r--sound/soc/stm/stm32_sai.h3
-rw-r--r--sound/soc/stm/stm32_sai_sub.c281
-rw-r--r--sound/soc/sunxi/Kconfig17
-rw-r--r--sound/soc/sunxi/Makefile2
-rw-r--r--sound/soc/sunxi/sun4i-i2s.c82
-rw-r--r--sound/soc/sunxi/sun50i-codec-analog.c444
-rw-r--r--sound/soc/sunxi/sun8i-adda-pr-regmap.c102
-rw-r--r--sound/soc/sunxi/sun8i-adda-pr-regmap.h7
-rw-r--r--sound/soc/sunxi/sun8i-codec-analog.c79
-rw-r--r--sound/soc/sunxi/sun8i-codec.c22
-rw-r--r--sound/soc/tegra/tegra_sgtl5000.c17
-rw-r--r--sound/soc/txx9/txx9aclc-ac97.c3
-rw-r--r--sound/usb/caiaq/device.c1
-rw-r--r--sound/usb/midi.c3
-rw-r--r--sound/usb/mixer_quirks.c381
-rw-r--r--sound/usb/quirks-table.h9
-rw-r--r--sound/x86/intel_hdmi_audio.c29
-rw-r--r--sound/xen/xen_snd_front_alsa.c46
244 files changed, 10779 insertions, 2627 deletions
diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.txt b/Documentation/devicetree/bindings/sound/adi,adau1977.txt
new file mode 100644
index 000000000000..e79aeef73f28
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,adau1977.txt
@@ -0,0 +1,54 @@
+Analog Devices ADAU1977/ADAU1978/ADAU1979
+
+Datasheets:
+http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf
+http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf
+http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf
+
+This driver supports both the I2C and SPI bus.
+
+Required properties:
+ - compatible: Should contain one of the following:
+ "adi,adau1977"
+ "adi,adau1978"
+ "adi,adau1979"
+
+ - AVDD-supply: analog power supply for the device, please consult
+ Documentation/devicetree/bindings/regulator/regulator.txt
+
+Optional properties:
+ - reset-gpio: the reset pin for the chip, for more details consult
+ Documentation/devicetree/bindings/gpio/gpio.txt
+
+ - DVDD-supply: supply voltage for the digital core, please consult
+ Documentation/devicetree/bindings/regulator/regulator.txt
+
+For required properties on SPI, please consult
+Documentation/devicetree/bindings/spi/spi-bus.txt
+
+Required properties on I2C:
+
+ - reg: The i2c address. Value depends on the state of ADDR0
+ and ADDR1, as wired in hardware.
+
+Examples:
+
+ adau1977_spi: adau1977@0 {
+ compatible = "adi,adau1977";
+ spi-max-frequency = <600000>;
+
+ AVDD-supply = <&regulator>;
+ DVDD-supply = <&regulator_digital>;
+
+ reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>;
+ };
+
+ adau1977_i2c: adau1977@11 {
+ compatible = "adi,adau1977";
+ reg = <0x11>;
+
+ AVDD-supply = <&regulator>;
+ DVDD-supply = <&regulator_digital>;
+
+ reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt
new file mode 100644
index 000000000000..5672d0bc5b16
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt
@@ -0,0 +1,24 @@
+* Amlogic Audio PDM input
+
+Required properties:
+- compatible: 'amlogic,axg-pdm'
+- reg: physical base address of the controller and length of memory
+ mapped region.
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+ * "pclk" : peripheral clock.
+ * "dclk" : pdm digital clock
+ * "sysclk" : dsp system clock
+- #sound-dai-cells: must be 0.
+
+Example of PDM on the A113 SoC:
+
+pdm: audio-controller@ff632000 {
+ compatible = "amlogic,axg-pdm";
+ reg = <0x0 0xff632000 0x0 0x34>;
+ #sound-dai-cells = <0>;
+ clocks = <&clkc_audio AUD_CLKID_PDM>,
+ <&clkc_audio AUD_CLKID_PDM_DCLK>,
+ <&clkc_audio AUD_CLKID_PDM_SYSCLK>;
+ clock-names = "pclk", "dclk", "sysclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/cs42l51.txt b/Documentation/devicetree/bindings/sound/cs42l51.txt
new file mode 100644
index 000000000000..4b5de33ce377
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs42l51.txt
@@ -0,0 +1,17 @@
+CS42L51 audio CODEC
+
+Optional properties:
+
+ - clocks : a list of phandles + clock-specifiers, one for each entry in
+ clock-names
+
+ - clock-names : must contain "MCLK"
+
+Example:
+
+cs42l51: cs42l51@4a {
+ compatible = "cirrus,cs42l51";
+ reg = <0x4a>;
+ clocks = <&mclk_prov>;
+ clock-names = "MCLK";
+};
diff --git a/Documentation/devicetree/bindings/sound/maxim,max98088.txt b/Documentation/devicetree/bindings/sound/maxim,max98088.txt
new file mode 100644
index 000000000000..da764d913319
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/maxim,max98088.txt
@@ -0,0 +1,23 @@
+MAX98088 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible: "maxim,max98088" or "maxim,max98089".
+- reg: The I2C address of the device.
+
+Optional properties:
+
+- clocks: the clock provider of MCLK, see ../clock/clock-bindings.txt section
+ "consumer" for more information.
+- clock-names: must be set to "mclk"
+
+Example:
+
+max98089: codec@10 {
+ compatible = "maxim,max98089";
+ reg = <0x10>;
+ clocks = <&clks IMX6QDL_CLK_CKO2>;
+ clock-names = "mclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt b/Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt
new file mode 100644
index 000000000000..912f8fae11c5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt
@@ -0,0 +1,23 @@
+Mikroe-PROTO audio board
+
+Required properties:
+ - compatible: "mikroe,mikroe-proto"
+ - dai-format: Must be "i2s".
+ - i2s-controller: The phandle of the I2S controller.
+ - audio-codec: The phandle of the WM8731 audio codec.
+Optional properties:
+ - model: The user-visible name of this sound complex.
+ - bitclock-master: Indicates dai-link bit clock master; for details see simple-card.txt (1).
+ - frame-master: Indicates dai-link frame master; for details see simple-card.txt (1).
+
+(1) : There must be the same master for both bit and frame clocks.
+
+Example:
+ sound {
+ compatible = "mikroe,mikroe-proto";
+ model = "wm8731 @ sama5d2_xplained";
+ i2s-controller = <&i2s0>;
+ audio-codec = <&wm8731>;
+ dai-format = "i2s";
+ };
+};
diff --git a/Documentation/devicetree/bindings/sound/nau8822.txt b/Documentation/devicetree/bindings/sound/nau8822.txt
new file mode 100644
index 000000000000..a471d162d4e5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nau8822.txt
@@ -0,0 +1,16 @@
+NAU8822 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+ - compatible : "nuvoton,nau8822"
+
+ - reg : the I2C address of the device.
+
+Example:
+
+codec: nau8822@1a {
+ compatible = "nuvoton,nau8822";
+ reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/pcm3060.txt b/Documentation/devicetree/bindings/sound/pcm3060.txt
new file mode 100644
index 000000000000..90fcb8523099
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/pcm3060.txt
@@ -0,0 +1,17 @@
+PCM3060 audio CODEC
+
+This driver supports both I2C and SPI.
+
+Required properties:
+
+- compatible: "ti,pcm3060"
+
+- reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Examples:
+
+ pcm3060: pcm3060@46 {
+ compatible = "ti,pcm3060";
+ reg = <0x46>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
index a8179409c194..d74888b9f1bb 100644
--- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
+++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
@@ -49,7 +49,7 @@ configuration of each dai. Must contain the following properties.
Usage: required for mi2s interface
Value type: <prop-encoded-array>
Definition: Must be list of serial data lines used by this dai.
- should be one or more of the 1-4 sd lines.
+ should be one or more of the 0-3 sd lines.
- qcom,tdm-sync-mode:
Usage: required for tdm interface
@@ -137,42 +137,42 @@ q6afe@4 {
prim-mi2s-rx@16 {
reg = <16>;
- qcom,sd-lines = <1 3>;
+ qcom,sd-lines = <0 2>;
};
prim-mi2s-tx@17 {
reg = <17>;
- qcom,sd-lines = <2>;
+ qcom,sd-lines = <1>;
};
sec-mi2s-rx@18 {
reg = <18>;
- qcom,sd-lines = <1 4>;
+ qcom,sd-lines = <0 3>;
};
sec-mi2s-tx@19 {
reg = <19>;
- qcom,sd-lines = <2>;
+ qcom,sd-lines = <1>;
};
tert-mi2s-rx@20 {
reg = <20>;
- qcom,sd-lines = <2 4>;
+ qcom,sd-lines = <1 3>;
};
tert-mi2s-tx@21 {
reg = <21>;
- qcom,sd-lines = <1>;
+ qcom,sd-lines = <0>;
};
quat-mi2s-rx@22 {
reg = <22>;
- qcom,sd-lines = <1>;
+ qcom,sd-lines = <0>;
};
quat-mi2s-tx@23 {
reg = <23>;
- qcom,sd-lines = <2>;
+ qcom,sd-lines = <1>;
};
};
};
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index 9e764270c36b..d92b705e7917 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -340,10 +340,12 @@ Required properties:
- compatible : "renesas,rcar_sound-<soctype>", fallbacks
"renesas,rcar_sound-gen1" if generation1, and
"renesas,rcar_sound-gen2" if generation2 (or RZ/G1)
- "renesas,rcar_sound-gen3" if generation3
+ "renesas,rcar_sound-gen3" if generation3 (or RZ/G2)
Examples with soctypes are:
- "renesas,rcar_sound-r8a7743" (RZ/G1M)
+ - "renesas,rcar_sound-r8a7744" (RZ/G1N)
- "renesas,rcar_sound-r8a7745" (RZ/G1E)
+ - "renesas,rcar_sound-r8a774a1" (RZ/G2M)
- "renesas,rcar_sound-r8a7778" (R-Car M1A)
- "renesas,rcar_sound-r8a7779" (R-Car H1)
- "renesas,rcar_sound-r8a7790" (R-Car H2)
@@ -353,6 +355,7 @@ Required properties:
- "renesas,rcar_sound-r8a7795" (R-Car H3)
- "renesas,rcar_sound-r8a7796" (R-Car M3-W)
- "renesas,rcar_sound-r8a77965" (R-Car M3-N)
+ - "renesas,rcar_sound-r8a77990" (R-Car E3)
- reg : Should contain the register physical address.
required register is
SRU/ADG/SSI if generation1
diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt
index 255de3ae5b2f..52265fb757c5 100644
--- a/Documentation/devicetree/bindings/sound/st,sta32x.txt
+++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt
@@ -19,6 +19,10 @@ Required properties:
Optional properties:
+ - clocks, clock-names: Clock specifier for XTI input clock.
+ If specified, the clock will be enabled when the codec is probed,
+ and disabled when it is removed. The 'clock-names' must be set to 'xti'.
+
- st,output-conf: number, Selects the output configuration:
0: 2-channel (full-bridge) power, 2-channel data-out
1: 2 (half-bridge). 1 (full-bridge) on-board power
@@ -39,6 +43,9 @@ Optional properties:
- st,thermal-warning-recover:
If present, thermal warning recovery is enabled.
+ - st,fault-detect-recovery:
+ If present, fault detect recovery is enabled.
+
- st,thermal-warning-adjustment:
If present, thermal warning adjustment is enabled.
@@ -76,6 +83,8 @@ Example:
codec: sta32x@38 {
compatible = "st,sta32x";
reg = <0x1c>;
+ clocks = <&clock>;
+ clock-names = "xti";
reset-gpios = <&gpio1 19 0>;
power-down-gpios = <&gpio1 16 0>;
st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel
diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt
index 3a3fc506e43a..3f4467ff0aa2 100644
--- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt
+++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt
@@ -31,7 +31,11 @@ SAI subnodes required properties:
- reg: Base address and size of SAI sub-block register set.
- clocks: Must contain one phandle and clock specifier pair
for sai_ck which feeds the internal clock generator.
+ If the SAI shares a master clock, with another SAI set as MCLK
+ clock provider, SAI provider phandle must be specified here.
- clock-names: Must contain "sai_ck".
+ Must also contain "MCLK", if SAI shares a master clock,
+ with a SAI set as MCLK clock provider.
- dmas: see Documentation/devicetree/bindings/dma/stm32-dma.txt
- dma-names: identifier string for each DMA request line
"tx": if sai sub-block is configured as playback DAI
@@ -51,6 +55,9 @@ SAI subnodes Optional properties:
configured according to protocol defined in related DAI link node,
such as i2s, left justified, right justified, dsp and pdm protocols.
Note: ac97 protocol is not supported by SAI driver
+ - #clock-cells: should be 0. This property must be present if the SAI device
+ is a master clock provider, according to clocks bindings, described in
+ Documentation/devicetree/bindings/clock/clock-bindings.txt.
The device node should contain one 'port' child node with one child 'endpoint'
node, according to the bindings defined in Documentation/devicetree/bindings/
diff --git a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
index b9d50d6cdef3..61e71c1729e0 100644
--- a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
+++ b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
@@ -10,6 +10,7 @@ Required properties:
- "allwinner,sun6i-a31-i2s"
- "allwinner,sun8i-a83t-i2s"
- "allwinner,sun8i-h3-i2s"
+ - "allwinner,sun50i-a64-codec-i2s"
- reg: physical base address of the controller and length of memory mapped
region.
- interrupts: should contain the I2S interrupt.
@@ -26,6 +27,7 @@ Required properties for the following compatibles:
- "allwinner,sun6i-a31-i2s"
- "allwinner,sun8i-a83t-i2s"
- "allwinner,sun8i-h3-i2s"
+ - "allwinner,sun50i-a64-codec-i2s"
- resets: phandle to the reset line for this codec
Example:
diff --git a/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt b/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt
new file mode 100644
index 000000000000..4f8ad0e04d20
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt
@@ -0,0 +1,12 @@
+* Allwinner A64 Codec Analog Controls
+
+Required properties:
+- compatible: must be one of the following compatibles:
+ - "allwinner,sun50i-a64-codec-analog"
+- reg: must contain the registers location and length
+
+Example:
+ codec_analog: codec-analog@1f015c0 {
+ compatible = "allwinner,sun50i-a64-codec-analog";
+ reg = <0x01f015c0 0x4>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt
index 3ed8359144d3..21ab45bc7e8f 100644
--- a/Documentation/devicetree/bindings/sound/ts3a227e.txt
+++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt
@@ -14,7 +14,7 @@ Required properties:
Optional properies:
- ti,micbias: Intended MICBIAS voltage (datasheet section 9.6.7).
- Select 0/1/2/3/4/5/6/7 to specify MACBIAS voltage
+ Select 0/1/2/3/4/5/6/7 to specify MICBIAS voltage
2.1V/2.2V/2.3V/2.4V/2.5V/2.6V/2.7V/2.8V
Default value is "1" (2.2V).
diff --git a/Documentation/devicetree/bindings/sound/wm8782.txt b/Documentation/devicetree/bindings/sound/wm8782.txt
new file mode 100644
index 000000000000..256cdec6ec4d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8782.txt
@@ -0,0 +1,17 @@
+WM8782 stereo ADC
+
+This device does not have any control interface or reset pins.
+
+Required properties:
+
+ - compatible : "wlf,wm8782"
+ - Vdda-supply : phandle to a regulator for the analog power supply (2.7V - 5.5V)
+ - Vdd-supply : phandle to a regulator for the digital power supply (2.7V - 3.6V)
+
+Example:
+
+wm8782: stereo-adc {
+ compatible = "wlf,wm8782";
+ Vdda-supply = <&vdda_supply>;
+ Vdd-supply = <&vdd_supply>;
+};
diff --git a/Documentation/devicetree/bindings/trivial-devices.txt b/Documentation/devicetree/bindings/trivial-devices.txt
index 763a2808a95c..69c934aec13b 100644
--- a/Documentation/devicetree/bindings/trivial-devices.txt
+++ b/Documentation/devicetree/bindings/trivial-devices.txt
@@ -35,7 +35,6 @@ at,24c08 i2c serial eeprom (24cxx)
atmel,at97sc3204t i2c trusted platform module (TPM)
capella,cm32181 CM32181: Ambient Light Sensor
capella,cm3232 CM3232: Ambient Light Sensor
-cirrus,cs42l51 Cirrus Logic CS42L51 audio codec
dallas,ds1374 I2C, 32-Bit Binary Counter Watchdog RTC with Trickle Charger and Reset Input/Output
dallas,ds1631 High-Precision Digital Thermometer
dallas,ds1672 Dallas DS1672 Real-time Clock
diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt
index 2c3fc512e746..f26bf667e530 100644
--- a/Documentation/devicetree/bindings/vendor-prefixes.txt
+++ b/Documentation/devicetree/bindings/vendor-prefixes.txt
@@ -235,6 +235,7 @@ micrel Micrel Inc.
microchip Microchip Technology Inc.
microcrystal Micro Crystal AG
micron Micron Technology Inc.
+mikroe MikroElektronika d.o.o.
minix MINIX Technology Ltd.
miramems MiraMEMS Sensing Technology Co., Ltd.
mitsubishi Mitsubishi Electric Corporation
diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst
index e06238131f77..368a07a165f5 100644
--- a/Documentation/sound/hd-audio/models.rst
+++ b/Documentation/sound/hd-audio/models.rst
@@ -309,6 +309,8 @@ asus-nx50
ASUS Nx50 fixups
asus-nx51
ASUS Nx51 fixups
+asus-g751
+ ASUS G751 fixups
alc891-headset
Headset mode support on ALC891
alc891-headset-multi
diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
index a0b268466cb1..b37234afdfa1 100644
--- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
+++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
@@ -3,8 +3,6 @@ Writing an ALSA Driver
======================
:Author: Takashi Iwai <tiwai@suse.de>
-:Date: Oct 15, 2007
-:Edition: 0.3.7
Preface
=======
@@ -21,11 +19,6 @@ explain the general topic of linux kernel coding and doesn't cover
low-level driver implementation details. It only describes the standard
way to write a PCI sound driver on ALSA.
-If you are already familiar with the older ALSA ver.0.5.x API, you can
-check the drivers such as ``sound/pci/es1938.c`` or
-``sound/pci/maestro3.c`` which have also almost the same code-base in
-the ALSA 0.5.x tree, so you can compare the differences.
-
This document is still a draft version. Any feedback and corrections,
please!!
@@ -35,24 +28,7 @@ File Tree Structure
General
-------
-The ALSA drivers are provided in two ways.
-
-One is the trees provided as a tarball or via cvs from the ALSA's ftp
-site, and another is the 2.6 (or later) Linux kernel tree. To
-synchronize both, the ALSA driver tree is split into two different
-trees: alsa-kernel and alsa-driver. The former contains purely the
-source code for the Linux 2.6 (or later) tree. This tree is designed
-only for compilation on 2.6 or later environment. The latter,
-alsa-driver, contains many subtle files for compiling ALSA drivers
-outside of the Linux kernel tree, wrapper functions for older 2.2 and
-2.4 kernels, to adapt the latest kernel API, and additional drivers
-which are still in development or in tests. The drivers in alsa-driver
-tree will be moved to alsa-kernel (and eventually to the 2.6 kernel
-tree) when they are finished and confirmed to work fine.
-
-The file tree structure of ALSA driver is depicted below. Both
-alsa-kernel and alsa-driver have almost the same file structure, except
-for “core” directory. It's named as “acore” in alsa-driver tree.
+The file tree structure of ALSA driver is depicted below.
::
@@ -61,14 +37,11 @@ for “core” directory. It's named as “acore” in alsa-driver tree.
/oss
/seq
/oss
- /instr
- /ioctl32
/include
/drivers
/mpu401
/opl3
/i2c
- /l3
/synth
/emux
/pci
@@ -80,6 +53,7 @@ for “core” directory. It's named as “acore” in alsa-driver tree.
/sparc
/usb
/pcmcia /(cards)
+ /soc
/oss
@@ -99,13 +73,6 @@ directory. The rawmidi OSS emulation is included in the ALSA rawmidi
code since it's quite small. The sequencer code is stored in
``core/seq/oss`` directory (see `below <#core-seq-oss>`__).
-core/ioctl32
-~~~~~~~~~~~~
-
-This directory contains the 32bit-ioctl wrappers for 64bit architectures
-such like x86-64, ppc64 and sparc64. For 32bit and alpha architectures,
-these are not compiled.
-
core/seq
~~~~~~~~
@@ -119,11 +86,6 @@ core/seq/oss
This contains the OSS sequencer emulation codes.
-core/seq/instr
-~~~~~~~~~~~~~~
-
-This directory contains the modules for the sequencer instrument layer.
-
include directory
-----------------
@@ -161,11 +123,6 @@ Although there is a standard i2c layer on Linux, ALSA has its own i2c
code for some cards, because the soundcard needs only a simple operation
and the standard i2c API is too complicated for such a purpose.
-i2c/l3
-~~~~~~
-
-This is a sub-directory for ARM L3 i2c.
-
synth directory
---------------
@@ -209,11 +166,19 @@ The PCMCIA, especially PCCard drivers will go here. CardBus drivers will
be in the pci directory, because their API is identical to that of
standard PCI cards.
+soc directory
+-------------
+
+This directory contains the codes for ASoC (ALSA System on Chip)
+layer including ASoC core, codec and machine drivers.
+
oss directory
-------------
-The OSS/Lite source files are stored here in Linux 2.6 (or later) tree.
-In the ALSA driver tarball, this directory is empty, of course :)
+Here contains OSS/Lite codes.
+All codes have been deprecated except for dmasound on m68k as of
+writing this.
+
Basic Flow for PCI Drivers
==========================
@@ -352,10 +317,8 @@ to details explained in the following section.
/* (3) */
err = snd_mychip_create(card, pci, &chip);
- if (err < 0) {
- snd_card_free(card);
- return err;
- }
+ if (err < 0)
+ goto error;
/* (4) */
strcpy(card->driver, "My Chip");
@@ -368,22 +331,23 @@ to details explained in the following section.
/* (6) */
err = snd_card_register(card);
- if (err < 0) {
- snd_card_free(card);
- return err;
- }
+ if (err < 0)
+ goto error;
/* (7) */
pci_set_drvdata(pci, card);
dev++;
return 0;
+
+ error:
+ snd_card_free(card);
+ return err;
}
/* destructor -- see the "Destructor" sub-section */
static void snd_mychip_remove(struct pci_dev *pci)
{
snd_card_free(pci_get_drvdata(pci));
- pci_set_drvdata(pci, NULL);
}
@@ -445,14 +409,26 @@ In this part, the PCI resources are allocated.
struct mychip *chip;
....
err = snd_mychip_create(card, pci, &chip);
- if (err < 0) {
- snd_card_free(card);
- return err;
- }
+ if (err < 0)
+ goto error;
The details will be explained in the section `PCI Resource
Management`_.
+When something goes wrong, the probe function needs to deal with the
+error. In this example, we have a single error handling path placed
+at the end of the function.
+
+::
+
+ error:
+ snd_card_free(card);
+ return err;
+
+Since each component can be properly freed, the single
+:c:func:`snd_card_free()` call should suffice in most cases.
+
+
4) Set the driver ID and name strings.
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
@@ -486,10 +462,8 @@ too.
::
err = snd_card_register(card);
- if (err < 0) {
- snd_card_free(card);
- return err;
- }
+ if (err < 0)
+ goto error;
Will be explained in the section `Management of Cards and
Components`_, too.
@@ -513,14 +487,13 @@ The destructor, remove callback, simply releases the card instance. Then
the ALSA middle layer will release all the attached components
automatically.
-It would be typically like the following:
+It would be typically just :c:func:`calling snd_card_free()`:
::
static void snd_mychip_remove(struct pci_dev *pci)
{
snd_card_free(pci_get_drvdata(pci));
- pci_set_drvdata(pci, NULL);
}
@@ -546,7 +519,7 @@ in the source file. If the code is split into several files, the files
without module options don't need them.
In addition to these headers, you'll need ``<linux/interrupt.h>`` for
-interrupt handling, and ``<asm/io.h>`` for I/O access. If you use the
+interrupt handling, and ``<linux/io.h>`` for I/O access. If you use the
:c:func:`mdelay()` or :c:func:`udelay()` functions, you'll need
to include ``<linux/delay.h>`` too.
@@ -720,6 +693,13 @@ function, which will call the real destructor.
where :c:func:`snd_mychip_free()` is the real destructor.
+The demerit of this method is the obviously more amount of codes.
+The merit is, however, you can trigger the own callback at registering
+and disconnecting the card via setting in snd_device_ops.
+About the registering and disconnecting the card, see the subsections
+below.
+
+
Registration and Release
------------------------
@@ -905,10 +885,8 @@ Resource Allocation
-------------------
The allocation of I/O ports and irqs is done via standard kernel
-functions. Unlike ALSA ver.0.5.x., there are no helpers for that. And
-these resources must be released in the destructor function (see below).
-Also, on ALSA 0.9.x, you don't need to allocate (pseudo-)DMA for PCI
-like in ALSA 0.5.x.
+functions. These resources must be released in the destructor
+function (see below).
Now assume that the PCI device has an I/O port with 8 bytes and an
interrupt. Then :c:type:`struct mychip <mychip>` will have the
@@ -1064,7 +1042,8 @@ and the allocation would be like below:
::
- if ((err = pci_request_regions(pci, "My Chip")) < 0) {
+ err = pci_request_regions(pci, "My Chip");
+ if (err < 0) {
kfree(chip);
return err;
}
@@ -1086,6 +1065,21 @@ and the corresponding destructor would be:
....
}
+Of course, a modern way with :c:func:`pci_iomap()` will make things a
+bit easier, too.
+
+::
+
+ err = pci_request_regions(pci, "My Chip");
+ if (err < 0) {
+ kfree(chip);
+ return err;
+ }
+ chip->iobase_virt = pci_iomap(pci, 0, 0);
+
+which is paired with :c:func:`pci_iounmap()` at destructor.
+
+
PCI Entries
-----------
@@ -1154,13 +1148,6 @@ And at last, the module entries:
Note that these module entries are tagged with ``__init`` and ``__exit``
prefixes.
-Oh, one thing was forgotten. If you have no exported symbols, you need
-to declare it in 2.2 or 2.4 kernels (it's not necessary in 2.6 kernels).
-
-::
-
- EXPORT_NO_SYMBOLS;
-
That's all!
PCM Interface
@@ -2113,6 +2100,16 @@ non-contiguous buffers. The mmap calls this callback to get the page
address. Some examples will be explained in the later section `Buffer
and Memory Management`_, too.
+mmap calllback
+~~~~~~~~~~~~~~
+
+This is another optional callback for controlling mmap behavior.
+Once when defined, PCM core calls this callback when a page is
+memory-mapped instead of dealing via the standard helper.
+If you need special handling (due to some architecture or
+device-specific issues), implement everything here as you like.
+
+
PCM Interrupt Handler
---------------------
@@ -2370,6 +2367,27 @@ to define the inverse rule:
hw_rule_format_by_channels, NULL,
SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+One typical usage of the hw constraints is to align the buffer size
+with the period size. As default, ALSA PCM core doesn't enforce the
+buffer size to be aligned with the period size. For example, it'd be
+possible to have a combination like 256 period bytes with 999 buffer
+bytes.
+
+Many device chips, however, require the buffer to be a multiple of
+periods. In such a case, call
+:c:func:`snd_pcm_hw_constraint_integer()` for
+``SNDRV_PCM_HW_PARAM_PERIODS``.
+
+::
+
+ snd_pcm_hw_constraint_integer(substream->runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+
+This assures that the number of periods is integer, hence the buffer
+size is aligned with the period size.
+
+The hw constraint is a very much powerful mechanism to define the
+preferred PCM configuration, and there are relevant helpers.
I won't give more details here, rather I would like to say, “Luke, use
the source.”
@@ -3712,7 +3730,14 @@ example, for an intermediate buffer. Since the allocated pages are not
contiguous, you need to set the ``page`` callback to obtain the physical
address at every offset.
-The implementation of ``page`` callback would be like this:
+The easiest way to achieve it would be to use
+:c:func:`snd_pcm_lib_alloc_vmalloc_buffer()` for allocating the buffer
+via :c:func:`vmalloc()`, and set :c:func:`snd_pcm_sgbuf_ops_page()` to
+the ``page`` callback. At release, you need to call
+:c:func:`snd_pcm_lib_free_vmalloc_buffer()`.
+
+If you want to implementation the ``page`` manually, it would be like
+this:
::
@@ -3848,7 +3873,9 @@ Power Management
If the chip is supposed to work with suspend/resume functions, you need
to add power-management code to the driver. The additional code for
-power-management should be ifdef-ed with ``CONFIG_PM``.
+power-management should be ifdef-ed with ``CONFIG_PM``, or annotated
+with __maybe_unused attribute; otherwise the compiler will complain
+you.
If the driver *fully* supports suspend/resume that is, the device can be
properly resumed to its state when suspend was called, you can set the
@@ -3879,18 +3906,16 @@ the case of PCI drivers, the callbacks look like below:
::
- #ifdef CONFIG_PM
- static int snd_my_suspend(struct pci_dev *pci, pm_message_t state)
+ static int __maybe_unused snd_my_suspend(struct device *dev)
{
.... /* do things for suspend */
return 0;
}
- static int snd_my_resume(struct pci_dev *pci)
+ static int __maybe_unused snd_my_resume(struct device *dev)
{
.... /* do things for suspend */
return 0;
}
- #endif
The scheme of the real suspend job is as follows.
@@ -3909,18 +3934,14 @@ The scheme of the real suspend job is as follows.
6. Stop the hardware if necessary.
-7. Disable the PCI device by calling
- :c:func:`pci_disable_device()`. Then, call
- :c:func:`pci_save_state()` at last.
-
A typical code would be like:
::
- static int mychip_suspend(struct pci_dev *pci, pm_message_t state)
+ static int __maybe_unused mychip_suspend(struct device *dev)
{
/* (1) */
- struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_card *card = dev_get_drvdata(dev);
struct mychip *chip = card->private_data;
/* (2) */
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -3932,9 +3953,6 @@ A typical code would be like:
snd_mychip_save_registers(chip);
/* (6) */
snd_mychip_stop_hardware(chip);
- /* (7) */
- pci_disable_device(pci);
- pci_save_state(pci);
return 0;
}
@@ -3943,44 +3961,35 @@ The scheme of the real resume job is as follows.
1. Retrieve the card and the chip data.
-2. Set up PCI. First, call :c:func:`pci_restore_state()`. Then
- enable the pci device again by calling
- :c:func:`pci_enable_device()`. Call
- :c:func:`pci_set_master()` if necessary, too.
+2. Re-initialize the chip.
-3. Re-initialize the chip.
+3. Restore the saved registers if necessary.
-4. Restore the saved registers if necessary.
+4. Resume the mixer, e.g. calling :c:func:`snd_ac97_resume()`.
-5. Resume the mixer, e.g. calling :c:func:`snd_ac97_resume()`.
+5. Restart the hardware (if any).
-6. Restart the hardware (if any).
-
-7. Call :c:func:`snd_power_change_state()` with
+6. Call :c:func:`snd_power_change_state()` with
``SNDRV_CTL_POWER_D0`` to notify the processes.
A typical code would be like:
::
- static int mychip_resume(struct pci_dev *pci)
+ static int __maybe_unused mychip_resume(struct pci_dev *pci)
{
/* (1) */
- struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_card *card = dev_get_drvdata(dev);
struct mychip *chip = card->private_data;
/* (2) */
- pci_restore_state(pci);
- pci_enable_device(pci);
- pci_set_master(pci);
- /* (3) */
snd_mychip_reinit_chip(chip);
- /* (4) */
+ /* (3) */
snd_mychip_restore_registers(chip);
- /* (5) */
+ /* (4) */
snd_ac97_resume(chip->ac97);
- /* (6) */
+ /* (5) */
snd_mychip_restart_chip(chip);
- /* (7) */
+ /* (6) */
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
return 0;
}
@@ -4046,15 +4055,14 @@ And next, set suspend/resume callbacks to the pci_driver.
::
+ static SIMPLE_DEV_PM_OPS(snd_my_pm_ops, mychip_suspend, mychip_resume);
+
static struct pci_driver driver = {
.name = KBUILD_MODNAME,
.id_table = snd_my_ids,
.probe = snd_my_probe,
.remove = snd_my_remove,
- #ifdef CONFIG_PM
- .suspend = snd_my_suspend,
- .resume = snd_my_resume,
- #endif
+ .driver.pm = &snd_my_pm_ops,
};
Module Parameters
@@ -4078,7 +4086,7 @@ variables, instead. ``enable`` option is not always necessary in this
case, but it would be better to have a dummy option for compatibility.
The module parameters must be declared with the standard
-``module_param()()``, ``module_param_array()()`` and
+``module_param()``, ``module_param_array()`` and
:c:func:`MODULE_PARM_DESC()` macros.
The typical coding would be like below:
@@ -4094,15 +4102,14 @@ The typical coding would be like below:
module_param_array(enable, bool, NULL, 0444);
MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
-Also, don't forget to define the module description, classes, license
-and devices. Especially, the recent modprobe requires to define the
+Also, don't forget to define the module description and the license.
+Especially, the recent modprobe requires to define the
module license as GPL, etc., otherwise the system is shown as “tainted”.
::
- MODULE_DESCRIPTION("My Chip");
+ MODULE_DESCRIPTION("Sound driver for My Chip");
MODULE_LICENSE("GPL");
- MODULE_SUPPORTED_DEVICE("{{Vendor,My Chip Name}}");
How To Put Your Driver Into ALSA Tree
@@ -4117,21 +4124,17 @@ a question now: how to put my own driver into the ALSA driver tree? Here
Suppose that you create a new PCI driver for the card “xyz”. The card
module name would be snd-xyz. The new driver is usually put into the
-alsa-driver tree, ``alsa-driver/pci`` directory in the case of PCI
-cards. Then the driver is evaluated, audited and tested by developers
-and users. After a certain time, the driver will go to the alsa-kernel
-tree (to the corresponding directory, such as ``alsa-kernel/pci``) and
-eventually will be integrated into the Linux 2.6 tree (the directory
-would be ``linux/sound/pci``).
+alsa-driver tree, ``sound/pci`` directory in the case of PCI
+cards.
In the following sections, the driver code is supposed to be put into
-alsa-driver tree. The two cases are covered: a driver consisting of a
+Linux kernel tree. The two cases are covered: a driver consisting of a
single source file and one consisting of several source files.
Driver with A Single Source File
--------------------------------
-1. Modify alsa-driver/pci/Makefile
+1. Modify sound/pci/Makefile
Suppose you have a file xyz.c. Add the following two lines
@@ -4160,52 +4163,43 @@ Driver with A Single Source File
For the details of Kconfig script, refer to the kbuild documentation.
-3. Run cvscompile script to re-generate the configure script and build
- the whole stuff again.
-
Drivers with Several Source Files
---------------------------------
Suppose that the driver snd-xyz have several source files. They are
-located in the new subdirectory, pci/xyz.
+located in the new subdirectory, sound/pci/xyz.
-1. Add a new directory (``xyz``) in ``alsa-driver/pci/Makefile`` as
- below
+1. Add a new directory (``sound/pci/xyz``) in ``sound/pci/Makefile``
+ as below
::
- obj-$(CONFIG_SND) += xyz/
+ obj-$(CONFIG_SND) += sound/pci/xyz/
-2. Under the directory ``xyz``, create a Makefile
+2. Under the directory ``sound/pci/xyz``, create a Makefile
::
- ifndef SND_TOPDIR
- SND_TOPDIR=../..
- endif
-
- include $(SND_TOPDIR)/toplevel.config
- include $(SND_TOPDIR)/Makefile.conf
-
snd-xyz-objs := xyz.o abc.o def.o
-
obj-$(CONFIG_SND_XYZ) += snd-xyz.o
- include $(SND_TOPDIR)/Rules.make
-
3. Create the Kconfig entry
This procedure is as same as in the last section.
-4. Run cvscompile script to re-generate the configure script and build
- the whole stuff again.
Useful Functions
================
:c:func:`snd_printk()` and friends
----------------------------------------
+----------------------------------
+
+.. note:: This subsection describes a few helper functions for
+ decorating a bit more on the standard :c:func:`printk()` & co.
+ However, in general, the use of such helpers is no longer recommended.
+ If possible, try to stick with the standard functions like
+ :c:func:`dev_err()` or :c:func:`pr_err()`.
ALSA provides a verbose version of the :c:func:`printk()` function.
If a kernel config ``CONFIG_SND_VERBOSE_PRINTK`` is set, this function
@@ -4221,13 +4215,10 @@ just like :c:func:`snd_printk()`. If the ALSA is compiled without
the debugging flag, it's ignored.
:c:func:`snd_printdd()` is compiled in only when
-``CONFIG_SND_DEBUG_VERBOSE`` is set. Please note that
-``CONFIG_SND_DEBUG_VERBOSE`` is not set as default even if you configure
-the alsa-driver with ``--with-debug=full`` option. You need to give
-explicitly ``--with-debug=detect`` option instead.
+``CONFIG_SND_DEBUG_VERBOSE`` is set.
:c:func:`snd_BUG()`
-------------------------
+-------------------
It shows the ``BUG?`` message and stack trace as well as
:c:func:`snd_BUG_ON()` at the point. It's useful to show that a
@@ -4236,7 +4227,7 @@ fatal error happens there.
When no debug flag is set, this macro is ignored.
:c:func:`snd_BUG_ON()`
-----------------------------
+----------------------
:c:func:`snd_BUG_ON()` macro is similar with
:c:func:`WARN_ON()` macro. For example, snd_BUG_ON(!pointer); or
diff --git a/MAINTAINERS b/MAINTAINERS
index 46a7656bf96b..2b928aa857ce 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -14759,6 +14759,13 @@ L: netdev@vger.kernel.org
S: Maintained
F: drivers/net/ethernet/ti/netcp*
+TI PCM3060 ASoC CODEC DRIVER
+M: Kirill Marinushkin <kmarinushkin@birdec.tech>
+L: alsa-devel@alsa-project.org (moderated for non-subscribers)
+S: Maintained
+F: Documentation/devicetree/bindings/sound/pcm3060.txt
+F: sound/soc/codecs/pcm3060*
+
TI TAS571X FAMILY ASoC CODEC DRIVER
M: Kevin Cernekee <cernekee@chromium.org>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
diff --git a/sound/pci/hda/hda_codec.h b/include/sound/hda_codec.h
index 0d98bb9068b1..0d98bb9068b1 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/include/sound/hda_codec.h
diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h
index 67561b997915..af3fa577fa06 100644
--- a/include/sound/memalloc.h
+++ b/include/sound/memalloc.h
@@ -47,10 +47,13 @@ struct snd_dma_device {
#define SNDRV_DMA_TYPE_UNKNOWN 0 /* not defined */
#define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */
#define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */
+#define SNDRV_DMA_TYPE_DEV_UC 5 /* continuous non-cahced */
#ifdef CONFIG_SND_DMA_SGBUF
#define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */
+#define SNDRV_DMA_TYPE_DEV_UC_SG 6 /* SG non-cached */
#else
#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */
+#define SNDRV_DMA_TYPE_DEV_UC_SG SNDRV_DMA_TYPE_DEV_UC
#endif
#ifdef CONFIG_GENERIC_ALLOCATOR
#define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */
diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h
index 6665cb29e1a2..3b5a061132b6 100644
--- a/include/sound/rawmidi.h
+++ b/include/sound/rawmidi.h
@@ -171,6 +171,7 @@ int __snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream,
unsigned char *buffer, int count);
int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream,
int count);
+int snd_rawmidi_proceed(struct snd_rawmidi_substream *substream);
/* main midi functions */
diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h
index 8bc5e2d8b13c..fb0318f9b10f 100644
--- a/include/sound/simple_card_utils.h
+++ b/include/sound/simple_card_utils.h
@@ -51,29 +51,35 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card,
#define asoc_simple_card_parse_clk_cpu(dev, node, dai_link, simple_dai) \
asoc_simple_card_parse_clk(dev, node, dai_link->cpu_of_node, simple_dai, \
- dai_link->cpu_dai_name)
+ dai_link->cpu_dai_name, NULL)
#define asoc_simple_card_parse_clk_codec(dev, node, dai_link, simple_dai) \
asoc_simple_card_parse_clk(dev, node, dai_link->codec_of_node, simple_dai,\
- dai_link->codec_dai_name)
+ dai_link->codec_dai_name, dai_link->codecs)
int asoc_simple_card_parse_clk(struct device *dev,
struct device_node *node,
struct device_node *dai_of_node,
struct asoc_simple_dai *simple_dai,
- const char *name);
+ const char *dai_name,
+ struct snd_soc_dai_link_component *dlc);
int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai);
void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai);
#define asoc_simple_card_parse_cpu(node, dai_link, \
list_name, cells_name, is_single_link) \
- asoc_simple_card_parse_dai(node, &dai_link->cpu_of_node, \
+ asoc_simple_card_parse_dai(node, NULL, \
+ &dai_link->cpu_of_node, \
&dai_link->cpu_dai_name, list_name, cells_name, is_single_link)
#define asoc_simple_card_parse_codec(node, dai_link, list_name, cells_name) \
- asoc_simple_card_parse_dai(node, &dai_link->codec_of_node, \
- &dai_link->codec_dai_name, list_name, cells_name, NULL)
+ asoc_simple_card_parse_dai(node, dai_link->codecs, \
+ &dai_link->codec_of_node, \
+ &dai_link->codec_dai_name, \
+ list_name, cells_name, NULL)
#define asoc_simple_card_parse_platform(node, dai_link, list_name, cells_name) \
- asoc_simple_card_parse_dai(node, &dai_link->platform_of_node, \
+ asoc_simple_card_parse_dai(node, dai_link->platform, \
+ &dai_link->platform_of_node, \
NULL, list_name, cells_name, NULL)
int asoc_simple_card_parse_dai(struct device_node *node,
+ struct snd_soc_dai_link_component *dlc,
struct device_node **endpoint_np,
const char **dai_name,
const char *list_name,
@@ -81,12 +87,15 @@ int asoc_simple_card_parse_dai(struct device_node *node,
int *is_single_links);
#define asoc_simple_card_parse_graph_cpu(ep, dai_link) \
- asoc_simple_card_parse_graph_dai(ep, &dai_link->cpu_of_node, \
+ asoc_simple_card_parse_graph_dai(ep, NULL, \
+ &dai_link->cpu_of_node, \
&dai_link->cpu_dai_name)
#define asoc_simple_card_parse_graph_codec(ep, dai_link) \
- asoc_simple_card_parse_graph_dai(ep, &dai_link->codec_of_node, \
+ asoc_simple_card_parse_graph_dai(ep, dai_link->codecs, \
+ &dai_link->codec_of_node, \
&dai_link->codec_dai_name)
int asoc_simple_card_parse_graph_dai(struct device_node *ep,
+ struct snd_soc_dai_link_component *dlc,
struct device_node **endpoint_np,
const char **dai_name);
diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h
index bb1d24b703fb..f48f59e5b7b0 100644
--- a/include/sound/soc-acpi-intel-match.h
+++ b/include/sound/soc-acpi-intel-match.h
@@ -25,4 +25,10 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[];
extern struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[];
extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[];
+/*
+ * generic table used for HDA codec-based platforms, possibly with
+ * additional ACPI-enumerated codecs
+ */
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[];
+
#endif
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index fdaaafdc7a00..bd8163f151cb 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -406,12 +406,6 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
struct snd_soc_dai *dai);
int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card);
void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card);
-int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
- struct snd_soc_pcm_runtime *rtd,
- const struct snd_soc_pcm_stream *params,
- unsigned int num_params,
- struct snd_soc_dapm_widget *source,
- struct snd_soc_dapm_widget *sink);
/* dapm path setup */
int snd_soc_dapm_new_widgets(struct snd_soc_card *card);
@@ -590,9 +584,6 @@ struct snd_soc_dapm_widget {
void *priv; /* widget specific data */
struct regulator *regulator; /* attached regulator */
struct pinctrl *pinctrl; /* attached pinctrl */
- const struct snd_soc_pcm_stream *params; /* params for dai links */
- unsigned int num_params; /* number of params for dai links */
- unsigned int params_select; /* currently selected param for dai link */
/* dapm control */
int reg; /* negative reg = no direct dapm */
diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h
index 9bb92f187af8..4be3a2b7c106 100644
--- a/include/sound/soc-dpcm.h
+++ b/include/sound/soc-dpcm.h
@@ -103,6 +103,16 @@ struct snd_soc_dpcm_runtime {
int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */
};
+#define for_each_dpcm_fe(be, stream, dpcm) \
+ list_for_each_entry(dpcm, &(be)->dpcm[stream].fe_clients, list_fe)
+
+#define for_each_dpcm_be(fe, stream, dpcm) \
+ list_for_each_entry(dpcm, &(fe)->dpcm[stream].be_clients, list_be)
+#define for_each_dpcm_be_safe(fe, stream, dpcm, _dpcm) \
+ list_for_each_entry_safe(dpcm, _dpcm, &(fe)->dpcm[stream].be_clients, list_be)
+#define for_each_dpcm_be_rollback(fe, stream, dpcm) \
+ list_for_each_entry_continue_reverse(dpcm, &(fe)->dpcm[stream].be_clients, list_be)
+
/* can this BE stop and free */
int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
struct snd_soc_pcm_runtime *be, int stream);
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 41cec42fb456..f1dab1f4b194 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -372,6 +372,11 @@
#define SND_SOC_COMP_ORDER_LATE 1
#define SND_SOC_COMP_ORDER_LAST 2
+#define for_each_comp_order(order) \
+ for (order = SND_SOC_COMP_ORDER_FIRST; \
+ order <= SND_SOC_COMP_ORDER_LAST; \
+ order++)
+
/*
* Bias levels
*
@@ -859,6 +864,11 @@ struct snd_soc_component {
#endif
};
+#define for_each_component_dais(component, dai)\
+ list_for_each_entry(dai, &(component)->dai_list, list)
+#define for_each_component_dais_safe(component, dai, _dai)\
+ list_for_each_entry_safe(dai, _dai, &(component)->dai_list, list)
+
struct snd_soc_rtdcom_list {
struct snd_soc_component *component;
struct list_head list; /* rtd::component_list */
@@ -915,6 +925,8 @@ struct snd_soc_dai_link {
*/
const char *platform_name;
struct device_node *platform_of_node;
+ struct snd_soc_dai_link_component *platform;
+
int id; /* optional ID for machine driver link identification */
const struct snd_soc_pcm_stream *params;
@@ -976,6 +988,10 @@ struct snd_soc_dai_link {
struct list_head list; /* DAI link list of the soc card */
struct snd_soc_dobj dobj; /* For topology */
};
+#define for_each_link_codecs(link, i, codec) \
+ for ((i) = 0; \
+ ((i) < link->num_codecs) && ((codec) = &link->codecs[i]); \
+ (i)++)
struct snd_soc_codec_conf {
/*
@@ -1054,7 +1070,6 @@ struct snd_soc_card {
struct snd_soc_dai_link *dai_link; /* predefined links only */
int num_links; /* predefined links only */
struct list_head dai_link_list; /* all links */
- int num_dai_links;
struct list_head rtd_list;
int num_rtd;
@@ -1092,6 +1107,7 @@ struct snd_soc_card {
/* lists of probed devices belonging to this card */
struct list_head component_dev_list;
+ struct list_head list;
struct list_head widgets;
struct list_head paths;
@@ -1114,6 +1130,23 @@ struct snd_soc_card {
void *drvdata;
};
+#define for_each_card_prelinks(card, i, link) \
+ for ((i) = 0; \
+ ((i) < (card)->num_links) && ((link) = &(card)->dai_link[i]); \
+ (i)++)
+
+#define for_each_card_links(card, link) \
+ list_for_each_entry(dai_link, &(card)->dai_link_list, list)
+#define for_each_card_links_safe(card, link, _link) \
+ list_for_each_entry_safe(link, _link, &(card)->dai_link_list, list)
+
+#define for_each_card_rtds(card, rtd) \
+ list_for_each_entry(rtd, &(card)->rtd_list, list)
+#define for_each_card_rtds_safe(card, rtd, _rtd) \
+ list_for_each_entry_safe(rtd, _rtd, &(card)->rtd_list, list)
+
+#define for_each_card_components(card, component) \
+ list_for_each_entry(component, &(card)->component_dev_list, card_list)
/* SoC machine DAI configuration, glues a codec and cpu DAI together */
struct snd_soc_pcm_runtime {
@@ -1124,6 +1157,8 @@ struct snd_soc_pcm_runtime {
enum snd_soc_pcm_subclass pcm_subclass;
struct snd_pcm_ops ops;
+ unsigned int params_select; /* currently selected param for dai link */
+
/* Dynamic PCM BE runtime data */
struct snd_soc_dpcm_runtime dpcm[2];
int fe_compr;
@@ -1152,6 +1187,13 @@ struct snd_soc_pcm_runtime {
unsigned int dev_registered:1;
unsigned int pop_wait:1;
};
+#define for_each_rtd_codec_dai(rtd, i, dai)\
+ for ((i) = 0; \
+ ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \
+ (i)++)
+#define for_each_rtd_codec_dai_rollback(rtd, i, dai) \
+ for (; ((i--) >= 0) && ((dai) = rtd->codec_dais[i]);)
+
/* mixer control */
struct soc_mixer_control {
@@ -1359,6 +1401,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
INIT_LIST_HEAD(&card->dapm_list);
INIT_LIST_HEAD(&card->aux_comp_list);
INIT_LIST_HEAD(&card->component_dev_list);
+ INIT_LIST_HEAD(&card->list);
}
static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc)
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index ed0a120d4f08..404d4b9ffe76 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -752,7 +752,7 @@ struct snd_timer_info {
#define SNDRV_TIMER_PSFLG_EARLY_EVENT (1<<2) /* write early event to the poll queue */
struct snd_timer_params {
- unsigned int flags; /* flags - SNDRV_MIXER_PSFLG_* */
+ unsigned int flags; /* flags - SNDRV_TIMER_PSFLG_* */
unsigned int ticks; /* requested resolution in ticks */
unsigned int queue_size; /* total size of queue (32-1024) */
unsigned int reserved0; /* reserved, was: failure locations */
diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c
index 000b58522106..bd7c5029fc59 100644
--- a/sound/aoa/soundbus/i2sbus/core.c
+++ b/sound/aoa/soundbus/i2sbus/core.c
@@ -157,18 +157,19 @@ static int i2sbus_add_dev(struct macio_dev *macio,
struct device_node *child = NULL, *sound = NULL;
struct resource *r;
int i, layout = 0, rlen, ok = force;
- static const char *rnames[] = { "i2sbus: %s (control)",
- "i2sbus: %s (tx)",
- "i2sbus: %s (rx)" };
+ char node_name[6];
+ static const char *rnames[] = { "i2sbus: %pOFn (control)",
+ "i2sbus: %pOFn (tx)",
+ "i2sbus: %pOFn (rx)" };
static irq_handler_t ints[] = {
i2sbus_bus_intr,
i2sbus_tx_intr,
i2sbus_rx_intr
};
- if (strlen(np->name) != 5)
+ if (snprintf(node_name, sizeof(node_name), "%pOFn", np) != 5)
return 0;
- if (strncmp(np->name, "i2s-", 4))
+ if (strncmp(node_name, "i2s-", 4))
return 0;
dev = kzalloc(sizeof(struct i2sbus_dev), GFP_KERNEL);
@@ -228,13 +229,13 @@ static int i2sbus_add_dev(struct macio_dev *macio,
dev->sound.pcmid = -1;
dev->macio = macio;
dev->control = control;
- dev->bus_number = np->name[4] - 'a';
+ dev->bus_number = node_name[4] - 'a';
INIT_LIST_HEAD(&dev->sound.codec_list);
for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) {
dev->interrupts[i] = -1;
snprintf(dev->rnames[i], sizeof(dev->rnames[i]),
- rnames[i], np->name);
+ rnames[i], np);
}
for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) {
int irq = irq_of_parse_and_map(np, i);
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index 5fbd47a9177e..28867732a318 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -31,7 +31,6 @@ endif # SND_ARM
config SND_PXA2XX_LIB
tristate
- select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
select SND_DMAENGINE_PCM
config SND_PXA2XX_LIB_AC97
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 753d5fc4b284..59a4adc286ed 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -25,6 +25,9 @@
#include <linux/mm.h>
#include <linux/dma-mapping.h>
#include <linux/genalloc.h>
+#ifdef CONFIG_X86
+#include <asm/set_memory.h>
+#endif
#include <sound/memalloc.h>
/*
@@ -82,31 +85,32 @@ EXPORT_SYMBOL(snd_free_pages);
#ifdef CONFIG_HAS_DMA
/* allocate the coherent DMA pages */
-static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma)
+static void snd_malloc_dev_pages(struct snd_dma_buffer *dmab, size_t size)
{
- int pg;
gfp_t gfp_flags;
- if (WARN_ON(!dma))
- return NULL;
- pg = get_order(size);
gfp_flags = GFP_KERNEL
| __GFP_COMP /* compound page lets parts be mapped */
| __GFP_NORETRY /* don't trigger OOM-killer */
| __GFP_NOWARN; /* no stack trace print - this call is non-critical */
- return dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags);
+ dmab->area = dma_alloc_coherent(dmab->dev.dev, size, &dmab->addr,
+ gfp_flags);
+#ifdef CONFIG_X86
+ if (dmab->area && dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC)
+ set_memory_wc((unsigned long)dmab->area,
+ PAGE_ALIGN(size) >> PAGE_SHIFT);
+#endif
}
/* free the coherent DMA pages */
-static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr,
- dma_addr_t dma)
+static void snd_free_dev_pages(struct snd_dma_buffer *dmab)
{
- int pg;
-
- if (ptr == NULL)
- return;
- pg = get_order(size);
- dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma);
+#ifdef CONFIG_X86
+ if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC)
+ set_memory_wb((unsigned long)dmab->area,
+ PAGE_ALIGN(dmab->bytes) >> PAGE_SHIFT);
+#endif
+ dma_free_coherent(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr);
}
#ifdef CONFIG_GENERIC_ALLOCATOR
@@ -199,12 +203,15 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
*/
dmab->dev.type = SNDRV_DMA_TYPE_DEV;
#endif /* CONFIG_GENERIC_ALLOCATOR */
+ /* fall through */
case SNDRV_DMA_TYPE_DEV:
- dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr);
+ case SNDRV_DMA_TYPE_DEV_UC:
+ snd_malloc_dev_pages(dmab, size);
break;
#endif
#ifdef CONFIG_SND_DMA_SGBUF
case SNDRV_DMA_TYPE_DEV_SG:
+ case SNDRV_DMA_TYPE_DEV_UC_SG:
snd_malloc_sgbuf_pages(device, size, dmab, NULL);
break;
#endif
@@ -275,11 +282,13 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab)
break;
#endif /* CONFIG_GENERIC_ALLOCATOR */
case SNDRV_DMA_TYPE_DEV:
- snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr);
+ case SNDRV_DMA_TYPE_DEV_UC:
+ snd_free_dev_pages(dmab);
break;
#endif
#ifdef CONFIG_SND_DMA_SGBUF
case SNDRV_DMA_TYPE_DEV_SG:
+ case SNDRV_DMA_TYPE_DEV_UC_SG:
snd_free_sgbuf_pages(dmab);
break;
#endif
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index 0391cb1a4f19..141c5f3a9575 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -111,7 +111,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames)
while (plugin->next) {
if (plugin->dst_frames)
frames = plugin->dst_frames(plugin, frames);
- if (snd_BUG_ON(frames <= 0))
+ if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0))
return -ENXIO;
plugin = plugin->next;
err = snd_pcm_plugin_alloc(plugin, frames);
@@ -123,7 +123,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames)
while (plugin->prev) {
if (plugin->src_frames)
frames = plugin->src_frames(plugin, frames);
- if (snd_BUG_ON(frames <= 0))
+ if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0))
return -ENXIO;
plugin = plugin->prev;
err = snd_pcm_plugin_alloc(plugin, frames);
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 4e6110d778bd..40013b26f671 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -2172,18 +2172,25 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream,
if (err < 0)
goto _end_unlock;
+ runtime->twake = runtime->control->avail_min ? : 1;
+ if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
+ snd_pcm_update_hw_ptr(substream);
+
if (!is_playback &&
- runtime->status->state == SNDRV_PCM_STATE_PREPARED &&
- size >= runtime->start_threshold) {
- err = snd_pcm_start(substream);
- if (err < 0)
+ runtime->status->state == SNDRV_PCM_STATE_PREPARED) {
+ if (size >= runtime->start_threshold) {
+ err = snd_pcm_start(substream);
+ if (err < 0)
+ goto _end_unlock;
+ } else {
+ /* nothing to do */
+ err = 0;
goto _end_unlock;
+ }
}
- runtime->twake = runtime->control->avail_min ? : 1;
- if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
- snd_pcm_update_hw_ptr(substream);
avail = snd_pcm_avail(substream);
+
while (size > 0) {
snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
snd_pcm_uframes_t cont;
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 08d5662039e3..ee601d7f0926 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -1236,6 +1236,28 @@ int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream,
}
EXPORT_SYMBOL(snd_rawmidi_transmit);
+/**
+ * snd_rawmidi_proceed - Discard the all pending bytes and proceed
+ * @substream: rawmidi substream
+ *
+ * Return: the number of discarded bytes
+ */
+int snd_rawmidi_proceed(struct snd_rawmidi_substream *substream)
+{
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+ unsigned long flags;
+ int count = 0;
+
+ spin_lock_irqsave(&runtime->lock, flags);
+ if (runtime->avail < runtime->buffer_size) {
+ count = runtime->buffer_size - runtime->avail;
+ __snd_rawmidi_transmit_ack(substream, count);
+ }
+ spin_unlock_irqrestore(&runtime->lock, flags);
+ return count;
+}
+EXPORT_SYMBOL(snd_rawmidi_proceed);
+
static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream,
const unsigned char __user *userbuf,
const unsigned char *kernelbuf,
diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c
index ba127c22539a..0778d28421da 100644
--- a/sound/core/seq/oss/seq_oss_timer.c
+++ b/sound/core/seq/oss/seq_oss_timer.c
@@ -92,7 +92,7 @@ snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev)
case TMR_WAIT_REL:
parm += rec->cur_tick;
rec->realtime = 0;
- /* fall through and continue to next */
+ /* fall through */
case TMR_WAIT_ABS:
if (parm == 0) {
rec->realtime = 1;
diff --git a/sound/core/seq/seq_system.c b/sound/core/seq/seq_system.c
index 8ce1d0b40dce..0dc5d5a45ecc 100644
--- a/sound/core/seq/seq_system.c
+++ b/sound/core/seq/seq_system.c
@@ -123,6 +123,7 @@ int __init snd_seq_system_client_init(void)
{
struct snd_seq_port_callback pcallbacks;
struct snd_seq_port_info *port;
+ int err;
port = kzalloc(sizeof(*port), GFP_KERNEL);
if (!port)
@@ -134,6 +135,10 @@ int __init snd_seq_system_client_init(void)
/* register client */
sysclient = snd_seq_create_kernel_client(NULL, 0, "System");
+ if (sysclient < 0) {
+ kfree(port);
+ return sysclient;
+ }
/* register timer */
strcpy(port->name, "Timer");
@@ -144,7 +149,10 @@ int __init snd_seq_system_client_init(void)
port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT;
port->addr.client = sysclient;
port->addr.port = SNDRV_SEQ_PORT_SYSTEM_TIMER;
- snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port);
+ err = snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT,
+ port);
+ if (err < 0)
+ goto error_port;
/* register announcement port */
strcpy(port->name, "Announce");
@@ -154,16 +162,24 @@ int __init snd_seq_system_client_init(void)
port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT;
port->addr.client = sysclient;
port->addr.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE;
- snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port);
+ err = snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT,
+ port);
+ if (err < 0)
+ goto error_port;
announce_port = port->addr.port;
kfree(port);
return 0;
+
+ error_port:
+ snd_seq_system_client_done();
+ kfree(port);
+ return err;
}
/* unregister our internal client */
-void __exit snd_seq_system_client_done(void)
+void snd_seq_system_client_done(void)
{
int oldsysclient = sysclient;
diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c
index cb988efd1ed0..e5a40795914a 100644
--- a/sound/core/seq/seq_virmidi.c
+++ b/sound/core/seq/seq_virmidi.c
@@ -149,9 +149,7 @@ static void snd_vmidi_output_work(struct work_struct *work)
/* discard the outputs in dispatch mode unless subscribed */
if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH &&
!(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) {
- char buf[32];
- while (snd_rawmidi_transmit(substream, buf, sizeof(buf)) > 0)
- ; /* ignored */
+ snd_rawmidi_proceed(substream);
return;
}
diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c
index 84fffabdd129..c1cfaa01a5cb 100644
--- a/sound/core/sgbuf.c
+++ b/sound/core/sgbuf.c
@@ -23,6 +23,7 @@
#include <linux/mm.h>
#include <linux/vmalloc.h>
#include <linux/export.h>
+#include <asm/pgtable.h>
#include <sound/memalloc.h>
@@ -43,6 +44,8 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab)
dmab->area = NULL;
tmpb.dev.type = SNDRV_DMA_TYPE_DEV;
+ if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC_SG)
+ tmpb.dev.type = SNDRV_DMA_TYPE_DEV_UC;
tmpb.dev.dev = sgbuf->dev;
for (i = 0; i < sgbuf->pages; i++) {
if (!(sgbuf->table[i].addr & ~PAGE_MASK))
@@ -72,12 +75,20 @@ void *snd_malloc_sgbuf_pages(struct device *device,
struct snd_dma_buffer tmpb;
struct snd_sg_page *table;
struct page **pgtable;
+ int type = SNDRV_DMA_TYPE_DEV;
+ pgprot_t prot = PAGE_KERNEL;
dmab->area = NULL;
dmab->addr = 0;
dmab->private_data = sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL);
if (! sgbuf)
return NULL;
+ if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC_SG) {
+ type = SNDRV_DMA_TYPE_DEV_UC;
+#ifdef pgprot_noncached
+ prot = pgprot_noncached(PAGE_KERNEL);
+#endif
+ }
sgbuf->dev = device;
pages = snd_sgbuf_aligned_pages(size);
sgbuf->tblsize = sgbuf_align_table(pages);
@@ -98,7 +109,7 @@ void *snd_malloc_sgbuf_pages(struct device *device,
if (chunk > maxpages)
chunk = maxpages;
chunk <<= PAGE_SHIFT;
- if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, device,
+ if (snd_dma_alloc_pages_fallback(type, device,
chunk, &tmpb) < 0) {
if (!sgbuf->pages)
goto _failed;
@@ -125,7 +136,7 @@ void *snd_malloc_sgbuf_pages(struct device *device,
}
sgbuf->size = size;
- dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, PAGE_KERNEL);
+ dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, prot);
if (! dmab->area)
goto _failed;
if (res_size)
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index 529d9f405fa9..8a146b039276 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -147,7 +147,9 @@ config SND_FIREWIRE_MOTU
help
Say Y here to enable support for FireWire devices which MOTU produced:
* 828mk2
+ * Traveler
* 828mk3
+ * Audio Express
To compile this driver as a module, choose M here: the module
will be called snd-firewire-motu.
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index cb9acfe60f6a..fcd965f1d69e 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -140,6 +140,59 @@ const unsigned int amdtp_rate_table[CIP_SFC_COUNT] = {
};
EXPORT_SYMBOL(amdtp_rate_table);
+static int apply_constraint_to_size(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *s = hw_param_interval(params, rule->var);
+ const struct snd_interval *r =
+ hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval t = {
+ .min = s->min, .max = s->max, .integer = 1,
+ };
+ int i;
+
+ for (i = 0; i < CIP_SFC_COUNT; ++i) {
+ unsigned int rate = amdtp_rate_table[i];
+ unsigned int step = amdtp_syt_intervals[i];
+
+ if (!snd_interval_test(r, rate))
+ continue;
+
+ t.min = roundup(t.min, step);
+ t.max = rounddown(t.max, step);
+ }
+
+ if (snd_interval_checkempty(&t))
+ return -EINVAL;
+
+ return snd_interval_refine(s, &t);
+}
+
+static int apply_constraint_to_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *r =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ const struct snd_interval *s = hw_param_interval_c(params, rule->deps[0]);
+ struct snd_interval t = {
+ .min = UINT_MAX, .max = 0, .integer = 1,
+ };
+ int i;
+
+ for (i = 0; i < CIP_SFC_COUNT; ++i) {
+ unsigned int step = amdtp_syt_intervals[i];
+ unsigned int rate = amdtp_rate_table[i];
+
+ if (s->min % step || s->max % step)
+ continue;
+
+ t.min = min(t.min, rate);
+ t.max = max(t.max, rate);
+ }
+
+ return snd_interval_refine(r, &t);
+}
+
/**
* amdtp_stream_add_pcm_hw_constraints - add hw constraints for PCM substream
* @s: the AMDTP stream, which must be initialized.
@@ -194,16 +247,27 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
* number equals to SYT_INTERVAL. So the number is 8, 16 or 32,
* depending on its sampling rate. For accurate period interrupt, it's
* preferrable to align period/buffer sizes to current SYT_INTERVAL.
- *
- * TODO: These constraints can be improved with proper rules.
- * Currently apply LCM of SYT_INTERVALs.
*/
- err = snd_pcm_hw_constraint_step(runtime, 0,
- SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32);
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ apply_constraint_to_size, NULL,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ if (err < 0)
+ goto end;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ apply_constraint_to_rate, NULL,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1);
+ if (err < 0)
+ goto end;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+ apply_constraint_to_size, NULL,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ if (err < 0)
+ goto end;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ apply_constraint_to_rate, NULL,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1);
if (err < 0)
goto end;
- err = snd_pcm_hw_constraint_step(runtime, 0,
- SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32);
end:
return err;
}
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 93676354f87f..672d13488454 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -126,23 +126,6 @@ end:
return err;
}
-static void bebob_free(struct snd_bebob *bebob)
-{
- snd_bebob_stream_destroy_duplex(bebob);
- fw_unit_put(bebob->unit);
-
- kfree(bebob->maudio_special_quirk);
-
- mutex_destroy(&bebob->mutex);
- kfree(bebob);
-}
-
-/*
- * This module releases the FireWire unit data after all ALSA character devices
- * are released by applications. This is for releasing stream data or finishing
- * transactions safely. Thus at returning from .remove(), this module still keep
- * references for the unit.
- */
static void
bebob_card_free(struct snd_card *card)
{
@@ -152,7 +135,7 @@ bebob_card_free(struct snd_card *card)
clear_bit(bebob->card_index, devices_used);
mutex_unlock(&devices_mutex);
- bebob_free(card->private_data);
+ snd_bebob_stream_destroy_duplex(bebob);
}
static const struct snd_bebob_spec *
@@ -192,7 +175,6 @@ do_registration(struct work_struct *work)
return;
mutex_lock(&devices_mutex);
-
for (card_index = 0; card_index < SNDRV_CARDS; card_index++) {
if (!test_bit(card_index, devices_used) && enable[card_index])
break;
@@ -208,6 +190,11 @@ do_registration(struct work_struct *work)
mutex_unlock(&devices_mutex);
return;
}
+ set_bit(card_index, devices_used);
+ mutex_unlock(&devices_mutex);
+
+ bebob->card->private_free = bebob_card_free;
+ bebob->card->private_data = bebob;
err = name_device(bebob);
if (err < 0)
@@ -248,23 +235,10 @@ do_registration(struct work_struct *work)
if (err < 0)
goto error;
- set_bit(card_index, devices_used);
- mutex_unlock(&devices_mutex);
-
- /*
- * After registered, bebob instance can be released corresponding to
- * releasing the sound card instance.
- */
- bebob->card->private_free = bebob_card_free;
- bebob->card->private_data = bebob;
bebob->registered = true;
return;
error:
- mutex_unlock(&devices_mutex);
- snd_bebob_stream_destroy_duplex(bebob);
- kfree(bebob->maudio_special_quirk);
- bebob->maudio_special_quirk = NULL;
snd_card_free(bebob->card);
dev_info(&bebob->unit->device,
"Sound card registration failed: %d\n", err);
@@ -295,15 +269,15 @@ bebob_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry)
}
/* Allocate this independent of sound card instance. */
- bebob = kzalloc(sizeof(struct snd_bebob), GFP_KERNEL);
- if (bebob == NULL)
+ bebob = devm_kzalloc(&unit->device, sizeof(struct snd_bebob),
+ GFP_KERNEL);
+ if (!bebob)
return -ENOMEM;
-
bebob->unit = fw_unit_get(unit);
- bebob->entry = entry;
- bebob->spec = spec;
dev_set_drvdata(&unit->device, bebob);
+ bebob->entry = entry;
+ bebob->spec = spec;
mutex_init(&bebob->mutex);
spin_lock_init(&bebob->lock);
init_waitqueue_head(&bebob->hwdep_wait);
@@ -379,12 +353,12 @@ static void bebob_remove(struct fw_unit *unit)
cancel_delayed_work_sync(&bebob->dwork);
if (bebob->registered) {
- /* No need to wait for releasing card object in this context. */
- snd_card_free_when_closed(bebob->card);
- } else {
- /* Don't forget this case. */
- bebob_free(bebob);
+ // Block till all of ALSA character devices are released.
+ snd_card_free(bebob->card);
}
+
+ mutex_destroy(&bebob->mutex);
+ fw_unit_put(bebob->unit);
}
static const struct snd_bebob_rate_spec normal_rate_spec = {
diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c
index c266997ad299..51152ca4af57 100644
--- a/sound/firewire/bebob/bebob_maudio.c
+++ b/sound/firewire/bebob/bebob_maudio.c
@@ -261,8 +261,9 @@ snd_bebob_maudio_special_discover(struct snd_bebob *bebob, bool is1814)
struct special_params *params;
int err;
- params = kzalloc(sizeof(struct special_params), GFP_KERNEL);
- if (params == NULL)
+ params = devm_kzalloc(&bebob->card->card_dev,
+ sizeof(struct special_params), GFP_KERNEL);
+ if (!params)
return -ENOMEM;
mutex_lock(&bebob->mutex);
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 774eb2205668..0f6dbcffe711 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -122,25 +122,12 @@ static void dice_card_strings(struct snd_dice *dice)
strcpy(card->mixername, "DICE");
}
-static void dice_free(struct snd_dice *dice)
+static void dice_card_free(struct snd_card *card)
{
+ struct snd_dice *dice = card->private_data;
+
snd_dice_stream_destroy_duplex(dice);
snd_dice_transaction_destroy(dice);
- fw_unit_put(dice->unit);
-
- mutex_destroy(&dice->mutex);
- kfree(dice);
-}
-
-/*
- * This module releases the FireWire unit data after all ALSA character devices
- * are released by applications. This is for releasing stream data or finishing
- * transactions safely. Thus at returning from .remove(), this module still keep
- * references for the unit.
- */
-static void dice_card_free(struct snd_card *card)
-{
- dice_free(card->private_data);
}
static void do_registration(struct work_struct *work)
@@ -155,6 +142,8 @@ static void do_registration(struct work_struct *work)
&dice->card);
if (err < 0)
return;
+ dice->card->private_free = dice_card_free;
+ dice->card->private_data = dice;
err = snd_dice_transaction_init(dice);
if (err < 0)
@@ -192,19 +181,10 @@ static void do_registration(struct work_struct *work)
if (err < 0)
goto error;
- /*
- * After registered, dice instance can be released corresponding to
- * releasing the sound card instance.
- */
- dice->card->private_free = dice_card_free;
- dice->card->private_data = dice;
dice->registered = true;
return;
error:
- snd_dice_stream_destroy_duplex(dice);
- snd_dice_transaction_destroy(dice);
- snd_dice_stream_destroy_duplex(dice);
snd_card_free(dice->card);
dev_info(&dice->unit->device,
"Sound card registration failed: %d\n", err);
@@ -223,10 +203,9 @@ static int dice_probe(struct fw_unit *unit,
}
/* Allocate this independent of sound card instance. */
- dice = kzalloc(sizeof(struct snd_dice), GFP_KERNEL);
- if (dice == NULL)
+ dice = devm_kzalloc(&unit->device, sizeof(struct snd_dice), GFP_KERNEL);
+ if (!dice)
return -ENOMEM;
-
dice->unit = fw_unit_get(unit);
dev_set_drvdata(&unit->device, dice);
@@ -263,10 +242,10 @@ static void dice_remove(struct fw_unit *unit)
if (dice->registered) {
/* No need to wait for releasing card object in this context. */
snd_card_free_when_closed(dice->card);
- } else {
- /* Don't forget this case. */
- dice_free(dice);
}
+
+ mutex_destroy(&dice->mutex);
+ fw_unit_put(dice->unit);
}
static void dice_bus_reset(struct fw_unit *unit)
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
index ef689997d6a5..6c6ea149ef6b 100644
--- a/sound/firewire/digi00x/digi00x.c
+++ b/sound/firewire/digi00x/digi00x.c
@@ -41,20 +41,12 @@ static int name_card(struct snd_dg00x *dg00x)
return 0;
}
-static void dg00x_free(struct snd_dg00x *dg00x)
+static void dg00x_card_free(struct snd_card *card)
{
+ struct snd_dg00x *dg00x = card->private_data;
+
snd_dg00x_stream_destroy_duplex(dg00x);
snd_dg00x_transaction_unregister(dg00x);
-
- fw_unit_put(dg00x->unit);
-
- mutex_destroy(&dg00x->mutex);
- kfree(dg00x);
-}
-
-static void dg00x_card_free(struct snd_card *card)
-{
- dg00x_free(card->private_data);
}
static void do_registration(struct work_struct *work)
@@ -70,6 +62,8 @@ static void do_registration(struct work_struct *work)
&dg00x->card);
if (err < 0)
return;
+ dg00x->card->private_free = dg00x_card_free;
+ dg00x->card->private_data = dg00x;
err = name_card(dg00x);
if (err < 0)
@@ -101,14 +95,10 @@ static void do_registration(struct work_struct *work)
if (err < 0)
goto error;
- dg00x->card->private_free = dg00x_card_free;
- dg00x->card->private_data = dg00x;
dg00x->registered = true;
return;
error:
- snd_dg00x_transaction_unregister(dg00x);
- snd_dg00x_stream_destroy_duplex(dg00x);
snd_card_free(dg00x->card);
dev_info(&dg00x->unit->device,
"Sound card registration failed: %d\n", err);
@@ -120,8 +110,9 @@ static int snd_dg00x_probe(struct fw_unit *unit,
struct snd_dg00x *dg00x;
/* Allocate this independent of sound card instance. */
- dg00x = kzalloc(sizeof(struct snd_dg00x), GFP_KERNEL);
- if (dg00x == NULL)
+ dg00x = devm_kzalloc(&unit->device, sizeof(struct snd_dg00x),
+ GFP_KERNEL);
+ if (!dg00x)
return -ENOMEM;
dg00x->unit = fw_unit_get(unit);
@@ -173,12 +164,12 @@ static void snd_dg00x_remove(struct fw_unit *unit)
cancel_delayed_work_sync(&dg00x->dwork);
if (dg00x->registered) {
- /* No need to wait for releasing card object in this context. */
- snd_card_free_when_closed(dg00x->card);
- } else {
- /* Don't forget this case. */
- dg00x_free(dg00x);
+ // Block till all of ALSA character devices are released.
+ snd_card_free(dg00x->card);
}
+
+ mutex_destroy(&dg00x->mutex);
+ fw_unit_put(dg00x->unit);
}
static const struct ieee1394_device_id snd_dg00x_id_table[] = {
diff --git a/sound/firewire/fireface/ff.c b/sound/firewire/fireface/ff.c
index 4974bc7980e9..3f61cfeace69 100644
--- a/sound/firewire/fireface/ff.c
+++ b/sound/firewire/fireface/ff.c
@@ -27,20 +27,12 @@ static void name_card(struct snd_ff *ff)
dev_name(&ff->unit->device), 100 << fw_dev->max_speed);
}
-static void ff_free(struct snd_ff *ff)
+static void ff_card_free(struct snd_card *card)
{
+ struct snd_ff *ff = card->private_data;
+
snd_ff_stream_destroy_duplex(ff);
snd_ff_transaction_unregister(ff);
-
- fw_unit_put(ff->unit);
-
- mutex_destroy(&ff->mutex);
- kfree(ff);
-}
-
-static void ff_card_free(struct snd_card *card)
-{
- ff_free(card->private_data);
}
static void do_registration(struct work_struct *work)
@@ -55,6 +47,8 @@ static void do_registration(struct work_struct *work)
&ff->card);
if (err < 0)
return;
+ ff->card->private_free = ff_card_free;
+ ff->card->private_data = ff;
err = snd_ff_transaction_register(ff);
if (err < 0)
@@ -84,14 +78,10 @@ static void do_registration(struct work_struct *work)
if (err < 0)
goto error;
- ff->card->private_free = ff_card_free;
- ff->card->private_data = ff;
ff->registered = true;
return;
error:
- snd_ff_transaction_unregister(ff);
- snd_ff_stream_destroy_duplex(ff);
snd_card_free(ff->card);
dev_info(&ff->unit->device,
"Sound card registration failed: %d\n", err);
@@ -102,11 +92,9 @@ static int snd_ff_probe(struct fw_unit *unit,
{
struct snd_ff *ff;
- ff = kzalloc(sizeof(struct snd_ff), GFP_KERNEL);
- if (ff == NULL)
+ ff = devm_kzalloc(&unit->device, sizeof(struct snd_ff), GFP_KERNEL);
+ if (!ff)
return -ENOMEM;
-
- /* initialize myself */
ff->unit = fw_unit_get(unit);
dev_set_drvdata(&unit->device, ff);
@@ -149,12 +137,12 @@ static void snd_ff_remove(struct fw_unit *unit)
cancel_work_sync(&ff->dwork.work);
if (ff->registered) {
- /* No need to wait for releasing card object in this context. */
- snd_card_free_when_closed(ff->card);
- } else {
- /* Don't forget this case. */
- ff_free(ff);
+ // Block till all of ALSA character devices are released.
+ snd_card_free(ff->card);
}
+
+ mutex_destroy(&ff->mutex);
+ fw_unit_put(ff->unit);
}
static const struct snd_ff_spec spec_ff400 = {
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index f2d073365cf6..faf0e001c4c5 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -184,36 +184,17 @@ end:
return err;
}
-static void efw_free(struct snd_efw *efw)
-{
- snd_efw_stream_destroy_duplex(efw);
- snd_efw_transaction_remove_instance(efw);
- fw_unit_put(efw->unit);
-
- kfree(efw->resp_buf);
-
- mutex_destroy(&efw->mutex);
- kfree(efw);
-}
-
-/*
- * This module releases the FireWire unit data after all ALSA character devices
- * are released by applications. This is for releasing stream data or finishing
- * transactions safely. Thus at returning from .remove(), this module still keep
- * references for the unit.
- */
static void
efw_card_free(struct snd_card *card)
{
struct snd_efw *efw = card->private_data;
- if (efw->card_index >= 0) {
- mutex_lock(&devices_mutex);
- clear_bit(efw->card_index, devices_used);
- mutex_unlock(&devices_mutex);
- }
+ mutex_lock(&devices_mutex);
+ clear_bit(efw->card_index, devices_used);
+ mutex_unlock(&devices_mutex);
- efw_free(card->private_data);
+ snd_efw_stream_destroy_duplex(efw);
+ snd_efw_transaction_remove_instance(efw);
}
static void
@@ -226,9 +207,8 @@ do_registration(struct work_struct *work)
if (efw->registered)
return;
- mutex_lock(&devices_mutex);
-
/* check registered cards */
+ mutex_lock(&devices_mutex);
for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) {
if (!test_bit(card_index, devices_used) && enable[card_index])
break;
@@ -244,12 +224,18 @@ do_registration(struct work_struct *work)
mutex_unlock(&devices_mutex);
return;
}
+ set_bit(card_index, devices_used);
+ mutex_unlock(&devices_mutex);
+
+ efw->card->private_free = efw_card_free;
+ efw->card->private_data = efw;
/* prepare response buffer */
snd_efw_resp_buf_size = clamp(snd_efw_resp_buf_size,
SND_EFW_RESPONSE_MAXIMUM_BYTES, 4096U);
- efw->resp_buf = kzalloc(snd_efw_resp_buf_size, GFP_KERNEL);
- if (efw->resp_buf == NULL) {
+ efw->resp_buf = devm_kzalloc(&efw->card->card_dev,
+ snd_efw_resp_buf_size, GFP_KERNEL);
+ if (!efw->resp_buf) {
err = -ENOMEM;
goto error;
}
@@ -284,25 +270,11 @@ do_registration(struct work_struct *work)
if (err < 0)
goto error;
- set_bit(card_index, devices_used);
- mutex_unlock(&devices_mutex);
-
- /*
- * After registered, efw instance can be released corresponding to
- * releasing the sound card instance.
- */
- efw->card->private_free = efw_card_free;
- efw->card->private_data = efw;
efw->registered = true;
return;
error:
- mutex_unlock(&devices_mutex);
- snd_efw_transaction_remove_instance(efw);
- snd_efw_stream_destroy_duplex(efw);
snd_card_free(efw->card);
- kfree(efw->resp_buf);
- efw->resp_buf = NULL;
dev_info(&efw->unit->device,
"Sound card registration failed: %d\n", err);
}
@@ -312,10 +284,9 @@ efw_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry)
{
struct snd_efw *efw;
- efw = kzalloc(sizeof(struct snd_efw), GFP_KERNEL);
+ efw = devm_kzalloc(&unit->device, sizeof(struct snd_efw), GFP_KERNEL);
if (efw == NULL)
return -ENOMEM;
-
efw->unit = fw_unit_get(unit);
dev_set_drvdata(&unit->device, efw);
@@ -363,12 +334,12 @@ static void efw_remove(struct fw_unit *unit)
cancel_delayed_work_sync(&efw->dwork);
if (efw->registered) {
- /* No need to wait for releasing card object in this context. */
- snd_card_free_when_closed(efw->card);
- } else {
- /* Don't forget this case. */
- efw_free(efw);
+ // Block till all of ALSA character devices are released.
+ snd_card_free(efw->card);
}
+
+ mutex_destroy(&efw->mutex);
+ fw_unit_put(efw->unit);
}
static const struct ieee1394_device_id efw_id_table[] = {
diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index 30957477e005..9ebe510ea26b 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -602,8 +602,6 @@ static void isight_card_free(struct snd_card *card)
struct isight *isight = card->private_data;
fw_iso_resources_destroy(&isight->resources);
- fw_unit_put(isight->unit);
- mutex_destroy(&isight->mutex);
}
static u64 get_unit_base(struct fw_unit *unit)
@@ -640,7 +638,7 @@ static int isight_probe(struct fw_unit *unit,
if (!isight->audio_base) {
dev_err(&unit->device, "audio unit base not found\n");
err = -ENXIO;
- goto err_unit;
+ goto error;
}
fw_iso_resources_init(&isight->resources, unit);
@@ -669,12 +667,12 @@ static int isight_probe(struct fw_unit *unit,
dev_set_drvdata(&unit->device, isight);
return 0;
-
-err_unit:
- fw_unit_put(isight->unit);
- mutex_destroy(&isight->mutex);
error:
snd_card_free(card);
+
+ mutex_destroy(&isight->mutex);
+ fw_unit_put(isight->unit);
+
return err;
}
@@ -703,7 +701,11 @@ static void isight_remove(struct fw_unit *unit)
isight_stop_streaming(isight);
mutex_unlock(&isight->mutex);
- snd_card_free_when_closed(isight->card);
+ // Block till all of ALSA character devices are released.
+ snd_card_free(isight->card);
+
+ mutex_destroy(&isight->mutex);
+ fw_unit_put(isight->unit);
}
static const struct ieee1394_device_id isight_id_table[] = {
diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c
index 300d31b6f191..220e61926ea4 100644
--- a/sound/firewire/motu/motu.c
+++ b/sound/firewire/motu/motu.c
@@ -52,26 +52,12 @@ static void name_card(struct snd_motu *motu)
dev_name(&motu->unit->device), 100 << fw_dev->max_speed);
}
-static void motu_free(struct snd_motu *motu)
+static void motu_card_free(struct snd_card *card)
{
- snd_motu_transaction_unregister(motu);
+ struct snd_motu *motu = card->private_data;
+ snd_motu_transaction_unregister(motu);
snd_motu_stream_destroy_duplex(motu);
- fw_unit_put(motu->unit);
-
- mutex_destroy(&motu->mutex);
- kfree(motu);
-}
-
-/*
- * This module releases the FireWire unit data after all ALSA character devices
- * are released by applications. This is for releasing stream data or finishing
- * transactions safely. Thus at returning from .remove(), this module still keep
- * references for the unit.
- */
-static void motu_card_free(struct snd_card *card)
-{
- motu_free(card->private_data);
}
static void do_registration(struct work_struct *work)
@@ -86,6 +72,8 @@ static void do_registration(struct work_struct *work)
&motu->card);
if (err < 0)
return;
+ motu->card->private_free = motu_card_free;
+ motu->card->private_data = motu;
name_card(motu);
@@ -120,18 +108,10 @@ static void do_registration(struct work_struct *work)
if (err < 0)
goto error;
- /*
- * After registered, motu instance can be released corresponding to
- * releasing the sound card instance.
- */
- motu->card->private_free = motu_card_free;
- motu->card->private_data = motu;
motu->registered = true;
return;
error:
- snd_motu_transaction_unregister(motu);
- snd_motu_stream_destroy_duplex(motu);
snd_card_free(motu->card);
dev_info(&motu->unit->device,
"Sound card registration failed: %d\n", err);
@@ -143,14 +123,13 @@ static int motu_probe(struct fw_unit *unit,
struct snd_motu *motu;
/* Allocate this independently of sound card instance. */
- motu = kzalloc(sizeof(struct snd_motu), GFP_KERNEL);
- if (motu == NULL)
+ motu = devm_kzalloc(&unit->device, sizeof(struct snd_motu), GFP_KERNEL);
+ if (!motu)
return -ENOMEM;
-
- motu->spec = (const struct snd_motu_spec *)entry->driver_data;
motu->unit = fw_unit_get(unit);
dev_set_drvdata(&unit->device, motu);
+ motu->spec = (const struct snd_motu_spec *)entry->driver_data;
mutex_init(&motu->mutex);
spin_lock_init(&motu->lock);
init_waitqueue_head(&motu->hwdep_wait);
@@ -174,12 +153,12 @@ static void motu_remove(struct fw_unit *unit)
cancel_delayed_work_sync(&motu->dwork);
if (motu->registered) {
- /* No need to wait for releasing card object in this context. */
- snd_card_free_when_closed(motu->card);
- } else {
- /* Don't forget this case. */
- motu_free(motu);
+ // Block till all of ALSA character devices are released.
+ snd_card_free(motu->card);
}
+
+ mutex_destroy(&motu->mutex);
+ fw_unit_put(motu->unit);
}
static void motu_bus_update(struct fw_unit *unit)
diff --git a/sound/firewire/oxfw/oxfw-scs1x.c b/sound/firewire/oxfw/oxfw-scs1x.c
index f33497cdc706..9d9545880a28 100644
--- a/sound/firewire/oxfw/oxfw-scs1x.c
+++ b/sound/firewire/oxfw/oxfw-scs1x.c
@@ -372,8 +372,9 @@ int snd_oxfw_scs1x_add(struct snd_oxfw *oxfw)
struct fw_scs1x *scs;
int err;
- scs = kzalloc(sizeof(struct fw_scs1x), GFP_KERNEL);
- if (scs == NULL)
+ scs = devm_kzalloc(&oxfw->card->card_dev, sizeof(struct fw_scs1x),
+ GFP_KERNEL);
+ if (!scs)
return -ENOMEM;
scs->fw_dev = fw_parent_device(oxfw->unit);
oxfw->spec = scs;
diff --git a/sound/firewire/oxfw/oxfw-spkr.c b/sound/firewire/oxfw/oxfw-spkr.c
index cb905af0660d..66d4b1f73f0f 100644
--- a/sound/firewire/oxfw/oxfw-spkr.c
+++ b/sound/firewire/oxfw/oxfw-spkr.c
@@ -270,8 +270,9 @@ int snd_oxfw_add_spkr(struct snd_oxfw *oxfw, bool is_lacie)
unsigned int i, first_ch;
int err;
- spkr = kzalloc(sizeof(struct fw_spkr), GFP_KERNEL);
- if (spkr == NULL)
+ spkr = devm_kzalloc(&oxfw->card->card_dev, sizeof(struct fw_spkr),
+ GFP_KERNEL);
+ if (!spkr)
return -ENOMEM;
oxfw->spec = spkr;
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index d9361f352133..f230a9e44c3c 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -517,8 +517,9 @@ assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir,
if (err < 0)
goto end;
- formats[eid] = kmemdup(buf, *len, GFP_KERNEL);
- if (formats[eid] == NULL) {
+ formats[eid] = devm_kmemdup(&oxfw->card->card_dev, buf, *len,
+ GFP_KERNEL);
+ if (!formats[eid]) {
err = -ENOMEM;
goto end;
}
@@ -535,7 +536,8 @@ assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir,
continue;
eid++;
- formats[eid] = kmemdup(buf, *len, GFP_KERNEL);
+ formats[eid] = devm_kmemdup(&oxfw->card->card_dev, buf, *len,
+ GFP_KERNEL);
if (formats[eid] == NULL) {
err = -ENOMEM;
goto end;
@@ -597,8 +599,9 @@ static int fill_stream_formats(struct snd_oxfw *oxfw,
if (err < 0)
break;
- formats[eid] = kmemdup(buf, len, GFP_KERNEL);
- if (formats[eid] == NULL) {
+ formats[eid] = devm_kmemdup(&oxfw->card->card_dev, buf, len,
+ GFP_KERNEL);
+ if (!formats[eid]) {
err = -ENOMEM;
break;
}
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 2ea8be6c8584..afb78d90384b 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -113,35 +113,13 @@ end:
return err;
}
-static void oxfw_free(struct snd_oxfw *oxfw)
+static void oxfw_card_free(struct snd_card *card)
{
- unsigned int i;
+ struct snd_oxfw *oxfw = card->private_data;
snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
if (oxfw->has_output)
snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream);
-
- fw_unit_put(oxfw->unit);
-
- for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
- kfree(oxfw->tx_stream_formats[i]);
- kfree(oxfw->rx_stream_formats[i]);
- }
-
- kfree(oxfw->spec);
- mutex_destroy(&oxfw->mutex);
- kfree(oxfw);
-}
-
-/*
- * This module releases the FireWire unit data after all ALSA character devices
- * are released by applications. This is for releasing stream data or finishing
- * transactions safely. Thus at returning from .remove(), this module still keep
- * references for the unit.
- */
-static void oxfw_card_free(struct snd_card *card)
-{
- oxfw_free(card->private_data);
}
static int detect_quirks(struct snd_oxfw *oxfw)
@@ -208,7 +186,6 @@ static int detect_quirks(struct snd_oxfw *oxfw)
static void do_registration(struct work_struct *work)
{
struct snd_oxfw *oxfw = container_of(work, struct snd_oxfw, dwork.work);
- int i;
int err;
if (oxfw->registered)
@@ -218,6 +195,8 @@ static void do_registration(struct work_struct *work)
&oxfw->card);
if (err < 0)
return;
+ oxfw->card->private_free = oxfw_card_free;
+ oxfw->card->private_data = oxfw;
err = name_card(oxfw);
if (err < 0)
@@ -258,28 +237,11 @@ static void do_registration(struct work_struct *work)
if (err < 0)
goto error;
- /*
- * After registered, oxfw instance can be released corresponding to
- * releasing the sound card instance.
- */
- oxfw->card->private_free = oxfw_card_free;
- oxfw->card->private_data = oxfw;
oxfw->registered = true;
return;
error:
- snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
- if (oxfw->has_output)
- snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream);
- for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; ++i) {
- kfree(oxfw->tx_stream_formats[i]);
- oxfw->tx_stream_formats[i] = NULL;
- kfree(oxfw->rx_stream_formats[i]);
- oxfw->rx_stream_formats[i] = NULL;
- }
snd_card_free(oxfw->card);
- kfree(oxfw->spec);
- oxfw->spec = NULL;
dev_info(&oxfw->unit->device,
"Sound card registration failed: %d\n", err);
}
@@ -293,14 +255,13 @@ static int oxfw_probe(struct fw_unit *unit,
return -ENODEV;
/* Allocate this independent of sound card instance. */
- oxfw = kzalloc(sizeof(struct snd_oxfw), GFP_KERNEL);
- if (oxfw == NULL)
+ oxfw = devm_kzalloc(&unit->device, sizeof(struct snd_oxfw), GFP_KERNEL);
+ if (!oxfw)
return -ENOMEM;
-
- oxfw->entry = entry;
oxfw->unit = fw_unit_get(unit);
dev_set_drvdata(&unit->device, oxfw);
+ oxfw->entry = entry;
mutex_init(&oxfw->mutex);
spin_lock_init(&oxfw->lock);
init_waitqueue_head(&oxfw->hwdep_wait);
@@ -347,12 +308,12 @@ static void oxfw_remove(struct fw_unit *unit)
cancel_delayed_work_sync(&oxfw->dwork);
if (oxfw->registered) {
- /* No need to wait for releasing card object in this context. */
- snd_card_free_when_closed(oxfw->card);
- } else {
- /* Don't forget this case. */
- oxfw_free(oxfw);
+ // Block till all of ALSA character devices are released.
+ snd_card_free(oxfw->card);
}
+
+ mutex_destroy(&oxfw->mutex);
+ fw_unit_put(oxfw->unit);
}
static const struct compat_info griffin_firewave = {
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index d3fdc463a884..ef57fa4db323 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -85,20 +85,12 @@ static int identify_model(struct snd_tscm *tscm)
return 0;
}
-static void tscm_free(struct snd_tscm *tscm)
+static void tscm_card_free(struct snd_card *card)
{
+ struct snd_tscm *tscm = card->private_data;
+
snd_tscm_transaction_unregister(tscm);
snd_tscm_stream_destroy_duplex(tscm);
-
- fw_unit_put(tscm->unit);
-
- mutex_destroy(&tscm->mutex);
- kfree(tscm);
-}
-
-static void tscm_card_free(struct snd_card *card)
-{
- tscm_free(card->private_data);
}
static void do_registration(struct work_struct *work)
@@ -110,6 +102,8 @@ static void do_registration(struct work_struct *work)
&tscm->card);
if (err < 0)
return;
+ tscm->card->private_free = tscm_card_free;
+ tscm->card->private_data = tscm;
err = identify_model(tscm);
if (err < 0)
@@ -141,18 +135,10 @@ static void do_registration(struct work_struct *work)
if (err < 0)
goto error;
- /*
- * After registered, tscm instance can be released corresponding to
- * releasing the sound card instance.
- */
- tscm->card->private_free = tscm_card_free;
- tscm->card->private_data = tscm;
tscm->registered = true;
return;
error:
- snd_tscm_transaction_unregister(tscm);
- snd_tscm_stream_destroy_duplex(tscm);
snd_card_free(tscm->card);
dev_info(&tscm->unit->device,
"Sound card registration failed: %d\n", err);
@@ -164,11 +150,9 @@ static int snd_tscm_probe(struct fw_unit *unit,
struct snd_tscm *tscm;
/* Allocate this independent of sound card instance. */
- tscm = kzalloc(sizeof(struct snd_tscm), GFP_KERNEL);
- if (tscm == NULL)
+ tscm = devm_kzalloc(&unit->device, sizeof(struct snd_tscm), GFP_KERNEL);
+ if (!tscm)
return -ENOMEM;
-
- /* initialize myself */
tscm->unit = fw_unit_get(unit);
dev_set_drvdata(&unit->device, tscm);
@@ -216,12 +200,12 @@ static void snd_tscm_remove(struct fw_unit *unit)
cancel_delayed_work_sync(&tscm->dwork);
if (tscm->registered) {
- /* No need to wait for releasing card object in this context. */
- snd_card_free_when_closed(tscm->card);
- } else {
- /* Don't forget this case. */
- tscm_free(tscm);
+ // Block till all of ALSA character devices are released.
+ snd_card_free(tscm->card);
}
+
+ mutex_destroy(&tscm->mutex);
+ fw_unit_put(tscm->unit);
}
static const struct ieee1394_device_id snd_tscm_id_table[] = {
diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c
index 5bc4a1d587d4..60cb00fd0c69 100644
--- a/sound/hda/ext/hdac_ext_controller.c
+++ b/sound/hda/ext/hdac_ext_controller.c
@@ -48,9 +48,11 @@ void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *bus, bool enable)
}
if (enable)
- snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_GPROCEN);
+ snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL,
+ AZX_PPCTL_GPROCEN, AZX_PPCTL_GPROCEN);
else
- snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_GPROCEN, 0);
+ snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL,
+ AZX_PPCTL_GPROCEN, 0);
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_enable);
@@ -68,9 +70,11 @@ void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *bus, bool enable)
}
if (enable)
- snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_PIE);
+ snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL,
+ AZX_PPCTL_PIE, AZX_PPCTL_PIE);
else
- snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_PIE, 0);
+ snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL,
+ AZX_PPCTL_PIE, 0);
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_int_enable);
@@ -194,7 +198,8 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable)
*/
int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link)
{
- snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA);
+ snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL,
+ AZX_MLCTL_SPA, AZX_MLCTL_SPA);
return check_hdac_link_power_active(link, true);
}
@@ -222,8 +227,8 @@ int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus)
int ret;
list_for_each_entry(hlink, &bus->hlink_list, list) {
- snd_hdac_updatel(hlink->ml_addr,
- AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA);
+ snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL,
+ AZX_MLCTL_SPA, AZX_MLCTL_SPA);
ret = check_hdac_link_power_active(hlink, true);
if (ret < 0)
return ret;
@@ -243,7 +248,8 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus)
int ret;
list_for_each_entry(hlink, &bus->hlink_list, list) {
- snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0);
+ snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL,
+ AZX_MLCTL_SPA, 0);
ret = check_hdac_link_power_active(hlink, false);
if (ret < 0)
return ret;
diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c
index 2647309bc675..8afa2f888466 100644
--- a/sound/i2c/cs8427.c
+++ b/sound/i2c/cs8427.c
@@ -118,7 +118,7 @@ static int snd_cs8427_send_corudata(struct snd_i2c_device *device,
struct cs8427 *chip = device->private_data;
char *hw_data = udata ?
chip->playback.hw_udata : chip->playback.hw_status;
- char data[32];
+ unsigned char data[32];
int err, idx;
if (!memcmp(hw_data, ndata, count))
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index ac0ab6eb40f0..47e0b2820ace 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -389,7 +389,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip,
case OPTi9XX_HW_82C931:
/* disable 3D sound (set GPIO1 as output, low) */
snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c);
- case OPTi9XX_HW_82C933: /* FALL THROUGH */
+ /* fall through */
+ case OPTi9XX_HW_82C933:
/*
* The BTC 1817DW has QS1000 wavetable which is connected
* to the serial digital input of the OPTI931.
@@ -400,7 +401,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip,
* or digital input signal.
*/
snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01);
- case OPTi9XX_HW_82C930: /* FALL THROUGH */
+ /* fall through */
+ case OPTi9XX_HW_82C930:
snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03);
snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff);
snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 |
diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c
index 481797744b3c..8288fae90085 100644
--- a/sound/isa/sb/sb8_main.c
+++ b/sound/isa/sb/sb8_main.c
@@ -130,13 +130,13 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream)
chip->playback_format = SB_DSP_HI_OUTPUT_AUTO;
break;
}
- /* fallthru */
+ /* fall through */
case SB_HW_201:
if (rate > 23000) {
chip->playback_format = SB_DSP_HI_OUTPUT_AUTO;
break;
}
- /* fallthru */
+ /* fall through */
case SB_HW_20:
chip->playback_format = SB_DSP_LO_OUTPUT_AUTO;
break;
@@ -287,7 +287,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream)
chip->capture_format = SB_DSP_HI_INPUT_AUTO;
break;
}
- /* fallthru */
+ /* fall through */
case SB_HW_20:
chip->capture_format = SB_DSP_LO_INPUT_AUTO;
break;
@@ -387,7 +387,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip)
case SB_MODE_PLAYBACK_16: /* ok.. playback is active */
if (chip->hardware != SB_HW_JAZZ16)
break;
- /* fallthru */
+ /* fall through */
case SB_MODE_PLAYBACK_8:
substream = chip->playback_substream;
if (chip->playback_format == SB_DSP_OUTPUT)
@@ -397,7 +397,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip)
case SB_MODE_CAPTURE_16:
if (chip->hardware != SB_HW_JAZZ16)
break;
- /* fallthru */
+ /* fall through */
case SB_MODE_CAPTURE_8:
substream = chip->capture_substream;
if (chip->capture_format == SB_DSP_INPUT)
diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c
index c8904e732aaa..a4ed54aeaf1d 100644
--- a/sound/mips/hal2.c
+++ b/sound/mips/hal2.c
@@ -500,7 +500,8 @@ static const struct snd_pcm_hardware hal2_pcm_hw = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_BLOCK_TRANSFER),
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_SYNC_APPLPTR),
.formats = SNDRV_PCM_FMTBIT_S16_BE,
.rates = SNDRV_PCM_RATE_8000_48000,
.rate_min = 8000,
@@ -563,6 +564,8 @@ static int hal2_playback_prepare(struct snd_pcm_substream *substream)
dac->sample_rate = hal2_compute_rate(dac, runtime->rate);
memset(&dac->pcm_indirect, 0, sizeof(dac->pcm_indirect));
dac->pcm_indirect.hw_buffer_size = H2_BUF_SIZE;
+ dac->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
+ dac->pcm_indirect.hw_io = dac->buffer_dma;
dac->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
dac->substream = substream;
hal2_setup_dac(hal2);
@@ -575,9 +578,6 @@ static int hal2_playback_trigger(struct snd_pcm_substream *substream, int cmd)
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- hal2->dac.pcm_indirect.hw_io = hal2->dac.buffer_dma;
- hal2->dac.pcm_indirect.hw_data = 0;
- substream->ops->ack(substream);
hal2_start_dac(hal2);
break;
case SNDRV_PCM_TRIGGER_STOP:
@@ -615,7 +615,6 @@ static int hal2_playback_ack(struct snd_pcm_substream *substream)
struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
struct hal2_codec *dac = &hal2->dac;
- dac->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
return snd_pcm_indirect_playback_transfer(substream,
&dac->pcm_indirect,
hal2_playback_transfer);
@@ -655,6 +654,7 @@ static int hal2_capture_prepare(struct snd_pcm_substream *substream)
memset(&adc->pcm_indirect, 0, sizeof(adc->pcm_indirect));
adc->pcm_indirect.hw_buffer_size = H2_BUF_SIZE;
adc->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
+ adc->pcm_indirect.hw_io = adc->buffer_dma;
adc->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
adc->substream = substream;
hal2_setup_adc(hal2);
@@ -667,9 +667,6 @@ static int hal2_capture_trigger(struct snd_pcm_substream *substream, int cmd)
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- hal2->adc.pcm_indirect.hw_io = hal2->adc.buffer_dma;
- hal2->adc.pcm_indirect.hw_data = 0;
- printk(KERN_DEBUG "buffer_dma %x\n", hal2->adc.buffer_dma);
hal2_start_adc(hal2);
break;
case SNDRV_PCM_TRIGGER_STOP:
diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c
index 5ef4fe964366..7c91330af719 100644
--- a/sound/pci/asihpi/hpios.c
+++ b/sound/pci/asihpi/hpios.c
@@ -49,7 +49,7 @@ u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
/*?? any benefit in using managed dmam_alloc_coherent? */
p_mem_area->vaddr =
dma_alloc_coherent(&pdev->dev, size, &p_mem_area->dma_handle,
- GFP_DMA32 | GFP_KERNEL);
+ GFP_KERNEL);
if (p_mem_area->vaddr) {
HPI_DEBUG_LOG(DEBUG, "allocated %d bytes, dma 0x%x vma %p\n",
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index a1e4944dcfe8..1a41f8c80243 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -903,15 +903,15 @@ static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream)
case 8:
data |= ATI_REG_OUT_DMA_SLOT_BIT(10) |
ATI_REG_OUT_DMA_SLOT_BIT(11);
- /* fallthru */
+ /* fall through */
case 6:
data |= ATI_REG_OUT_DMA_SLOT_BIT(7) |
ATI_REG_OUT_DMA_SLOT_BIT(8);
- /* fallthru */
+ /* fall through */
case 4:
data |= ATI_REG_OUT_DMA_SLOT_BIT(6) |
ATI_REG_OUT_DMA_SLOT_BIT(9);
- /* fallthru */
+ /* fall through */
default:
data |= ATI_REG_OUT_DMA_SLOT_BIT(3) |
ATI_REG_OUT_DMA_SLOT_BIT(4);
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 2e5b460a847c..96ece1a71cf1 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1115,6 +1115,7 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma,
hwwrite(vortex->mmio,
VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0xc,
snd_pcm_sgbuf_get_addr(dma->substream, psize * 3));
+ /* fall through */
/* 3 pages */
case 3:
dma->cfg0 |= 0x12000000;
@@ -1122,12 +1123,14 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma,
hwwrite(vortex->mmio,
VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0x8,
snd_pcm_sgbuf_get_addr(dma->substream, psize * 2));
+ /* fall through */
/* 2 pages */
case 2:
dma->cfg0 |= 0x88000000 | 0x44000000 | 0x10000000 | (psize - 1);
hwwrite(vortex->mmio,
VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0x4,
snd_pcm_sgbuf_get_addr(dma->substream, psize));
+ /* fall through */
/* 1 page */
case 1:
dma->cfg0 |= 0x80000000 | 0x40000000 | ((psize - 1) << 0xc);
@@ -1390,17 +1393,20 @@ vortex_wtdma_setbuffers(vortex_t * vortex, int wtdma,
dma->cfg1 |= 0x88000000 | 0x44000000 | 0x30000000 | (psize-1);
hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0xc,
snd_pcm_sgbuf_get_addr(dma->substream, psize * 3));
+ /* fall through */
/* 3 pages */
case 3:
dma->cfg0 |= 0x12000000;
dma->cfg1 |= 0x80000000 | 0x40000000 | ((psize-1) << 0xc);
hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0x8,
snd_pcm_sgbuf_get_addr(dma->substream, psize * 2));
+ /* fall through */
/* 2 pages */
case 2:
dma->cfg0 |= 0x88000000 | 0x44000000 | 0x10000000 | (psize-1);
hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0x4,
snd_pcm_sgbuf_get_addr(dma->substream, psize));
+ /* fall through */
/* 1 page */
case 1:
dma->cfg0 |= 0x80000000 | 0x40000000 | ((psize-1) << 0xc);
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 146e1a3498c7..750eec437a79 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -1443,7 +1443,8 @@ static const struct snd_pcm_hardware snd_cs46xx_playback =
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER /*|*/
- /*SNDRV_PCM_INFO_RESUME*/),
+ /*SNDRV_PCM_INFO_RESUME*/ |
+ SNDRV_PCM_INFO_SYNC_APPLPTR),
.formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |
SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE),
@@ -1465,7 +1466,8 @@ static const struct snd_pcm_hardware snd_cs46xx_capture =
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER /*|*/
- /*SNDRV_PCM_INFO_RESUME*/),
+ /*SNDRV_PCM_INFO_RESUME*/ |
+ SNDRV_PCM_INFO_SYNC_APPLPTR),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
.rate_min = 5500,
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index 9f2b6097f486..30b3472d0b75 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -1753,7 +1753,8 @@ static const struct snd_pcm_hardware snd_emu10k1_fx8010_playback =
{
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_RESUME |
- /* SNDRV_PCM_INFO_MMAP_VALID | */ SNDRV_PCM_INFO_PAUSE),
+ /* SNDRV_PCM_INFO_MMAP_VALID | */ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_APPLPTR),
.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_48000,
.rate_min = 48000,
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index b9a6b66aeb0e..df0d636145f8 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -13,7 +13,7 @@
#include <linux/export.h>
#include <linux/sort.h>
#include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_auto_parser.h"
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index d1a6a9c1329a..f1457c6b3969 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -9,7 +9,7 @@
#ifndef __SOUND_HDA_BEEP_H
#define __SOUND_HDA_BEEP_H
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#define HDA_BEEP_MODE_OFF 0
#define HDA_BEEP_MODE_ON 1
diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index d361bb77ca00..9174f1b3a987 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -11,7 +11,7 @@
#include <linux/pm.h>
#include <linux/pm_runtime.h>
#include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
/*
@@ -81,6 +81,12 @@ static int hda_codec_driver_probe(struct device *dev)
hda_codec_patch_t patch;
int err;
+ if (codec->bus->core.ext_ops) {
+ if (WARN_ON(!codec->bus->core.ext_ops->hdev_attach))
+ return -EINVAL;
+ return codec->bus->core.ext_ops->hdev_attach(&codec->core);
+ }
+
if (WARN_ON(!codec->preset))
return -EINVAL;
@@ -134,6 +140,12 @@ static int hda_codec_driver_remove(struct device *dev)
{
struct hda_codec *codec = dev_to_hda_codec(dev);
+ if (codec->bus->core.ext_ops) {
+ if (WARN_ON(!codec->bus->core.ext_ops->hdev_detach))
+ return -EINVAL;
+ return codec->bus->core.ext_ops->hdev_detach(&codec->core);
+ }
+
if (codec->patch_ops.free)
codec->patch_ops.free(codec);
snd_hda_codec_cleanup_for_unbind(codec);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 26d348b47867..0957813939e5 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -27,7 +27,7 @@
#include <linux/pm.h>
#include <linux/pm_runtime.h>
#include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include <sound/asoundef.h>
#include <sound/tlv.h>
#include <sound/initval.h>
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index a12e594d4e3b..fe2506672a72 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -130,8 +130,9 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
azx_dev->core.bufsize = 0;
azx_dev->core.period_bytes = 0;
azx_dev->core.format_val = 0;
- ret = chip->ops->substream_alloc_pages(chip, substream,
- params_buffer_bytes(hw_params));
+ ret = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+
unlock:
dsp_unlock(azx_dev);
return ret;
@@ -141,7 +142,6 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
{
struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
struct azx_dev *azx_dev = get_azx_dev(substream);
- struct azx *chip = apcm->chip;
struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream);
int err;
@@ -152,7 +152,7 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
snd_hda_codec_cleanup(apcm->codec, hinfo, substream);
- err = chip->ops->substream_free_pages(chip, substream);
+ err = snd_pcm_lib_free_pages(substream);
azx_stream(azx_dev)->prepared = 0;
dsp_unlock(azx_dev);
return err;
@@ -732,6 +732,7 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec,
int pcm_dev = cpcm->device;
unsigned int size;
int s, err;
+ int type = SNDRV_DMA_TYPE_DEV_SG;
list_for_each_entry(apcm, &chip->pcm_list, list) {
if (apcm->pcm->device == pcm_dev) {
@@ -770,7 +771,9 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec,
size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024;
if (size > MAX_PREALLOC_SIZE)
size = MAX_PREALLOC_SIZE;
- snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+ if (chip->uc_buffer)
+ type = SNDRV_DMA_TYPE_DEV_UC_SG;
+ snd_pcm_lib_preallocate_pages_for_all(pcm, type,
chip->card->dev,
size, MAX_PREALLOC_SIZE);
return 0;
@@ -1220,27 +1223,6 @@ void snd_hda_bus_reset(struct hda_bus *bus)
bus->in_reset = 0;
}
-static int get_jackpoll_interval(struct azx *chip)
-{
- int i;
- unsigned int j;
-
- if (!chip->jackpoll_ms)
- return 0;
-
- i = chip->jackpoll_ms[chip->dev_index];
- if (i == 0)
- return 0;
- if (i < 50 || i > 60000)
- j = 0;
- else
- j = msecs_to_jiffies(i);
- if (j == 0)
- dev_warn(chip->card->dev,
- "jackpoll_ms value out of range: %d\n", i);
- return j;
-}
-
/* HD-audio bus initialization */
int azx_bus_init(struct azx *chip, const char *model,
const struct hdac_io_ops *io_ops)
@@ -1323,7 +1305,7 @@ int azx_probe_codecs(struct azx *chip, unsigned int max_slots)
err = snd_hda_codec_new(&chip->bus, chip->card, c, &codec);
if (err < 0)
continue;
- codec->jackpoll_interval = get_jackpoll_interval(chip);
+ codec->jackpoll_interval = chip->jackpoll_interval;
codec->beep_mode = chip->beep_mode;
codecs++;
}
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index a68e75b00ea3..c95097bb5a0c 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -20,7 +20,7 @@
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/initval.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include <sound/hda_register.h>
#define AZX_MAX_CODECS HDA_MAX_CODECS
@@ -76,7 +76,6 @@ struct azx_dev {
* when link position is not greater than FIFO size
*/
unsigned int insufficient:1;
- unsigned int wc_marked:1;
};
#define azx_stream(dev) (&(dev)->core)
@@ -88,11 +87,6 @@ struct azx;
struct hda_controller_ops {
/* Disable msi if supported, PCI only */
int (*disable_msi_reset_irq)(struct azx *);
- int (*substream_alloc_pages)(struct azx *chip,
- struct snd_pcm_substream *substream,
- size_t size);
- int (*substream_free_pages)(struct azx *chip,
- struct snd_pcm_substream *substream);
void (*pcm_mmap_prepare)(struct snd_pcm_substream *substream,
struct vm_area_struct *area);
/* Check if current position is acceptable */
@@ -127,7 +121,7 @@ struct azx {
int capture_streams;
int capture_index_offset;
int num_streams;
- const int *jackpoll_ms; /* per-card jack poll interval */
+ int jackpoll_interval; /* jack poll interval in jiffies */
/* Register interaction. */
const struct hda_controller_ops *ops;
@@ -160,6 +154,7 @@ struct azx {
unsigned int msi:1;
unsigned int probing:1; /* codec probing phase */
unsigned int snoop:1;
+ unsigned int uc_buffer:1; /* non-cached pages for stream buffers */
unsigned int align_buffer_size:1;
unsigned int region_requested:1;
unsigned int disabled:1; /* disabled by vga_switcheroo */
@@ -175,11 +170,10 @@ struct azx {
#define azx_bus(chip) (&(chip)->bus.core)
#define bus_to_azx(_bus) container_of(_bus, struct azx, bus.core)
-#ifdef CONFIG_X86
-#define azx_snoop(chip) ((chip)->snoop)
-#else
-#define azx_snoop(chip) true
-#endif
+static inline bool azx_snoop(struct azx *chip)
+{
+ return !IS_ENABLED(CONFIG_X86) || chip->snoop;
+}
/*
* macros for easy use
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index ba7fe9b6655c..806b12ed44a2 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -27,7 +27,7 @@
#include <sound/core.h>
#include <asm/unaligned.h>
#include <sound/hda_chmap.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
enum eld_versions {
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 579984ecdec3..276150f29cda 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -32,7 +32,7 @@
#include <sound/core.h>
#include <sound/jack.h>
#include <sound/tlv.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_auto_parser.h"
#include "hda_jack.h"
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index cc009a4a3d1d..268bba6ec985 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -23,7 +23,7 @@
#include <linux/compat.h>
#include <linux/nospec.h>
#include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include <sound/hda_hwdep.h>
#include <sound/minors.h>
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index aa4c672dbaf7..d8eb2b5f51ae 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -63,7 +63,7 @@
#include <linux/vgaarb.h>
#include <linux/vga_switcheroo.h>
#include <linux/firmware.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_controller.h"
#include "hda_intel.h"
@@ -399,61 +399,6 @@ static char *driver_short_names[] = {
[AZX_DRIVER_GENERIC] = "HD-Audio Generic",
};
-#ifdef CONFIG_X86
-static void __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, bool on)
-{
- int pages;
-
- if (azx_snoop(chip))
- return;
- if (!dmab || !dmab->area || !dmab->bytes)
- return;
-
-#ifdef CONFIG_SND_DMA_SGBUF
- if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_SG) {
- struct snd_sg_buf *sgbuf = dmab->private_data;
- if (chip->driver_type == AZX_DRIVER_CMEDIA)
- return; /* deal with only CORB/RIRB buffers */
- if (on)
- set_pages_array_wc(sgbuf->page_table, sgbuf->pages);
- else
- set_pages_array_wb(sgbuf->page_table, sgbuf->pages);
- return;
- }
-#endif
-
- pages = (dmab->bytes + PAGE_SIZE - 1) >> PAGE_SHIFT;
- if (on)
- set_memory_wc((unsigned long)dmab->area, pages);
- else
- set_memory_wb((unsigned long)dmab->area, pages);
-}
-
-static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf,
- bool on)
-{
- __mark_pages_wc(chip, buf, on);
-}
-static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev,
- struct snd_pcm_substream *substream, bool on)
-{
- if (azx_dev->wc_marked != on) {
- __mark_pages_wc(chip, snd_pcm_get_dma_buf(substream), on);
- azx_dev->wc_marked = on;
- }
-}
-#else
-/* NOP for other archs */
-static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf,
- bool on)
-{
-}
-static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev,
- struct snd_pcm_substream *substream, bool on)
-{
-}
-#endif
-
static int azx_acquire_irq(struct azx *chip, int do_disconnect);
static void set_default_power_save(struct azx *chip);
@@ -1678,6 +1623,7 @@ static void azx_check_snoop_available(struct azx *chip)
dev_info(chip->card->dev, "Force to %s mode by module option\n",
snoop ? "snoop" : "non-snoop");
chip->snoop = snoop;
+ chip->uc_buffer = !snoop;
return;
}
@@ -1698,8 +1644,12 @@ static void azx_check_snoop_available(struct azx *chip)
snoop = false;
chip->snoop = snoop;
- if (!snoop)
+ if (!snoop) {
dev_info(chip->card->dev, "Force to non-snoop mode\n");
+ /* C-Media requires non-cached pages only for CORB/RIRB */
+ if (chip->driver_type != AZX_DRIVER_CMEDIA)
+ chip->uc_buffer = true;
+ }
}
static void azx_probe_work(struct work_struct *work)
@@ -1767,7 +1717,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci,
chip->driver_type = driver_caps & 0xff;
check_msi(chip);
chip->dev_index = dev;
- chip->jackpoll_ms = jackpoll_ms;
+ if (jackpoll_ms[dev] >= 50 && jackpoll_ms[dev] <= 60000)
+ chip->jackpoll_interval = msecs_to_jiffies(jackpoll_ms[dev]);
INIT_LIST_HEAD(&chip->pcm_list);
INIT_WORK(&hda->irq_pending_work, azx_irq_pending_work);
INIT_LIST_HEAD(&hda->list);
@@ -2090,55 +2041,24 @@ static int dma_alloc_pages(struct hdac_bus *bus,
struct snd_dma_buffer *buf)
{
struct azx *chip = bus_to_azx(bus);
- int err;
- err = snd_dma_alloc_pages(type,
- bus->dev,
- size, buf);
- if (err < 0)
- return err;
- mark_pages_wc(chip, buf, true);
- return 0;
+ if (!azx_snoop(chip) && type == SNDRV_DMA_TYPE_DEV)
+ type = SNDRV_DMA_TYPE_DEV_UC;
+ return snd_dma_alloc_pages(type, bus->dev, size, buf);
}
static void dma_free_pages(struct hdac_bus *bus, struct snd_dma_buffer *buf)
{
- struct azx *chip = bus_to_azx(bus);
-
- mark_pages_wc(chip, buf, false);
snd_dma_free_pages(buf);
}
-static int substream_alloc_pages(struct azx *chip,
- struct snd_pcm_substream *substream,
- size_t size)
-{
- struct azx_dev *azx_dev = get_azx_dev(substream);
- int ret;
-
- mark_runtime_wc(chip, azx_dev, substream, false);
- ret = snd_pcm_lib_malloc_pages(substream, size);
- if (ret < 0)
- return ret;
- mark_runtime_wc(chip, azx_dev, substream, true);
- return 0;
-}
-
-static int substream_free_pages(struct azx *chip,
- struct snd_pcm_substream *substream)
-{
- struct azx_dev *azx_dev = get_azx_dev(substream);
- mark_runtime_wc(chip, azx_dev, substream, false);
- return snd_pcm_lib_free_pages(substream);
-}
-
static void pcm_mmap_prepare(struct snd_pcm_substream *substream,
struct vm_area_struct *area)
{
#ifdef CONFIG_X86
struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
struct azx *chip = apcm->chip;
- if (!azx_snoop(chip) && chip->driver_type != AZX_DRIVER_CMEDIA)
+ if (chip->uc_buffer)
area->vm_page_prot = pgprot_writecombine(area->vm_page_prot);
#endif
}
@@ -2156,8 +2076,6 @@ static const struct hdac_io_ops pci_hda_io_ops = {
static const struct hda_controller_ops pci_hda_ops = {
.disable_msi_reset_irq = disable_msi_reset_irq,
- .substream_alloc_pages = substream_alloc_pages,
- .substream_free_pages = substream_free_pages,
.pcm_mmap_prepare = pcm_mmap_prepare,
.position_check = azx_position_check,
.link_power = azx_intel_link_power,
@@ -2257,8 +2175,12 @@ static struct snd_pci_quirk power_save_blacklist[] = {
/* https://bugzilla.redhat.com/show_bug.cgi?id=1581607 */
SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", 0),
/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
+ SND_PCI_QUIRK(0x1028, 0x0497, "Dell Precision T3600", 0),
+ /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
/* Note the P55A-UD3 and Z87-D3HP share the subsys id for the HDA dev */
SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P55A-UD3 / Z87-D3HP", 0),
+ /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
+ SND_PCI_QUIRK(0x8086, 0x2040, "Intel DZ77BH-55K", 0),
/* https://bugzilla.kernel.org/show_bug.cgi?id=199607 */
SND_PCI_QUIRK(0x8086, 0x2057, "Intel NUC5i7RYB", 0),
/* https://bugzilla.redhat.com/show_bug.cgi?id=1520902 */
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index a33234e04d4f..c499727920e6 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -15,7 +15,7 @@
#include <sound/core.h>
#include <sound/control.h>
#include <sound/jack.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_auto_parser.h"
#include "hda_jack.h"
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index c6b778b2580c..a65740419650 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -25,7 +25,7 @@
#include <linux/slab.h>
#include <sound/core.h>
#include <linux/module.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
static int dump_coef = -1;
diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c
index 6ec79c58d48d..c154b19a0c45 100644
--- a/sound/pci/hda/hda_sysfs.c
+++ b/sound/pci/hda/hda_sysfs.c
@@ -14,7 +14,7 @@
#include <linux/string.h>
#include <linux/export.h>
#include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include <sound/hda_hwdep.h>
#include <sound/minors.h>
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 0621920f7617..dd7d4242d6d2 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -35,7 +35,7 @@
#include <sound/core.h>
#include <sound/initval.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_controller.h"
/* Defines for Nvidia Tegra HDA support */
@@ -99,19 +99,6 @@ static void dma_free_pages(struct hdac_bus *bus, struct snd_dma_buffer *buf)
snd_dma_free_pages(buf);
}
-static int substream_alloc_pages(struct azx *chip,
- struct snd_pcm_substream *substream,
- size_t size)
-{
- return snd_pcm_lib_malloc_pages(substream, size);
-}
-
-static int substream_free_pages(struct azx *chip,
- struct snd_pcm_substream *substream)
-{
- return snd_pcm_lib_free_pages(substream);
-}
-
/*
* Register access ops. Tegra HDA register access is DWORD only.
*/
@@ -180,10 +167,7 @@ static const struct hdac_io_ops hda_tegra_io_ops = {
.dma_free_pages = dma_free_pages,
};
-static const struct hda_controller_ops hda_tegra_ops = {
- .substream_alloc_pages = substream_alloc_pages,
- .substream_free_pages = substream_free_pages,
-};
+static const struct hda_controller_ops hda_tegra_ops; /* nothing special */
static void hda_tegra_init(struct hda_tegra *hda)
{
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index fd476fb40e1b..ebfd0be885b3 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -24,7 +24,7 @@
#include <linux/module.h>
#include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_auto_parser.h"
#include "hda_beep.h"
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index c2d9ee9cfdc0..21d0f0610913 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -22,7 +22,7 @@
#include <linux/slab.h>
#include <linux/module.h>
#include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_auto_parser.h"
#include "hda_jack.h"
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 0166a3d7cd55..0a24037184c3 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -31,8 +31,9 @@
#include <linux/types.h>
#include <linux/io.h>
#include <linux/pci.h>
+#include <asm/io.h>
#include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_auto_parser.h"
#include "hda_jack.h"
@@ -81,12 +82,12 @@
#define SCP_GET 1
#define EFX_FILE "ctefx.bin"
-#define SBZ_EFX_FILE "ctefx-sbz.bin"
+#define DESKTOP_EFX_FILE "ctefx-desktop.bin"
#define R3DI_EFX_FILE "ctefx-r3di.bin"
#ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP
MODULE_FIRMWARE(EFX_FILE);
-MODULE_FIRMWARE(SBZ_EFX_FILE);
+MODULE_FIRMWARE(DESKTOP_EFX_FILE);
MODULE_FIRMWARE(R3DI_EFX_FILE);
#endif
@@ -152,7 +153,10 @@ enum {
XBASS_XOVER,
EQ_PRESET_ENUM,
SMART_VOLUME_ENUM,
- MIC_BOOST_ENUM
+ MIC_BOOST_ENUM,
+ AE5_HEADPHONE_GAIN_ENUM,
+ AE5_SOUND_FILTER_ENUM,
+ ZXR_HEADPHONE_GAIN
#define EFFECTS_COUNT (EFFECT_END_NID - EFFECT_START_NID)
};
@@ -666,6 +670,65 @@ static const struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = {
}
};
+/* Values for ca0113_mmio_command_set for selecting output. */
+#define AE5_CA0113_OUT_SET_COMMANDS 6
+struct ae5_ca0113_output_set {
+ unsigned int group[AE5_CA0113_OUT_SET_COMMANDS];
+ unsigned int target[AE5_CA0113_OUT_SET_COMMANDS];
+ unsigned int vals[AE5_CA0113_OUT_SET_COMMANDS];
+};
+
+static const struct ae5_ca0113_output_set ae5_ca0113_output_presets[] = {
+ { .group = { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+ .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+ .vals = { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f }
+ },
+ { .group = { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+ .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+ .vals = { 0x3f, 0x3f, 0x00, 0x00, 0x00, 0x00 }
+ },
+ { .group = { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+ .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+ .vals = { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f }
+ }
+};
+
+/* ae5 ca0113 command sequences to set headphone gain levels. */
+#define AE5_HEADPHONE_GAIN_PRESET_MAX_COMMANDS 4
+struct ae5_headphone_gain_set {
+ char *name;
+ unsigned int vals[AE5_HEADPHONE_GAIN_PRESET_MAX_COMMANDS];
+};
+
+static const struct ae5_headphone_gain_set ae5_headphone_gain_presets[] = {
+ { .name = "Low (16-31",
+ .vals = { 0xff, 0x2c, 0xf5, 0x32 }
+ },
+ { .name = "Medium (32-149",
+ .vals = { 0x38, 0xa8, 0x3e, 0x4c }
+ },
+ { .name = "High (150-600",
+ .vals = { 0xff, 0xff, 0xff, 0x7f }
+ }
+};
+
+struct ae5_filter_set {
+ char *name;
+ unsigned int val;
+};
+
+static const struct ae5_filter_set ae5_filter_presets[] = {
+ { .name = "Slow Roll Off",
+ .val = 0xa0
+ },
+ { .name = "Minimum Phase",
+ .val = 0xc0
+ },
+ { .name = "Fast Roll Off",
+ .val = 0x80
+ }
+};
+
enum hda_cmd_vendor_io {
/* for DspIO node */
VENDOR_DSPIO_SCP_WRITE_DATA_LOW = 0x000,
@@ -685,6 +748,9 @@ enum hda_cmd_vendor_io {
VENDOR_CHIPIO_DATA_LOW = 0x300,
VENDOR_CHIPIO_DATA_HIGH = 0x400,
+ VENDOR_CHIPIO_8051_WRITE_DIRECT = 0x500,
+ VENDOR_CHIPIO_8051_READ_DIRECT = 0xD00,
+
VENDOR_CHIPIO_GET_PARAMETER = 0xF00,
VENDOR_CHIPIO_STATUS = 0xF01,
VENDOR_CHIPIO_HIC_POST_READ = 0x702,
@@ -692,6 +758,9 @@ enum hda_cmd_vendor_io {
VENDOR_CHIPIO_8051_DATA_WRITE = 0x707,
VENDOR_CHIPIO_8051_DATA_READ = 0xF07,
+ VENDOR_CHIPIO_8051_PMEM_READ = 0xF08,
+ VENDOR_CHIPIO_8051_IRAM_WRITE = 0x709,
+ VENDOR_CHIPIO_8051_IRAM_READ = 0xF09,
VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE = 0x70A,
VENDOR_CHIPIO_CT_EXTENSIONS_GET = 0xF0A,
@@ -798,6 +867,12 @@ enum control_param_id {
* impedance is selected*/
CONTROL_PARAM_PORTD_160OHM_GAIN = 10,
+ /*
+ * This control param name was found in the 8051 memory, and makes
+ * sense given the fact the AE-5 uses it and has the ASI flag set.
+ */
+ CONTROL_PARAM_ASI = 23,
+
/* Stream Control */
/* Select stream with the given ID */
@@ -955,7 +1030,11 @@ struct ca0132_spec {
long eq_preset_val;
unsigned int tlv[4];
struct hda_vmaster_mute_hook vmaster_mute;
-
+ /* AE-5 Control values */
+ unsigned char ae5_headphone_gain_val;
+ unsigned char ae5_filter_val;
+ /* ZxR Control Values */
+ unsigned char zxr_gain_set;
struct hda_codec *codec;
struct delayed_work unsol_hp_work;
@@ -995,8 +1074,11 @@ enum {
QUIRK_ALIENWARE,
QUIRK_ALIENWARE_M17XR4,
QUIRK_SBZ,
+ QUIRK_ZXR,
+ QUIRK_ZXR_DBPRO,
QUIRK_R3DI,
QUIRK_R3D,
+ QUIRK_AE5,
};
static const struct hda_pintbl alienware_pincfgs[] = {
@@ -1028,6 +1110,21 @@ static const struct hda_pintbl sbz_pincfgs[] = {
{}
};
+/* Sound Blaster ZxR pin configs taken from Windows Driver */
+static const struct hda_pintbl zxr_pincfgs[] = {
+ { 0x0b, 0x01047110 }, /* Port G -- Lineout FRONT L/R */
+ { 0x0c, 0x414510f0 }, /* SPDIF Out 1 - Disabled*/
+ { 0x0d, 0x014510f0 }, /* Digital Out */
+ { 0x0e, 0x41c520f0 }, /* SPDIF In - Disabled*/
+ { 0x0f, 0x0122711f }, /* Port A -- BackPanel HP */
+ { 0x10, 0x01017111 }, /* Port D -- Center/LFE */
+ { 0x11, 0x01017114 }, /* Port B -- LineMicIn2 / Rear L/R */
+ { 0x12, 0x01a271f0 }, /* Port C -- LineIn1 */
+ { 0x13, 0x908700f0 }, /* What U Hear In*/
+ { 0x18, 0x50d000f0 }, /* N/A */
+ {}
+};
+
/* Recon3D pin configs taken from Windows Driver */
static const struct hda_pintbl r3d_pincfgs[] = {
{ 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */
@@ -1043,6 +1140,21 @@ static const struct hda_pintbl r3d_pincfgs[] = {
{}
};
+/* Sound Blaster AE-5 pin configs taken from Windows Driver */
+static const struct hda_pintbl ae5_pincfgs[] = {
+ { 0x0b, 0x01017010 }, /* Port G -- Lineout FRONT L/R */
+ { 0x0c, 0x014510f0 }, /* SPDIF Out 1 */
+ { 0x0d, 0x014510f0 }, /* Digital Out */
+ { 0x0e, 0x01c510f0 }, /* SPDIF In */
+ { 0x0f, 0x01017114 }, /* Port A -- Rear L/R. */
+ { 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */
+ { 0x11, 0x01a170ff }, /* Port B -- LineMicIn2 / Rear Headphone */
+ { 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */
+ { 0x13, 0x908700f0 }, /* What U Hear In*/
+ { 0x18, 0x50d000f0 }, /* N/A */
+ {}
+};
+
/* Recon3D integrated pin configs taken from Windows Driver */
static const struct hda_pintbl r3di_pincfgs[] = {
{ 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */
@@ -1069,6 +1181,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI),
SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI),
SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D),
+ SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5),
{}
};
@@ -1454,6 +1567,20 @@ static void chipio_set_conn_rate(struct hda_codec *codec,
}
/*
+ * Writes to the 8051's internal address space directly instead of indirectly,
+ * giving access to the special function registers located at addresses
+ * 0x80-0xFF.
+ */
+static void chipio_8051_write_direct(struct hda_codec *codec,
+ unsigned int addr, unsigned int data)
+{
+ unsigned int verb;
+
+ verb = VENDOR_CHIPIO_8051_WRITE_DIRECT | data;
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, verb, addr);
+}
+
+/*
* Enable clocks.
*/
static void chipio_enable_clocks(struct hda_codec *codec)
@@ -3088,7 +3215,9 @@ static bool dspload_wait_loaded(struct hda_codec *codec)
}
/*
- * Setup GPIO for the other variants of Core3D.
+ * ca0113 related functions. The ca0113 acts as the HDA bus for the pci-e
+ * based cards, and has a second mmio region, region2, that's used for special
+ * commands.
*/
/*
@@ -3096,8 +3225,11 @@ static bool dspload_wait_loaded(struct hda_codec *codec)
* the mmio address 0x320 is used to set GPIO pins. The format for the data
* The first eight bits are just the number of the pin. So far, I've only seen
* this number go to 7.
+ * AE-5 note: The AE-5 seems to use pins 2 and 3 to somehow set the color value
+ * of the on-card LED. It seems to use pin 2 for data, then toggles 3 to on and
+ * then off to send that bit.
*/
-static void ca0132_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin,
+static void ca0113_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin,
bool enable)
{
struct ca0132_spec *spec = codec->spec;
@@ -3110,6 +3242,89 @@ static void ca0132_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin,
}
/*
+ * Special pci region2 commands that are only used by the AE-5. They follow
+ * a set format, and require reads at certain points to seemingly 'clear'
+ * the response data. My first tests didn't do these reads, and would cause
+ * the card to get locked up until the memory was read. These commands
+ * seem to work with three distinct values that I've taken to calling group,
+ * target-id, and value.
+ */
+static void ca0113_mmio_command_set(struct hda_codec *codec, unsigned int group,
+ unsigned int target, unsigned int value)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int write_val;
+
+ writel(0x0000007e, spec->mem_base + 0x210);
+ readl(spec->mem_base + 0x210);
+ writel(0x0000005a, spec->mem_base + 0x210);
+ readl(spec->mem_base + 0x210);
+ readl(spec->mem_base + 0x210);
+
+ writel(0x00800005, spec->mem_base + 0x20c);
+ writel(group, spec->mem_base + 0x804);
+
+ writel(0x00800005, spec->mem_base + 0x20c);
+ write_val = (target & 0xff);
+ write_val |= (value << 8);
+
+
+ writel(write_val, spec->mem_base + 0x204);
+ /*
+ * Need delay here or else it goes too fast and works inconsistently.
+ */
+ msleep(20);
+
+ readl(spec->mem_base + 0x860);
+ readl(spec->mem_base + 0x854);
+ readl(spec->mem_base + 0x840);
+
+ writel(0x00800004, spec->mem_base + 0x20c);
+ writel(0x00000000, spec->mem_base + 0x210);
+ readl(spec->mem_base + 0x210);
+ readl(spec->mem_base + 0x210);
+}
+
+/*
+ * This second type of command is used for setting the sound filter type.
+ */
+static void ca0113_mmio_command_set_type2(struct hda_codec *codec,
+ unsigned int group, unsigned int target, unsigned int value)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int write_val;
+
+ writel(0x0000007e, spec->mem_base + 0x210);
+ readl(spec->mem_base + 0x210);
+ writel(0x0000005a, spec->mem_base + 0x210);
+ readl(spec->mem_base + 0x210);
+ readl(spec->mem_base + 0x210);
+
+ writel(0x00800003, spec->mem_base + 0x20c);
+ writel(group, spec->mem_base + 0x804);
+
+ writel(0x00800005, spec->mem_base + 0x20c);
+ write_val = (target & 0xff);
+ write_val |= (value << 8);
+
+
+ writel(write_val, spec->mem_base + 0x204);
+ msleep(20);
+ readl(spec->mem_base + 0x860);
+ readl(spec->mem_base + 0x854);
+ readl(spec->mem_base + 0x840);
+
+ writel(0x00800004, spec->mem_base + 0x20c);
+ writel(0x00000000, spec->mem_base + 0x210);
+ readl(spec->mem_base + 0x210);
+ readl(spec->mem_base + 0x210);
+}
+
+/*
+ * Setup GPIO for the other variants of Core3D.
+ */
+
+/*
* Sets up the GPIO pins so that they are discoverable. If this isn't done,
* the card shows as having no GPIO pins.
*/
@@ -3119,6 +3334,7 @@ static void ca0132_gpio_init(struct hda_codec *codec)
switch (spec->quirk) {
case QUIRK_SBZ:
+ case QUIRK_AE5:
snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
snd_hda_codec_write(codec, 0x01, 0, 0x790, 0x23);
@@ -3928,6 +4144,138 @@ exit:
return err < 0 ? err : 0;
}
+static int ae5_headphone_gain_set(struct hda_codec *codec, long val);
+static int zxr_headphone_gain_set(struct hda_codec *codec, long val);
+static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val);
+
+static void ae5_mmio_select_out(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int i;
+
+ for (i = 0; i < AE5_CA0113_OUT_SET_COMMANDS; i++)
+ ca0113_mmio_command_set(codec,
+ ae5_ca0113_output_presets[spec->cur_out_type].group[i],
+ ae5_ca0113_output_presets[spec->cur_out_type].target[i],
+ ae5_ca0113_output_presets[spec->cur_out_type].vals[i]);
+}
+
+/*
+ * These are the commands needed to setup output on each of the different card
+ * types.
+ */
+static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int tmp;
+
+ switch (spec->cur_out_type) {
+ case SPEAKER_OUT:
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ ca0113_mmio_gpio_set(codec, 7, false);
+ ca0113_mmio_gpio_set(codec, 4, true);
+ ca0113_mmio_gpio_set(codec, 1, true);
+ chipio_set_control_param(codec, 0x0d, 0x18);
+ break;
+ case QUIRK_ZXR:
+ ca0113_mmio_gpio_set(codec, 2, true);
+ ca0113_mmio_gpio_set(codec, 3, true);
+ ca0113_mmio_gpio_set(codec, 5, false);
+ zxr_headphone_gain_set(codec, 0);
+ chipio_set_control_param(codec, 0x0d, 0x24);
+ break;
+ case QUIRK_R3DI:
+ chipio_set_control_param(codec, 0x0d, 0x24);
+ r3di_gpio_out_set(codec, R3DI_LINE_OUT);
+ break;
+ case QUIRK_R3D:
+ chipio_set_control_param(codec, 0x0d, 0x24);
+ ca0113_mmio_gpio_set(codec, 1, true);
+ break;
+ case QUIRK_AE5:
+ ae5_mmio_select_out(codec);
+ ae5_headphone_gain_set(codec, 2);
+ tmp = FLOAT_ZERO;
+ dspio_set_uint_param(codec, 0x96, 0x29, tmp);
+ dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
+ chipio_set_control_param(codec, 0x0d, 0xa4);
+ chipio_write(codec, 0x18b03c, 0x00000012);
+ break;
+ }
+ break;
+ case HEADPHONE_OUT:
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ ca0113_mmio_gpio_set(codec, 7, true);
+ ca0113_mmio_gpio_set(codec, 4, true);
+ ca0113_mmio_gpio_set(codec, 1, false);
+ chipio_set_control_param(codec, 0x0d, 0x12);
+ break;
+ case QUIRK_ZXR:
+ ca0113_mmio_gpio_set(codec, 2, false);
+ ca0113_mmio_gpio_set(codec, 3, false);
+ ca0113_mmio_gpio_set(codec, 5, true);
+ zxr_headphone_gain_set(codec, spec->zxr_gain_set);
+ chipio_set_control_param(codec, 0x0d, 0x21);
+ break;
+ case QUIRK_R3DI:
+ chipio_set_control_param(codec, 0x0d, 0x21);
+ r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT);
+ break;
+ case QUIRK_R3D:
+ chipio_set_control_param(codec, 0x0d, 0x21);
+ ca0113_mmio_gpio_set(codec, 0x1, false);
+ break;
+ case QUIRK_AE5:
+ ae5_mmio_select_out(codec);
+ ae5_headphone_gain_set(codec,
+ spec->ae5_headphone_gain_val);
+ tmp = FLOAT_ONE;
+ dspio_set_uint_param(codec, 0x96, 0x29, tmp);
+ dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
+ chipio_set_control_param(codec, 0x0d, 0xa1);
+ chipio_write(codec, 0x18b03c, 0x00000012);
+ break;
+ }
+ break;
+ case SURROUND_OUT:
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ ca0113_mmio_gpio_set(codec, 7, false);
+ ca0113_mmio_gpio_set(codec, 4, true);
+ ca0113_mmio_gpio_set(codec, 1, true);
+ chipio_set_control_param(codec, 0x0d, 0x18);
+ break;
+ case QUIRK_ZXR:
+ ca0113_mmio_gpio_set(codec, 2, true);
+ ca0113_mmio_gpio_set(codec, 3, true);
+ ca0113_mmio_gpio_set(codec, 5, false);
+ zxr_headphone_gain_set(codec, 0);
+ chipio_set_control_param(codec, 0x0d, 0x24);
+ break;
+ case QUIRK_R3DI:
+ chipio_set_control_param(codec, 0x0d, 0x24);
+ r3di_gpio_out_set(codec, R3DI_LINE_OUT);
+ break;
+ case QUIRK_R3D:
+ ca0113_mmio_gpio_set(codec, 1, true);
+ chipio_set_control_param(codec, 0x0d, 0x24);
+ break;
+ case QUIRK_AE5:
+ ae5_mmio_select_out(codec);
+ ae5_headphone_gain_set(codec, 2);
+ tmp = FLOAT_ZERO;
+ dspio_set_uint_param(codec, 0x96, 0x29, tmp);
+ dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
+ chipio_set_control_param(codec, 0x0d, 0xa4);
+ chipio_write(codec, 0x18b03c, 0x00000012);
+ break;
+ }
+ break;
+ }
+}
+
/*
* This function behaves similarly to the ca0132_select_out funciton above,
* except with a few differences. It adds the ability to select the current
@@ -3978,26 +4326,11 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
if (err < 0)
goto exit;
+ ca0132_alt_select_out_quirk_handler(codec);
+
switch (spec->cur_out_type) {
case SPEAKER_OUT:
codec_dbg(codec, "%s speaker\n", __func__);
- /*speaker out config*/
- switch (spec->quirk) {
- case QUIRK_SBZ:
- ca0132_mmio_gpio_set(codec, 7, false);
- ca0132_mmio_gpio_set(codec, 4, true);
- ca0132_mmio_gpio_set(codec, 1, true);
- chipio_set_control_param(codec, 0x0D, 0x18);
- break;
- case QUIRK_R3DI:
- chipio_set_control_param(codec, 0x0D, 0x24);
- r3di_gpio_out_set(codec, R3DI_LINE_OUT);
- break;
- case QUIRK_R3D:
- chipio_set_control_param(codec, 0x0D, 0x24);
- ca0132_mmio_gpio_set(codec, 1, true);
- break;
- }
/* disable headphone node */
pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0,
@@ -4021,23 +4354,6 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
break;
case HEADPHONE_OUT:
codec_dbg(codec, "%s hp\n", __func__);
- /* Headphone out config*/
- switch (spec->quirk) {
- case QUIRK_SBZ:
- ca0132_mmio_gpio_set(codec, 7, true);
- ca0132_mmio_gpio_set(codec, 4, true);
- ca0132_mmio_gpio_set(codec, 1, false);
- chipio_set_control_param(codec, 0x0D, 0x12);
- break;
- case QUIRK_R3DI:
- chipio_set_control_param(codec, 0x0D, 0x21);
- r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT);
- break;
- case QUIRK_R3D:
- chipio_set_control_param(codec, 0x0D, 0x21);
- ca0132_mmio_gpio_set(codec, 0x1, false);
- break;
- }
snd_hda_codec_write(codec, spec->out_pins[0], 0,
AC_VERB_SET_EAPD_BTLENABLE, 0x00);
@@ -4067,23 +4383,7 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
break;
case SURROUND_OUT:
codec_dbg(codec, "%s surround\n", __func__);
- /* Surround out config*/
- switch (spec->quirk) {
- case QUIRK_SBZ:
- ca0132_mmio_gpio_set(codec, 7, false);
- ca0132_mmio_gpio_set(codec, 4, true);
- ca0132_mmio_gpio_set(codec, 1, true);
- chipio_set_control_param(codec, 0x0D, 0x18);
- break;
- case QUIRK_R3DI:
- chipio_set_control_param(codec, 0x0D, 0x24);
- r3di_gpio_out_set(codec, R3DI_LINE_OUT);
- break;
- case QUIRK_R3D:
- ca0132_mmio_gpio_set(codec, 1, true);
- chipio_set_control_param(codec, 0x0D, 0x24);
- break;
- }
+
/* enable line out node */
pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
@@ -4108,14 +4408,21 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
snd_hda_set_pin_ctl(codec, spec->out_pins[3],
pin_ctl | PIN_OUT);
- if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
- dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
- else
- dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
+ dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
break;
}
+ /*
+ * Surround always sets it's scp command to req 0x04 to FLOAT_EIGHT.
+ * With this set though, X_BASS cannot be enabled. So, if we have OutFX
+ * enabled, we need to make sure X_BASS is off, otherwise everything
+ * sounds all muffled. Running ca0132_effects_set with X_BASS as the
+ * effect should sort this out.
+ */
+ if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
+ ca0132_effects_set(codec, X_BASS,
+ spec->effects_switch[X_BASS - EFFECT_START_NID]);
- /* run through the output dsp commands for line-out */
+ /* run through the output dsp commands for the selected output. */
for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; i++) {
err = dspio_set_uint_param(codec,
alt_out_presets[spec->cur_out_type].mids[i],
@@ -4152,7 +4459,6 @@ static void ca0132_unsol_hp_delayed(struct work_struct *work)
static void ca0132_set_dmic(struct hda_codec *codec, int enable);
static int ca0132_mic_boost_set(struct hda_codec *codec, long val);
-static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val);
static void resume_mic1(struct hda_codec *codec, unsigned int oldval);
static int stop_mic1(struct hda_codec *codec);
static int ca0132_cvoice_switch_set(struct hda_codec *codec);
@@ -4341,13 +4647,20 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
switch (spec->quirk) {
case QUIRK_SBZ:
case QUIRK_R3D:
- ca0132_mmio_gpio_set(codec, 0, false);
+ ca0113_mmio_gpio_set(codec, 0, false);
+ tmp = FLOAT_THREE;
+ break;
+ case QUIRK_ZXR:
tmp = FLOAT_THREE;
break;
case QUIRK_R3DI:
r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
tmp = FLOAT_ONE;
break;
+ case QUIRK_AE5:
+ ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00);
+ tmp = FLOAT_THREE;
+ break;
default:
tmp = FLOAT_ONE;
break;
@@ -4362,10 +4675,19 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
chipio_set_stream_control(codec, 0x03, 1);
chipio_set_stream_control(codec, 0x04, 1);
-
- if (spec->quirk == QUIRK_SBZ) {
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
chipio_write(codec, 0x18B098, 0x0000000C);
chipio_write(codec, 0x18B09C, 0x0000000C);
+ break;
+ case QUIRK_ZXR:
+ chipio_write(codec, 0x18B098, 0x0000000C);
+ chipio_write(codec, 0x18B09C, 0x000000CC);
+ break;
+ case QUIRK_AE5:
+ chipio_write(codec, 0x18B098, 0x0000000C);
+ chipio_write(codec, 0x18B09C, 0x0000004C);
+ break;
}
ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
break;
@@ -4374,11 +4696,14 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
switch (spec->quirk) {
case QUIRK_SBZ:
case QUIRK_R3D:
- ca0132_mmio_gpio_set(codec, 0, false);
+ ca0113_mmio_gpio_set(codec, 0, false);
break;
case QUIRK_R3DI:
r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
break;
+ case QUIRK_AE5:
+ ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00);
+ break;
}
chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
@@ -4389,11 +4714,13 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
tmp = FLOAT_ZERO;
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
- if (spec->quirk == QUIRK_SBZ) {
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ case QUIRK_AE5:
chipio_write(codec, 0x18B098, 0x00000000);
chipio_write(codec, 0x18B09C, 0x00000000);
+ break;
}
-
chipio_set_stream_control(codec, 0x03, 1);
chipio_set_stream_control(codec, 0x04, 1);
break;
@@ -4401,14 +4728,18 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
switch (spec->quirk) {
case QUIRK_SBZ:
case QUIRK_R3D:
- ca0132_mmio_gpio_set(codec, 0, true);
- ca0132_mmio_gpio_set(codec, 5, false);
+ ca0113_mmio_gpio_set(codec, 0, true);
+ ca0113_mmio_gpio_set(codec, 5, false);
tmp = FLOAT_THREE;
break;
case QUIRK_R3DI:
r3di_gpio_mic_set(codec, R3DI_FRONT_MIC);
tmp = FLOAT_ONE;
break;
+ case QUIRK_AE5:
+ ca0113_mmio_command_set(codec, 0x48, 0x28, 0x3f);
+ tmp = FLOAT_THREE;
+ break;
default:
tmp = FLOAT_ONE;
break;
@@ -4424,9 +4755,15 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
chipio_set_stream_control(codec, 0x03, 1);
chipio_set_stream_control(codec, 0x04, 1);
- if (spec->quirk == QUIRK_SBZ) {
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
chipio_write(codec, 0x18B098, 0x0000000C);
chipio_write(codec, 0x18B09C, 0x000000CC);
+ break;
+ case QUIRK_AE5:
+ chipio_write(codec, 0x18B098, 0x0000000C);
+ chipio_write(codec, 0x18B09C, 0x0000004C);
+ break;
}
ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
break;
@@ -4435,7 +4772,6 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
snd_hda_power_down_pm(codec);
return 0;
-
}
/*
@@ -4507,6 +4843,8 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
/* if PE if off, turn off out effects. */
if (!spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
val = 0;
+ if (spec->cur_out_type == SURROUND_OUT && nid == X_BASS)
+ val = 0;
}
/* for in effect, qualify with CrystalVoice */
@@ -4520,7 +4858,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
val = 0;
/* If Voice Focus on SBZ, set to two channel. */
- if ((nid == VOICE_FOCUS) && (spec->quirk == QUIRK_SBZ)
+ if ((nid == VOICE_FOCUS) && (spec->use_pci_mmio)
&& (spec->cur_mic_type != REAR_LINE_IN)) {
if (spec->effects_switch[CRYSTAL_VOICE -
EFFECT_START_NID]) {
@@ -4539,7 +4877,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
* For SBZ noise reduction, there's an extra command
* to module ID 0x47. No clue why.
*/
- if ((nid == NOISE_REDUCTION) && (spec->quirk == QUIRK_SBZ)
+ if ((nid == NOISE_REDUCTION) && (spec->use_pci_mmio)
&& (spec->cur_mic_type != REAR_LINE_IN)) {
if (spec->effects_switch[CRYSTAL_VOICE -
EFFECT_START_NID]) {
@@ -4678,6 +5016,27 @@ static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val)
return ret;
}
+static int ae5_headphone_gain_set(struct hda_codec *codec, long val)
+{
+ unsigned int i;
+
+ for (i = 0; i < 4; i++)
+ ca0113_mmio_command_set(codec, 0x48, 0x11 + i,
+ ae5_headphone_gain_presets[val].vals[i]);
+ return 0;
+}
+
+/*
+ * gpio pin 1 is a relay that switches on/off, apparently setting the headphone
+ * amplifier to handle a 600 ohm load.
+ */
+static int zxr_headphone_gain_set(struct hda_codec *codec, long val)
+{
+ ca0113_mmio_gpio_set(codec, 1, val);
+
+ return 0;
+}
+
static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -4942,6 +5301,112 @@ static int ca0132_alt_mic_boost_put(struct snd_kcontrol *kcontrol,
return 1;
}
+/*
+ * Sound BlasterX AE-5 Headphone Gain Controls.
+ */
+#define AE5_HEADPHONE_GAIN_MAX 3
+static int ae5_headphone_gain_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ char *sfx = " Ohms)";
+ char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = AE5_HEADPHONE_GAIN_MAX;
+ if (uinfo->value.enumerated.item >= AE5_HEADPHONE_GAIN_MAX)
+ uinfo->value.enumerated.item = AE5_HEADPHONE_GAIN_MAX - 1;
+ sprintf(namestr, "%s %s",
+ ae5_headphone_gain_presets[uinfo->value.enumerated.item].name,
+ sfx);
+ strcpy(uinfo->value.enumerated.name, namestr);
+ return 0;
+}
+
+static int ae5_headphone_gain_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ca0132_spec *spec = codec->spec;
+
+ ucontrol->value.enumerated.item[0] = spec->ae5_headphone_gain_val;
+ return 0;
+}
+
+static int ae5_headphone_gain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ca0132_spec *spec = codec->spec;
+ int sel = ucontrol->value.enumerated.item[0];
+ unsigned int items = AE5_HEADPHONE_GAIN_MAX;
+
+ if (sel >= items)
+ return 0;
+
+ codec_dbg(codec, "ae5_headphone_gain: boost=%d\n",
+ sel);
+
+ spec->ae5_headphone_gain_val = sel;
+
+ if (spec->out_enum_val == HEADPHONE_OUT)
+ ae5_headphone_gain_set(codec, spec->ae5_headphone_gain_val);
+
+ return 1;
+}
+
+/*
+ * Sound BlasterX AE-5 sound filter enumerated control.
+ */
+#define AE5_SOUND_FILTER_MAX 3
+
+static int ae5_sound_filter_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = AE5_SOUND_FILTER_MAX;
+ if (uinfo->value.enumerated.item >= AE5_SOUND_FILTER_MAX)
+ uinfo->value.enumerated.item = AE5_SOUND_FILTER_MAX - 1;
+ sprintf(namestr, "%s",
+ ae5_filter_presets[uinfo->value.enumerated.item].name);
+ strcpy(uinfo->value.enumerated.name, namestr);
+ return 0;
+}
+
+static int ae5_sound_filter_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ca0132_spec *spec = codec->spec;
+
+ ucontrol->value.enumerated.item[0] = spec->ae5_filter_val;
+ return 0;
+}
+
+static int ae5_sound_filter_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ca0132_spec *spec = codec->spec;
+ int sel = ucontrol->value.enumerated.item[0];
+ unsigned int items = AE5_SOUND_FILTER_MAX;
+
+ if (sel >= items)
+ return 0;
+
+ codec_dbg(codec, "ae5_sound_filter: %s\n",
+ ae5_filter_presets[sel].name);
+
+ spec->ae5_filter_val = sel;
+
+ ca0113_mmio_command_set_type2(codec, 0x48, 0x07,
+ ae5_filter_presets[sel].val);
+
+ return 1;
+}
/*
* Input Select Control for alternative ca0132 codecs. This exists because
@@ -5330,6 +5795,16 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
goto exit;
}
+ if (nid == ZXR_HEADPHONE_GAIN) {
+ spec->zxr_gain_set = *valp;
+ if (spec->cur_out_type == HEADPHONE_OUT)
+ changed = zxr_headphone_gain_set(codec, *valp);
+ else
+ changed = 0;
+
+ goto exit;
+ }
+
exit:
snd_hda_power_down(codec);
return changed;
@@ -5705,6 +6180,50 @@ static int ca0132_alt_add_mic_boost_enum(struct hda_codec *codec)
}
/*
+ * Add headphone gain enumerated control for the AE-5. This switches between
+ * three modes, low, medium, and high. When non-headphone outputs are selected,
+ * it is automatically set to high. This is the same behavior as Windows.
+ */
+static int ae5_add_headphone_gain_enum(struct hda_codec *codec)
+{
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_MUTE_MONO("AE-5: Headphone Gain",
+ AE5_HEADPHONE_GAIN_ENUM, 1, 0, HDA_OUTPUT);
+ knew.info = ae5_headphone_gain_info;
+ knew.get = ae5_headphone_gain_get;
+ knew.put = ae5_headphone_gain_put;
+ return snd_hda_ctl_add(codec, AE5_HEADPHONE_GAIN_ENUM,
+ snd_ctl_new1(&knew, codec));
+}
+
+/*
+ * Add sound filter enumerated control for the AE-5. This adds three different
+ * settings: Slow Roll Off, Minimum Phase, and Fast Roll Off. From what I've
+ * read into it, it changes the DAC's interpolation filter.
+ */
+static int ae5_add_sound_filter_enum(struct hda_codec *codec)
+{
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_MUTE_MONO("AE-5: Sound Filter",
+ AE5_SOUND_FILTER_ENUM, 1, 0, HDA_OUTPUT);
+ knew.info = ae5_sound_filter_info;
+ knew.get = ae5_sound_filter_get;
+ knew.put = ae5_sound_filter_put;
+ return snd_hda_ctl_add(codec, AE5_SOUND_FILTER_ENUM,
+ snd_ctl_new1(&knew, codec));
+}
+
+static int zxr_add_headphone_gain_switch(struct hda_codec *codec)
+{
+ struct snd_kcontrol_new knew =
+ CA0132_CODEC_MUTE_MONO("ZxR: 600 Ohm Gain",
+ ZXR_HEADPHONE_GAIN, 1, HDA_OUTPUT);
+
+ return snd_hda_ctl_add(codec, ZXR_HEADPHONE_GAIN,
+ snd_ctl_new1(&knew, codec));
+}
+
+/*
* Need to create slave controls for the alternate codecs that have surround
* capabilities.
*/
@@ -5847,7 +6366,8 @@ static int ca0132_build_controls(struct hda_codec *codec)
NULL, ca0132_alt_slave_pfxs,
"Playback Switch",
true, &spec->vmaster_mute.sw_kctl);
-
+ if (err < 0)
+ return err;
}
/* Add in and out effects controls.
@@ -5855,8 +6375,8 @@ static int ca0132_build_controls(struct hda_codec *codec)
*/
num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT;
for (i = 0; i < num_fx; i++) {
- /* SBZ and R3D break if Echo Cancellation is used. */
- if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3D) {
+ /* Desktop cards break if Echo Cancellation is used. */
+ if (spec->use_pci_mmio) {
if (i == (ECHO_CANCELLATION - IN_EFFECT_START_NID +
OUT_EFFECTS_COUNT))
continue;
@@ -5874,8 +6394,14 @@ static int ca0132_build_controls(struct hda_codec *codec)
* prefix, and change PlayEnhancement and CrystalVoice to match.
*/
if (spec->use_alt_controls) {
- ca0132_alt_add_svm_enum(codec);
- add_ca0132_alt_eq_presets(codec);
+ err = ca0132_alt_add_svm_enum(codec);
+ if (err < 0)
+ return err;
+
+ err = add_ca0132_alt_eq_presets(codec);
+ if (err < 0)
+ return err;
+
err = add_fx_switch(codec, PLAY_ENHANCEMENT,
"Enable OutFX", 0);
if (err < 0)
@@ -5912,7 +6438,9 @@ static int ca0132_build_controls(struct hda_codec *codec)
if (err < 0)
return err;
}
- add_voicefx(codec);
+ err = add_voicefx(codec);
+ if (err < 0)
+ return err;
/*
* If the codec uses alt_functions, you need the enumerated controls
@@ -5920,9 +6448,36 @@ static int ca0132_build_controls(struct hda_codec *codec)
* setting control.
*/
if (spec->use_alt_functions) {
- ca0132_alt_add_output_enum(codec);
- ca0132_alt_add_input_enum(codec);
- ca0132_alt_add_mic_boost_enum(codec);
+ err = ca0132_alt_add_output_enum(codec);
+ if (err < 0)
+ return err;
+ err = ca0132_alt_add_mic_boost_enum(codec);
+ if (err < 0)
+ return err;
+ /*
+ * ZxR only has microphone input, there is no front panel
+ * header on the card, and aux-in is handled by the DBPro board.
+ */
+ if (spec->quirk != QUIRK_ZXR) {
+ err = ca0132_alt_add_input_enum(codec);
+ if (err < 0)
+ return err;
+ }
+ }
+
+ if (spec->quirk == QUIRK_AE5) {
+ err = ae5_add_headphone_gain_enum(codec);
+ if (err < 0)
+ return err;
+ err = ae5_add_sound_filter_enum(codec);
+ if (err < 0)
+ return err;
+ }
+
+ if (spec->quirk == QUIRK_ZXR) {
+ err = zxr_add_headphone_gain_switch(codec);
+ if (err < 0)
+ return err;
}
#ifdef ENABLE_TUNING_CONTROLS
add_tuning_ctls(codec);
@@ -5955,6 +6510,27 @@ static int ca0132_build_controls(struct hda_codec *codec)
return 0;
}
+static int dbpro_build_controls(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ int err = 0;
+
+ if (spec->dig_out) {
+ err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out,
+ spec->dig_out);
+ if (err < 0)
+ return err;
+ }
+
+ if (spec->dig_in) {
+ err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
/*
* PCM
*/
@@ -6058,6 +6634,40 @@ static int ca0132_build_pcms(struct hda_codec *codec)
return 0;
}
+static int dbpro_build_pcms(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ struct hda_pcm *info;
+
+ info = snd_hda_codec_pcm_new(codec, "CA0132 Alt Analog");
+ if (!info)
+ return -ENOMEM;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0];
+
+
+ if (!spec->dig_out && !spec->dig_in)
+ return 0;
+
+ info = snd_hda_codec_pcm_new(codec, "CA0132 Digital");
+ if (!info)
+ return -ENOMEM;
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
+ if (spec->dig_out) {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+ ca0132_pcm_digital_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dig_out;
+ }
+ if (spec->dig_in) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ ca0132_pcm_digital_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in;
+ }
+
+ return 0;
+}
+
static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
{
if (pin) {
@@ -6238,69 +6848,48 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec)
}
/*
- * Recon3D r3d_setup_defaults sub functions.
+ * Creates a dummy stream to bind the output to. This seems to have to be done
+ * after changing the main outputs source and destination streams.
*/
-
-static void r3d_dsp_scp_startup(struct hda_codec *codec)
+static void ca0132_alt_create_dummy_stream(struct hda_codec *codec)
{
- unsigned int tmp;
-
- tmp = 0x00000000;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
-
- tmp = 0x00000001;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
-
- tmp = 0x00000004;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
- tmp = 0x00000005;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
- tmp = 0x00000000;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
-}
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int stream_format;
-static void r3d_dsp_initial_mic_setup(struct hda_codec *codec)
-{
- unsigned int tmp;
+ stream_format = snd_hdac_calc_stream_format(48000, 2,
+ SNDRV_PCM_FORMAT_S32_LE, 32, 0);
- /* Mic 1 Setup */
- chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
- chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
- /* This ConnPointID is unique to Recon3Di. Haven't seen it elsewhere */
- chipio_set_conn_rate(codec, 0x0F, SR_96_000);
- tmp = FLOAT_ONE;
- dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+ snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
+ 0, stream_format);
- /* Mic 2 Setup, even though it isn't connected on SBZ */
- chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000);
- chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000);
- chipio_set_conn_rate(codec, 0x0F, SR_96_000);
- tmp = FLOAT_ZERO;
- dspio_set_uint_param(codec, 0x80, 0x01, tmp);
+ snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
}
/*
- * Initialize Sound Blaster Z analog microphones.
+ * Initialize mic for non-chromebook ca0132 implementations.
*/
-static void sbz_init_analog_mics(struct hda_codec *codec)
+static void ca0132_alt_init_analog_mics(struct hda_codec *codec)
{
+ struct ca0132_spec *spec = codec->spec;
unsigned int tmp;
/* Mic 1 Setup */
chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
- tmp = FLOAT_THREE;
+ if (spec->quirk == QUIRK_R3DI) {
+ chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+ tmp = FLOAT_ONE;
+ } else
+ tmp = FLOAT_THREE;
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
- /* Mic 2 Setup, even though it isn't connected on SBZ */
+ /* Mic 2 setup (not present on desktop cards) */
chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000);
chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000);
+ if (spec->quirk == QUIRK_R3DI)
+ chipio_set_conn_rate(codec, 0x0F, SR_96_000);
tmp = FLOAT_ZERO;
dspio_set_uint_param(codec, 0x80, 0x01, tmp);
-
}
/*
@@ -6333,7 +6922,6 @@ static void sbz_connect_streams(struct hda_codec *codec)
codec_dbg(codec, "Connect Streams exited, mutex released.\n");
mutex_unlock(&spec->chipio_mutex);
-
}
/*
@@ -6360,19 +6948,29 @@ static void sbz_chipio_startup_data(struct hda_codec *codec)
chipio_set_stream_channels(codec, 0x0C, 6);
chipio_set_stream_control(codec, 0x0C, 1);
/* No clue what these control */
- chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0);
- chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1);
- chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2);
- chipio_write_no_mutex(codec, 0x19003c, 0x0001e5c3);
- chipio_write_no_mutex(codec, 0x190040, 0x0001e2c4);
- chipio_write_no_mutex(codec, 0x190044, 0x0001e3c5);
- chipio_write_no_mutex(codec, 0x190048, 0x0001e8c6);
- chipio_write_no_mutex(codec, 0x19004c, 0x0001e9c7);
- chipio_write_no_mutex(codec, 0x190050, 0x0001ecc8);
- chipio_write_no_mutex(codec, 0x190054, 0x0001edc9);
- chipio_write_no_mutex(codec, 0x190058, 0x0001eaca);
- chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb);
-
+ if (spec->quirk == QUIRK_SBZ) {
+ chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0);
+ chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1);
+ chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2);
+ chipio_write_no_mutex(codec, 0x19003c, 0x0001e5c3);
+ chipio_write_no_mutex(codec, 0x190040, 0x0001e2c4);
+ chipio_write_no_mutex(codec, 0x190044, 0x0001e3c5);
+ chipio_write_no_mutex(codec, 0x190048, 0x0001e8c6);
+ chipio_write_no_mutex(codec, 0x19004c, 0x0001e9c7);
+ chipio_write_no_mutex(codec, 0x190050, 0x0001ecc8);
+ chipio_write_no_mutex(codec, 0x190054, 0x0001edc9);
+ chipio_write_no_mutex(codec, 0x190058, 0x0001eaca);
+ chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb);
+ } else if (spec->quirk == QUIRK_ZXR) {
+ chipio_write_no_mutex(codec, 0x190038, 0x000140c2);
+ chipio_write_no_mutex(codec, 0x19003c, 0x000141c3);
+ chipio_write_no_mutex(codec, 0x190040, 0x000150c4);
+ chipio_write_no_mutex(codec, 0x190044, 0x000151c5);
+ chipio_write_no_mutex(codec, 0x190050, 0x000142c8);
+ chipio_write_no_mutex(codec, 0x190054, 0x000143c9);
+ chipio_write_no_mutex(codec, 0x190058, 0x000152ca);
+ chipio_write_no_mutex(codec, 0x19005c, 0x000153cb);
+ }
chipio_write_no_mutex(codec, 0x19042c, 0x00000001);
codec_dbg(codec, "Startup Data exited, mutex released.\n");
@@ -6380,35 +6978,56 @@ static void sbz_chipio_startup_data(struct hda_codec *codec)
}
/*
- * Sound Blaster Z uses these after DSP is loaded. Weird SCP commands
- * without a 0x20 source like normal.
+ * Custom DSP SCP commands where the src value is 0x00 instead of 0x20. This is
+ * done after the DSP is loaded.
*/
-static void sbz_dsp_scp_startup(struct hda_codec *codec)
+static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec)
{
- unsigned int tmp;
-
- tmp = 0x00000003;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
- tmp = 0x00000000;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
-
- tmp = 0x00000001;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
-
- tmp = 0x00000004;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
- tmp = 0x00000005;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
- tmp = 0x00000000;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int tmp, i;
+ /*
+ * Gotta run these twice, or else mic works inconsistently. Not clear
+ * why this is, but multiple tests have confirmed it.
+ */
+ for (i = 0; i < 2; i++) {
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ case QUIRK_AE5:
+ tmp = 0x00000003;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+ tmp = 0x00000000;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
+ tmp = 0x00000001;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
+ tmp = 0x00000004;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+ tmp = 0x00000005;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+ tmp = 0x00000000;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+ break;
+ case QUIRK_R3D:
+ case QUIRK_R3DI:
+ tmp = 0x00000000;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
+ tmp = 0x00000001;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
+ tmp = 0x00000004;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+ tmp = 0x00000005;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+ tmp = 0x00000000;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+ break;
+ }
+ msleep(100);
+ }
}
-static void sbz_dsp_initial_mic_setup(struct hda_codec *codec)
+static void ca0132_alt_dsp_initial_mic_setup(struct hda_codec *codec)
{
+ struct ca0132_spec *spec = codec->spec;
unsigned int tmp;
chipio_set_stream_control(codec, 0x03, 0);
@@ -6423,8 +7042,161 @@ static void sbz_dsp_initial_mic_setup(struct hda_codec *codec)
chipio_set_stream_control(codec, 0x03, 1);
chipio_set_stream_control(codec, 0x04, 1);
- chipio_write(codec, 0x18b098, 0x0000000c);
- chipio_write(codec, 0x18b09C, 0x0000000c);
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ chipio_write(codec, 0x18b098, 0x0000000c);
+ chipio_write(codec, 0x18b09C, 0x0000000c);
+ break;
+ case QUIRK_AE5:
+ chipio_write(codec, 0x18b098, 0x0000000c);
+ chipio_write(codec, 0x18b09c, 0x0000004c);
+ break;
+ }
+}
+
+static void ae5_post_dsp_register_set(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ chipio_8051_write_direct(codec, 0x93, 0x10);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2);
+
+ writeb(0xff, spec->mem_base + 0x304);
+ writeb(0xff, spec->mem_base + 0x304);
+ writeb(0xff, spec->mem_base + 0x304);
+ writeb(0xff, spec->mem_base + 0x304);
+ writeb(0x00, spec->mem_base + 0x100);
+ writeb(0xff, spec->mem_base + 0x304);
+ writeb(0x00, spec->mem_base + 0x100);
+ writeb(0xff, spec->mem_base + 0x304);
+ writeb(0x00, spec->mem_base + 0x100);
+ writeb(0xff, spec->mem_base + 0x304);
+ writeb(0x00, spec->mem_base + 0x100);
+ writeb(0xff, spec->mem_base + 0x304);
+
+ ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x3f);
+ ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f);
+ ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+}
+
+static void ae5_post_dsp_param_setup(struct hda_codec *codec)
+{
+ /*
+ * Param3 in the 8051's memory is represented by the ascii string 'mch'
+ * which seems to be 'multichannel'. This is also mentioned in the
+ * AE-5's registry values in Windows.
+ */
+ chipio_set_control_param(codec, 3, 0);
+ /*
+ * I believe ASI is 'audio serial interface' and that it's used to
+ * change colors on the external LED strip connected to the AE-5.
+ */
+ chipio_set_control_flag(codec, CONTROL_FLAG_ASI_96KHZ, 1);
+
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x724, 0x83);
+ chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0);
+
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x92);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0xfa);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_DATA_WRITE, 0x22);
+}
+
+static void ae5_post_dsp_pll_setup(struct hda_codec *codec)
+{
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x41);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc8);
+
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x45);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_PLL_PMU_WRITE, 0xcc);
+
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x40);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_PLL_PMU_WRITE, 0xcb);
+
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7);
+
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x51);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_PLL_PMU_WRITE, 0x8d);
+}
+
+static void ae5_post_dsp_stream_setup(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ mutex_lock(&spec->chipio_mutex);
+
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x725, 0x81);
+
+ chipio_set_conn_rate_no_mutex(codec, 0x70, SR_96_000);
+
+ chipio_set_stream_channels(codec, 0x0C, 6);
+ chipio_set_stream_control(codec, 0x0C, 1);
+
+ chipio_set_stream_source_dest(codec, 0x5, 0x43, 0x0);
+
+ chipio_set_stream_source_dest(codec, 0x18, 0x9, 0xd0);
+ chipio_set_conn_rate_no_mutex(codec, 0xd0, SR_96_000);
+ chipio_set_stream_channels(codec, 0x18, 6);
+ chipio_set_stream_control(codec, 0x18, 1);
+
+ chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 4);
+
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7);
+
+ ca0113_mmio_command_set(codec, 0x48, 0x01, 0x80);
+
+ mutex_unlock(&spec->chipio_mutex);
+}
+
+static void ae5_post_dsp_startup_data(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ mutex_lock(&spec->chipio_mutex);
+
+ chipio_write_no_mutex(codec, 0x189000, 0x0001f101);
+ chipio_write_no_mutex(codec, 0x189004, 0x0001f101);
+ chipio_write_no_mutex(codec, 0x189024, 0x00014004);
+ chipio_write_no_mutex(codec, 0x189028, 0x0002000f);
+
+ ca0113_mmio_command_set(codec, 0x48, 0x0a, 0x05);
+ chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 7);
+ ca0113_mmio_command_set(codec, 0x48, 0x0b, 0x12);
+ ca0113_mmio_command_set(codec, 0x48, 0x04, 0x00);
+ ca0113_mmio_command_set(codec, 0x48, 0x06, 0x48);
+ ca0113_mmio_command_set(codec, 0x48, 0x0a, 0x05);
+ ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+ ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00);
+ ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00);
+ ca0113_mmio_gpio_set(codec, 0, true);
+ ca0113_mmio_gpio_set(codec, 1, true);
+ ca0113_mmio_command_set(codec, 0x48, 0x07, 0x80);
+
+ chipio_write_no_mutex(codec, 0x18b03c, 0x00000012);
+
+ ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00);
+ ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00);
+
+ mutex_unlock(&spec->chipio_mutex);
}
/*
@@ -6485,9 +7257,8 @@ static void r3d_setup_defaults(struct hda_codec *codec)
if (spec->dsp_state != DSP_DOWNLOADED)
return;
- r3d_dsp_scp_startup(codec);
-
- r3d_dsp_initial_mic_setup(codec);
+ ca0132_alt_dsp_scp_startup(codec);
+ ca0132_alt_init_analog_mics(codec);
/*remove DSP headroom*/
tmp = FLOAT_ZERO;
@@ -6523,19 +7294,16 @@ static void r3d_setup_defaults(struct hda_codec *codec)
static void sbz_setup_defaults(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
- unsigned int tmp, stream_format;
+ unsigned int tmp;
int num_fx;
int idx, i;
if (spec->dsp_state != DSP_DOWNLOADED)
return;
- sbz_dsp_scp_startup(codec);
-
- sbz_init_analog_mics(codec);
-
+ ca0132_alt_dsp_scp_startup(codec);
+ ca0132_alt_init_analog_mics(codec);
sbz_connect_streams(codec);
-
sbz_chipio_startup_data(codec);
chipio_set_stream_control(codec, 0x03, 1);
@@ -6561,8 +7329,7 @@ static void sbz_setup_defaults(struct hda_codec *codec)
/* Set speaker source? */
dspio_set_uint_param(codec, 0x32, 0x00, tmp);
- sbz_dsp_initial_mic_setup(codec);
-
+ ca0132_alt_dsp_initial_mic_setup(codec);
/* out, in effects + voicefx */
num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1;
@@ -6575,23 +7342,74 @@ static void sbz_setup_defaults(struct hda_codec *codec)
}
}
- /*
- * Have to make a stream to bind the sound output to, otherwise
- * you'll get dead audio. Before I did this, it would bind to an
- * audio input, and would never work
- */
- stream_format = snd_hdac_calc_stream_format(48000, 2,
- SNDRV_PCM_FORMAT_S32_LE, 32, 0);
+ ca0132_alt_create_dummy_stream(codec);
+}
- snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
- 0, stream_format);
+/*
+ * Setup default parameters for the Sound BlasterX AE-5 DSP.
+ */
+static void ae5_setup_defaults(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int tmp;
+ int num_fx;
+ int idx, i;
- snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
+ if (spec->dsp_state != DSP_DOWNLOADED)
+ return;
- snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
- 0, stream_format);
+ ca0132_alt_dsp_scp_startup(codec);
+ ca0132_alt_init_analog_mics(codec);
+ chipio_set_stream_control(codec, 0x03, 1);
+ chipio_set_stream_control(codec, 0x04, 1);
- snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
+ /* New, unknown SCP req's */
+ tmp = FLOAT_ZERO;
+ dspio_set_uint_param(codec, 0x96, 0x29, tmp);
+ dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
+ dspio_set_uint_param(codec, 0x80, 0x0d, tmp);
+ dspio_set_uint_param(codec, 0x80, 0x0e, tmp);
+
+ ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f);
+ ca0113_mmio_gpio_set(codec, 0, false);
+ ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
+
+ /* Internal loopback off */
+ tmp = FLOAT_ONE;
+ dspio_set_uint_param(codec, 0x37, 0x08, tmp);
+ dspio_set_uint_param(codec, 0x37, 0x10, tmp);
+
+ /*remove DSP headroom*/
+ tmp = FLOAT_ZERO;
+ dspio_set_uint_param(codec, 0x96, 0x3C, tmp);
+
+ /* set WUH source */
+ tmp = FLOAT_TWO;
+ dspio_set_uint_param(codec, 0x31, 0x00, tmp);
+ chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000);
+
+ /* Set speaker source? */
+ dspio_set_uint_param(codec, 0x32, 0x00, tmp);
+
+ ca0132_alt_dsp_initial_mic_setup(codec);
+ ae5_post_dsp_register_set(codec);
+ ae5_post_dsp_param_setup(codec);
+ ae5_post_dsp_pll_setup(codec);
+ ae5_post_dsp_stream_setup(codec);
+ ae5_post_dsp_startup_data(codec);
+
+ /* out, in effects + voicefx */
+ num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1;
+ for (idx = 0; idx < num_fx; idx++) {
+ for (i = 0; i <= ca0132_effects[idx].params; i++) {
+ dspio_set_uint_param(codec,
+ ca0132_effects[idx].mid,
+ ca0132_effects[idx].reqs[i],
+ ca0132_effects[idx].def_vals[i]);
+ }
+ }
+
+ ca0132_alt_create_dummy_stream(codec);
}
/*
@@ -6673,12 +7491,14 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec)
*/
switch (spec->quirk) {
case QUIRK_SBZ:
- if (request_firmware(&fw_entry, SBZ_EFX_FILE,
+ case QUIRK_R3D:
+ case QUIRK_AE5:
+ if (request_firmware(&fw_entry, DESKTOP_EFX_FILE,
codec->card->dev) != 0) {
- codec_dbg(codec, "SBZ alt firmware not detected. ");
+ codec_dbg(codec, "Desktop firmware not found.");
spec->alt_firmware_present = false;
} else {
- codec_dbg(codec, "Sound Blaster Z firmware selected.");
+ codec_dbg(codec, "Desktop firmware selected.");
spec->alt_firmware_present = true;
}
break;
@@ -6921,6 +7741,14 @@ static void ca0132_init_chip(struct hda_codec *codec)
spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID] = 1;
spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] = 0;
+ /*
+ * The ZxR doesn't have a front panel header, and it's line-in is on
+ * the daughter board. So, there is no input enum control, and we need
+ * to make sure that spec->in_enum_val is set properly.
+ */
+ if (spec->quirk == QUIRK_ZXR)
+ spec->in_enum_val = REAR_MIC;
+
#ifdef ENABLE_TUNING_CONTROLS
ca0132_init_tuning_defaults(codec);
#endif
@@ -6948,11 +7776,11 @@ static void sbz_region2_exit(struct hda_codec *codec)
for (i = 0; i < 8; i++)
writeb(0xb3, spec->mem_base + 0x304);
- ca0132_mmio_gpio_set(codec, 0, false);
- ca0132_mmio_gpio_set(codec, 1, false);
- ca0132_mmio_gpio_set(codec, 4, true);
- ca0132_mmio_gpio_set(codec, 5, false);
- ca0132_mmio_gpio_set(codec, 7, false);
+ ca0113_mmio_gpio_set(codec, 0, false);
+ ca0113_mmio_gpio_set(codec, 1, false);
+ ca0113_mmio_gpio_set(codec, 4, true);
+ ca0113_mmio_gpio_set(codec, 5, false);
+ ca0113_mmio_gpio_set(codec, 7, false);
}
static void sbz_set_pin_ctl_default(struct hda_codec *codec)
@@ -6995,6 +7823,16 @@ static void sbz_gpio_shutdown_commands(struct hda_codec *codec, int dir,
AC_VERB_SET_GPIO_DATA, data);
}
+static void zxr_dbpro_power_state_shutdown(struct hda_codec *codec)
+{
+ hda_nid_t pins[7] = {0x05, 0x0c, 0x09, 0x0e, 0x08, 0x11, 0x01};
+ unsigned int i;
+
+ for (i = 0; i < 7; i++)
+ snd_hda_codec_write(codec, pins[i], 0,
+ AC_VERB_SET_POWER_STATE, 0x03);
+}
+
static void sbz_exit_chip(struct hda_codec *codec)
{
chipio_set_stream_control(codec, 0x03, 0);
@@ -7037,6 +7875,61 @@ static void r3d_exit_chip(struct hda_codec *codec)
snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5b);
}
+static void ae5_exit_chip(struct hda_codec *codec)
+{
+ chipio_set_stream_control(codec, 0x03, 0);
+ chipio_set_stream_control(codec, 0x04, 0);
+
+ ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
+ ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+ ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+ ca0113_mmio_command_set(codec, 0x30, 0x30, 0x00);
+ ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00);
+ ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x00);
+ ca0113_mmio_gpio_set(codec, 0, false);
+ ca0113_mmio_gpio_set(codec, 1, false);
+
+ snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
+ snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
+
+ chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0);
+
+ chipio_set_stream_control(codec, 0x18, 0);
+ chipio_set_stream_control(codec, 0x0c, 0);
+
+ snd_hda_codec_write(codec, 0x01, 0, 0x724, 0x83);
+}
+
+static void zxr_exit_chip(struct hda_codec *codec)
+{
+ chipio_set_stream_control(codec, 0x03, 0);
+ chipio_set_stream_control(codec, 0x04, 0);
+ chipio_set_stream_control(codec, 0x14, 0);
+ chipio_set_stream_control(codec, 0x0C, 0);
+
+ chipio_set_conn_rate(codec, 0x41, SR_192_000);
+ chipio_set_conn_rate(codec, 0x91, SR_192_000);
+
+ chipio_write(codec, 0x18a020, 0x00000083);
+
+ snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
+ snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
+
+ ca0132_clear_unsolicited(codec);
+ sbz_set_pin_ctl_default(codec);
+ snd_hda_codec_write(codec, 0x0B, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x00);
+
+ ca0113_mmio_gpio_set(codec, 5, false);
+ ca0113_mmio_gpio_set(codec, 2, false);
+ ca0113_mmio_gpio_set(codec, 3, false);
+ ca0113_mmio_gpio_set(codec, 0, false);
+ ca0113_mmio_gpio_set(codec, 4, true);
+ ca0113_mmio_gpio_set(codec, 0, true);
+ ca0113_mmio_gpio_set(codec, 5, true);
+ ca0113_mmio_gpio_set(codec, 2, false);
+ ca0113_mmio_gpio_set(codec, 3, false);
+}
+
static void ca0132_exit_chip(struct hda_codec *codec)
{
/* put any chip cleanup stuffs here. */
@@ -7140,11 +8033,6 @@ static void sbz_pre_dsp_setup(struct hda_codec *codec)
writel(0x00820680, spec->mem_base + 0x01C);
writel(0x00820680, spec->mem_base + 0x01C);
- snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfc);
- snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfd);
- snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfe);
- snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xff);
-
chipio_write(codec, 0x18b0a4, 0x000000c2);
snd_hda_codec_write(codec, 0x11, 0,
@@ -7153,12 +8041,6 @@ static void sbz_pre_dsp_setup(struct hda_codec *codec)
static void r3d_pre_dsp_setup(struct hda_codec *codec)
{
-
- snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfc);
- snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfd);
- snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfe);
- snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xff);
-
chipio_write(codec, 0x18b0a4, 0x000000c2);
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
@@ -7205,23 +8087,116 @@ static void ca0132_mmio_init(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
- writel(0x00000000, spec->mem_base + 0x400);
- writel(0x00000000, spec->mem_base + 0x408);
- writel(0x00000000, spec->mem_base + 0x40C);
- writel(0x00880680, spec->mem_base + 0x01C);
- writel(0x00000083, spec->mem_base + 0xC0C);
+ if (spec->quirk == QUIRK_AE5)
+ writel(0x00000001, spec->mem_base + 0x400);
+ else
+ writel(0x00000000, spec->mem_base + 0x400);
+
+ if (spec->quirk == QUIRK_AE5)
+ writel(0x00000001, spec->mem_base + 0x408);
+ else
+ writel(0x00000000, spec->mem_base + 0x408);
+
+ if (spec->quirk == QUIRK_AE5)
+ writel(0x00000001, spec->mem_base + 0x40c);
+ else
+ writel(0x00000000, spec->mem_base + 0x40C);
+
+ if (spec->quirk == QUIRK_ZXR)
+ writel(0x00880640, spec->mem_base + 0x01C);
+ else
+ writel(0x00880680, spec->mem_base + 0x01C);
+
+ if (spec->quirk == QUIRK_AE5)
+ writel(0x00000080, spec->mem_base + 0xC0C);
+ else
+ writel(0x00000083, spec->mem_base + 0xC0C);
+
writel(0x00000030, spec->mem_base + 0xC00);
writel(0x00000000, spec->mem_base + 0xC04);
+
+ if (spec->quirk == QUIRK_AE5)
+ writel(0x00000000, spec->mem_base + 0xC0C);
+ else
+ writel(0x00000003, spec->mem_base + 0xC0C);
+
writel(0x00000003, spec->mem_base + 0xC0C);
writel(0x00000003, spec->mem_base + 0xC0C);
writel(0x00000003, spec->mem_base + 0xC0C);
- writel(0x00000003, spec->mem_base + 0xC0C);
- writel(0x000000C1, spec->mem_base + 0xC08);
+
+ if (spec->quirk == QUIRK_AE5)
+ writel(0x00000001, spec->mem_base + 0xC08);
+ else
+ writel(0x000000C1, spec->mem_base + 0xC08);
+
writel(0x000000F1, spec->mem_base + 0xC08);
writel(0x00000001, spec->mem_base + 0xC08);
writel(0x000000C7, spec->mem_base + 0xC08);
writel(0x000000C1, spec->mem_base + 0xC08);
writel(0x00000080, spec->mem_base + 0xC04);
+
+ if (spec->quirk == QUIRK_AE5) {
+ writel(0x00000000, spec->mem_base + 0x42c);
+ writel(0x00000000, spec->mem_base + 0x46c);
+ writel(0x00000000, spec->mem_base + 0x4ac);
+ writel(0x00000000, spec->mem_base + 0x4ec);
+ writel(0x00000000, spec->mem_base + 0x43c);
+ writel(0x00000000, spec->mem_base + 0x47c);
+ writel(0x00000000, spec->mem_base + 0x4bc);
+ writel(0x00000000, spec->mem_base + 0x4fc);
+ writel(0x00000600, spec->mem_base + 0x100);
+ writel(0x00000014, spec->mem_base + 0x410);
+ writel(0x0000060f, spec->mem_base + 0x100);
+ writel(0x0000070f, spec->mem_base + 0x100);
+ writel(0x00000aff, spec->mem_base + 0x830);
+ writel(0x00000000, spec->mem_base + 0x86c);
+ writel(0x0000006b, spec->mem_base + 0x800);
+ writel(0x00000001, spec->mem_base + 0x86c);
+ writel(0x0000006b, spec->mem_base + 0x800);
+ writel(0x00000057, spec->mem_base + 0x804);
+ writel(0x00800000, spec->mem_base + 0x20c);
+ }
+}
+
+/*
+ * This function writes to some SFR's, does some region2 writes, and then
+ * eventually resets the codec with the 0x7ff verb. Not quite sure why it does
+ * what it does.
+ */
+static void ae5_register_set(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ chipio_8051_write_direct(codec, 0x93, 0x10);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2);
+
+ writeb(0x0f, spec->mem_base + 0x304);
+ writeb(0x0f, spec->mem_base + 0x304);
+ writeb(0x0f, spec->mem_base + 0x304);
+ writeb(0x0f, spec->mem_base + 0x304);
+ writeb(0x0e, spec->mem_base + 0x100);
+ writeb(0x1f, spec->mem_base + 0x304);
+ writeb(0x0c, spec->mem_base + 0x100);
+ writeb(0x3f, spec->mem_base + 0x304);
+ writeb(0x08, spec->mem_base + 0x100);
+ writeb(0x7f, spec->mem_base + 0x304);
+ writeb(0x00, spec->mem_base + 0x100);
+ writeb(0xff, spec->mem_base + 0x304);
+
+ ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f);
+
+ chipio_8051_write_direct(codec, 0x90, 0x00);
+ chipio_8051_write_direct(codec, 0x90, 0x10);
+
+ ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+
+ chipio_write(codec, 0x18b0a4, 0x000000c2);
+
+ snd_hda_codec_write(codec, 0x01, 0, 0x7ff, 0x00);
+ snd_hda_codec_write(codec, 0x01, 0, 0x7ff, 0x00);
}
/*
@@ -7257,6 +8232,21 @@ static void ca0132_alt_init(struct hda_codec *codec)
snd_hda_sequence_write(codec, spec->chip_init_verbs);
snd_hda_sequence_write(codec, spec->desktop_init_verbs);
break;
+ case QUIRK_AE5:
+ ca0132_gpio_init(codec);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x49);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_PLL_PMU_WRITE, 0x88);
+ chipio_write(codec, 0x18b030, 0x00000020);
+ snd_hda_sequence_write(codec, spec->chip_init_verbs);
+ snd_hda_sequence_write(codec, spec->desktop_init_verbs);
+ ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
+ break;
+ case QUIRK_ZXR:
+ snd_hda_sequence_write(codec, spec->chip_init_verbs);
+ snd_hda_sequence_write(codec, spec->desktop_init_verbs);
+ break;
}
}
@@ -7298,6 +8288,9 @@ static int ca0132_init(struct hda_codec *codec)
snd_hda_power_up_pm(codec);
+ if (spec->quirk == QUIRK_AE5)
+ ae5_register_set(codec);
+
ca0132_init_unsol(codec);
ca0132_init_params(codec);
ca0132_init_flags(codec);
@@ -7317,8 +8310,12 @@ static int ca0132_init(struct hda_codec *codec)
r3d_setup_defaults(codec);
break;
case QUIRK_SBZ:
+ case QUIRK_ZXR:
sbz_setup_defaults(codec);
break;
+ case QUIRK_AE5:
+ ae5_setup_defaults(codec);
+ break;
default:
ca0132_setup_defaults(codec);
ca0132_init_analog_mic2(codec);
@@ -7372,6 +8369,21 @@ static int ca0132_init(struct hda_codec *codec)
return 0;
}
+static int dbpro_init(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ unsigned int i;
+
+ init_output(codec, cfg->dig_out_pins[0], spec->dig_out);
+ init_input(codec, cfg->dig_in_pin, spec->dig_in);
+
+ for (i = 0; i < spec->num_inputs; i++)
+ init_input(codec, spec->input_pins[i], spec->adcs[i]);
+
+ return 0;
+}
+
static void ca0132_free(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
@@ -7382,9 +8394,15 @@ static void ca0132_free(struct hda_codec *codec)
case QUIRK_SBZ:
sbz_exit_chip(codec);
break;
+ case QUIRK_ZXR:
+ zxr_exit_chip(codec);
+ break;
case QUIRK_R3D:
r3d_exit_chip(codec);
break;
+ case QUIRK_AE5:
+ ae5_exit_chip(codec);
+ break;
case QUIRK_R3DI:
r3di_gpio_shutdown(codec);
break;
@@ -7400,6 +8418,16 @@ static void ca0132_free(struct hda_codec *codec)
kfree(codec->spec);
}
+static void dbpro_free(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ zxr_dbpro_power_state_shutdown(codec);
+
+ kfree(spec->spec_init_verbs);
+ kfree(codec->spec);
+}
+
static void ca0132_reboot_notify(struct hda_codec *codec)
{
codec->patch_ops.free(codec);
@@ -7414,6 +8442,13 @@ static const struct hda_codec_ops ca0132_patch_ops = {
.reboot_notify = ca0132_reboot_notify,
};
+static const struct hda_codec_ops dbpro_patch_ops = {
+ .build_controls = dbpro_build_controls,
+ .build_pcms = dbpro_build_pcms,
+ .init = dbpro_init,
+ .free = dbpro_free,
+};
+
static void ca0132_config(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
@@ -7432,9 +8467,33 @@ static void ca0132_config(struct hda_codec *codec)
switch (spec->quirk) {
case QUIRK_ALIENWARE:
- codec_dbg(codec, "ca0132_config: QUIRK_ALIENWARE applied.\n");
+ codec_dbg(codec, "%s: QUIRK_ALIENWARE applied.\n", __func__);
snd_hda_apply_pincfgs(codec, alienware_pincfgs);
+ break;
+ case QUIRK_SBZ:
+ codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__);
+ snd_hda_apply_pincfgs(codec, sbz_pincfgs);
+ break;
+ case QUIRK_ZXR:
+ codec_dbg(codec, "%s: QUIRK_ZXR applied.\n", __func__);
+ snd_hda_apply_pincfgs(codec, zxr_pincfgs);
+ break;
+ case QUIRK_R3D:
+ codec_dbg(codec, "%s: QUIRK_R3D applied.\n", __func__);
+ snd_hda_apply_pincfgs(codec, r3d_pincfgs);
+ break;
+ case QUIRK_R3DI:
+ codec_dbg(codec, "%s: QUIRK_R3DI applied.\n", __func__);
+ snd_hda_apply_pincfgs(codec, r3di_pincfgs);
+ break;
+ case QUIRK_AE5:
+ codec_dbg(codec, "%s: QUIRK_AE5 applied.\n", __func__);
+ snd_hda_apply_pincfgs(codec, r3di_pincfgs);
+ break;
+ }
+ switch (spec->quirk) {
+ case QUIRK_ALIENWARE:
spec->num_outputs = 2;
spec->out_pins[0] = 0x0b; /* speaker out */
spec->out_pins[1] = 0x0f;
@@ -7454,15 +8513,6 @@ static void ca0132_config(struct hda_codec *codec)
break;
case QUIRK_SBZ:
case QUIRK_R3D:
- if (spec->quirk == QUIRK_SBZ) {
- codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__);
- snd_hda_apply_pincfgs(codec, sbz_pincfgs);
- }
- if (spec->quirk == QUIRK_R3D) {
- codec_dbg(codec, "%s: QUIRK_R3D applied.\n", __func__);
- snd_hda_apply_pincfgs(codec, r3d_pincfgs);
- }
-
spec->num_outputs = 2;
spec->out_pins[0] = 0x0B; /* Line out */
spec->out_pins[1] = 0x0F; /* Rear headphone out */
@@ -7487,10 +8537,62 @@ static void ca0132_config(struct hda_codec *codec)
spec->multiout.dig_out_nid = spec->dig_out;
spec->dig_in = 0x09;
break;
- case QUIRK_R3DI:
- codec_dbg(codec, "%s: QUIRK_R3DI applied.\n", __func__);
- snd_hda_apply_pincfgs(codec, r3di_pincfgs);
+ case QUIRK_ZXR:
+ spec->num_outputs = 2;
+ spec->out_pins[0] = 0x0B; /* Line out */
+ spec->out_pins[1] = 0x0F; /* Rear headphone out */
+ spec->out_pins[2] = 0x10; /* Center/LFE */
+ spec->out_pins[3] = 0x11; /* Rear surround */
+ spec->shared_out_nid = 0x2;
+ spec->unsol_tag_hp = spec->out_pins[1];
+ spec->unsol_tag_front_hp = spec->out_pins[2];
+ spec->adcs[0] = 0x7; /* Rear Mic / Line-in */
+ spec->adcs[1] = 0x8; /* Not connected, no front mic */
+ spec->adcs[2] = 0xa; /* what u hear */
+
+ spec->num_inputs = 2;
+ spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */
+ spec->input_pins[1] = 0x13; /* What U Hear */
+ spec->shared_mic_nid = 0x7;
+ spec->unsol_tag_amic1 = spec->input_pins[0];
+ break;
+ case QUIRK_ZXR_DBPRO:
+ spec->adcs[0] = 0x8; /* ZxR DBPro Aux In */
+
+ spec->num_inputs = 1;
+ spec->input_pins[0] = 0x11; /* RCA Line-in */
+
+ spec->dig_out = 0x05;
+ spec->multiout.dig_out_nid = spec->dig_out;
+
+ spec->dig_in = 0x09;
+ break;
+ case QUIRK_AE5:
+ spec->num_outputs = 2;
+ spec->out_pins[0] = 0x0B; /* Line out */
+ spec->out_pins[1] = 0x11; /* Rear headphone out */
+ spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/
+ spec->out_pins[3] = 0x0F; /* Rear surround */
+ spec->shared_out_nid = 0x2;
+ spec->unsol_tag_hp = spec->out_pins[1];
+ spec->unsol_tag_front_hp = spec->out_pins[2];
+
+ spec->adcs[0] = 0x7; /* Rear Mic / Line-in */
+ spec->adcs[1] = 0x8; /* Front Mic, but only if no DSP */
+ spec->adcs[2] = 0xa; /* what u hear */
+
+ spec->num_inputs = 2;
+ spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */
+ spec->input_pins[1] = 0x13; /* What U Hear */
+ spec->shared_mic_nid = 0x7;
+ spec->unsol_tag_amic1 = spec->input_pins[0];
+
+ /* SPDIF I/O */
+ spec->dig_out = 0x05;
+ spec->multiout.dig_out_nid = spec->dig_out;
+ break;
+ case QUIRK_R3DI:
spec->num_outputs = 2;
spec->out_pins[0] = 0x0B; /* Line out */
spec->out_pins[1] = 0x0F; /* Rear headphone out */
@@ -7547,7 +8649,11 @@ static int ca0132_prepare_verbs(struct hda_codec *codec)
struct ca0132_spec *spec = codec->spec;
spec->chip_init_verbs = ca0132_init_verbs0;
- if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3D)
+ /*
+ * Since desktop cards use pci_mmio, this can be used to determine
+ * whether or not to use these verbs instead of a separate bool.
+ */
+ if (spec->use_pci_mmio)
spec->desktop_init_verbs = ca0132_init_verbs1;
spec->spec_init_verbs = kcalloc(NUM_SPEC_VERBS,
sizeof(struct hda_verb),
@@ -7579,6 +8685,29 @@ static int ca0132_prepare_verbs(struct hda_codec *codec)
return 0;
}
+/*
+ * The Sound Blaster ZxR shares the same PCI subsystem ID as some regular
+ * Sound Blaster Z cards. However, they have different HDA codec subsystem
+ * ID's. So, we check for the ZxR's subsystem ID, as well as the DBPro
+ * daughter boards ID.
+ */
+static void sbz_detect_quirk(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ switch (codec->core.subsystem_id) {
+ case 0x11020033:
+ spec->quirk = QUIRK_ZXR;
+ break;
+ case 0x1102003f:
+ spec->quirk = QUIRK_ZXR_DBPRO;
+ break;
+ default:
+ spec->quirk = QUIRK_SBZ;
+ break;
+ }
+}
+
static int patch_ca0132(struct hda_codec *codec)
{
struct ca0132_spec *spec;
@@ -7593,10 +8722,6 @@ static int patch_ca0132(struct hda_codec *codec)
codec->spec = spec;
spec->codec = codec;
- codec->patch_ops = ca0132_patch_ops;
- codec->pcm_format_first = 1;
- codec->no_sticky_stream = 1;
-
/* Detect codec quirk */
quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks);
if (quirk)
@@ -7604,6 +8729,18 @@ static int patch_ca0132(struct hda_codec *codec)
else
spec->quirk = QUIRK_NONE;
+ if (spec->quirk == QUIRK_SBZ)
+ sbz_detect_quirk(codec);
+
+ if (spec->quirk == QUIRK_ZXR_DBPRO)
+ codec->patch_ops = dbpro_patch_ops;
+ else
+ codec->patch_ops = ca0132_patch_ops;
+
+ codec->pcm_format_first = 1;
+ codec->no_sticky_stream = 1;
+
+
spec->dsp_state = DSP_DOWNLOAD_INIT;
spec->num_mixers = 1;
@@ -7613,6 +8750,12 @@ static int patch_ca0132(struct hda_codec *codec)
spec->mixers[0] = desktop_mixer;
snd_hda_codec_set_name(codec, "Sound Blaster Z");
break;
+ case QUIRK_ZXR:
+ spec->mixers[0] = desktop_mixer;
+ snd_hda_codec_set_name(codec, "Sound Blaster ZxR");
+ break;
+ case QUIRK_ZXR_DBPRO:
+ break;
case QUIRK_R3D:
spec->mixers[0] = desktop_mixer;
snd_hda_codec_set_name(codec, "Recon3D");
@@ -7621,6 +8764,10 @@ static int patch_ca0132(struct hda_codec *codec)
spec->mixers[0] = r3di_mixer;
snd_hda_codec_set_name(codec, "Recon3Di");
break;
+ case QUIRK_AE5:
+ spec->mixers[0] = desktop_mixer;
+ snd_hda_codec_set_name(codec, "Sound BlasterX AE-5");
+ break;
default:
spec->mixers[0] = ca0132_mixer;
break;
@@ -7630,6 +8777,8 @@ static int patch_ca0132(struct hda_codec *codec)
switch (spec->quirk) {
case QUIRK_SBZ:
case QUIRK_R3D:
+ case QUIRK_AE5:
+ case QUIRK_ZXR:
spec->use_alt_controls = true;
spec->use_alt_functions = true;
spec->use_pci_mmio = true;
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index a7f91be45194..64fa5a82bb9f 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -23,7 +23,7 @@
#include <linux/module.h>
#include <sound/core.h>
#include <sound/tlv.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_auto_parser.h"
#include "hda_jack.h"
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 1b2195dd2b26..52642ba3e2c0 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -25,7 +25,7 @@
#include <linux/slab.h>
#include <linux/module.h>
#include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_auto_parser.h"
#include "hda_jack.h"
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index cfd4e4f97f8f..950e02e71766 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -27,7 +27,7 @@
#include <sound/core.h>
#include <sound/jack.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_auto_parser.h"
#include "hda_beep.h"
@@ -943,6 +943,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD),
+ SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC),
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index cb587dce67a9..67099cbb6be2 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -41,7 +41,7 @@
#include <sound/hdaudio.h>
#include <sound/hda_i915.h>
#include <sound/hda_chmap.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_jack.h"
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3ac7ba9b342d..fa61674a5605 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -32,7 +32,7 @@
#include <linux/input.h>
#include <sound/core.h>
#include <sound/jack.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_auto_parser.h"
#include "hda_jack.h"
@@ -6843,6 +6843,12 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x21, 0x0221101f}),
SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC294_FIXUP_LENOVO_MIC_LOCATION,
{0x14, 0x90170110},
+ {0x19, 0x02a11030},
+ {0x1a, 0x02a11040},
+ {0x1b, 0x01011020},
+ {0x21, 0x0221101f}),
+ SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC294_FIXUP_LENOVO_MIC_LOCATION,
+ {0x14, 0x90170110},
{0x19, 0x02a11020},
{0x1a, 0x02a11030},
{0x21, 0x0221101f}),
@@ -7738,6 +7744,8 @@ enum {
ALC662_FIXUP_ASUS_Nx50,
ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE,
ALC668_FIXUP_ASUS_Nx51,
+ ALC668_FIXUP_MIC_COEF,
+ ALC668_FIXUP_ASUS_G751,
ALC891_FIXUP_HEADSET_MODE,
ALC891_FIXUP_DELL_MIC_NO_PRESENCE,
ALC662_FIXUP_ACER_VERITON,
@@ -8007,6 +8015,23 @@ static const struct hda_fixup alc662_fixups[] = {
.chained = true,
.chain_id = ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE,
},
+ [ALC668_FIXUP_MIC_COEF] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0xc3 },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x4000 },
+ {}
+ },
+ },
+ [ALC668_FIXUP_ASUS_G751] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x16, 0x0421101f }, /* HP */
+ {}
+ },
+ .chained = true,
+ .chain_id = ALC668_FIXUP_MIC_COEF
+ },
[ALC891_FIXUP_HEADSET_MODE] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_headset_mode,
@@ -8080,6 +8105,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50),
SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A),
SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50),
+ SND_PCI_QUIRK(0x1043, 0x12ff, "ASUS G751", ALC668_FIXUP_ASUS_G751),
SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP),
SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16),
SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51),
@@ -8184,6 +8210,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = {
{.id = ALC668_FIXUP_DELL_XPS13, .name = "dell-xps13"},
{.id = ALC662_FIXUP_ASUS_Nx50, .name = "asus-nx50"},
{.id = ALC668_FIXUP_ASUS_Nx51, .name = "asus-nx51"},
+ {.id = ALC668_FIXUP_ASUS_G751, .name = "asus-g751"},
{.id = ALC891_FIXUP_HEADSET_MODE, .name = "alc891-headset"},
{.id = ALC891_FIXUP_DELL_MIC_NO_PRESENCE, .name = "alc891-headset-multi"},
{.id = ALC662_FIXUP_ACER_VERITON, .name = "acer-veriton"},
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c
index f63acb1b965c..c49d25bcd7f2 100644
--- a/sound/pci/hda/patch_si3054.c
+++ b/sound/pci/hda/patch_si3054.c
@@ -27,7 +27,7 @@
#include <linux/slab.h>
#include <linux/module.h>
#include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
/* si3054 verbs */
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 046705b4691a..1b6ecfb01759 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -32,7 +32,7 @@
#include <linux/module.h>
#include <sound/core.h>
#include <sound/jack.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_auto_parser.h"
#include "hda_beep.h"
@@ -77,6 +77,7 @@ enum {
STAC_DELL_M6_BOTH,
STAC_DELL_EQ,
STAC_ALIENWARE_M17X,
+ STAC_ELO_VUPOINT_15MX,
STAC_92HD89XX_HP_FRONT_JACK,
STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK,
STAC_92HD73XX_ASUS_MOBO,
@@ -1879,6 +1880,18 @@ static void stac92hd73xx_fixup_no_jd(struct hda_codec *codec,
codec->no_jack_detect = 1;
}
+
+static void stac92hd73xx_disable_automute(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct sigmatel_spec *spec = codec->spec;
+
+ if (action != HDA_FIXUP_ACT_PRE_PROBE)
+ return;
+
+ spec->gen.suppress_auto_mute = 1;
+}
+
static const struct hda_fixup stac92hd73xx_fixups[] = {
[STAC_92HD73XX_REF] = {
.type = HDA_FIXUP_FUNC,
@@ -1904,6 +1917,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = stac92hd73xx_fixup_alienware_m17x,
},
+ [STAC_ELO_VUPOINT_15MX] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = stac92hd73xx_disable_automute,
+ },
[STAC_92HD73XX_INTEL] = {
.type = HDA_FIXUP_PINS,
.v.pins = intel_dg45id_pin_configs,
@@ -1942,6 +1959,7 @@ static const struct hda_model_fixup stac92hd73xx_models[] = {
{ .id = STAC_DELL_M6_BOTH, .name = "dell-m6" },
{ .id = STAC_DELL_EQ, .name = "dell-eq" },
{ .id = STAC_ALIENWARE_M17X, .name = "alienware" },
+ { .id = STAC_ELO_VUPOINT_15MX, .name = "elo-vupoint-15mx" },
{ .id = STAC_92HD73XX_ASUS_MOBO, .name = "asus-mobo" },
{}
};
@@ -1991,6 +2009,8 @@ 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(0x1059, 0x1011,
+ "ELO VuPoint 15MX", STAC_ELO_VUPOINT_15MX),
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,
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 6b9617aee0e6..9f6f13e25145 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -52,7 +52,7 @@
#include <linux/module.h>
#include <sound/core.h>
#include <sound/asoundef.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_auto_parser.h"
#include "hda_jack.h"
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 5ee468d1aefe..ffddcdfe0c66 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -38,11 +38,6 @@
#include <sound/ac97_codec.h>
#include <sound/info.h>
#include <sound/initval.h>
-/* for 440MX workaround */
-#include <asm/pgtable.h>
-#ifdef CONFIG_X86
-#include <asm/set_memory.h>
-#endif
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455");
@@ -374,7 +369,6 @@ struct ichdev {
unsigned int ali_slot; /* ALI DMA slot */
struct ac97_pcm *pcm;
int pcm_open_flag;
- unsigned int page_attr_changed: 1;
unsigned int suspended: 1;
};
@@ -724,25 +718,6 @@ static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ich
iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI);
}
-#ifdef __i386__
-/*
- * Intel 82443MX running a 100MHz processor system bus has a hardware bug,
- * which aborts PCI busmaster for audio transfer. A workaround is to set
- * the pages as non-cached. For details, see the errata in
- * http://download.intel.com/design/chipsets/specupdt/24505108.pdf
- */
-static void fill_nocache(void *buf, int size, int nocache)
-{
- size = (size + PAGE_SIZE - 1) >> PAGE_SHIFT;
- if (nocache)
- set_pages_uc(virt_to_page(buf), size);
- else
- set_pages_wb(virt_to_page(buf), size);
-}
-#else
-#define fill_nocache(buf, size, nocache) do { ; } while (0)
-#endif
-
/*
* Interrupt handler
*/
@@ -850,7 +825,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd
switch (cmd) {
case SNDRV_PCM_TRIGGER_RESUME:
ichdev->suspended = 0;
- /* fallthru */
+ /* fall through */
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
val = ICH_IOCE | ICH_STARTBM;
@@ -858,7 +833,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
ichdev->suspended = 1;
- /* fallthru */
+ /* fall through */
case SNDRV_PCM_TRIGGER_STOP:
val = 0;
break;
@@ -892,7 +867,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd
switch (cmd) {
case SNDRV_PCM_TRIGGER_RESUME:
ichdev->suspended = 0;
- /* fallthru */
+ /* fall through */
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -909,7 +884,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
ichdev->suspended = 1;
- /* fallthru */
+ /* fall through */
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
/* pause */
@@ -938,23 +913,12 @@ static int snd_intel8x0_hw_params(struct snd_pcm_substream *substream,
{
struct intel8x0 *chip = snd_pcm_substream_chip(substream);
struct ichdev *ichdev = get_ichdev(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
int dbl = params_rate(hw_params) > 48000;
int err;
- if (chip->fix_nocache && ichdev->page_attr_changed) {
- fill_nocache(runtime->dma_area, runtime->dma_bytes, 0); /* clear */
- ichdev->page_attr_changed = 0;
- }
err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
if (err < 0)
return err;
- if (chip->fix_nocache) {
- if (runtime->dma_area && ! ichdev->page_attr_changed) {
- fill_nocache(runtime->dma_area, runtime->dma_bytes, 1);
- ichdev->page_attr_changed = 1;
- }
- }
if (ichdev->pcm_open_flag) {
snd_ac97_pcm_close(ichdev->pcm);
ichdev->pcm_open_flag = 0;
@@ -974,17 +938,12 @@ static int snd_intel8x0_hw_params(struct snd_pcm_substream *substream,
static int snd_intel8x0_hw_free(struct snd_pcm_substream *substream)
{
- struct intel8x0 *chip = snd_pcm_substream_chip(substream);
struct ichdev *ichdev = get_ichdev(substream);
if (ichdev->pcm_open_flag) {
snd_ac97_pcm_close(ichdev->pcm);
ichdev->pcm_open_flag = 0;
}
- if (chip->fix_nocache && ichdev->page_attr_changed) {
- fill_nocache(substream->runtime->dma_area, substream->runtime->dma_bytes, 0);
- ichdev->page_attr_changed = 0;
- }
return snd_pcm_lib_free_pages(substream);
}
@@ -1510,6 +1469,9 @@ struct ich_pcm_table {
int ac97_idx;
};
+#define intel8x0_dma_type(chip) \
+ ((chip)->fix_nocache ? SNDRV_DMA_TYPE_DEV_UC : SNDRV_DMA_TYPE_DEV)
+
static int snd_intel8x0_pcm1(struct intel8x0 *chip, int device,
struct ich_pcm_table *rec)
{
@@ -1540,7 +1502,7 @@ static int snd_intel8x0_pcm1(struct intel8x0 *chip, int device,
strcpy(pcm->name, chip->card->shortname);
chip->pcm[device] = pcm;
- snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_pcm_lib_preallocate_pages_for_all(pcm, intel8x0_dma_type(chip),
snd_dma_pci_data(chip->pci),
rec->prealloc_size, rec->prealloc_max_size);
@@ -2629,11 +2591,8 @@ static int snd_intel8x0_free(struct intel8x0 *chip)
__hw_end:
if (chip->irq >= 0)
free_irq(chip->irq, chip);
- if (chip->bdbars.area) {
- if (chip->fix_nocache)
- fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 0);
+ if (chip->bdbars.area)
snd_dma_free_pages(&chip->bdbars);
- }
if (chip->addr)
pci_iounmap(chip->pci, chip->addr);
if (chip->bmaddr)
@@ -2657,17 +2616,6 @@ static int intel8x0_suspend(struct device *dev)
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
for (i = 0; i < chip->pcm_devs; i++)
snd_pcm_suspend_all(chip->pcm[i]);
- /* clear nocache */
- if (chip->fix_nocache) {
- for (i = 0; i < chip->bdbars_count; i++) {
- struct ichdev *ichdev = &chip->ichd[i];
- if (ichdev->substream && ichdev->page_attr_changed) {
- struct snd_pcm_runtime *runtime = ichdev->substream->runtime;
- if (runtime->dma_area)
- fill_nocache(runtime->dma_area, runtime->dma_bytes, 0);
- }
- }
- }
for (i = 0; i < chip->ncodecs; i++)
snd_ac97_suspend(chip->ac97[i]);
if (chip->device_type == DEVICE_INTEL_ICH4)
@@ -2708,25 +2656,9 @@ static int intel8x0_resume(struct device *dev)
ICH_PCM_SPDIF_1011);
}
- /* refill nocache */
- if (chip->fix_nocache)
- fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 1);
-
for (i = 0; i < chip->ncodecs; i++)
snd_ac97_resume(chip->ac97[i]);
- /* refill nocache */
- if (chip->fix_nocache) {
- for (i = 0; i < chip->bdbars_count; i++) {
- struct ichdev *ichdev = &chip->ichd[i];
- if (ichdev->substream && ichdev->page_attr_changed) {
- struct snd_pcm_runtime *runtime = ichdev->substream->runtime;
- if (runtime->dma_area)
- fill_nocache(runtime->dma_area, runtime->dma_bytes, 1);
- }
- }
- }
-
/* resume status */
for (i = 0; i < chip->bdbars_count; i++) {
struct ichdev *ichdev = &chip->ichd[i];
@@ -3057,6 +2989,12 @@ static int snd_intel8x0_create(struct snd_card *card,
chip->inside_vm = snd_intel8x0_inside_vm(pci);
+ /*
+ * Intel 82443MX running a 100MHz processor system bus has a hardware
+ * bug, which aborts PCI busmaster for audio transfer. A workaround
+ * is to set the pages as non-cached. For details, see the errata in
+ * http://download.intel.com/design/chipsets/specupdt/24505108.pdf
+ */
if (pci->vendor == PCI_VENDOR_ID_INTEL &&
pci->device == PCI_DEVICE_ID_INTEL_440MX)
chip->fix_nocache = 1; /* enable workaround */
@@ -3128,7 +3066,7 @@ static int snd_intel8x0_create(struct snd_card *card,
/* allocate buffer descriptor lists */
/* the start of each lists must be aligned to 8 bytes */
- if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ if (snd_dma_alloc_pages(intel8x0_dma_type(chip), snd_dma_pci_data(pci),
chip->bdbars_count * sizeof(u32) * ICH_MAX_FRAGS * 2,
&chip->bdbars) < 0) {
snd_intel8x0_free(chip);
@@ -3137,9 +3075,6 @@ static int snd_intel8x0_create(struct snd_card *card,
}
/* tables must be aligned to 8 bytes here, but the kernel pages
are much bigger, so we don't care (on i386) */
- /* workaround for 440MX */
- if (chip->fix_nocache)
- fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 1);
int_sta_masks = 0;
for (i = 0; i < chip->bdbars_count; i++) {
ichdev = &chip->ichd[i];
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 943a726b1c1b..c84629190cba 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -1171,16 +1171,6 @@ static int snd_intel8x0m_create(struct snd_card *card,
}
port_inited:
- if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED,
- KBUILD_MODNAME, chip)) {
- dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq);
- snd_intel8x0m_free(chip);
- return -EBUSY;
- }
- chip->irq = pci->irq;
- pci_set_master(pci);
- synchronize_irq(chip->irq);
-
/* initialize offsets */
chip->bdbars_count = 2;
tbl = intel_regs;
@@ -1224,11 +1214,21 @@ static int snd_intel8x0m_create(struct snd_card *card,
chip->int_sta_reg = ICH_REG_GLOB_STA;
chip->int_sta_mask = int_sta_masks;
+ pci_set_master(pci);
+
if ((err = snd_intel8x0m_chip_init(chip, 1)) < 0) {
snd_intel8x0m_free(chip);
return err;
}
+ if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED,
+ KBUILD_MODNAME, chip)) {
+ dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq);
+ snd_intel8x0m_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+
if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
snd_intel8x0m_free(chip);
return err;
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index f0906ba416d4..3ac8c71d567c 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -319,7 +319,8 @@ static const struct snd_pcm_hardware snd_rme32_spdif_info = {
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_SYNC_START),
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_SYNC_APPLPTR),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S32_LE),
.rates = (SNDRV_PCM_RATE_32000 |
@@ -346,7 +347,8 @@ static const struct snd_pcm_hardware snd_rme32_adat_info =
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_SYNC_START),
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_SYNC_APPLPTR),
.formats= SNDRV_PCM_FMTBIT_S16_LE,
.rates = (SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000),
@@ -370,7 +372,8 @@ static const struct snd_pcm_hardware snd_rme32_spdif_fd_info = {
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_SYNC_START),
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_SYNC_APPLPTR),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S32_LE),
.rates = (SNDRV_PCM_RATE_32000 |
@@ -397,7 +400,8 @@ static const struct snd_pcm_hardware snd_rme32_adat_fd_info =
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_SYNC_START),
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_SYNC_APPLPTR),
.formats= SNDRV_PCM_FMTBIT_S16_LE,
.rates = (SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000),
@@ -1104,16 +1108,6 @@ snd_rme32_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
snd_pcm_trigger_done(s, substream);
}
- /* prefill playback buffer */
- if (cmd == SNDRV_PCM_TRIGGER_START && rme32->fullduplex_mode) {
- snd_pcm_group_for_each_entry(s, substream) {
- if (s == rme32->playback_substream) {
- s->ops->ack(s);
- break;
- }
- }
- }
-
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
if (rme32->running && ! RME32_ISWORKING(rme32))
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 11b5b5e0e058..679ad0415e3b 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6534,7 +6534,7 @@ static int snd_hdspm_create_alsa_devices(struct snd_card *card,
dev_dbg(card->dev, "Update mixer controls...\n");
hdspm_update_simple_mixer_controls(hdspm);
- dev_dbg(card->dev, "Initializeing complete ???\n");
+ dev_dbg(card->dev, "Initializing complete?\n");
err = snd_card_register(card);
if (err < 0) {
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index 8e3275a96a82..3f813ea5210a 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -42,7 +42,7 @@
#include "../codecs/da7219.h"
#include "../codecs/da7219-aad.h"
-#define CZ_PLAT_CLK 25000000
+#define CZ_PLAT_CLK 48000000
#define DUAL_CHANNEL 2
static struct snd_soc_jack cz_jack;
@@ -75,7 +75,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd)
da7219_dai_clk = clk_get(component->dev, "da7219-dai-clks");
ret = snd_soc_card_jack_new(card, "Headset Jack",
- SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
+ SND_JACK_HEADSET | SND_JACK_LINEOUT |
SND_JACK_BTN_0 | SND_JACK_BTN_1 |
SND_JACK_BTN_2 | SND_JACK_BTN_3,
&cz_jack, NULL, 0);
@@ -133,7 +133,7 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = {
.mask = 0,
};
-static int cz_da7219_startup(struct snd_pcm_substream *substream)
+static int cz_da7219_play_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -150,7 +150,28 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream)
snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
&constraints_rates);
- machine->i2s_instance = I2S_SP_INSTANCE;
+ machine->play_i2s_instance = I2S_SP_INSTANCE;
+ return da7219_clk_enable(substream);
+}
+
+static int cz_da7219_cap_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *card = rtd->card;
+ struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+ /*
+ * On this platform for PCM device we support stereo
+ */
+
+ runtime->hw.channels_max = DUAL_CHANNEL;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_channels);
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_rates);
+
+ machine->cap_i2s_instance = I2S_SP_INSTANCE;
machine->capture_channel = CAP_CHANNEL1;
return da7219_clk_enable(substream);
}
@@ -162,11 +183,22 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream)
static int cz_max_startup(struct snd_pcm_substream *substream)
{
+ struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_card *card = rtd->card;
struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
- machine->i2s_instance = I2S_BT_INSTANCE;
+ /*
+ * On this platform for PCM device we support stereo
+ */
+
+ runtime->hw.channels_max = DUAL_CHANNEL;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_channels);
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_rates);
+
+ machine->play_i2s_instance = I2S_BT_INSTANCE;
return da7219_clk_enable(substream);
}
@@ -177,21 +209,43 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream)
static int cz_dmic0_startup(struct snd_pcm_substream *substream)
{
+ struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_card *card = rtd->card;
struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
- machine->i2s_instance = I2S_BT_INSTANCE;
+ /*
+ * On this platform for PCM device we support stereo
+ */
+
+ runtime->hw.channels_max = DUAL_CHANNEL;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_channels);
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_rates);
+
+ machine->cap_i2s_instance = I2S_BT_INSTANCE;
return da7219_clk_enable(substream);
}
static int cz_dmic1_startup(struct snd_pcm_substream *substream)
{
+ struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_card *card = rtd->card;
struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
- machine->i2s_instance = I2S_SP_INSTANCE;
+ /*
+ * On this platform for PCM device we support stereo
+ */
+
+ runtime->hw.channels_max = DUAL_CHANNEL;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_channels);
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_rates);
+
+ machine->cap_i2s_instance = I2S_SP_INSTANCE;
machine->capture_channel = CAP_CHANNEL0;
return da7219_clk_enable(substream);
}
@@ -201,8 +255,13 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream)
da7219_clk_disable();
}
+static const struct snd_soc_ops cz_da7219_play_ops = {
+ .startup = cz_da7219_play_startup,
+ .shutdown = cz_da7219_shutdown,
+};
+
static const struct snd_soc_ops cz_da7219_cap_ops = {
- .startup = cz_da7219_startup,
+ .startup = cz_da7219_cap_startup,
.shutdown = cz_da7219_shutdown,
};
@@ -233,7 +292,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
| SND_SOC_DAIFMT_CBM_CFM,
.init = cz_da7219_init,
.dpcm_playback = 1,
- .ops = &cz_da7219_cap_ops,
+ .ops = &cz_da7219_play_ops,
},
{
.name = "amd-da7219-cap",
diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c
index 77b265bd0505..cdebab2f8ce5 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -867,8 +867,12 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
if (pinfo) {
- rtd->i2s_instance = pinfo->i2s_instance;
- rtd->capture_channel = pinfo->capture_channel;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ rtd->i2s_instance = pinfo->play_i2s_instance;
+ } else {
+ rtd->i2s_instance = pinfo->cap_i2s_instance;
+ rtd->capture_channel = pinfo->capture_channel;
+ }
}
if (adata->asic_type == CHIP_STONEY) {
val = acp_reg_read(adata->acp_mmio,
@@ -1036,16 +1040,22 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream)
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
period_bytes = frames_to_bytes(runtime, runtime->period_size);
- dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr);
- if (dscr == rtd->dma_dscr_idx_1)
- pos = period_bytes;
- else
- pos = 0;
bytescount = acp_get_byte_count(rtd);
- if (bytescount > rtd->bytescount)
+ if (bytescount >= rtd->bytescount)
bytescount -= rtd->bytescount;
- delay = do_div(bytescount, period_bytes);
- runtime->delay = bytes_to_frames(runtime, delay);
+ if (bytescount < period_bytes) {
+ pos = 0;
+ } else {
+ dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr);
+ if (dscr == rtd->dma_dscr_idx_1)
+ pos = period_bytes;
+ else
+ pos = 0;
+ }
+ if (bytescount > 0) {
+ delay = do_div(bytescount, period_bytes);
+ runtime->delay = bytes_to_frames(runtime, delay);
+ }
} else {
buffersize = frames_to_bytes(runtime, runtime->buffer_size);
bytescount = acp_get_byte_count(rtd);
diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h
index be3963e8f4fa..dbbb1a85638d 100644
--- a/sound/soc/amd/acp.h
+++ b/sound/soc/amd/acp.h
@@ -158,7 +158,8 @@ struct audio_drv_data {
* and dma driver
*/
struct acp_platform_info {
- u16 i2s_instance;
+ u16 play_i2s_instance;
+ u16 cap_i2s_instance;
u16 capture_channel;
};
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index 64b784e96f84..64f86f0b87e5 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -97,4 +97,16 @@ config SND_ATMEL_SOC_I2S
help
Say Y or M if you want to add support for Atmel ASoc driver for boards
using I2S.
+
+config SND_SOC_MIKROE_PROTO
+ tristate "Support for Mikroe-PROTO board"
+ depends on OF
+ depends on SND_SOC_I2C_AND_SPI
+ select SND_SOC_WM8731
+ help
+ Say Y or M if you want to add support for MikroElektronika PROTO Audio
+ Board. This board contains the WM8731 codec, which can be configured
+ using I2C over SDA (MPU Data Input) and SCL (MPU Clock Input) pins.
+ Both playback and capture are supported.
+
endif
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index cd87cb4bcff5..9f41bfa0fea3 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -17,6 +17,7 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
snd-atmel-soc-classd-objs := atmel-classd.o
snd-atmel-soc-pdmic-objs := atmel-pdmic.o
snd-atmel-soc-tse850-pcm5142-objs := tse850-pcm5142.o
+snd-soc-mikroe-proto-objs := mikroe-proto.o
obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
@@ -24,3 +25,4 @@ obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o
obj-$(CONFIG_SND_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o
obj-$(CONFIG_SND_ATMEL_SOC_PDMIC) += snd-atmel-soc-pdmic.o
obj-$(CONFIG_SND_ATMEL_SOC_TSE850_PCM5142) += snd-atmel-soc-tse850-pcm5142.o
+obj-$(CONFIG_SND_SOC_MIKROE_PROTO) += snd-soc-mikroe-proto.o
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index d3b69682d9c2..6291ec7f9dd6 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -1005,11 +1005,11 @@ static int asoc_ssc_init(struct device *dev)
struct ssc_device *ssc = dev_get_drvdata(dev);
int ret;
- ret = snd_soc_register_component(dev, &atmel_ssc_component,
+ ret = devm_snd_soc_register_component(dev, &atmel_ssc_component,
&atmel_ssc_dai, 1);
if (ret) {
dev_err(dev, "Could not register DAI: %d\n", ret);
- goto err;
+ return ret;
}
if (ssc->pdata->use_dma)
@@ -1019,15 +1019,10 @@ static int asoc_ssc_init(struct device *dev)
if (ret) {
dev_err(dev, "Could not register PCM: %d\n", ret);
- goto err_unregister_dai;
+ return ret;
}
return 0;
-
-err_unregister_dai:
- snd_soc_unregister_component(dev);
-err:
- return ret;
}
static void asoc_ssc_exit(struct device *dev)
@@ -1038,8 +1033,6 @@ static void asoc_ssc_exit(struct device *dev)
atmel_pcm_dma_platform_unregister(dev);
else
atmel_pcm_pdc_platform_unregister(dev);
-
- snd_soc_unregister_component(dev);
}
/**
diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c
new file mode 100644
index 000000000000..d47aaa5bf75a
--- /dev/null
+++ b/sound/soc/atmel/mikroe-proto.c
@@ -0,0 +1,165 @@
+/*
+ * ASoC driver for PROTO AudioCODEC (with a WM8731)
+ *
+ * Author: Florian Meier, <koalo@koalo.de>
+ * Copyright 2013
+ *
+ * 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/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "../codecs/wm8731.h"
+
+#define XTAL_RATE 12288000 /* This is fixed on this board */
+
+static int snd_proto_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+ /* Set proto sysclk */
+ int ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
+ XTAL_RATE, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(card->dev, "Failed to set WM8731 SYSCLK: %d\n",
+ ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget snd_proto_widget[] = {
+ SND_SOC_DAPM_MIC("Microphone Jack", NULL),
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route snd_proto_route[] = {
+ /* speaker connected to LHPOUT/RHPOUT */
+ {"Headphone Jack", NULL, "LHPOUT"},
+ {"Headphone Jack", NULL, "RHPOUT"},
+
+ /* mic is connected to Mic Jack, with WM8731 Mic Bias */
+ {"MICIN", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Microphone Jack"},
+};
+
+/* audio machine driver */
+static struct snd_soc_card snd_proto = {
+ .name = "snd_mikroe_proto",
+ .owner = THIS_MODULE,
+ .dapm_widgets = snd_proto_widget,
+ .num_dapm_widgets = ARRAY_SIZE(snd_proto_widget),
+ .dapm_routes = snd_proto_route,
+ .num_dapm_routes = ARRAY_SIZE(snd_proto_route),
+};
+
+static int snd_proto_probe(struct platform_device *pdev)
+{
+ struct snd_soc_dai_link *dai;
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *codec_np, *cpu_np;
+ struct device_node *bitclkmaster = NULL;
+ struct device_node *framemaster = NULL;
+ unsigned int dai_fmt;
+ int ret = 0;
+
+ if (!np) {
+ dev_err(&pdev->dev, "No device node supplied\n");
+ return -EINVAL;
+ }
+
+ snd_proto.dev = &pdev->dev;
+ ret = snd_soc_of_parse_card_name(&snd_proto, "model");
+ if (ret)
+ return ret;
+
+ dai = devm_kzalloc(&pdev->dev, sizeof(*dai), GFP_KERNEL);
+ if (!dai)
+ return -ENOMEM;
+
+ snd_proto.dai_link = dai;
+ snd_proto.num_links = 1;
+
+ dai->name = "WM8731";
+ dai->stream_name = "WM8731 HiFi";
+ dai->codec_dai_name = "wm8731-hifi";
+ dai->init = &snd_proto_init;
+
+ codec_np = of_parse_phandle(np, "audio-codec", 0);
+ if (!codec_np) {
+ dev_err(&pdev->dev, "audio-codec node missing\n");
+ return -EINVAL;
+ }
+ dai->codec_of_node = codec_np;
+
+ cpu_np = of_parse_phandle(np, "i2s-controller", 0);
+ if (!cpu_np) {
+ dev_err(&pdev->dev, "i2s-controller missing\n");
+ return -EINVAL;
+ }
+ dai->cpu_of_node = cpu_np;
+ dai->platform_of_node = cpu_np;
+
+ dai_fmt = snd_soc_of_parse_daifmt(np, NULL,
+ &bitclkmaster, &framemaster);
+ if (bitclkmaster != framemaster) {
+ dev_err(&pdev->dev, "Must be the same bitclock and frame master\n");
+ return -EINVAL;
+ }
+ if (bitclkmaster) {
+ dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+ if (codec_np == bitclkmaster)
+ dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+ else
+ dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+ }
+ of_node_put(bitclkmaster);
+ of_node_put(framemaster);
+ dai->dai_fmt = dai_fmt;
+
+ of_node_put(codec_np);
+ of_node_put(cpu_np);
+
+ ret = snd_soc_register_card(&snd_proto);
+ if (ret && ret != -EPROBE_DEFER)
+ dev_err(&pdev->dev,
+ "snd_soc_register_card() failed: %d\n", ret);
+
+ return ret;
+}
+
+static int snd_proto_remove(struct platform_device *pdev)
+{
+ return snd_soc_unregister_card(&snd_proto);
+}
+
+static const struct of_device_id snd_proto_of_match[] = {
+ { .compatible = "mikroe,mikroe-proto", },
+ {},
+};
+MODULE_DEVICE_TABLE(of, snd_proto_of_match);
+
+static struct platform_driver snd_proto_driver = {
+ .driver = {
+ .name = "snd-mikroe-proto",
+ .of_match_table = snd_proto_of_match,
+ },
+ .probe = snd_proto_probe,
+ .remove = snd_proto_remove,
+};
+
+module_platform_driver(snd_proto_driver);
+
+MODULE_AUTHOR("Florian Meier");
+MODULE_DESCRIPTION("ASoC Driver for PROTO board (WM8731)");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c
index 3a1393283156..214adcad5419 100644
--- a/sound/soc/atmel/tse850-pcm5142.c
+++ b/sound/soc/atmel/tse850-pcm5142.c
@@ -1,44 +1,38 @@
-/*
- * TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec
- *
- * Copyright (C) 2016 Axentia Technologies AB
- *
- * Author: Peter Rosin <peda@axentia.se>
- *
- * 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.
- */
-
-/*
- * loop1 relays
- * IN1 +---o +------------+ o---+ OUT1
- * \ /
- * + +
- * | / |
- * +--o +--. |
- * | add | |
- * | V |
- * | .---. |
- * DAC +----------->|Sum|---+
- * | '---' |
- * | |
- * + +
- *
- * IN2 +---o--+------------+--o---+ OUT2
- * loop2 relays
- *
- * The 'loop1' gpio pin controlls two relays, which are either in loop
- * position, meaning that input and output are directly connected, or
- * they are in mixer position, meaning that the signal is passed through
- * the 'Sum' mixer. Similarly for 'loop2'.
- *
- * In the above, the 'loop1' relays are inactive, thus feeding IN1 to the
- * mixer (if 'add' is active) and feeding the mixer output to OUT1. The
- * 'loop2' relays are active, short-cutting the TSE-850 from channel 2.
- * IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name
- * of the (filtered) output from the PCM5142 codec.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec
+//
+// Copyright (C) 2016 Axentia Technologies AB
+//
+// Author: Peter Rosin <peda@axentia.se>
+//
+// loop1 relays
+// IN1 +---o +------------+ o---+ OUT1
+// \ /
+// + +
+// | / |
+// +--o +--. |
+// | add | |
+// | V |
+// | .---. |
+// DAC +----------->|Sum|---+
+// | '---' |
+// | |
+// + +
+//
+// IN2 +---o--+------------+--o---+ OUT2
+// loop2 relays
+//
+// The 'loop1' gpio pin controlls two relays, which are either in loop
+// position, meaning that input and output are directly connected, or
+// they are in mixer position, meaning that the signal is passed through
+// the 'Sum' mixer. Similarly for 'loop2'.
+//
+// In the above, the 'loop1' relays are inactive, thus feeding IN1 to the
+// mixer (if 'add' is active) and feeding the mixer output to OUT1. The
+// 'loop2' relays are active, short-cutting the TSE-850 from channel 2.
+// IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name
+// of the (filtered) output from the PCM5142 codec.
#include <linux/clk.h>
#include <linux/gpio.h>
@@ -452,4 +446,4 @@ module_platform_driver(tse850_driver);
/* Module information */
MODULE_AUTHOR("Peter Rosin <peda@axentia.se>");
MODULE_DESCRIPTION("ALSA SoC driver for TSE-850 with PCM5142 codec");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c
index b733f1446353..b7c358b48d8d 100644
--- a/sound/soc/bcm/cygnus-ssp.c
+++ b/sound/soc/bcm/cygnus-ssp.c
@@ -1334,7 +1334,7 @@ static int cygnus_ssp_probe(struct platform_device *pdev)
cygaud->active_ports = 0;
dev_dbg(dev, "Registering %d DAIs\n", active_port_count);
- err = snd_soc_register_component(dev, &cygnus_ssp_component,
+ err = devm_snd_soc_register_component(dev, &cygnus_ssp_component,
cygnus_ssp_dai, active_port_count);
if (err) {
dev_err(dev, "snd_soc_register_dai failed\n");
@@ -1345,32 +1345,27 @@ static int cygnus_ssp_probe(struct platform_device *pdev)
if (cygaud->irq_num <= 0) {
dev_err(dev, "platform_get_irq failed\n");
err = cygaud->irq_num;
- goto err_irq;
+ return err;
}
err = audio_clk_init(pdev, cygaud);
if (err) {
dev_err(dev, "audio clock initialization failed\n");
- goto err_irq;
+ return err;
}
err = cygnus_soc_platform_register(dev, cygaud);
if (err) {
dev_err(dev, "platform reg error %d\n", err);
- goto err_irq;
+ return err;
}
return 0;
-
-err_irq:
- snd_soc_unregister_component(dev);
- return err;
}
static int cygnus_ssp_remove(struct platform_device *pdev)
{
cygnus_soc_platform_unregister(&pdev->dev);
- snd_soc_unregister_component(&pdev->dev);
return 0;
}
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index efb095dbcd71..9cc4f1848c9b 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -82,6 +82,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_ES7241
select SND_SOC_GTM601
select SND_SOC_HDAC_HDMI
+ select SND_SOC_HDAC_HDA
select SND_SOC_ICS43432
select SND_SOC_INNO_RK3036
select SND_SOC_ISABELLE if I2C
@@ -109,6 +110,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_MT6351 if MTK_PMIC_WRAP
select SND_SOC_NAU8540 if I2C
select SND_SOC_NAU8810 if I2C
+ select SND_SOC_NAU8822 if I2C
select SND_SOC_NAU8824 if I2C
select SND_SOC_NAU8825 if I2C
select SND_SOC_HDMI_CODEC
@@ -119,6 +121,8 @@ config SND_SOC_ALL_CODECS
select SND_SOC_PCM186X_I2C if I2C
select SND_SOC_PCM186X_SPI if SPI_MASTER
select SND_SOC_PCM3008
+ select SND_SOC_PCM3060_I2C if I2C
+ select SND_SOC_PCM3060_SPI if SPI_MASTER
select SND_SOC_PCM3168A_I2C if I2C
select SND_SOC_PCM3168A_SPI if SPI_MASTER
select SND_SOC_PCM5102A
@@ -575,7 +579,11 @@ config SND_SOC_DA9055
tristate
config SND_SOC_DMIC
- tristate
+ tristate "Generic Digital Microphone CODEC"
+ depends on GPIOLIB
+ help
+ Enable support for the Generic Digital Microphone CODEC.
+ Select this if your sound card has DMICs.
config SND_SOC_HDMI_CODEC
tristate
@@ -615,6 +623,10 @@ config SND_SOC_HDAC_HDMI
select SND_PCM_ELD
select HDMI
+config SND_SOC_HDAC_HDA
+ tristate
+ select SND_HDA
+
config SND_SOC_ICS43432
tristate
@@ -629,7 +641,8 @@ config SND_SOC_LM49453
tristate
config SND_SOC_MAX98088
- tristate
+ tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec"
+ depends on I2C
config SND_SOC_MAX98090
tristate
@@ -732,6 +745,21 @@ config SND_SOC_PCM186X_SPI
config SND_SOC_PCM3008
tristate
+config SND_SOC_PCM3060
+ tristate
+
+config SND_SOC_PCM3060_I2C
+ tristate "Texas Instruments PCM3060 CODEC - I2C"
+ depends on I2C
+ select SND_SOC_PCM3060
+ select REGMAP_I2C
+
+config SND_SOC_PCM3060_SPI
+ tristate "Texas Instruments PCM3060 CODEC - SPI"
+ depends on SPI_MASTER
+ select SND_SOC_PCM3060
+ select REGMAP_SPI
+
config SND_SOC_PCM3168A
tristate
@@ -1299,6 +1327,10 @@ config SND_SOC_NAU8810
tristate "Nuvoton Technology Corporation NAU88C10 CODEC"
depends on I2C
+config SND_SOC_NAU8822
+ tristate "Nuvoton Technology Corporation NAU88C22 CODEC"
+ depends on I2C
+
config SND_SOC_NAU8824
tristate "Nuvoton Technology Corporation NAU88L24 CODEC"
depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 7ae7c85e8219..8ffab8c8dbfa 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -78,6 +78,7 @@ snd-soc-es8328-i2c-objs := es8328-i2c.o
snd-soc-es8328-spi-objs := es8328-spi.o
snd-soc-gtm601-objs := gtm601.o
snd-soc-hdac-hdmi-objs := hdac_hdmi.o
+snd-soc-hdac-hda-objs := hdac_hda.o
snd-soc-ics43432-objs := ics43432.o
snd-soc-inno-rk3036-objs := inno_rk3036.o
snd-soc-isabelle-objs := isabelle.o
@@ -106,6 +107,7 @@ snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o
snd-soc-mt6351-objs := mt6351.o
snd-soc-nau8540-objs := nau8540.o
snd-soc-nau8810-objs := nau8810.o
+snd-soc-nau8822-objs := nau8822.o
snd-soc-nau8824-objs := nau8824.o
snd-soc-nau8825-objs := nau8825.o
snd-soc-hdmi-codec-objs := hdmi-codec.o
@@ -119,6 +121,9 @@ snd-soc-pcm186x-objs := pcm186x.o
snd-soc-pcm186x-i2c-objs := pcm186x-i2c.o
snd-soc-pcm186x-spi-objs := pcm186x-spi.o
snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-pcm3060-objs := pcm3060.o
+snd-soc-pcm3060-i2c-objs := pcm3060-i2c.o
+snd-soc-pcm3060-spi-objs := pcm3060-spi.o
snd-soc-pcm3168a-objs := pcm3168a.o
snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o
snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o
@@ -338,6 +343,7 @@ obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o
obj-$(CONFIG_SND_SOC_GTM601) += snd-soc-gtm601.o
obj-$(CONFIG_SND_SOC_HDAC_HDMI) += snd-soc-hdac-hdmi.o
+obj-$(CONFIG_SND_SOC_HDAC_HDA) += snd-soc-hdac-hda.o
obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o
obj-$(CONFIG_SND_SOC_INNO_RK3036) += snd-soc-inno-rk3036.o
obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o
@@ -366,6 +372,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o
obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o
obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o
obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o
+obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o
obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o
obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o
obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o
@@ -379,6 +386,9 @@ obj-$(CONFIG_SND_SOC_PCM186X) += snd-soc-pcm186x.o
obj-$(CONFIG_SND_SOC_PCM186X_I2C) += snd-soc-pcm186x-i2c.o
obj-$(CONFIG_SND_SOC_PCM186X_SPI) += snd-soc-pcm186x-spi.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_PCM3060) += snd-soc-pcm3060.o
+obj-$(CONFIG_SND_SOC_PCM3060_I2C) += snd-soc-pcm3060-i2c.o
+obj-$(CONFIG_SND_SOC_PCM3060_SPI) += snd-soc-pcm3060-spi.o
obj-$(CONFIG_SND_SOC_PCM3168A) += snd-soc-pcm3168a.o
obj-$(CONFIG_SND_SOC_PCM3168A_I2C) += snd-soc-pcm3168a-i2c.o
obj-$(CONFIG_SND_SOC_PCM3168A_SPI) += snd-soc-pcm3168a-spi.o
diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c
index be136e981653..bef3e9e74c26 100644
--- a/sound/soc/codecs/adau1761.c
+++ b/sound/soc/codecs/adau1761.c
@@ -518,7 +518,8 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_component *component)
ARRAY_SIZE(adau1761_jack_detect_controls));
if (ret)
return ret;
- case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */
+ /* fall through */
+ case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE:
ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes,
ARRAY_SIZE(adau1761_no_dmic_routes));
if (ret)
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 57169b8ff14e..3959e6ad113d 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -21,11 +21,18 @@
#include <linux/i2c.h>
#include <linux/spi/spi.h>
#include <linux/regmap.h>
+#include <asm/unaligned.h>
#include "sigmadsp.h"
#include "adau17x1.h"
#include "adau-utils.h"
+#define ADAU17X1_SAFELOAD_TARGET_ADDRESS 0x0006
+#define ADAU17X1_SAFELOAD_TRIGGER 0x0007
+#define ADAU17X1_SAFELOAD_DATA 0x0001
+#define ADAU17X1_SAFELOAD_DATA_SIZE 20
+#define ADAU17X1_WORD_SIZE 4
+
static const char * const adau17x1_capture_mixer_boost_text[] = {
"Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3",
};
@@ -60,6 +67,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = {
SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum),
};
+static int adau17x1_setup_firmware(struct snd_soc_component *component,
+ unsigned int rate);
+
static int adau17x1_pll_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -313,7 +323,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = {
{ "Capture", NULL, "Right Decimator" },
};
-bool adau17x1_has_dsp(struct adau *adau)
+static bool adau17x1_has_dsp(struct adau *adau)
{
switch (adau->type) {
case ADAU1761:
@@ -324,7 +334,17 @@ bool adau17x1_has_dsp(struct adau *adau)
return false;
}
}
-EXPORT_SYMBOL_GPL(adau17x1_has_dsp);
+
+static bool adau17x1_has_safeload(struct adau *adau)
+{
+ switch (adau->type) {
+ case ADAU1761:
+ case ADAU1781:
+ return true;
+ default:
+ return false;
+ }
+}
static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id,
int source, unsigned int freq_in, unsigned int freq_out)
@@ -836,7 +856,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg)
}
EXPORT_SYMBOL_GPL(adau17x1_volatile_register);
-int adau17x1_setup_firmware(struct snd_soc_component *component,
+static int adau17x1_setup_firmware(struct snd_soc_component *component,
unsigned int rate)
{
int ret;
@@ -880,7 +900,6 @@ err:
return ret;
}
-EXPORT_SYMBOL_GPL(adau17x1_setup_firmware);
int adau17x1_add_widgets(struct snd_soc_component *component)
{
@@ -957,6 +976,56 @@ int adau17x1_resume(struct snd_soc_component *component)
}
EXPORT_SYMBOL_GPL(adau17x1_resume);
+static int adau17x1_safeload(struct sigmadsp *sigmadsp, unsigned int addr,
+ const uint8_t bytes[], size_t len)
+{
+ uint8_t buf[ADAU17X1_WORD_SIZE];
+ uint8_t data[ADAU17X1_SAFELOAD_DATA_SIZE];
+ unsigned int addr_offset;
+ unsigned int nbr_words;
+ int ret;
+
+ /* write data to safeload addresses. Check if len is not a multiple of
+ * 4 bytes, if so we need to zero pad.
+ */
+ nbr_words = len / ADAU17X1_WORD_SIZE;
+ if ((len - nbr_words * ADAU17X1_WORD_SIZE) == 0) {
+ ret = regmap_raw_write(sigmadsp->control_data,
+ ADAU17X1_SAFELOAD_DATA, bytes, len);
+ } else {
+ nbr_words++;
+ memset(data, 0, ADAU17X1_SAFELOAD_DATA_SIZE);
+ memcpy(data, bytes, len);
+ ret = regmap_raw_write(sigmadsp->control_data,
+ ADAU17X1_SAFELOAD_DATA, data,
+ nbr_words * ADAU17X1_WORD_SIZE);
+ }
+
+ if (ret < 0)
+ return ret;
+
+ /* Write target address, target address is offset by 1 */
+ addr_offset = addr - 1;
+ put_unaligned_be32(addr_offset, buf);
+ ret = regmap_raw_write(sigmadsp->control_data,
+ ADAU17X1_SAFELOAD_TARGET_ADDRESS, buf, ADAU17X1_WORD_SIZE);
+ if (ret < 0)
+ return ret;
+
+ /* write nbr of words to trigger address */
+ put_unaligned_be32(nbr_words, buf);
+ ret = regmap_raw_write(sigmadsp->control_data,
+ ADAU17X1_SAFELOAD_TRIGGER, buf, ADAU17X1_WORD_SIZE);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static const struct sigmadsp_ops adau17x1_sigmadsp_ops = {
+ .safeload = adau17x1_safeload,
+};
+
int adau17x1_probe(struct device *dev, struct regmap *regmap,
enum adau17x1_type type, void (*switch_mode)(struct device *dev),
const char *firmware_name)
@@ -1002,8 +1071,13 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap,
dev_set_drvdata(dev, adau);
if (firmware_name) {
- adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, NULL,
- firmware_name);
+ if (adau17x1_has_safeload(adau)) {
+ adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap,
+ &adau17x1_sigmadsp_ops, firmware_name);
+ } else {
+ adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap,
+ NULL, firmware_name);
+ }
if (IS_ERR(adau->sigmadsp)) {
dev_warn(dev, "Could not find firmware file: %ld\n",
PTR_ERR(adau->sigmadsp));
diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h
index e6fe87beec07..98a3b6f5bc96 100644
--- a/sound/soc/codecs/adau17x1.h
+++ b/sound/soc/codecs/adau17x1.h
@@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component);
extern const struct snd_soc_dai_ops adau17x1_dai_ops;
-int adau17x1_setup_firmware(struct snd_soc_component *component,
- unsigned int rate);
-bool adau17x1_has_dsp(struct adau *adau);
-
#define ADAU17X1_CLOCK_CONTROL 0x4000
#define ADAU17X1_PLL_CONTROL 0x4002
#define ADAU17X1_REC_POWER_MGMT 0x4009
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index 407554175282..ab27d2b94d02 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -154,11 +154,11 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = {
SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1,
6, 1, 0),
SOC_ENUM("C Data Access", cam_mode_enum),
+ SOC_SINGLE("SPDIF Switch", CS4265_SPDIF_CTL2, 5, 1, 1),
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", CS4265_SPDIF_CTL2,
- 0, 1, 0),
+ SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, 0, 1, 0),
SOC_ENUM("Mono Channel Select", spdif_mono_select_enum),
SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24),
};
@@ -221,10 +221,11 @@ static const struct snd_soc_dapm_route cs4265_audio_map[] = {
{"LINEOUTR", NULL, "DAC"},
{"SPDIFOUT", NULL, "SPDIF"},
+ {"Pre-amp MIC", NULL, "MICL"},
+ {"Pre-amp MIC", NULL, "MICR"},
+ {"ADC Mux", "MIC", "Pre-amp MIC"},
{"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"},
@@ -496,7 +497,8 @@ static int cs4265_set_bias_level(struct snd_soc_component *component,
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)
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE)
static const struct snd_soc_dai_ops cs4265_ops = {
.hw_params = cs4265_pcm_hw_params,
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 5080d7a3c279..fd2bd74024c1 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -21,6 +21,7 @@
* - master mode *NOT* supported
*/
+#include <linux/clk.h>
#include <linux/module.h>
#include <linux/slab.h>
#include <sound/core.h>
@@ -41,6 +42,7 @@ enum master_slave_mode {
struct cs42l51_private {
unsigned int mclk;
+ struct clk *mclk_handle;
unsigned int audio_mode; /* The mode (I2S or left-justified) */
enum master_slave_mode func;
};
@@ -237,6 +239,10 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = {
&cs42l51_adcr_mux_controls),
};
+static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = {
+ SND_SOC_DAPM_CLOCK_SUPPLY("MCLK")
+};
+
static const struct snd_soc_dapm_route cs42l51_routes[] = {
{"HPL", NULL, "Left DAC"},
{"HPR", NULL, "Right DAC"},
@@ -487,6 +493,14 @@ static struct snd_soc_dai_driver cs42l51_dai = {
static int cs42l51_component_probe(struct snd_soc_component *component)
{
int ret, reg;
+ struct snd_soc_dapm_context *dapm;
+ struct cs42l51_private *cs42l51;
+
+ cs42l51 = snd_soc_component_get_drvdata(component);
+ dapm = snd_soc_component_get_dapm(component);
+
+ if (cs42l51->mclk_handle)
+ snd_soc_dapm_new_controls(dapm, cs42l51_dapm_mclk_widgets, 1);
/*
* DAC configuration
@@ -540,6 +554,13 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap)
dev_set_drvdata(dev, cs42l51);
+ cs42l51->mclk_handle = devm_clk_get(dev, "MCLK");
+ if (IS_ERR(cs42l51->mclk_handle)) {
+ if (PTR_ERR(cs42l51->mclk_handle) != -ENOENT)
+ return PTR_ERR(cs42l51->mclk_handle);
+ cs42l51->mclk_handle = NULL;
+ }
+
/* Verify that we have a CS42L51 */
ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val);
if (ret < 0) {
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index 8c4926df9286..71322e0410ee 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -148,6 +148,7 @@ static const struct of_device_id dmic_dev_match[] = {
{.compatible = "dmic-codec"},
{}
};
+MODULE_DEVICE_TABLE(of, dmic_dev_match);
static struct platform_driver dmic_driver = {
.driver = {
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 4b5827dc23aa..04a3aa770722 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -566,14 +566,14 @@ static int es8328_set_sysclk(struct snd_soc_dai *codec_dai,
break;
case 22579200:
mclkdiv2 = 1;
- /* fallthru */
+ /* fall through */
case 11289600:
es8328->sysclk_constraints = &constraints_11289;
es8328->mclk_ratios = ratios_11289;
break;
case 24576000:
mclkdiv2 = 1;
- /* fallthru */
+ /* fall through */
case 12288000:
es8328->sysclk_constraints = &constraints_12288;
es8328->mclk_ratios = ratios_12288;
diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c
new file mode 100644
index 000000000000..2aaa83028e55
--- /dev/null
+++ b/sound/soc/codecs/hdac_hda.c
@@ -0,0 +1,483 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2015-18 Intel Corporation.
+
+/*
+ * hdac_hda.c - ASoC extensions to reuse the legacy HDA codec drivers
+ * with ASoC platform drivers. These APIs are called by the legacy HDA
+ * codec drivers using hdac_ext_bus_ops ops.
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/pm_runtime.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/hdaudio_ext.h>
+#include <sound/hda_codec.h>
+#include <sound/hda_register.h>
+#include "hdac_hda.h"
+
+#define HDAC_ANALOG_DAI_ID 0
+#define HDAC_DIGITAL_DAI_ID 1
+#define HDAC_ALT_ANALOG_DAI_ID 2
+
+#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_U16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_U24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE | \
+ SNDRV_PCM_FMTBIT_U32_LE | \
+ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
+
+static int hdac_hda_dai_open(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai);
+static void hdac_hda_dai_close(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai);
+static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai);
+static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai);
+static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask,
+ int slots, int slot_width);
+static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt,
+ struct snd_soc_dai *dai);
+
+static struct snd_soc_dai_ops hdac_hda_dai_ops = {
+ .startup = hdac_hda_dai_open,
+ .shutdown = hdac_hda_dai_close,
+ .prepare = hdac_hda_dai_prepare,
+ .hw_free = hdac_hda_dai_hw_free,
+ .set_tdm_slot = hdac_hda_dai_set_tdm_slot,
+};
+
+static struct snd_soc_dai_driver hdac_hda_dais[] = {
+{
+ .id = HDAC_ANALOG_DAI_ID,
+ .name = "Analog Codec DAI",
+ .ops = &hdac_hda_dai_ops,
+ .playback = {
+ .stream_name = "Analog Codec Playback",
+ .channels_min = 1,
+ .channels_max = 16,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = STUB_FORMATS,
+ .sig_bits = 24,
+ },
+ .capture = {
+ .stream_name = "Analog Codec Capture",
+ .channels_min = 1,
+ .channels_max = 16,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = STUB_FORMATS,
+ .sig_bits = 24,
+ },
+},
+{
+ .id = HDAC_DIGITAL_DAI_ID,
+ .name = "Digital Codec DAI",
+ .ops = &hdac_hda_dai_ops,
+ .playback = {
+ .stream_name = "Digital Codec Playback",
+ .channels_min = 1,
+ .channels_max = 16,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = STUB_FORMATS,
+ .sig_bits = 24,
+ },
+ .capture = {
+ .stream_name = "Digital Codec Capture",
+ .channels_min = 1,
+ .channels_max = 16,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = STUB_FORMATS,
+ .sig_bits = 24,
+ },
+},
+{
+ .id = HDAC_ALT_ANALOG_DAI_ID,
+ .name = "Alt Analog Codec DAI",
+ .ops = &hdac_hda_dai_ops,
+ .playback = {
+ .stream_name = "Alt Analog Codec Playback",
+ .channels_min = 1,
+ .channels_max = 16,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = STUB_FORMATS,
+ .sig_bits = 24,
+ },
+ .capture = {
+ .stream_name = "Alt Analog Codec Capture",
+ .channels_min = 1,
+ .channels_max = 16,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = STUB_FORMATS,
+ .sig_bits = 24,
+ },
+}
+
+};
+
+static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask,
+ int slots, int slot_width)
+{
+ struct snd_soc_component *component = dai->component;
+ struct hdac_hda_priv *hda_pvt;
+ struct hdac_hda_pcm *pcm;
+
+ hda_pvt = snd_soc_component_get_drvdata(component);
+ pcm = &hda_pvt->pcm[dai->id];
+ if (tx_mask)
+ pcm[dai->id].stream_tag[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask;
+ else
+ pcm[dai->id].stream_tag[SNDRV_PCM_STREAM_CAPTURE] = rx_mask;
+
+ return 0;
+}
+
+static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct hdac_hda_priv *hda_pvt;
+ struct hda_pcm_stream *hda_stream;
+ struct hda_pcm *pcm;
+
+ hda_pvt = snd_soc_component_get_drvdata(component);
+ pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+ if (!pcm)
+ return -EINVAL;
+
+ hda_stream = &pcm->stream[substream->stream];
+ snd_hda_codec_cleanup(&hda_pvt->codec, hda_stream, substream);
+
+ return 0;
+}
+
+static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct hdac_hda_priv *hda_pvt;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct hdac_device *hdev;
+ struct hda_pcm_stream *hda_stream;
+ unsigned int format_val;
+ struct hda_pcm *pcm;
+ unsigned int stream;
+ int ret = 0;
+
+ hda_pvt = snd_soc_component_get_drvdata(component);
+ hdev = &hda_pvt->codec.core;
+ pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+ if (!pcm)
+ return -EINVAL;
+
+ hda_stream = &pcm->stream[substream->stream];
+
+ format_val = snd_hdac_calc_stream_format(runtime->rate,
+ runtime->channels,
+ runtime->format,
+ hda_stream->maxbps,
+ 0);
+ if (!format_val) {
+ dev_err(&hdev->dev,
+ "invalid format_val, rate=%d, ch=%d, format=%d\n",
+ runtime->rate, runtime->channels, runtime->format);
+ return -EINVAL;
+ }
+
+ stream = hda_pvt->pcm[dai->id].stream_tag[substream->stream];
+
+ ret = snd_hda_codec_prepare(&hda_pvt->codec, hda_stream,
+ stream, format_val, substream);
+ if (ret < 0)
+ dev_err(&hdev->dev, "codec prepare failed %d\n", ret);
+
+ return ret;
+}
+
+static int hdac_hda_dai_open(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct hdac_hda_priv *hda_pvt;
+ struct hda_pcm_stream *hda_stream;
+ struct hda_pcm *pcm;
+ int ret;
+
+ hda_pvt = snd_soc_component_get_drvdata(component);
+ pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+ if (!pcm)
+ return -EINVAL;
+
+ snd_hda_codec_pcm_get(pcm);
+
+ hda_stream = &pcm->stream[substream->stream];
+
+ ret = hda_stream->ops.open(hda_stream, &hda_pvt->codec, substream);
+ if (ret < 0)
+ snd_hda_codec_pcm_put(pcm);
+
+ return ret;
+}
+
+static void hdac_hda_dai_close(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct hdac_hda_priv *hda_pvt;
+ struct hda_pcm_stream *hda_stream;
+ struct hda_pcm *pcm;
+
+ hda_pvt = snd_soc_component_get_drvdata(component);
+ pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+ if (!pcm)
+ return;
+
+ hda_stream = &pcm->stream[substream->stream];
+
+ hda_stream->ops.close(hda_stream, &hda_pvt->codec, substream);
+
+ snd_hda_codec_pcm_put(pcm);
+}
+
+static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt,
+ struct snd_soc_dai *dai)
+{
+ struct hda_codec *hcodec = &hda_pvt->codec;
+ struct hda_pcm *cpcm;
+ const char *pcm_name;
+
+ switch (dai->id) {
+ case HDAC_ANALOG_DAI_ID:
+ pcm_name = "Analog";
+ break;
+ case HDAC_DIGITAL_DAI_ID:
+ pcm_name = "Digital";
+ break;
+ case HDAC_ALT_ANALOG_DAI_ID:
+ pcm_name = "Alt Analog";
+ break;
+ default:
+ dev_err(&hcodec->core.dev, "invalid dai id %d\n", dai->id);
+ return NULL;
+ }
+
+ list_for_each_entry(cpcm, &hcodec->pcm_list_head, list) {
+ if (strpbrk(cpcm->name, pcm_name))
+ return cpcm;
+ }
+
+ dev_err(&hcodec->core.dev, "didn't find PCM for DAI %s\n", dai->name);
+ return NULL;
+}
+
+static int hdac_hda_codec_probe(struct snd_soc_component *component)
+{
+ struct hdac_hda_priv *hda_pvt =
+ snd_soc_component_get_drvdata(component);
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_component_get_dapm(component);
+ struct hdac_device *hdev = &hda_pvt->codec.core;
+ struct hda_codec *hcodec = &hda_pvt->codec;
+ struct hdac_ext_link *hlink;
+ hda_codec_patch_t patch;
+ int ret;
+
+ hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+ if (!hlink) {
+ dev_err(&hdev->dev, "hdac link not found\n");
+ return -EIO;
+ }
+
+ snd_hdac_ext_bus_link_get(hdev->bus, hlink);
+
+ ret = snd_hda_codec_device_new(hcodec->bus, component->card->snd_card,
+ hdev->addr, hcodec);
+ if (ret < 0) {
+ dev_err(&hdev->dev, "failed to create hda codec %d\n", ret);
+ goto error_no_pm;
+ }
+
+ /*
+ * snd_hda_codec_device_new decrements the usage count so call get pm
+ * else the device will be powered off
+ */
+ pm_runtime_get_noresume(&hdev->dev);
+
+ hcodec->bus->card = dapm->card->snd_card;
+
+ ret = snd_hda_codec_set_name(hcodec, hcodec->preset->name);
+ if (ret < 0) {
+ dev_err(&hdev->dev, "name failed %s\n", hcodec->preset->name);
+ goto error;
+ }
+
+ ret = snd_hdac_regmap_init(&hcodec->core);
+ if (ret < 0) {
+ dev_err(&hdev->dev, "regmap init failed\n");
+ goto error;
+ }
+
+ patch = (hda_codec_patch_t)hcodec->preset->driver_data;
+ if (patch) {
+ ret = patch(hcodec);
+ if (ret < 0) {
+ dev_err(&hdev->dev, "patch failed %d\n", ret);
+ goto error;
+ }
+ } else {
+ dev_dbg(&hdev->dev, "no patch file found\n");
+ }
+
+ ret = snd_hda_codec_parse_pcms(hcodec);
+ if (ret < 0) {
+ dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret);
+ goto error;
+ }
+
+ ret = snd_hda_codec_build_controls(hcodec);
+ if (ret < 0) {
+ dev_err(&hdev->dev, "unable to create controls %d\n", ret);
+ goto error;
+ }
+
+ hcodec->core.lazy_cache = true;
+
+ /*
+ * hdac_device core already sets the state to active and calls
+ * get_noresume. So enable runtime and set the device to suspend.
+ * pm_runtime_enable is also called during codec registeration
+ */
+ pm_runtime_put(&hdev->dev);
+ pm_runtime_suspend(&hdev->dev);
+
+ return 0;
+
+error:
+ pm_runtime_put(&hdev->dev);
+error_no_pm:
+ snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+ return ret;
+}
+
+static void hdac_hda_codec_remove(struct snd_soc_component *component)
+{
+ struct hdac_hda_priv *hda_pvt =
+ snd_soc_component_get_drvdata(component);
+ struct hdac_device *hdev = &hda_pvt->codec.core;
+ struct hdac_ext_link *hlink = NULL;
+
+ hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+ if (!hlink) {
+ dev_err(&hdev->dev, "hdac link not found\n");
+ return;
+ }
+
+ snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+ pm_runtime_disable(&hdev->dev);
+}
+
+static const struct snd_soc_dapm_route hdac_hda_dapm_routes[] = {
+ {"AIF1TX", NULL, "Codec Input Pin1"},
+ {"AIF2TX", NULL, "Codec Input Pin2"},
+ {"AIF3TX", NULL, "Codec Input Pin3"},
+
+ {"Codec Output Pin1", NULL, "AIF1RX"},
+ {"Codec Output Pin2", NULL, "AIF2RX"},
+ {"Codec Output Pin3", NULL, "AIF3RX"},
+};
+
+static const struct snd_soc_dapm_widget hdac_hda_dapm_widgets[] = {
+ /* Audio Interface */
+ SND_SOC_DAPM_AIF_IN("AIF1RX", "Analog Codec Playback", 0,
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIF2RX", "Digital Codec Playback", 0,
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIF3RX", "Alt Analog Codec Playback", 0,
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AIF1TX", "Analog Codec Capture", 0,
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AIF2TX", "Digital Codec Capture", 0,
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AIF3TX", "Alt Analog Codec Capture", 0,
+ SND_SOC_NOPM, 0, 0),
+
+ /* Input Pins */
+ SND_SOC_DAPM_INPUT("Codec Input Pin1"),
+ SND_SOC_DAPM_INPUT("Codec Input Pin2"),
+ SND_SOC_DAPM_INPUT("Codec Input Pin3"),
+
+ /* Output Pins */
+ SND_SOC_DAPM_OUTPUT("Codec Output Pin1"),
+ SND_SOC_DAPM_OUTPUT("Codec Output Pin2"),
+ SND_SOC_DAPM_OUTPUT("Codec Output Pin3"),
+};
+
+static const struct snd_soc_component_driver hdac_hda_codec = {
+ .probe = hdac_hda_codec_probe,
+ .remove = hdac_hda_codec_remove,
+ .idle_bias_on = false,
+ .dapm_widgets = hdac_hda_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(hdac_hda_dapm_widgets),
+ .dapm_routes = hdac_hda_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(hdac_hda_dapm_routes),
+};
+
+static int hdac_hda_dev_probe(struct hdac_device *hdev)
+{
+ struct hdac_ext_link *hlink;
+ struct hdac_hda_priv *hda_pvt;
+ int ret;
+
+ /* hold the ref while we probe */
+ hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+ if (!hlink) {
+ dev_err(&hdev->dev, "hdac link not found\n");
+ return -EIO;
+ }
+ snd_hdac_ext_bus_link_get(hdev->bus, hlink);
+
+ hda_pvt = hdac_to_hda_priv(hdev);
+ if (!hda_pvt)
+ return -ENOMEM;
+
+ /* ASoC specific initialization */
+ ret = devm_snd_soc_register_component(&hdev->dev,
+ &hdac_hda_codec, hdac_hda_dais,
+ ARRAY_SIZE(hdac_hda_dais));
+ if (ret < 0) {
+ dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret);
+ return ret;
+ }
+
+ dev_set_drvdata(&hdev->dev, hda_pvt);
+ snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+
+ return ret;
+}
+
+static int hdac_hda_dev_remove(struct hdac_device *hdev)
+{
+ return 0;
+}
+
+static struct hdac_ext_bus_ops hdac_ops = {
+ .hdev_attach = hdac_hda_dev_probe,
+ .hdev_detach = hdac_hda_dev_remove,
+};
+
+struct hdac_ext_bus_ops *snd_soc_hdac_hda_get_ops(void)
+{
+ return &hdac_ops;
+}
+EXPORT_SYMBOL_GPL(snd_soc_hdac_hda_get_ops);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("ASoC Extensions for legacy HDA Drivers");
+MODULE_AUTHOR("Rakesh Ughreja<rakesh.a.ughreja@intel.com>");
diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h
new file mode 100644
index 000000000000..e444ef593360
--- /dev/null
+++ b/sound/soc/codecs/hdac_hda.h
@@ -0,0 +1,24 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright(c) 2015-18 Intel Corporation.
+ */
+
+#ifndef __HDAC_HDA_H__
+#define __HDAC_HDA_H__
+
+struct hdac_hda_pcm {
+ int stream_tag[2];
+};
+
+struct hdac_hda_priv {
+ struct hda_codec codec;
+ struct hdac_hda_pcm pcm[2];
+};
+
+#define hdac_to_hda_priv(_hdac) \
+ container_of(_hdac, struct hdac_hda_priv, codec.core)
+#define hdac_to_hda_codec(_hdac) container_of(_hdac, struct hda_codec, core)
+
+struct hdac_ext_bus_ops *snd_soc_hdac_hda_get_ops(void);
+
+#endif /* __HDAC_HDA_H__ */
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 7b8533abf637..4e9854889a95 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1410,6 +1410,12 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev,
if (ret)
return ret;
+ /* Filter out 44.1, 88.2 and 176.4Khz */
+ rates &= ~(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_176400);
+ if (!rates)
+ return -EINVAL;
+
sprintf(dai_name, "intel-hdmi-hifi%d", i+1);
hdmi_dais[i].name = devm_kstrdup(&hdev->dev,
dai_name, GFP_KERNEL);
@@ -1598,7 +1604,7 @@ static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card,
{
struct snd_soc_pcm_runtime *rtd;
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
if (rtd->pcm && (rtd->pcm->device == device))
return rtd->pcm;
}
@@ -1961,9 +1967,6 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdev, int pcm_idx)
port = list_first_entry(&pcm->port_list, struct hdac_hdmi_port, head);
- if (!port)
- return 0;
-
if (!port || !port->eld.eld_valid)
return 0;
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index fb515aaa54fc..ca172a4b6849 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -16,6 +16,7 @@
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/regmap.h>
+#include <linux/clk.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -42,6 +43,7 @@ struct max98088_priv {
struct regmap *regmap;
enum max98088_type devtype;
struct max98088_pdata *pdata;
+ struct clk *mclk;
unsigned int sysclk;
struct max98088_cdata dai[2];
int eq_textcnt;
@@ -1103,6 +1105,11 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai,
if (freq == max98088->sysclk)
return 0;
+ if (!IS_ERR(max98088->mclk)) {
+ freq = clk_round_rate(max98088->mclk, freq);
+ clk_set_rate(max98088->mclk, freq);
+ }
+
/* Setup clocks for slave mode, and using the PLL
* PSCLK = 0x01 (when master clk is 10MHz to 20MHz)
* 0x02 (when master clk is 20MHz to 30MHz)..
@@ -1310,6 +1317,20 @@ static int max98088_set_bias_level(struct snd_soc_component *component,
break;
case SND_SOC_BIAS_PREPARE:
+ /*
+ * SND_SOC_BIAS_PREPARE is called while preparing for a
+ * transition to ON or away from ON. If current bias_level
+ * is SND_SOC_BIAS_ON, then it is preparing for a transition
+ * away from ON. Disable the clock in that case, otherwise
+ * enable it.
+ */
+ if (!IS_ERR(max98088->mclk)) {
+ if (snd_soc_component_get_bias_level(component) ==
+ SND_SOC_BIAS_ON)
+ clk_disable_unprepare(max98088->mclk);
+ else
+ clk_prepare_enable(max98088->mclk);
+ }
break;
case SND_SOC_BIAS_STANDBY:
@@ -1725,6 +1746,11 @@ static int max98088_i2c_probe(struct i2c_client *i2c,
if (IS_ERR(max98088->regmap))
return PTR_ERR(max98088->regmap);
+ max98088->mclk = devm_clk_get(&i2c->dev, "mclk");
+ if (IS_ERR(max98088->mclk))
+ if (PTR_ERR(max98088->mclk) == -EPROBE_DEFER)
+ return PTR_ERR(max98088->mclk);
+
max98088->devtype = id->driver_data;
i2c_set_clientdata(i2c, max98088);
@@ -1742,9 +1768,19 @@ static const struct i2c_device_id max98088_i2c_id[] = {
};
MODULE_DEVICE_TABLE(i2c, max98088_i2c_id);
+#if defined(CONFIG_OF)
+static const struct of_device_id max98088_of_match[] = {
+ { .compatible = "maxim,max98088" },
+ { .compatible = "maxim,max98089" },
+ { }
+};
+MODULE_DEVICE_TABLE(of, max98088_of_match);
+#endif
+
static struct i2c_driver max98088_i2c_driver = {
.driver = {
.name = "max98088",
+ .of_match_table = of_match_ptr(max98088_of_match),
},
.probe = max98088_i2c_probe,
.id_table = max98088_i2c_id,
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index 1093f766d0d2..a09d01318f79 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -2,6 +2,7 @@
// Copyright (c) 2017, Maxim Integrated
#include <linux/acpi.h>
+#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/module.h>
#include <linux/regmap.h>
@@ -454,7 +455,7 @@ SND_SOC_DAPM_SIGGEN("IMON"),
SND_SOC_DAPM_SIGGEN("FBMON"),
};
-static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, 0, -50, 0);
+static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, -6350, 50, 1);
static const DECLARE_TLV_DB_RANGE(max98373_spk_tlv,
0, 8, TLV_DB_SCALE_ITEM(0, 50, 0),
9, 10, TLV_DB_SCALE_ITEM(500, 100, 0),
@@ -470,19 +471,19 @@ static const DECLARE_TLV_DB_RANGE(max98373_dht_spkgain_min_tlv,
0, 9, TLV_DB_SCALE_ITEM(800, 100, 0),
);
static const DECLARE_TLV_DB_RANGE(max98373_dht_rotation_point_tlv,
- 0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0),
- 2, 7, TLV_DB_SCALE_ITEM(-200, -100, 0),
- 8, 9, TLV_DB_SCALE_ITEM(-1000, -200, 0),
- 10, 11, TLV_DB_SCALE_ITEM(-1500, -300, 0),
- 12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0),
- 14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0),
+ 0, 1, TLV_DB_SCALE_ITEM(-3000, 500, 0),
+ 2, 4, TLV_DB_SCALE_ITEM(-2200, 200, 0),
+ 5, 6, TLV_DB_SCALE_ITEM(-1500, 300, 0),
+ 7, 9, TLV_DB_SCALE_ITEM(-1000, 200, 0),
+ 10, 13, TLV_DB_SCALE_ITEM(-500, 100, 0),
+ 14, 15, TLV_DB_SCALE_ITEM(-100, 50, 0),
);
static const DECLARE_TLV_DB_RANGE(max98373_limiter_thresh_tlv,
- 0, 15, TLV_DB_SCALE_ITEM(0, -100, 0),
+ 0, 15, TLV_DB_SCALE_ITEM(-1500, 100, 0),
);
static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv,
- 0, 60, TLV_DB_SCALE_ITEM(0, -25, 0),
+ 0, 60, TLV_DB_SCALE_ITEM(-1500, 25, 0),
);
static bool max98373_readable_register(struct device *dev, unsigned int reg)
@@ -604,7 +605,7 @@ SOC_SINGLE("Dither Switch", MAX98373_R203F_AMP_DSP_CFG,
SOC_SINGLE("DC Blocker Switch", MAX98373_R203F_AMP_DSP_CFG,
MAX98373_AMP_DSP_CFG_DCBLK_SHIFT, 1, 0),
SOC_SINGLE_TLV("Digital Volume", MAX98373_R203D_AMP_DIG_VOL_CTRL,
- 0, 0x7F, 0, max98373_digital_tlv),
+ 0, 0x7F, 1, max98373_digital_tlv),
SOC_SINGLE_TLV("Speaker Volume", MAX98373_R203E_AMP_PATH_GAIN,
MAX98373_SPK_DIGI_GAIN_SHIFT, 10, 0, max98373_spk_tlv),
SOC_SINGLE_TLV("FS Max Volume", MAX98373_R203E_AMP_PATH_GAIN,
@@ -616,7 +617,7 @@ SOC_SINGLE("DHT Switch", MAX98373_R20D4_DHT_EN,
SOC_SINGLE_TLV("DHT Min Volume", MAX98373_R20D1_DHT_CFG,
MAX98373_DHT_SPK_GAIN_MIN_SHIFT, 9, 0, max98373_dht_spkgain_min_tlv),
SOC_SINGLE_TLV("DHT Rot Pnt Volume", MAX98373_R20D1_DHT_CFG,
- MAX98373_DHT_ROT_PNT_SHIFT, 15, 0, max98373_dht_rotation_point_tlv),
+ MAX98373_DHT_ROT_PNT_SHIFT, 15, 1, max98373_dht_rotation_point_tlv),
SOC_SINGLE_TLV("DHT Attack Step Volume", MAX98373_R20D2_DHT_ATTACK_CFG,
MAX98373_DHT_ATTACK_STEP_SHIFT, 4, 0, max98373_dht_step_size_tlv),
SOC_SINGLE_TLV("DHT Release Step Volume", MAX98373_R20D3_DHT_RELEASE_CFG,
@@ -653,29 +654,29 @@ SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0),
SOC_SINGLE("BDE Attack Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 4, 0xF, 0),
SOC_SINGLE("BDE Release Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 0, 0xF, 0),
SOC_SINGLE_TLV("BDE LVL1 Clip Thresh Volume", MAX98373_R20A9_BDE_L1_CFG_2,
- 0, 0x3C, 0, max98373_bde_gain_tlv),
+ 0, 0x3C, 1, max98373_bde_gain_tlv),
SOC_SINGLE_TLV("BDE LVL2 Clip Thresh Volume", MAX98373_R20AC_BDE_L2_CFG_2,
- 0, 0x3C, 0, max98373_bde_gain_tlv),
+ 0, 0x3C, 1, max98373_bde_gain_tlv),
SOC_SINGLE_TLV("BDE LVL3 Clip Thresh Volume", MAX98373_R20AF_BDE_L3_CFG_2,
- 0, 0x3C, 0, max98373_bde_gain_tlv),
+ 0, 0x3C, 1, max98373_bde_gain_tlv),
SOC_SINGLE_TLV("BDE LVL4 Clip Thresh Volume", MAX98373_R20B2_BDE_L4_CFG_2,
- 0, 0x3C, 0, max98373_bde_gain_tlv),
+ 0, 0x3C, 1, max98373_bde_gain_tlv),
SOC_SINGLE_TLV("BDE LVL1 Clip Reduction Volume", MAX98373_R20AA_BDE_L1_CFG_3,
- 0, 0x3C, 0, max98373_bde_gain_tlv),
+ 0, 0x3C, 1, max98373_bde_gain_tlv),
SOC_SINGLE_TLV("BDE LVL2 Clip Reduction Volume", MAX98373_R20AD_BDE_L2_CFG_3,
- 0, 0x3C, 0, max98373_bde_gain_tlv),
+ 0, 0x3C, 1, max98373_bde_gain_tlv),
SOC_SINGLE_TLV("BDE LVL3 Clip Reduction Volume", MAX98373_R20B0_BDE_L3_CFG_3,
- 0, 0x3C, 0, max98373_bde_gain_tlv),
+ 0, 0x3C, 1, max98373_bde_gain_tlv),
SOC_SINGLE_TLV("BDE LVL4 Clip Reduction Volume", MAX98373_R20B3_BDE_L4_CFG_3,
- 0, 0x3C, 0, max98373_bde_gain_tlv),
+ 0, 0x3C, 1, max98373_bde_gain_tlv),
SOC_SINGLE_TLV("BDE LVL1 Limiter Thresh Volume", MAX98373_R20A8_BDE_L1_CFG_1,
- 0, 0xF, 0, max98373_limiter_thresh_tlv),
+ 0, 0xF, 1, max98373_limiter_thresh_tlv),
SOC_SINGLE_TLV("BDE LVL2 Limiter Thresh Volume", MAX98373_R20AB_BDE_L2_CFG_1,
- 0, 0xF, 0, max98373_limiter_thresh_tlv),
+ 0, 0xF, 1, max98373_limiter_thresh_tlv),
SOC_SINGLE_TLV("BDE LVL3 Limiter Thresh Volume", MAX98373_R20AE_BDE_L3_CFG_1,
- 0, 0xF, 0, max98373_limiter_thresh_tlv),
+ 0, 0xF, 1, max98373_limiter_thresh_tlv),
SOC_SINGLE_TLV("BDE LVL4 Limiter Thresh Volume", MAX98373_R20B1_BDE_L4_CFG_1,
- 0, 0xF, 0, max98373_limiter_thresh_tlv),
+ 0, 0xF, 1, max98373_limiter_thresh_tlv),
/* Limiter */
SOC_SINGLE("Limiter Switch", MAX98373_R20E2_LIMITER_EN,
MAX98373_LIMITER_EN_SHIFT, 1, 0),
diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c
new file mode 100644
index 000000000000..622ce947f134
--- /dev/null
+++ b/sound/soc/codecs/nau8822.c
@@ -0,0 +1,1136 @@
+/*
+ * nau8822.c -- NAU8822 ALSA Soc Audio Codec driver
+ *
+ * Copyright 2017 Nuvoton Technology Corp.
+ *
+ * Author: David Lin <ctlin0@nuvoton.com>
+ * Co-author: John Hsu <kchsu0@nuvoton.com>
+ * Co-author: Seven Li <wtli@nuvoton.com>
+ *
+ * Based on WM8974.c
+ *
+ * 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/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <asm/div64.h>
+#include "nau8822.h"
+
+#define NAU_PLL_FREQ_MAX 100000000
+#define NAU_PLL_FREQ_MIN 90000000
+#define NAU_PLL_REF_MAX 33000000
+#define NAU_PLL_REF_MIN 8000000
+#define NAU_PLL_OPTOP_MIN 6
+
+static const int nau8822_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 };
+
+static const struct reg_default nau8822_reg_defaults[] = {
+ { NAU8822_REG_POWER_MANAGEMENT_1, 0x0000 },
+ { NAU8822_REG_POWER_MANAGEMENT_2, 0x0000 },
+ { NAU8822_REG_POWER_MANAGEMENT_3, 0x0000 },
+ { NAU8822_REG_AUDIO_INTERFACE, 0x0050 },
+ { NAU8822_REG_COMPANDING_CONTROL, 0x0000 },
+ { NAU8822_REG_CLOCKING, 0x0140 },
+ { NAU8822_REG_ADDITIONAL_CONTROL, 0x0000 },
+ { NAU8822_REG_GPIO_CONTROL, 0x0000 },
+ { NAU8822_REG_JACK_DETECT_CONTROL_1, 0x0000 },
+ { NAU8822_REG_DAC_CONTROL, 0x0000 },
+ { NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, 0x00ff },
+ { NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0x00ff },
+ { NAU8822_REG_JACK_DETECT_CONTROL_2, 0x0000 },
+ { NAU8822_REG_ADC_CONTROL, 0x0100 },
+ { NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, 0x00ff },
+ { NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0x00ff },
+ { NAU8822_REG_EQ1, 0x012c },
+ { NAU8822_REG_EQ2, 0x002c },
+ { NAU8822_REG_EQ3, 0x002c },
+ { NAU8822_REG_EQ4, 0x002c },
+ { NAU8822_REG_EQ5, 0x002c },
+ { NAU8822_REG_DAC_LIMITER_1, 0x0032 },
+ { NAU8822_REG_DAC_LIMITER_2, 0x0000 },
+ { NAU8822_REG_NOTCH_FILTER_1, 0x0000 },
+ { NAU8822_REG_NOTCH_FILTER_2, 0x0000 },
+ { NAU8822_REG_NOTCH_FILTER_3, 0x0000 },
+ { NAU8822_REG_NOTCH_FILTER_4, 0x0000 },
+ { NAU8822_REG_ALC_CONTROL_1, 0x0038 },
+ { NAU8822_REG_ALC_CONTROL_2, 0x000b },
+ { NAU8822_REG_ALC_CONTROL_3, 0x0032 },
+ { NAU8822_REG_NOISE_GATE, 0x0010 },
+ { NAU8822_REG_PLL_N, 0x0008 },
+ { NAU8822_REG_PLL_K1, 0x000c },
+ { NAU8822_REG_PLL_K2, 0x0093 },
+ { NAU8822_REG_PLL_K3, 0x00e9 },
+ { NAU8822_REG_3D_CONTROL, 0x0000 },
+ { NAU8822_REG_RIGHT_SPEAKER_CONTROL, 0x0000 },
+ { NAU8822_REG_INPUT_CONTROL, 0x0033 },
+ { NAU8822_REG_LEFT_INP_PGA_CONTROL, 0x0010 },
+ { NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0x0010 },
+ { NAU8822_REG_LEFT_ADC_BOOST_CONTROL, 0x0100 },
+ { NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0x0100 },
+ { NAU8822_REG_OUTPUT_CONTROL, 0x0002 },
+ { NAU8822_REG_LEFT_MIXER_CONTROL, 0x0001 },
+ { NAU8822_REG_RIGHT_MIXER_CONTROL, 0x0001 },
+ { NAU8822_REG_LHP_VOLUME, 0x0039 },
+ { NAU8822_REG_RHP_VOLUME, 0x0039 },
+ { NAU8822_REG_LSPKOUT_VOLUME, 0x0039 },
+ { NAU8822_REG_RSPKOUT_VOLUME, 0x0039 },
+ { NAU8822_REG_AUX2_MIXER, 0x0001 },
+ { NAU8822_REG_AUX1_MIXER, 0x0001 },
+ { NAU8822_REG_POWER_MANAGEMENT_4, 0x0000 },
+ { NAU8822_REG_LEFT_TIME_SLOT, 0x0000 },
+ { NAU8822_REG_MISC, 0x0020 },
+ { NAU8822_REG_RIGHT_TIME_SLOT, 0x0000 },
+ { NAU8822_REG_DEVICE_REVISION, 0x007f },
+ { NAU8822_REG_DEVICE_ID, 0x001a },
+ { NAU8822_REG_DAC_DITHER, 0x0114 },
+ { NAU8822_REG_ALC_ENHANCE_1, 0x0000 },
+ { NAU8822_REG_ALC_ENHANCE_2, 0x0000 },
+ { NAU8822_REG_192KHZ_SAMPLING, 0x0008 },
+ { NAU8822_REG_MISC_CONTROL, 0x0000 },
+ { NAU8822_REG_INPUT_TIEOFF, 0x0000 },
+ { NAU8822_REG_POWER_REDUCTION, 0x0000 },
+ { NAU8822_REG_AGC_PEAK2PEAK, 0x0000 },
+ { NAU8822_REG_AGC_PEAK_DETECT, 0x0000 },
+ { NAU8822_REG_AUTOMUTE_CONTROL, 0x0000 },
+ { NAU8822_REG_OUTPUT_TIEOFF, 0x0000 },
+};
+
+static bool nau8822_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1:
+ case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME:
+ case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME:
+ case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5:
+ case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2:
+ case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4:
+ case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3:
+ case NAU8822_REG_3D_CONTROL:
+ case NAU8822_REG_RIGHT_SPEAKER_CONTROL:
+ case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL:
+ case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER:
+ case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID:
+ case NAU8822_REG_DAC_DITHER:
+ case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL:
+ case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool nau8822_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1:
+ case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME:
+ case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME:
+ case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5:
+ case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2:
+ case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4:
+ case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3:
+ case NAU8822_REG_3D_CONTROL:
+ case NAU8822_REG_RIGHT_SPEAKER_CONTROL:
+ case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL:
+ case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER:
+ case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID:
+ case NAU8822_REG_DAC_DITHER:
+ case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL:
+ case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool nau8822_volatile(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case NAU8822_REG_RESET:
+ case NAU8822_REG_DEVICE_REVISION:
+ case NAU8822_REG_DEVICE_ID:
+ case NAU8822_REG_AGC_PEAK2PEAK:
+ case NAU8822_REG_AGC_PEAK_DETECT:
+ case NAU8822_REG_AUTOMUTE_CONTROL:
+ return true;
+ default:
+ return false;
+ }
+}
+
+/* The EQ parameters get function is to get the 5 band equalizer control.
+ * The regmap raw read can't work here because regmap doesn't provide
+ * value format for value width of 9 bits. Therefore, the driver reads data
+ * from cache and makes value format according to the endianness of
+ * bytes type control element.
+ */
+static int nau8822_eq_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component =
+ snd_soc_kcontrol_component(kcontrol);
+ struct soc_bytes_ext *params = (void *)kcontrol->private_value;
+ int i, reg;
+ u16 reg_val, *val;
+
+ val = (u16 *)ucontrol->value.bytes.data;
+ reg = NAU8822_REG_EQ1;
+ for (i = 0; i < params->max / sizeof(u16); i++) {
+ reg_val = snd_soc_component_read32(component, reg + i);
+ /* conversion of 16-bit integers between native CPU format
+ * and big endian format
+ */
+ reg_val = cpu_to_be16(reg_val);
+ memcpy(val + i, &reg_val, sizeof(reg_val));
+ }
+
+ return 0;
+}
+
+/* The EQ parameters put function is to make configuration of 5 band equalizer
+ * control. These configuration includes central frequency, equalizer gain,
+ * cut-off frequency, bandwidth control, and equalizer path.
+ * The regmap raw write can't work here because regmap doesn't provide
+ * register and value format for register with address 7 bits and value 9 bits.
+ * Therefore, the driver makes value format according to the endianness of
+ * bytes type control element and writes data to codec.
+ */
+static int nau8822_eq_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component =
+ snd_soc_kcontrol_component(kcontrol);
+ struct soc_bytes_ext *params = (void *)kcontrol->private_value;
+ void *data;
+ u16 *val, value;
+ int i, reg, ret;
+
+ data = kmemdup(ucontrol->value.bytes.data,
+ params->max, GFP_KERNEL | GFP_DMA);
+ if (!data)
+ return -ENOMEM;
+
+ val = (u16 *)data;
+ reg = NAU8822_REG_EQ1;
+ for (i = 0; i < params->max / sizeof(u16); i++) {
+ /* conversion of 16-bit integers between native CPU format
+ * and big endian format
+ */
+ value = be16_to_cpu(*(val + i));
+ ret = snd_soc_component_write(component, reg + i, value);
+ if (ret) {
+ dev_err(component->dev,
+ "EQ configuration fail, register: %x ret: %d\n",
+ reg + i, ret);
+ kfree(data);
+ return ret;
+ }
+ }
+ kfree(data);
+
+ return 0;
+}
+
+static const char * const nau8822_companding[] = {
+ "Off", "NC", "u-law", "A-law"};
+
+static const struct soc_enum nau8822_companding_adc_enum =
+ SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_ADCCM_SFT,
+ ARRAY_SIZE(nau8822_companding), nau8822_companding);
+
+static const struct soc_enum nau8822_companding_dac_enum =
+ SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_DACCM_SFT,
+ ARRAY_SIZE(nau8822_companding), nau8822_companding);
+
+static const char * const nau8822_eqmode[] = {"Capture", "Playback"};
+
+static const struct soc_enum nau8822_eqmode_enum =
+ SOC_ENUM_SINGLE(NAU8822_REG_EQ1, NAU8822_EQM_SFT,
+ ARRAY_SIZE(nau8822_eqmode), nau8822_eqmode);
+
+static const char * const nau8822_alc1[] = {"Off", "Right", "Left", "Both"};
+static const char * const nau8822_alc3[] = {"Normal", "Limiter"};
+
+static const struct soc_enum nau8822_alc_enable_enum =
+ SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_1, NAU8822_ALCEN_SFT,
+ ARRAY_SIZE(nau8822_alc1), nau8822_alc1);
+
+static const struct soc_enum nau8822_alc_mode_enum =
+ SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_3, NAU8822_ALCM_SFT,
+ ARRAY_SIZE(nau8822_alc3), nau8822_alc3);
+
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0);
+static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0);
+static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0);
+static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1);
+static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0);
+
+static const struct snd_kcontrol_new nau8822_snd_controls[] = {
+ SOC_ENUM("ADC Companding", nau8822_companding_adc_enum),
+ SOC_ENUM("DAC Companding", nau8822_companding_dac_enum),
+
+ SOC_ENUM("EQ Function", nau8822_eqmode_enum),
+ SND_SOC_BYTES_EXT("EQ Parameters", 10,
+ nau8822_eq_get, nau8822_eq_put),
+
+ SOC_DOUBLE("DAC Inversion Switch",
+ NAU8822_REG_DAC_CONTROL, 0, 1, 1, 0),
+ SOC_DOUBLE_R_TLV("PCM Volume",
+ NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME,
+ NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv),
+
+ SOC_SINGLE("High Pass Filter Switch",
+ NAU8822_REG_ADC_CONTROL, 8, 1, 0),
+ SOC_SINGLE("High Pass Cut Off",
+ NAU8822_REG_ADC_CONTROL, 4, 7, 0),
+
+ SOC_DOUBLE("ADC Inversion Switch",
+ NAU8822_REG_ADC_CONTROL, 0, 1, 1, 0),
+ SOC_DOUBLE_R_TLV("ADC Volume",
+ NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME,
+ NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv),
+
+ SOC_SINGLE("DAC Limiter Switch",
+ NAU8822_REG_DAC_LIMITER_1, 8, 1, 0),
+ SOC_SINGLE("DAC Limiter Decay",
+ NAU8822_REG_DAC_LIMITER_1, 4, 15, 0),
+ SOC_SINGLE("DAC Limiter Attack",
+ NAU8822_REG_DAC_LIMITER_1, 0, 15, 0),
+ SOC_SINGLE("DAC Limiter Threshold",
+ NAU8822_REG_DAC_LIMITER_2, 4, 7, 0),
+ SOC_SINGLE_TLV("DAC Limiter Volume",
+ NAU8822_REG_DAC_LIMITER_2, 0, 12, 0, limiter_tlv),
+
+ SOC_ENUM("ALC Mode", nau8822_alc_mode_enum),
+ SOC_ENUM("ALC Enable Switch", nau8822_alc_enable_enum),
+ SOC_SINGLE("ALC Min Gain",
+ NAU8822_REG_ALC_CONTROL_1, 0, 7, 0),
+ SOC_SINGLE("ALC Max Gain",
+ NAU8822_REG_ALC_CONTROL_1, 3, 7, 0),
+ SOC_SINGLE("ALC Hold",
+ NAU8822_REG_ALC_CONTROL_2, 4, 10, 0),
+ SOC_SINGLE("ALC Target",
+ NAU8822_REG_ALC_CONTROL_2, 0, 15, 0),
+ SOC_SINGLE("ALC Decay",
+ NAU8822_REG_ALC_CONTROL_3, 4, 10, 0),
+ SOC_SINGLE("ALC Attack",
+ NAU8822_REG_ALC_CONTROL_3, 0, 10, 0),
+ SOC_SINGLE("ALC Noise Gate Switch",
+ NAU8822_REG_NOISE_GATE, 3, 1, 0),
+ SOC_SINGLE("ALC Noise Gate Threshold",
+ NAU8822_REG_NOISE_GATE, 0, 7, 0),
+
+ SOC_DOUBLE_R("PGA ZC Switch",
+ NAU8822_REG_LEFT_INP_PGA_CONTROL,
+ NAU8822_REG_RIGHT_INP_PGA_CONTROL,
+ 7, 1, 0),
+ SOC_DOUBLE_R_TLV("PGA Volume",
+ NAU8822_REG_LEFT_INP_PGA_CONTROL,
+ NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0, 63, 0, inpga_tlv),
+
+ SOC_DOUBLE_R("Headphone ZC Switch",
+ NAU8822_REG_LHP_VOLUME,
+ NAU8822_REG_RHP_VOLUME, 7, 1, 0),
+ SOC_DOUBLE_R("Headphone Playback Switch",
+ NAU8822_REG_LHP_VOLUME,
+ NAU8822_REG_RHP_VOLUME, 6, 1, 1),
+ SOC_DOUBLE_R_TLV("Headphone Volume",
+ NAU8822_REG_LHP_VOLUME,
+ NAU8822_REG_RHP_VOLUME, 0, 63, 0, spk_tlv),
+
+ SOC_DOUBLE_R("Speaker ZC Switch",
+ NAU8822_REG_LSPKOUT_VOLUME,
+ NAU8822_REG_RSPKOUT_VOLUME, 7, 1, 0),
+ SOC_DOUBLE_R("Speaker Playback Switch",
+ NAU8822_REG_LSPKOUT_VOLUME,
+ NAU8822_REG_RSPKOUT_VOLUME, 6, 1, 1),
+ SOC_DOUBLE_R_TLV("Speaker Volume",
+ NAU8822_REG_LSPKOUT_VOLUME,
+ NAU8822_REG_RSPKOUT_VOLUME, 0, 63, 0, spk_tlv),
+
+ SOC_DOUBLE_R("AUXOUT Playback Switch",
+ NAU8822_REG_AUX2_MIXER,
+ NAU8822_REG_AUX1_MIXER, 6, 1, 1),
+
+ SOC_DOUBLE_R_TLV("PGA Boost Volume",
+ NAU8822_REG_LEFT_ADC_BOOST_CONTROL,
+ NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 8, 1, 0, pga_boost_tlv),
+ SOC_DOUBLE_R_TLV("L2/R2 Boost Volume",
+ NAU8822_REG_LEFT_ADC_BOOST_CONTROL,
+ NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 4, 7, 0, boost_tlv),
+ SOC_DOUBLE_R_TLV("Aux Boost Volume",
+ NAU8822_REG_LEFT_ADC_BOOST_CONTROL,
+ NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0, 7, 0, boost_tlv),
+
+ SOC_SINGLE("DAC 128x Oversampling Switch",
+ NAU8822_REG_DAC_CONTROL, 5, 1, 0),
+ SOC_SINGLE("ADC 128x Oversampling Switch",
+ NAU8822_REG_ADC_CONTROL, 5, 1, 0),
+};
+
+/* LMAIN and RMAIN Mixer */
+static const struct snd_kcontrol_new nau8822_left_out_mixer[] = {
+ SOC_DAPM_SINGLE("LINMIX Switch",
+ NAU8822_REG_LEFT_MIXER_CONTROL, 1, 1, 0),
+ SOC_DAPM_SINGLE("LAUX Switch",
+ NAU8822_REG_LEFT_MIXER_CONTROL, 5, 1, 0),
+ SOC_DAPM_SINGLE("LDAC Switch",
+ NAU8822_REG_LEFT_MIXER_CONTROL, 0, 1, 0),
+ SOC_DAPM_SINGLE("RDAC Switch",
+ NAU8822_REG_OUTPUT_CONTROL, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new nau8822_right_out_mixer[] = {
+ SOC_DAPM_SINGLE("RINMIX Switch",
+ NAU8822_REG_RIGHT_MIXER_CONTROL, 1, 1, 0),
+ SOC_DAPM_SINGLE("RAUX Switch",
+ NAU8822_REG_RIGHT_MIXER_CONTROL, 5, 1, 0),
+ SOC_DAPM_SINGLE("RDAC Switch",
+ NAU8822_REG_RIGHT_MIXER_CONTROL, 0, 1, 0),
+ SOC_DAPM_SINGLE("LDAC Switch",
+ NAU8822_REG_OUTPUT_CONTROL, 6, 1, 0),
+};
+
+/* AUX1 and AUX2 Mixer */
+static const struct snd_kcontrol_new nau8822_auxout1_mixer[] = {
+ SOC_DAPM_SINGLE("RDAC Switch", NAU8822_REG_AUX1_MIXER, 0, 1, 0),
+ SOC_DAPM_SINGLE("RMIX Switch", NAU8822_REG_AUX1_MIXER, 1, 1, 0),
+ SOC_DAPM_SINGLE("RINMIX Switch", NAU8822_REG_AUX1_MIXER, 2, 1, 0),
+ SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX1_MIXER, 3, 1, 0),
+ SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX1_MIXER, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new nau8822_auxout2_mixer[] = {
+ SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX2_MIXER, 0, 1, 0),
+ SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX2_MIXER, 1, 1, 0),
+ SOC_DAPM_SINGLE("LINMIX Switch", NAU8822_REG_AUX2_MIXER, 2, 1, 0),
+ SOC_DAPM_SINGLE("AUX1MIX Output Switch",
+ NAU8822_REG_AUX2_MIXER, 3, 1, 0),
+};
+
+/* Input PGA */
+static const struct snd_kcontrol_new nau8822_left_input_mixer[] = {
+ SOC_DAPM_SINGLE("L2 Switch", NAU8822_REG_INPUT_CONTROL, 2, 1, 0),
+ SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 1, 1, 0),
+ SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 0, 1, 0),
+};
+static const struct snd_kcontrol_new nau8822_right_input_mixer[] = {
+ SOC_DAPM_SINGLE("R2 Switch", NAU8822_REG_INPUT_CONTROL, 6, 1, 0),
+ SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 5, 1, 0),
+ SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 4, 1, 0),
+};
+
+/* Loopback Switch */
+static const struct snd_kcontrol_new nau8822_loopback =
+ SOC_DAPM_SINGLE("Switch", NAU8822_REG_COMPANDING_CONTROL,
+ NAU8822_ADDAP_SFT, 1, 0);
+
+static int check_mclk_select_pll(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(source->dapm);
+ unsigned int value;
+
+ value = snd_soc_component_read32(component, NAU8822_REG_CLOCKING);
+
+ return (value & NAU8822_CLKM_MASK);
+}
+
+static const struct snd_soc_dapm_widget nau8822_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
+ NAU8822_REG_POWER_MANAGEMENT_3, 0, 0),
+ SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
+ NAU8822_REG_POWER_MANAGEMENT_3, 1, 0),
+ SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
+ NAU8822_REG_POWER_MANAGEMENT_2, 0, 0),
+ SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
+ NAU8822_REG_POWER_MANAGEMENT_2, 1, 0),
+
+ SOC_MIXER_ARRAY("Left Output Mixer",
+ NAU8822_REG_POWER_MANAGEMENT_3, 2, 0, nau8822_left_out_mixer),
+ SOC_MIXER_ARRAY("Right Output Mixer",
+ NAU8822_REG_POWER_MANAGEMENT_3, 3, 0, nau8822_right_out_mixer),
+ SOC_MIXER_ARRAY("AUX1 Output Mixer",
+ NAU8822_REG_POWER_MANAGEMENT_1, 7, 0, nau8822_auxout1_mixer),
+ SOC_MIXER_ARRAY("AUX2 Output Mixer",
+ NAU8822_REG_POWER_MANAGEMENT_1, 6, 0, nau8822_auxout2_mixer),
+
+ SOC_MIXER_ARRAY("Left Input Mixer",
+ NAU8822_REG_POWER_MANAGEMENT_2,
+ 2, 0, nau8822_left_input_mixer),
+ SOC_MIXER_ARRAY("Right Input Mixer",
+ NAU8822_REG_POWER_MANAGEMENT_2,
+ 3, 0, nau8822_right_input_mixer),
+
+ SND_SOC_DAPM_PGA("Left Boost Mixer",
+ NAU8822_REG_POWER_MANAGEMENT_2, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right Boost Mixer",
+ NAU8822_REG_POWER_MANAGEMENT_2, 5, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA("Left Capture PGA",
+ NAU8822_REG_LEFT_INP_PGA_CONTROL, 6, 1, NULL, 0),
+ SND_SOC_DAPM_PGA("Right Capture PGA",
+ NAU8822_REG_RIGHT_INP_PGA_CONTROL, 6, 1, NULL, 0),
+
+ SND_SOC_DAPM_PGA("Left Headphone Out",
+ NAU8822_REG_POWER_MANAGEMENT_2, 7, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right Headphone Out",
+ NAU8822_REG_POWER_MANAGEMENT_2, 8, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA("Left Speaker Out",
+ NAU8822_REG_POWER_MANAGEMENT_3, 6, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right Speaker Out",
+ NAU8822_REG_POWER_MANAGEMENT_3, 5, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA("AUX1 Out",
+ NAU8822_REG_POWER_MANAGEMENT_3, 8, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("AUX2 Out",
+ NAU8822_REG_POWER_MANAGEMENT_3, 7, 0, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("Mic Bias",
+ NAU8822_REG_POWER_MANAGEMENT_1, 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLL",
+ NAU8822_REG_POWER_MANAGEMENT_1, 5, 0, NULL, 0),
+
+ SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0,
+ &nau8822_loopback),
+
+ SND_SOC_DAPM_INPUT("LMICN"),
+ SND_SOC_DAPM_INPUT("LMICP"),
+ SND_SOC_DAPM_INPUT("RMICN"),
+ SND_SOC_DAPM_INPUT("RMICP"),
+ SND_SOC_DAPM_INPUT("LAUX"),
+ SND_SOC_DAPM_INPUT("RAUX"),
+ SND_SOC_DAPM_INPUT("L2"),
+ SND_SOC_DAPM_INPUT("R2"),
+ SND_SOC_DAPM_OUTPUT("LHP"),
+ SND_SOC_DAPM_OUTPUT("RHP"),
+ SND_SOC_DAPM_OUTPUT("LSPK"),
+ SND_SOC_DAPM_OUTPUT("RSPK"),
+ SND_SOC_DAPM_OUTPUT("AUXOUT1"),
+ SND_SOC_DAPM_OUTPUT("AUXOUT2"),
+};
+
+static const struct snd_soc_dapm_route nau8822_dapm_routes[] = {
+ {"Right DAC", NULL, "PLL", check_mclk_select_pll},
+ {"Left DAC", NULL, "PLL", check_mclk_select_pll},
+
+ /* LMAIN and RMAIN Mixer */
+ {"Right Output Mixer", "LDAC Switch", "Left DAC"},
+ {"Right Output Mixer", "RDAC Switch", "Right DAC"},
+ {"Right Output Mixer", "RAUX Switch", "RAUX"},
+ {"Right Output Mixer", "RINMIX Switch", "Right Boost Mixer"},
+
+ {"Left Output Mixer", "LDAC Switch", "Left DAC"},
+ {"Left Output Mixer", "RDAC Switch", "Right DAC"},
+ {"Left Output Mixer", "LAUX Switch", "LAUX"},
+ {"Left Output Mixer", "LINMIX Switch", "Left Boost Mixer"},
+
+ /* AUX1 and AUX2 Mixer */
+ {"AUX1 Output Mixer", "RDAC Switch", "Right DAC"},
+ {"AUX1 Output Mixer", "RMIX Switch", "Right Output Mixer"},
+ {"AUX1 Output Mixer", "RINMIX Switch", "Right Boost Mixer"},
+ {"AUX1 Output Mixer", "LDAC Switch", "Left DAC"},
+ {"AUX1 Output Mixer", "LMIX Switch", "Left Output Mixer"},
+
+ {"AUX2 Output Mixer", "LDAC Switch", "Left DAC"},
+ {"AUX2 Output Mixer", "LMIX Switch", "Left Output Mixer"},
+ {"AUX2 Output Mixer", "LINMIX Switch", "Left Boost Mixer"},
+ {"AUX2 Output Mixer", "AUX1MIX Output Switch", "AUX1 Output Mixer"},
+
+ /* Outputs */
+ {"Right Headphone Out", NULL, "Right Output Mixer"},
+ {"RHP", NULL, "Right Headphone Out"},
+
+ {"Left Headphone Out", NULL, "Left Output Mixer"},
+ {"LHP", NULL, "Left Headphone Out"},
+
+ {"Right Speaker Out", NULL, "Right Output Mixer"},
+ {"RSPK", NULL, "Right Speaker Out"},
+
+ {"Left Speaker Out", NULL, "Left Output Mixer"},
+ {"LSPK", NULL, "Left Speaker Out"},
+
+ {"AUX1 Out", NULL, "AUX1 Output Mixer"},
+ {"AUX2 Out", NULL, "AUX2 Output Mixer"},
+ {"AUXOUT1", NULL, "AUX1 Out"},
+ {"AUXOUT2", NULL, "AUX2 Out"},
+
+ /* Boost Mixer */
+ {"Right ADC", NULL, "PLL", check_mclk_select_pll},
+ {"Left ADC", NULL, "PLL", check_mclk_select_pll},
+
+ {"Right ADC", NULL, "Right Boost Mixer"},
+
+ {"Right Boost Mixer", NULL, "RAUX"},
+ {"Right Boost Mixer", NULL, "Right Capture PGA"},
+ {"Right Boost Mixer", NULL, "R2"},
+
+ {"Left ADC", NULL, "Left Boost Mixer"},
+
+ {"Left Boost Mixer", NULL, "LAUX"},
+ {"Left Boost Mixer", NULL, "Left Capture PGA"},
+ {"Left Boost Mixer", NULL, "L2"},
+
+ /* Input PGA */
+ {"Right Capture PGA", NULL, "Right Input Mixer"},
+ {"Left Capture PGA", NULL, "Left Input Mixer"},
+
+ /* Enable Microphone Power */
+ {"Right Capture PGA", NULL, "Mic Bias"},
+ {"Left Capture PGA", NULL, "Mic Bias"},
+
+ {"Right Input Mixer", "R2 Switch", "R2"},
+ {"Right Input Mixer", "MicN Switch", "RMICN"},
+ {"Right Input Mixer", "MicP Switch", "RMICP"},
+
+ {"Left Input Mixer", "L2 Switch", "L2"},
+ {"Left Input Mixer", "MicN Switch", "LMICN"},
+ {"Left Input Mixer", "MicP Switch", "LMICP"},
+
+ /* Digital Loopback */
+ {"Digital Loopback", "Switch", "Left ADC"},
+ {"Digital Loopback", "Switch", "Right ADC"},
+ {"Left DAC", NULL, "Digital Loopback"},
+ {"Right DAC", NULL, "Digital Loopback"},
+};
+
+static int nau8822_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct snd_soc_component *component = dai->component;
+ struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+
+ nau8822->div_id = clk_id;
+ nau8822->sysclk = freq;
+ dev_dbg(component->dev, "master sysclk %dHz, source %s\n", freq,
+ clk_id == NAU8822_CLK_PLL ? "PLL" : "MCLK");
+
+ return 0;
+}
+
+static int nau8822_calc_pll(unsigned int pll_in, unsigned int fs,
+ struct nau8822_pll *pll_param)
+{
+ u64 f2, f2_max, pll_ratio;
+ int i, scal_sel;
+
+ if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN)
+ return -EINVAL;
+ f2_max = 0;
+ scal_sel = ARRAY_SIZE(nau8822_mclk_scaler);
+
+ for (i = 0; i < scal_sel; i++) {
+ f2 = 256 * fs * 4 * nau8822_mclk_scaler[i] / 10;
+ if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX &&
+ f2_max < f2) {
+ f2_max = f2;
+ scal_sel = i;
+ }
+ }
+
+ if (ARRAY_SIZE(nau8822_mclk_scaler) == scal_sel)
+ return -EINVAL;
+ pll_param->mclk_scaler = scal_sel;
+ f2 = f2_max;
+
+ /* Calculate the PLL 4-bit integer input and the PLL 24-bit fractional
+ * input; round up the 24+4bit.
+ */
+ pll_ratio = div_u64(f2 << 28, pll_in);
+ pll_param->pre_factor = 0;
+ if (((pll_ratio >> 28) & 0xF) < NAU_PLL_OPTOP_MIN) {
+ pll_ratio <<= 1;
+ pll_param->pre_factor = 1;
+ }
+ pll_param->pll_int = (pll_ratio >> 28) & 0xF;
+ pll_param->pll_frac = ((pll_ratio & 0xFFFFFFF) >> 4);
+
+ return 0;
+}
+
+static int nau8822_config_clkdiv(struct snd_soc_dai *dai, int div, int rate)
+{
+ struct snd_soc_component *component = dai->component;
+ struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+ struct nau8822_pll *pll = &nau8822->pll;
+ int i, sclk, imclk;
+
+ switch (nau8822->div_id) {
+ case NAU8822_CLK_MCLK:
+ /* Configure the master clock prescaler div to make system
+ * clock to approximate the internal master clock (IMCLK);
+ * and large or equal to IMCLK.
+ */
+ div = 0;
+ imclk = rate * 256;
+ for (i = 1; i < ARRAY_SIZE(nau8822_mclk_scaler); i++) {
+ sclk = (nau8822->sysclk * 10) / nau8822_mclk_scaler[i];
+ if (sclk < imclk)
+ break;
+ div = i;
+ }
+ dev_dbg(component->dev, "master clock prescaler %x for fs %d\n",
+ div, rate);
+
+ /* master clock from MCLK and disable PLL */
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK,
+ (div << NAU8822_MCLKSEL_SFT));
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK,
+ NAU8822_CLKM_MCLK);
+ break;
+
+ case NAU8822_CLK_PLL:
+ /* master clock from PLL and enable PLL */
+ if (pll->mclk_scaler != div) {
+ dev_err(component->dev,
+ "master clock prescaler not meet PLL parameters\n");
+ return -EINVAL;
+ }
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK,
+ (div << NAU8822_MCLKSEL_SFT));
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK,
+ NAU8822_CLKM_PLL);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
+ unsigned int freq_in, unsigned int freq_out)
+{
+ struct snd_soc_component *component = dai->component;
+ struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+ struct nau8822_pll *pll_param = &nau8822->pll;
+ int ret, fs;
+
+ fs = freq_out / 256;
+
+ ret = nau8822_calc_pll(freq_in, fs, pll_param);
+ if (ret < 0) {
+ dev_err(component->dev, "Unsupported input clock %d\n",
+ freq_in);
+ return ret;
+ }
+
+ dev_info(component->dev,
+ "pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n",
+ pll_param->pll_int, pll_param->pll_frac,
+ pll_param->mclk_scaler, pll_param->pre_factor);
+
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK,
+ (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) |
+ pll_param->pll_int);
+ snd_soc_component_write(component,
+ NAU8822_REG_PLL_K1, (pll_param->pll_frac >> NAU8822_PLLK1_SFT) &
+ NAU8822_PLLK1_MASK);
+ snd_soc_component_write(component,
+ NAU8822_REG_PLL_K2, (pll_param->pll_frac >> NAU8822_PLLK2_SFT) &
+ NAU8822_PLLK2_MASK);
+ snd_soc_component_write(component,
+ NAU8822_REG_PLL_K3, pll_param->pll_frac & NAU8822_PLLK3_MASK);
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK,
+ pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT);
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL);
+
+ return 0;
+}
+
+static int nau8822_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_component *component = dai->component;
+ u16 ctrl1_val = 0, ctrl2_val = 0;
+
+ dev_dbg(component->dev, "%s\n", __func__);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ ctrl2_val |= 1;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ ctrl2_val &= ~1;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ ctrl1_val |= 0x10;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ ctrl1_val |= 0x8;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ ctrl1_val |= 0x18;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ ctrl1_val |= 0x180;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ ctrl1_val |= 0x100;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ ctrl1_val |= 0x80;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_AUDIO_INTERFACE,
+ NAU8822_AIFMT_MASK | NAU8822_LRP_MASK | NAU8822_BCLKP_MASK,
+ ctrl1_val);
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_CLOCKING, NAU8822_CLKIOEN_MASK, ctrl2_val);
+
+ return 0;
+}
+
+static int nau8822_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+ int val_len = 0, val_rate = 0;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val_len |= NAU8822_WLEN_20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val_len |= NAU8822_WLEN_24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ val_len |= NAU8822_WLEN_32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (params_rate(params)) {
+ case 8000:
+ val_rate |= NAU8822_SMPLR_8K;
+ break;
+ case 11025:
+ val_rate |= NAU8822_SMPLR_12K;
+ break;
+ case 16000:
+ val_rate |= NAU8822_SMPLR_16K;
+ break;
+ case 22050:
+ val_rate |= NAU8822_SMPLR_24K;
+ break;
+ case 32000:
+ val_rate |= NAU8822_SMPLR_32K;
+ break;
+ case 44100:
+ case 48000:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_AUDIO_INTERFACE, NAU8822_WLEN_MASK, val_len);
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_ADDITIONAL_CONTROL, NAU8822_SMPLR_MASK, val_rate);
+
+ /* If the master clock is from MCLK, provide the runtime FS for driver
+ * to get the master clock prescaler configuration.
+ */
+ if (nau8822->div_id == NAU8822_CLK_MCLK)
+ nau8822_config_clkdiv(dai, 0, params_rate(params));
+
+ return 0;
+}
+
+static int nau8822_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_component *component = dai->component;
+
+ dev_dbg(component->dev, "%s: %d\n", __func__, mute);
+
+ if (mute)
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_DAC_CONTROL, 0x40, 0x40);
+ else
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_DAC_CONTROL, 0x40, 0);
+
+ return 0;
+}
+
+static int nau8822_set_bias_level(struct snd_soc_component *component,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_POWER_MANAGEMENT_1,
+ NAU8822_REFIMP_MASK, NAU8822_REFIMP_80K);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_POWER_MANAGEMENT_1,
+ NAU8822_IOBUF_EN | NAU8822_ABIAS_EN,
+ NAU8822_IOBUF_EN | NAU8822_ABIAS_EN);
+
+ if (snd_soc_component_get_bias_level(component) ==
+ SND_SOC_BIAS_OFF) {
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_POWER_MANAGEMENT_1,
+ NAU8822_REFIMP_MASK, NAU8822_REFIMP_3K);
+ mdelay(100);
+ }
+ snd_soc_component_update_bits(component,
+ NAU8822_REG_POWER_MANAGEMENT_1,
+ NAU8822_REFIMP_MASK, NAU8822_REFIMP_300K);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ snd_soc_component_write(component,
+ NAU8822_REG_POWER_MANAGEMENT_1, 0);
+ snd_soc_component_write(component,
+ NAU8822_REG_POWER_MANAGEMENT_2, 0);
+ snd_soc_component_write(component,
+ NAU8822_REG_POWER_MANAGEMENT_3, 0);
+ break;
+ }
+
+ dev_dbg(component->dev, "%s: %d\n", __func__, level);
+
+ return 0;
+}
+
+#define NAU8822_RATES (SNDRV_PCM_RATE_8000_48000)
+
+#define NAU8822_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static const struct snd_soc_dai_ops nau8822_dai_ops = {
+ .hw_params = nau8822_hw_params,
+ .digital_mute = nau8822_mute,
+ .set_fmt = nau8822_set_dai_fmt,
+ .set_sysclk = nau8822_set_dai_sysclk,
+ .set_pll = nau8822_set_pll,
+};
+
+static struct snd_soc_dai_driver nau8822_dai = {
+ .name = "nau8822-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = NAU8822_RATES,
+ .formats = NAU8822_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = NAU8822_RATES,
+ .formats = NAU8822_FORMATS,
+ },
+ .ops = &nau8822_dai_ops,
+ .symmetric_rates = 1,
+};
+
+static int nau8822_suspend(struct snd_soc_component *component)
+{
+ struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+
+ snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF);
+
+ regcache_mark_dirty(nau8822->regmap);
+
+ return 0;
+}
+
+static int nau8822_resume(struct snd_soc_component *component)
+{
+ struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+
+ regcache_sync(nau8822->regmap);
+
+ snd_soc_component_force_bias_level(component, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+
+/*
+ * These registers contain an "update" bit - bit 8. This means, for example,
+ * that one can write new DAC digital volume for both channels, but only when
+ * the update bit is set, will also the volume be updated - simultaneously for
+ * both channels.
+ */
+static const int update_reg[] = {
+ NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME,
+ NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME,
+ NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME,
+ NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME,
+ NAU8822_REG_LEFT_INP_PGA_CONTROL,
+ NAU8822_REG_RIGHT_INP_PGA_CONTROL,
+ NAU8822_REG_LHP_VOLUME,
+ NAU8822_REG_RHP_VOLUME,
+ NAU8822_REG_LSPKOUT_VOLUME,
+ NAU8822_REG_RSPKOUT_VOLUME,
+};
+
+static int nau8822_probe(struct snd_soc_component *component)
+{
+ int i;
+
+ /*
+ * Set the update bit in all registers, that have one. This way all
+ * writes to those registers will also cause the update bit to be
+ * written.
+ */
+ for (i = 0; i < ARRAY_SIZE(update_reg); i++)
+ snd_soc_component_update_bits(component,
+ update_reg[i], 0x100, 0x100);
+
+ return 0;
+}
+
+static const struct snd_soc_component_driver soc_component_dev_nau8822 = {
+ .probe = nau8822_probe,
+ .suspend = nau8822_suspend,
+ .resume = nau8822_resume,
+ .set_bias_level = nau8822_set_bias_level,
+ .controls = nau8822_snd_controls,
+ .num_controls = ARRAY_SIZE(nau8822_snd_controls),
+ .dapm_widgets = nau8822_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(nau8822_dapm_widgets),
+ .dapm_routes = nau8822_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(nau8822_dapm_routes),
+ .idle_bias_on = 1,
+ .use_pmdown_time = 1,
+ .endianness = 1,
+ .non_legacy_dai_naming = 1,
+};
+
+static const struct regmap_config nau8822_regmap_config = {
+ .reg_bits = 7,
+ .val_bits = 9,
+
+ .max_register = NAU8822_REG_MAX_REGISTER,
+ .volatile_reg = nau8822_volatile,
+
+ .readable_reg = nau8822_readable_reg,
+ .writeable_reg = nau8822_writeable_reg,
+
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults = nau8822_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(nau8822_reg_defaults),
+};
+
+static int nau8822_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct device *dev = &i2c->dev;
+ struct nau8822 *nau8822 = dev_get_platdata(dev);
+ int ret;
+
+ if (!nau8822) {
+ nau8822 = devm_kzalloc(dev, sizeof(*nau8822), GFP_KERNEL);
+ if (nau8822 == NULL)
+ return -ENOMEM;
+ }
+ i2c_set_clientdata(i2c, nau8822);
+
+ nau8822->regmap = devm_regmap_init_i2c(i2c, &nau8822_regmap_config);
+ if (IS_ERR(nau8822->regmap)) {
+ ret = PTR_ERR(nau8822->regmap);
+ dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
+ return ret;
+ }
+ nau8822->dev = dev;
+
+ /* Reset the codec */
+ ret = regmap_write(nau8822->regmap, NAU8822_REG_RESET, 0x00);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
+ return ret;
+ }
+
+ ret = devm_snd_soc_register_component(dev, &soc_component_dev_nau8822,
+ &nau8822_dai, 1);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static const struct i2c_device_id nau8822_i2c_id[] = {
+ { "nau8822", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, nau8822_i2c_id);
+
+#ifdef CONFIG_OF
+static const struct of_device_id nau8822_of_match[] = {
+ { .compatible = "nuvoton,nau8822", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, nau8822_of_match);
+#endif
+
+static struct i2c_driver nau8822_i2c_driver = {
+ .driver = {
+ .name = "nau8822",
+ .of_match_table = of_match_ptr(nau8822_of_match),
+ },
+ .probe = nau8822_i2c_probe,
+ .id_table = nau8822_i2c_id,
+};
+module_i2c_driver(nau8822_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC NAU8822 codec driver");
+MODULE_AUTHOR("David Lin <ctlin0@nuvoton.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h
new file mode 100644
index 000000000000..aa79c969cd44
--- /dev/null
+++ b/sound/soc/codecs/nau8822.h
@@ -0,0 +1,204 @@
+/*
+ * nau8822.h -- NAU8822 Soc Audio Codec driver
+ *
+ * Author: David Lin <ctlin0@nuvoton.com>
+ * Co-author: John Hsu <kchsu0@nuvoton.com>
+ * Co-author: Seven Li <wtli@nuvoton.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 __NAU8822_H__
+#define __NAU8822_H__
+
+#define NAU8822_REG_RESET 0x00
+#define NAU8822_REG_POWER_MANAGEMENT_1 0x01
+#define NAU8822_REG_POWER_MANAGEMENT_2 0x02
+#define NAU8822_REG_POWER_MANAGEMENT_3 0x03
+#define NAU8822_REG_AUDIO_INTERFACE 0x04
+#define NAU8822_REG_COMPANDING_CONTROL 0x05
+#define NAU8822_REG_CLOCKING 0x06
+#define NAU8822_REG_ADDITIONAL_CONTROL 0x07
+#define NAU8822_REG_GPIO_CONTROL 0x08
+#define NAU8822_REG_JACK_DETECT_CONTROL_1 0x09
+#define NAU8822_REG_DAC_CONTROL 0x0A
+#define NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME 0x0B
+#define NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME 0x0C
+#define NAU8822_REG_JACK_DETECT_CONTROL_2 0x0D
+#define NAU8822_REG_ADC_CONTROL 0x0E
+#define NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME 0x0F
+#define NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME 0x10
+#define NAU8822_REG_EQ1 0x12
+#define NAU8822_REG_EQ2 0x13
+#define NAU8822_REG_EQ3 0x14
+#define NAU8822_REG_EQ4 0x15
+#define NAU8822_REG_EQ5 0x16
+#define NAU8822_REG_DAC_LIMITER_1 0x18
+#define NAU8822_REG_DAC_LIMITER_2 0x19
+#define NAU8822_REG_NOTCH_FILTER_1 0x1B
+#define NAU8822_REG_NOTCH_FILTER_2 0x1C
+#define NAU8822_REG_NOTCH_FILTER_3 0x1D
+#define NAU8822_REG_NOTCH_FILTER_4 0x1E
+#define NAU8822_REG_ALC_CONTROL_1 0x20
+#define NAU8822_REG_ALC_CONTROL_2 0x21
+#define NAU8822_REG_ALC_CONTROL_3 0x22
+#define NAU8822_REG_NOISE_GATE 0x23
+#define NAU8822_REG_PLL_N 0x24
+#define NAU8822_REG_PLL_K1 0x25
+#define NAU8822_REG_PLL_K2 0x26
+#define NAU8822_REG_PLL_K3 0x27
+#define NAU8822_REG_3D_CONTROL 0x29
+#define NAU8822_REG_RIGHT_SPEAKER_CONTROL 0x2B
+#define NAU8822_REG_INPUT_CONTROL 0x2C
+#define NAU8822_REG_LEFT_INP_PGA_CONTROL 0x2D
+#define NAU8822_REG_RIGHT_INP_PGA_CONTROL 0x2E
+#define NAU8822_REG_LEFT_ADC_BOOST_CONTROL 0x2F
+#define NAU8822_REG_RIGHT_ADC_BOOST_CONTROL 0x30
+#define NAU8822_REG_OUTPUT_CONTROL 0x31
+#define NAU8822_REG_LEFT_MIXER_CONTROL 0x32
+#define NAU8822_REG_RIGHT_MIXER_CONTROL 0x33
+#define NAU8822_REG_LHP_VOLUME 0x34
+#define NAU8822_REG_RHP_VOLUME 0x35
+#define NAU8822_REG_LSPKOUT_VOLUME 0x36
+#define NAU8822_REG_RSPKOUT_VOLUME 0x37
+#define NAU8822_REG_AUX2_MIXER 0x38
+#define NAU8822_REG_AUX1_MIXER 0x39
+#define NAU8822_REG_POWER_MANAGEMENT_4 0x3A
+#define NAU8822_REG_LEFT_TIME_SLOT 0x3B
+#define NAU8822_REG_MISC 0x3C
+#define NAU8822_REG_RIGHT_TIME_SLOT 0x3D
+#define NAU8822_REG_DEVICE_REVISION 0x3E
+#define NAU8822_REG_DEVICE_ID 0x3F
+#define NAU8822_REG_DAC_DITHER 0x41
+#define NAU8822_REG_ALC_ENHANCE_1 0x46
+#define NAU8822_REG_ALC_ENHANCE_2 0x47
+#define NAU8822_REG_192KHZ_SAMPLING 0x48
+#define NAU8822_REG_MISC_CONTROL 0x49
+#define NAU8822_REG_INPUT_TIEOFF 0x4A
+#define NAU8822_REG_POWER_REDUCTION 0x4B
+#define NAU8822_REG_AGC_PEAK2PEAK 0x4C
+#define NAU8822_REG_AGC_PEAK_DETECT 0x4D
+#define NAU8822_REG_AUTOMUTE_CONTROL 0x4E
+#define NAU8822_REG_OUTPUT_TIEOFF 0x4F
+#define NAU8822_REG_MAX_REGISTER NAU8822_REG_OUTPUT_TIEOFF
+
+/* NAU8822_REG_POWER_MANAGEMENT_1 (0x1) */
+#define NAU8822_REFIMP_MASK 0x3
+#define NAU8822_REFIMP_80K 0x1
+#define NAU8822_REFIMP_300K 0x2
+#define NAU8822_REFIMP_3K 0x3
+#define NAU8822_IOBUF_EN (0x1 << 2)
+#define NAU8822_ABIAS_EN (0x1 << 3)
+
+/* NAU8822_REG_AUDIO_INTERFACE (0x4) */
+#define NAU8822_AIFMT_MASK (0x3 << 3)
+#define NAU8822_WLEN_MASK (0x3 << 5)
+#define NAU8822_WLEN_20 (0x1 << 5)
+#define NAU8822_WLEN_24 (0x2 << 5)
+#define NAU8822_WLEN_32 (0x3 << 5)
+#define NAU8822_LRP_MASK (0x1 << 7)
+#define NAU8822_BCLKP_MASK (0x1 << 8)
+
+/* NAU8822_REG_COMPANDING_CONTROL (0x5) */
+#define NAU8822_ADDAP_SFT 0
+#define NAU8822_ADCCM_SFT 1
+#define NAU8822_DACCM_SFT 3
+
+/* NAU8822_REG_CLOCKING (0x6) */
+#define NAU8822_CLKIOEN_MASK 0x1
+#define NAU8822_MCLKSEL_SFT 5
+#define NAU8822_MCLKSEL_MASK (0x7 << 5)
+#define NAU8822_BCLKSEL_SFT 2
+#define NAU8822_BCLKSEL_MASK (0x7 << 2)
+#define NAU8822_CLKM_MASK (0x1 << 8)
+#define NAU8822_CLKM_MCLK (0x0 << 8)
+#define NAU8822_CLKM_PLL (0x1 << 8)
+
+/* NAU8822_REG_ADDITIONAL_CONTROL (0x08) */
+#define NAU8822_SMPLR_SFT 1
+#define NAU8822_SMPLR_MASK (0x7 << 1)
+#define NAU8822_SMPLR_48K (0x0 << 1)
+#define NAU8822_SMPLR_32K (0x1 << 1)
+#define NAU8822_SMPLR_24K (0x2 << 1)
+#define NAU8822_SMPLR_16K (0x3 << 1)
+#define NAU8822_SMPLR_12K (0x4 << 1)
+#define NAU8822_SMPLR_8K (0x5 << 1)
+
+/* NAU8822_REG_EQ1 (0x12) */
+#define NAU8822_EQ1GC_SFT 0
+#define NAU8822_EQ1CF_SFT 5
+#define NAU8822_EQM_SFT 8
+
+/* NAU8822_REG_EQ2 (0x13) */
+#define NAU8822_EQ2GC_SFT 0
+#define NAU8822_EQ2CF_SFT 5
+#define NAU8822_EQ2BW_SFT 8
+
+/* NAU8822_REG_EQ3 (0x14) */
+#define NAU8822_EQ3GC_SFT 0
+#define NAU8822_EQ3CF_SFT 5
+#define NAU8822_EQ3BW_SFT 8
+
+/* NAU8822_REG_EQ4 (0x15) */
+#define NAU8822_EQ4GC_SFT 0
+#define NAU8822_EQ4CF_SFT 5
+#define NAU8822_EQ4BW_SFT 8
+
+/* NAU8822_REG_EQ5 (0x16) */
+#define NAU8822_EQ5GC_SFT 0
+#define NAU8822_EQ5CF_SFT 5
+
+/* NAU8822_REG_ALC_CONTROL_1 (0x20) */
+#define NAU8822_ALCMINGAIN_SFT 0
+#define NAU8822_ALCMXGAIN_SFT 3
+#define NAU8822_ALCEN_SFT 7
+
+/* NAU8822_REG_ALC_CONTROL_2 (0x21) */
+#define NAU8822_ALCSL_SFT 0
+#define NAU8822_ALCHT_SFT 4
+
+/* NAU8822_REG_ALC_CONTROL_3 (0x22) */
+#define NAU8822_ALCATK_SFT 0
+#define NAU8822_ALCDCY_SFT 4
+#define NAU8822_ALCM_SFT 8
+
+/* NAU8822_REG_PLL_N (0x24) */
+#define NAU8822_PLLMCLK_DIV2 (0x1 << 4)
+#define NAU8822_PLLN_MASK 0xF
+
+#define NAU8822_PLLK1_SFT 18
+#define NAU8822_PLLK1_MASK 0x3F
+
+/* NAU8822_REG_PLL_K2 (0x26) */
+#define NAU8822_PLLK2_SFT 9
+#define NAU8822_PLLK2_MASK 0x1FF
+
+/* NAU8822_REG_PLL_K3 (0x27) */
+#define NAU8822_PLLK3_MASK 0x1FF
+
+/* System Clock Source */
+enum {
+ NAU8822_CLK_MCLK,
+ NAU8822_CLK_PLL,
+};
+
+struct nau8822_pll {
+ int pre_factor;
+ int mclk_scaler;
+ int pll_frac;
+ int pll_int;
+};
+
+/* Codec Private Data */
+struct nau8822 {
+ struct device *dev;
+ struct regmap *regmap;
+ int mclk_idx;
+ struct nau8822_pll pll;
+ int sysclk;
+ int div_id;
+};
+
+#endif /* __NAU8822_H__ */
diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c
index 690c26e7389e..809b7e9f03ca 100644
--- a/sound/soc/codecs/pcm186x.c
+++ b/sound/soc/codecs/pcm186x.c
@@ -401,7 +401,8 @@ static int pcm186x_set_fmt(struct snd_soc_dai *dai, unsigned int format)
break;
case SND_SOC_DAIFMT_DSP_A:
priv->tdm_offset += 1;
- /* Fall through... DSP_A uses the same basic config as DSP_B
+ /* fall through */
+ /* DSP_A uses the same basic config as DSP_B
* except we need to shift the TDM output by one BCK cycle
*/
case SND_SOC_DAIFMT_DSP_B:
diff --git a/sound/soc/codecs/pcm3060-i2c.c b/sound/soc/codecs/pcm3060-i2c.c
new file mode 100644
index 000000000000..cdc8314882bc
--- /dev/null
+++ b/sound/soc/codecs/pcm3060-i2c.c
@@ -0,0 +1,60 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 I2C driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <sound/soc.h>
+
+#include "pcm3060.h"
+
+static int pcm3060_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct pcm3060_priv *priv;
+
+ priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, priv);
+
+ priv->regmap = devm_regmap_init_i2c(i2c, &pcm3060_regmap);
+ if (IS_ERR(priv->regmap))
+ return PTR_ERR(priv->regmap);
+
+ return pcm3060_probe(&i2c->dev);
+}
+
+static const struct i2c_device_id pcm3060_i2c_id[] = {
+ { .name = "pcm3060" },
+ { },
+};
+MODULE_DEVICE_TABLE(i2c, pcm3060_i2c_id);
+
+#ifdef CONFIG_OF
+static const struct of_device_id pcm3060_of_match[] = {
+ { .compatible = "ti,pcm3060" },
+ { },
+};
+MODULE_DEVICE_TABLE(of, pcm3060_of_match);
+#endif /* CONFIG_OF */
+
+static struct i2c_driver pcm3060_i2c_driver = {
+ .driver = {
+ .name = "pcm3060",
+#ifdef CONFIG_OF
+ .of_match_table = pcm3060_of_match,
+#endif /* CONFIG_OF */
+ },
+ .id_table = pcm3060_i2c_id,
+ .probe = pcm3060_i2c_probe,
+};
+
+module_i2c_driver(pcm3060_i2c_driver);
+
+MODULE_DESCRIPTION("PCM3060 I2C driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060-spi.c b/sound/soc/codecs/pcm3060-spi.c
new file mode 100644
index 000000000000..f6f19fa80932
--- /dev/null
+++ b/sound/soc/codecs/pcm3060-spi.c
@@ -0,0 +1,59 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 SPI driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+
+#include "pcm3060.h"
+
+static int pcm3060_spi_probe(struct spi_device *spi)
+{
+ struct pcm3060_priv *priv;
+
+ priv = devm_kzalloc(&spi->dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ spi_set_drvdata(spi, priv);
+
+ priv->regmap = devm_regmap_init_spi(spi, &pcm3060_regmap);
+ if (IS_ERR(priv->regmap))
+ return PTR_ERR(priv->regmap);
+
+ return pcm3060_probe(&spi->dev);
+}
+
+static const struct spi_device_id pcm3060_spi_id[] = {
+ { .name = "pcm3060" },
+ { },
+};
+MODULE_DEVICE_TABLE(spi, pcm3060_spi_id);
+
+#ifdef CONFIG_OF
+static const struct of_device_id pcm3060_of_match[] = {
+ { .compatible = "ti,pcm3060" },
+ { },
+};
+MODULE_DEVICE_TABLE(of, pcm3060_of_match);
+#endif /* CONFIG_OF */
+
+static struct spi_driver pcm3060_spi_driver = {
+ .driver = {
+ .name = "pcm3060",
+#ifdef CONFIG_OF
+ .of_match_table = pcm3060_of_match,
+#endif /* CONFIG_OF */
+ },
+ .id_table = pcm3060_spi_id,
+ .probe = pcm3060_spi_probe,
+};
+
+module_spi_driver(pcm3060_spi_driver);
+
+MODULE_DESCRIPTION("PCM3060 SPI driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c
new file mode 100644
index 000000000000..494d9d662be8
--- /dev/null
+++ b/sound/soc/codecs/pcm3060.c
@@ -0,0 +1,295 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 codec driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/module.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "pcm3060.h"
+
+/* dai */
+
+static int pcm3060_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct snd_soc_component *comp = dai->component;
+ struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+
+ if (dir != SND_SOC_CLOCK_IN) {
+ dev_err(comp->dev, "unsupported sysclock dir: %d\n", dir);
+ return -EINVAL;
+ }
+
+ priv->dai[dai->id].sclk_freq = freq;
+
+ return 0;
+}
+
+static int pcm3060_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_component *comp = dai->component;
+ struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+ unsigned int reg;
+ unsigned int val;
+
+ if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
+ dev_err(comp->dev, "unsupported DAI polarity: 0x%x\n", fmt);
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ priv->dai[dai->id].is_master = true;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ priv->dai[dai->id].is_master = false;
+ break;
+ default:
+ dev_err(comp->dev, "unsupported DAI master mode: 0x%x\n", fmt);
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ val = PCM3060_REG_FMT_I2S;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ val = PCM3060_REG_FMT_RJ;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ val = PCM3060_REG_FMT_LJ;
+ break;
+ default:
+ dev_err(comp->dev, "unsupported DAI format: 0x%x\n", fmt);
+ return -EINVAL;
+ }
+
+ if (dai->id == PCM3060_DAI_ID_DAC)
+ reg = PCM3060_REG67;
+ else
+ reg = PCM3060_REG72;
+
+ regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_FMT, val);
+
+ return 0;
+}
+
+static int pcm3060_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *comp = dai->component;
+ struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+ unsigned int rate;
+ unsigned int ratio;
+ unsigned int reg;
+ unsigned int val;
+
+ if (!priv->dai[dai->id].is_master) {
+ val = PCM3060_REG_MS_S;
+ goto val_ready;
+ }
+
+ rate = params_rate(params);
+ if (!rate) {
+ dev_err(comp->dev, "rate is not configured\n");
+ return -EINVAL;
+ }
+
+ ratio = priv->dai[dai->id].sclk_freq / rate;
+
+ switch (ratio) {
+ case 768:
+ val = PCM3060_REG_MS_M768;
+ break;
+ case 512:
+ val = PCM3060_REG_MS_M512;
+ break;
+ case 384:
+ val = PCM3060_REG_MS_M384;
+ break;
+ case 256:
+ val = PCM3060_REG_MS_M256;
+ break;
+ case 192:
+ val = PCM3060_REG_MS_M192;
+ break;
+ case 128:
+ val = PCM3060_REG_MS_M128;
+ break;
+ default:
+ dev_err(comp->dev, "unsupported ratio: %d\n", ratio);
+ return -EINVAL;
+ }
+
+val_ready:
+ if (dai->id == PCM3060_DAI_ID_DAC)
+ reg = PCM3060_REG67;
+ else
+ reg = PCM3060_REG72;
+
+ regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_MS, val);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops pcm3060_dai_ops = {
+ .set_sysclk = pcm3060_set_sysclk,
+ .set_fmt = pcm3060_set_fmt,
+ .hw_params = pcm3060_hw_params,
+};
+
+#define PCM3060_DAI_RATES_ADC (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define PCM3060_DAI_RATES_DAC (PCM3060_DAI_RATES_ADC | \
+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
+static struct snd_soc_dai_driver pcm3060_dai[] = {
+ {
+ .name = "pcm3060-dac",
+ .id = PCM3060_DAI_ID_DAC,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PCM3060_DAI_RATES_DAC,
+ .formats = SNDRV_PCM_FMTBIT_S24_LE,
+ },
+ .ops = &pcm3060_dai_ops,
+ },
+ {
+ .name = "pcm3060-adc",
+ .id = PCM3060_DAI_ID_ADC,
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PCM3060_DAI_RATES_ADC,
+ .formats = SNDRV_PCM_FMTBIT_S24_LE,
+ },
+ .ops = &pcm3060_dai_ops,
+ },
+};
+
+/* dapm */
+
+static DECLARE_TLV_DB_SCALE(pcm3060_dapm_tlv, -10050, 50, 1);
+
+static const struct snd_kcontrol_new pcm3060_dapm_controls[] = {
+ SOC_DOUBLE_R_RANGE_TLV("Master Playback Volume",
+ PCM3060_REG65, PCM3060_REG66, 0,
+ PCM3060_REG_AT2_MIN, PCM3060_REG_AT2_MAX,
+ 0, pcm3060_dapm_tlv),
+ SOC_DOUBLE("Master Playback Switch", PCM3060_REG68,
+ PCM3060_REG_SHIFT_MUT21, PCM3060_REG_SHIFT_MUT22, 1, 1),
+
+ SOC_DOUBLE_R_RANGE_TLV("Master Capture Volume",
+ PCM3060_REG70, PCM3060_REG71, 0,
+ PCM3060_REG_AT1_MIN, PCM3060_REG_AT1_MAX,
+ 0, pcm3060_dapm_tlv),
+ SOC_DOUBLE("Master Capture Switch", PCM3060_REG73,
+ PCM3060_REG_SHIFT_MUT11, PCM3060_REG_SHIFT_MUT12, 1, 1),
+};
+
+static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = {
+ SND_SOC_DAPM_OUTPUT("OUTL+"),
+ SND_SOC_DAPM_OUTPUT("OUTR+"),
+ SND_SOC_DAPM_OUTPUT("OUTL-"),
+ SND_SOC_DAPM_OUTPUT("OUTR-"),
+
+ SND_SOC_DAPM_INPUT("INL"),
+ SND_SOC_DAPM_INPUT("INR"),
+};
+
+static const struct snd_soc_dapm_route pcm3060_dapm_map[] = {
+ { "OUTL+", NULL, "Playback" },
+ { "OUTR+", NULL, "Playback" },
+ { "OUTL-", NULL, "Playback" },
+ { "OUTR-", NULL, "Playback" },
+
+ { "Capture", NULL, "INL" },
+ { "Capture", NULL, "INR" },
+};
+
+/* soc component */
+
+static const struct snd_soc_component_driver pcm3060_soc_comp_driver = {
+ .controls = pcm3060_dapm_controls,
+ .num_controls = ARRAY_SIZE(pcm3060_dapm_controls),
+ .dapm_widgets = pcm3060_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(pcm3060_dapm_widgets),
+ .dapm_routes = pcm3060_dapm_map,
+ .num_dapm_routes = ARRAY_SIZE(pcm3060_dapm_map),
+};
+
+/* regmap */
+
+static bool pcm3060_reg_writeable(struct device *dev, unsigned int reg)
+{
+ return (reg >= PCM3060_REG64);
+}
+
+static bool pcm3060_reg_readable(struct device *dev, unsigned int reg)
+{
+ return (reg >= PCM3060_REG64);
+}
+
+static bool pcm3060_reg_volatile(struct device *dev, unsigned int reg)
+{
+ /* PCM3060_REG64 is volatile */
+ return (reg == PCM3060_REG64);
+}
+
+static const struct reg_default pcm3060_reg_defaults[] = {
+ { PCM3060_REG64, 0xF0 },
+ { PCM3060_REG65, 0xFF },
+ { PCM3060_REG66, 0xFF },
+ { PCM3060_REG67, 0x00 },
+ { PCM3060_REG68, 0x00 },
+ { PCM3060_REG69, 0x00 },
+ { PCM3060_REG70, 0xD7 },
+ { PCM3060_REG71, 0xD7 },
+ { PCM3060_REG72, 0x00 },
+ { PCM3060_REG73, 0x00 },
+};
+
+const struct regmap_config pcm3060_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .writeable_reg = pcm3060_reg_writeable,
+ .readable_reg = pcm3060_reg_readable,
+ .volatile_reg = pcm3060_reg_volatile,
+ .max_register = PCM3060_REG73,
+ .reg_defaults = pcm3060_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(pcm3060_reg_defaults),
+ .cache_type = REGCACHE_RBTREE,
+};
+EXPORT_SYMBOL(pcm3060_regmap);
+
+/* device */
+
+int pcm3060_probe(struct device *dev)
+{
+ int rc;
+
+ rc = devm_snd_soc_register_component(dev, &pcm3060_soc_comp_driver,
+ pcm3060_dai,
+ ARRAY_SIZE(pcm3060_dai));
+ if (rc) {
+ dev_err(dev, "failed to register component, rc=%d\n", rc);
+ return rc;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL(pcm3060_probe);
+
+MODULE_DESCRIPTION("PCM3060 codec driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h
new file mode 100644
index 000000000000..fd89a68aa8a7
--- /dev/null
+++ b/sound/soc/codecs/pcm3060.h
@@ -0,0 +1,88 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * PCM3060 codec driver
+ *
+ * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+ */
+
+#ifndef _SND_SOC_PCM3060_H
+#define _SND_SOC_PCM3060_H
+
+#include <linux/device.h>
+#include <linux/regmap.h>
+
+extern const struct regmap_config pcm3060_regmap;
+
+#define PCM3060_DAI_ID_DAC 0
+#define PCM3060_DAI_ID_ADC 1
+#define PCM3060_DAI_IDS_NUM 2
+
+struct pcm3060_priv_dai {
+ bool is_master;
+ unsigned int sclk_freq;
+};
+
+struct pcm3060_priv {
+ struct regmap *regmap;
+ struct pcm3060_priv_dai dai[PCM3060_DAI_IDS_NUM];
+};
+
+int pcm3060_probe(struct device *dev);
+int pcm3060_remove(struct device *dev);
+
+/* registers */
+
+#define PCM3060_REG64 0x40
+#define PCM3060_REG_MRST 0x80
+#define PCM3060_REG_SRST 0x40
+#define PCM3060_REG_ADPSV 0x20
+#define PCM3060_REG_DAPSV 0x10
+#define PCM3060_REG_SE 0x01
+
+#define PCM3060_REG65 0x41
+#define PCM3060_REG66 0x42
+#define PCM3060_REG_AT2_MIN 0x36
+#define PCM3060_REG_AT2_MAX 0xFF
+
+#define PCM3060_REG67 0x43
+#define PCM3060_REG72 0x48
+#define PCM3060_REG_CSEL 0x80
+#define PCM3060_REG_MASK_MS 0x70
+#define PCM3060_REG_MS_S 0x00
+#define PCM3060_REG_MS_M768 (0x01 << 4)
+#define PCM3060_REG_MS_M512 (0x02 << 4)
+#define PCM3060_REG_MS_M384 (0x03 << 4)
+#define PCM3060_REG_MS_M256 (0x04 << 4)
+#define PCM3060_REG_MS_M192 (0x05 << 4)
+#define PCM3060_REG_MS_M128 (0x06 << 4)
+#define PCM3060_REG_MASK_FMT 0x03
+#define PCM3060_REG_FMT_I2S 0x00
+#define PCM3060_REG_FMT_LJ 0x01
+#define PCM3060_REG_FMT_RJ 0x02
+
+#define PCM3060_REG68 0x44
+#define PCM3060_REG_OVER 0x40
+#define PCM3060_REG_DREV2 0x04
+#define PCM3060_REG_SHIFT_MUT21 0x00
+#define PCM3060_REG_SHIFT_MUT22 0x01
+
+#define PCM3060_REG69 0x45
+#define PCM3060_REG_FLT 0x80
+#define PCM3060_REG_MASK_DMF 0x60
+#define PCM3060_REG_DMC 0x10
+#define PCM3060_REG_ZREV 0x02
+#define PCM3060_REG_AZRO 0x01
+
+#define PCM3060_REG70 0x46
+#define PCM3060_REG71 0x47
+#define PCM3060_REG_AT1_MIN 0x0E
+#define PCM3060_REG_AT1_MAX 0xFF
+
+#define PCM3060_REG73 0x49
+#define PCM3060_REG_ZCDD 0x10
+#define PCM3060_REG_BYP 0x08
+#define PCM3060_REG_DREV1 0x04
+#define PCM3060_REG_SHIFT_MUT11 0x00
+#define PCM3060_REG_SHIFT_MUT12 0x01
+
+#endif /* _SND_SOC_PCM3060_H */
diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 3356c91f55b0..52cc950c9fd1 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -33,6 +33,8 @@
#define PCM3168A_FMT_RIGHT_J_16 0x3
#define PCM3168A_FMT_DSP_A 0x4
#define PCM3168A_FMT_DSP_B 0x5
+#define PCM3168A_FMT_I2S_TDM 0x6
+#define PCM3168A_FMT_LEFT_J_TDM 0x7
#define PCM3168A_FMT_DSP_MASK 0x4
#define PCM3168A_NUM_SUPPLIES 6
@@ -401,9 +403,11 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
bool tx, master_mode;
u32 val, mask, shift, reg;
unsigned int rate, fmt, ratio, max_ratio;
+ unsigned int chan;
int i, min_frame_size;
rate = params_rate(params);
+ chan = params_channels(params);
ratio = pcm3168a->sysclk / rate;
@@ -456,6 +460,21 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
+ /* for TDM */
+ if (chan > 2) {
+ switch (fmt) {
+ case PCM3168A_FMT_I2S:
+ fmt = PCM3168A_FMT_I2S_TDM;
+ break;
+ case PCM3168A_FMT_LEFT_J:
+ fmt = PCM3168A_FMT_LEFT_J_TDM;
+ break;
+ default:
+ dev_err(component->dev, "TDM is supported under I2S/Left_J only\n");
+ return -EINVAL;
+ }
+ }
+
if (master_mode)
val = ((i + 1) << shift);
else
@@ -476,7 +495,69 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
return 0;
}
+static int pcm3168a_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component);
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ unsigned int fmt;
+ unsigned int sample_min;
+ unsigned int channel_max;
+
+ if (tx)
+ fmt = pcm3168a->dac_fmt;
+ else
+ fmt = pcm3168a->adc_fmt;
+
+ /*
+ * Available Data Bits
+ *
+ * RIGHT_J : 24 / 16
+ * LEFT_J : 24
+ * I2S : 24
+ *
+ * TDM available
+ *
+ * I2S
+ * LEFT_J
+ */
+ switch (fmt) {
+ case PCM3168A_FMT_RIGHT_J:
+ sample_min = 16;
+ channel_max = 2;
+ break;
+ case PCM3168A_FMT_LEFT_J:
+ sample_min = 24;
+ if (tx)
+ channel_max = 8;
+ else
+ channel_max = 6;
+ break;
+ case PCM3168A_FMT_I2S:
+ sample_min = 24;
+ if (tx)
+ channel_max = 8;
+ else
+ channel_max = 6;
+ break;
+ default:
+ sample_min = 24;
+ channel_max = 2;
+ }
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ sample_min, 32);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ 2, channel_max);
+
+ return 0;
+}
static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = {
+ .startup = pcm3168a_startup,
.set_fmt = pcm3168a_set_dai_fmt_dac,
.set_sysclk = pcm3168a_set_dai_sysclk,
.hw_params = pcm3168a_hw_params,
@@ -484,6 +565,7 @@ static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = {
};
static const struct snd_soc_dai_ops pcm3168a_adc_dai_ops = {
+ .startup = pcm3168a_startup,
.set_fmt = pcm3168a_set_dai_fmt_adc,
.set_sysclk = pcm3168a_set_dai_sysclk,
.hw_params = pcm3168a_hw_params
diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c
index d88e67341083..0ef966d56bac 100644
--- a/sound/soc/codecs/rt274.c
+++ b/sound/soc/codecs/rt274.c
@@ -755,6 +755,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
break;
default:
dev_warn(component->dev, "invalid pll source, use BCLK\n");
+ /* fall through */
case RT274_PLL2_S_BCLK:
snd_soc_component_update_bits(component, RT274_PLL2_CTRL,
RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_BCLK);
@@ -782,6 +783,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
break;
default:
dev_warn(component->dev, "invalid freq_in, assume 4.8M\n");
+ /* fall through */
case 100:
snd_soc_component_write(component, 0x7a, 0xaab6);
snd_soc_component_write(component, 0x7b, 0x0301);
diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c
index 6478d10c4f4a..4d46f4567c3a 100644
--- a/sound/soc/codecs/rt5514-spi.c
+++ b/sound/soc/codecs/rt5514-spi.c
@@ -91,6 +91,14 @@ static void rt5514_spi_copy_work(struct work_struct *work)
runtime = rt5514_dsp->substream->runtime;
period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream);
+ if (!period_bytes) {
+ schedule_delayed_work(&rt5514_dsp->copy_work, 5);
+ goto done;
+ }
+
+ if (rt5514_dsp->buf_size % period_bytes)
+ rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) *
+ period_bytes;
if (rt5514_dsp->get_size >= rt5514_dsp->buf_size) {
rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf,
@@ -149,13 +157,11 @@ done:
static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp)
{
- size_t period_bytes;
u8 buf[8];
if (!rt5514_dsp->substream)
return;
- period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream);
rt5514_dsp->get_size = 0;
/**
@@ -183,10 +189,6 @@ static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp)
rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base;
- if (rt5514_dsp->buf_size % period_bytes)
- rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) *
- period_bytes;
-
if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit &&
rt5514_dsp->buf_rp && rt5514_dsp->buf_size)
schedule_delayed_work(&rt5514_dsp->copy_work, 0);
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 5bcedbc7eb4a..b7ba64350a07 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -10,7 +10,6 @@
*/
#include <linux/module.h>
-#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index 70441661ea4a..7eb2cbd39d6e 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -72,6 +72,7 @@ struct rt5663_priv {
static const struct reg_sequence rt5663_patch_list[] = {
{ 0x002a, 0x8020 },
{ 0x0086, 0x0028 },
+ { 0x0100, 0xa020 },
{ 0x0117, 0x0f28 },
{ 0x02fb, 0x8089 },
};
@@ -580,7 +581,7 @@ static const struct reg_default rt5663_reg[] = {
{ 0x00fd, 0x0001 },
{ 0x00fe, 0x10ec },
{ 0x00ff, 0x6406 },
- { 0x0100, 0xa0a0 },
+ { 0x0100, 0xa020 },
{ 0x0108, 0x4444 },
{ 0x0109, 0x4444 },
{ 0x010a, 0xaaaa },
@@ -2337,6 +2338,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w,
0x8000);
snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000,
0x3000);
+ snd_soc_component_update_bits(component,
+ RT5663_DIG_VOL_ZCD, 0x00c0, 0x0080);
}
break;
@@ -2351,6 +2354,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w,
RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN);
snd_soc_component_update_bits(component,
RT5663_DACREF_LDO, 0x3e0e, 0);
+ snd_soc_component_update_bits(component,
+ RT5663_DIG_VOL_ZCD, 0x00c0, 0);
}
break;
diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c
index 3f6046a66b56..230a21c93b6b 100644
--- a/sound/soc/codecs/rt5668.c
+++ b/sound/soc/codecs/rt5668.c
@@ -2588,17 +2588,10 @@ static int rt5668_i2c_probe(struct i2c_client *i2c,
}
- return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668,
+ return devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668,
rt5668_dai, ARRAY_SIZE(rt5668_dai));
}
-static int rt5668_i2c_remove(struct i2c_client *i2c)
-{
- snd_soc_unregister_component(&i2c->dev);
-
- return 0;
-}
-
static void rt5668_i2c_shutdown(struct i2c_client *client)
{
struct rt5668_priv *rt5668 = i2c_get_clientdata(client);
@@ -2629,7 +2622,6 @@ static struct i2c_driver rt5668_i2c_driver = {
.acpi_match_table = ACPI_PTR(rt5668_acpi_match),
},
.probe = rt5668_i2c_probe,
- .remove = rt5668_i2c_remove,
.shutdown = rt5668_i2c_shutdown,
.id_table = rt5668_i2c_id,
};
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index f0f8debc2829..453328c988c0 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -2878,6 +2878,18 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = {
},
{
.callback = rt5670_quirk_cb,
+ .ident = "Lenovo Thinkpad Tablet 8",
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+ DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 8"),
+ },
+ .driver_data = (unsigned long *)(RT5670_DMIC_EN |
+ RT5670_DMIC2_INR |
+ RT5670_DEV_GPIO |
+ RT5670_JD_MODE1),
+ },
+ {
+ .callback = rt5670_quirk_cb,
.ident = "Lenovo Thinkpad Tablet 10",
.matches = {
DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c
index bd51f3655ee3..84501c2020c7 100644
--- a/sound/soc/codecs/rt5677-spi.c
+++ b/sound/soc/codecs/rt5677-spi.c
@@ -18,7 +18,6 @@
#include <linux/interrupt.h>
#include <linux/irq.h>
#include <linux/slab.h>
-#include <linux/gpio.h>
#include <linux/sched.h>
#include <linux/uaccess.h>
#include <linux/regulator/consumer.h>
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 8a3052874c29..34cfaf8f6f34 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -67,7 +67,8 @@ struct rt5682_priv {
};
static const struct reg_sequence patch_list[] = {
- {0x01c1, 0x1000},
+ {RT5682_HP_IMP_SENS_CTRL_19, 0x1000},
+ {RT5682_DAC_ADC_DIG_VOL1, 0xa020},
};
static const struct reg_default rt5682_reg[] = {
@@ -749,7 +750,6 @@ static bool rt5682_readable_register(struct device *dev, unsigned int reg)
}
}
-static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0);
static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0);
static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0);
static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
@@ -1108,10 +1108,6 @@ static void rt5682_jack_detect_handler(struct work_struct *work)
}
static const struct snd_kcontrol_new rt5682_snd_controls[] = {
- /* Headphone Output Volume */
- SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5682_HPL_GAIN,
- RT5682_HPR_GAIN, RT5682_G_HP_SFT, 15, 1, hp_vol_tlv),
-
/* DAC Digital Volume */
SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL,
RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv),
@@ -1437,6 +1433,28 @@ static const struct snd_kcontrol_new hpor_switch =
SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1,
RT5682_R_MUTE_SFT, 1, 1);
+static int rt5682_charge_pump_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(w->dapm);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_component_update_bits(component,
+ RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_component_update_bits(component,
+ RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_LV);
+ break;
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -1449,10 +1467,10 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
RT5682_HP_LOGIC_CTRL_2, 0x0012);
snd_soc_component_write(component,
RT5682_HP_CTRL_2, 0x6000);
- snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1,
- RT5682_NG2_EN_MASK, RT5682_NG2_EN);
snd_soc_component_update_bits(component,
RT5682_DEPOP_1, 0x60, 0x60);
+ snd_soc_component_update_bits(component,
+ RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080);
break;
case SND_SOC_DAPM_POST_PMD:
@@ -1460,6 +1478,8 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
RT5682_DEPOP_1, 0x60, 0x0);
snd_soc_component_write(component,
RT5682_HP_CTRL_2, 0x0000);
+ snd_soc_component_update_bits(component,
+ RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0000);
break;
default:
@@ -1723,7 +1743,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1,
RT5682_PWR_HA_R_BIT, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1,
- RT5682_PUMP_EN_SFT, 0, NULL, 0),
+ RT5682_PUMP_EN_SFT, 0, rt5682_charge_pump_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1,
RT5682_CAPLESS_EN_SFT, 0, NULL, 0),
@@ -1884,6 +1905,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
{"HP Amp", NULL, "Charge Pump"},
{"HP Amp", NULL, "CLKDET SYS"},
{"HP Amp", NULL, "CBJ Power"},
+ {"HP Amp", NULL, "Vref1"},
{"HP Amp", NULL, "Vref2"},
{"HPOL Playback", "Switch", "HP Amp"},
{"HPOR Playback", "Switch", "HP Amp"},
@@ -2452,30 +2474,23 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
mutex_lock(&rt5682->calibrate_mutex);
rt5682_reset(rt5682->regmap);
- regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf);
+ regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af);
usleep_range(15000, 20000);
- regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf);
- regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380);
- regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8001);
- regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000);
- regmap_write(rt5682->regmap, RT5682_STO1_DAC_MIXER, 0x2080);
- regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x4040);
- regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0069);
+ regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af);
+ regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0300);
+ regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x8000);
+ regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100);
+ regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800);
regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000);
- regmap_write(rt5682->regmap, RT5682_HP_CTRL_2, 0x6000);
- regmap_write(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, 0x0f26);
- regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7f05);
+ regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005);
regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c);
regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d);
- regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_9, 0x000f);
- regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8d01);
regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321);
regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004);
regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00);
regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1);
regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311);
- regmap_write(rt5682->regmap, RT5682_RESET_HPF_CTRL, 0x0000);
- regmap_write(rt5682->regmap, RT5682_ADC_STO1_HP_CTRL_1, 0x3320);
+ regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00);
regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00);
@@ -2491,8 +2506,12 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
pr_err("HP Calibration Failure\n");
/* restore settings */
- regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4);
+ regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0x02af);
+ regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0080);
+ regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000);
regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000);
+ regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000);
+ regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005);
mutex_unlock(&rt5682->calibrate_mutex);
@@ -2566,7 +2585,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
rt5682_calibrate(rt5682);
- ret = regmap_register_patch(rt5682->regmap, patch_list,
+ ret = regmap_multi_reg_write(rt5682->regmap, patch_list,
ARRAY_SIZE(patch_list));
if (ret != 0)
dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret);
@@ -2620,6 +2639,10 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK,
RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1);
regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000);
+ regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8,
+ RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA);
+ regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1,
+ RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ);
INIT_DELAYED_WORK(&rt5682->jack_detect_work,
rt5682_jack_detect_handler);
@@ -2637,11 +2660,17 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
}
- return devm_snd_soc_register_component(&i2c->dev,
- &soc_component_dev_rt5682,
+ return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682,
rt5682_dai, ARRAY_SIZE(rt5682_dai));
}
+static int rt5682_i2c_remove(struct i2c_client *i2c)
+{
+ snd_soc_unregister_component(&i2c->dev);
+
+ return 0;
+}
+
static void rt5682_i2c_shutdown(struct i2c_client *client)
{
struct rt5682_priv *rt5682 = i2c_get_clientdata(client);
@@ -2672,6 +2701,7 @@ static struct i2c_driver rt5682_i2c_driver = {
.acpi_match_table = ACPI_PTR(rt5682_acpi_match),
},
.probe = rt5682_i2c_probe,
+ .remove = rt5682_i2c_remove,
.shutdown = rt5682_i2c_shutdown,
.id_table = rt5682_i2c_id,
};
diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h
index 8068140ebe3f..d82a8301fd74 100644
--- a/sound/soc/codecs/rt5682.h
+++ b/sound/soc/codecs/rt5682.h
@@ -1214,6 +1214,20 @@
#define RT5682_JDH_NO_PLUG (0x1 << 4)
#define RT5682_JDH_PLUG (0x0 << 4)
+/* Bias current control 8 (0x0111) */
+#define RT5682_HPA_CP_BIAS_CTRL_MASK (0x3 << 2)
+#define RT5682_HPA_CP_BIAS_2UA (0x0 << 2)
+#define RT5682_HPA_CP_BIAS_3UA (0x1 << 2)
+#define RT5682_HPA_CP_BIAS_4UA (0x2 << 2)
+#define RT5682_HPA_CP_BIAS_6UA (0x3 << 2)
+
+/* Charge Pump Internal Register1 (0x0125) */
+#define RT5682_CP_CLK_HP_MASK (0x3 << 4)
+#define RT5682_CP_CLK_HP_100KHZ (0x0 << 4)
+#define RT5682_CP_CLK_HP_200KHZ (0x1 << 4)
+#define RT5682_CP_CLK_HP_300KHZ (0x2 << 4)
+#define RT5682_CP_CLK_HP_600KHZ (0x3 << 4)
+
/* Chopper and Clock control for DAC (0x013a)*/
#define RT5682_CKXEN_DAC1_MASK (0x1 << 13)
#define RT5682_CKXEN_DAC1_SFT 13
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 60764f6201b1..add18d6d77da 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1218,7 +1218,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_component *component)
* Searching for a suitable index solving this formula:
* idx = 40 * log10(vag_val / lo_cagcntrl) + 15
*/
- vol_quot = (vag * 100) / lo_vag;
+ vol_quot = lo_vag ? (vag * 100) / lo_vag : 0;
lo_vol = 0;
for (i = 0; i < ARRAY_SIZE(vol_quot_table); i++) {
if (vol_quot >= vol_quot_table[i])
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index d5035f2f2b2b..f753d2db0a5a 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -21,6 +21,7 @@
#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/i2c.h>
@@ -142,6 +143,7 @@ static const char *sta32x_supply_names[] = {
/* codec private data */
struct sta32x_priv {
struct regmap *regmap;
+ struct clk *xti_clk;
struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)];
struct snd_soc_component *component;
struct sta32x_platform_data *pdata;
@@ -879,6 +881,18 @@ static int sta32x_probe(struct snd_soc_component *component)
struct sta32x_priv *sta32x = snd_soc_component_get_drvdata(component);
struct sta32x_platform_data *pdata = sta32x->pdata;
int i, ret = 0, thermal = 0;
+
+ sta32x->component = component;
+
+ if (sta32x->xti_clk) {
+ ret = clk_prepare_enable(sta32x->xti_clk);
+ if (ret != 0) {
+ dev_err(component->dev,
+ "Failed to enable clock: %d\n", ret);
+ return ret;
+ }
+ }
+
ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies),
sta32x->supplies);
if (ret != 0) {
@@ -981,6 +995,9 @@ static void sta32x_remove(struct snd_soc_component *component)
sta32x_watchdog_stop(sta32x);
regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+
+ if (sta32x->xti_clk)
+ clk_disable_unprepare(sta32x->xti_clk);
}
static const struct snd_soc_component_driver sta32x_component = {
@@ -1038,6 +1055,8 @@ static int sta32x_probe_dt(struct device *dev, struct sta32x_priv *sta32x)
of_property_read_u8(np, "st,ch3-output-mapping",
&pdata->ch3_output_mapping);
+ if (of_get_property(np, "st,fault-detect-recovery", NULL))
+ pdata->fault_detect_recovery = 1;
if (of_get_property(np, "st,thermal-warning-recovery", NULL))
pdata->thermal_warning_recovery = 1;
if (of_get_property(np, "st,thermal-warning-adjustment", NULL))
@@ -1095,6 +1114,17 @@ static int sta32x_i2c_probe(struct i2c_client *i2c,
}
#endif
+ /* Clock */
+ sta32x->xti_clk = devm_clk_get(dev, "xti");
+ if (IS_ERR(sta32x->xti_clk)) {
+ ret = PTR_ERR(sta32x->xti_clk);
+
+ if (ret == -EPROBE_DEFER)
+ return ret;
+
+ sta32x->xti_clk = NULL;
+ }
+
/* GPIOs */
sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset",
GPIOD_OUT_LOW);
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
index ae3d032ac35a..6bd0e5d5347f 100644
--- a/sound/soc/codecs/tas5720.c
+++ b/sound/soc/codecs/tas5720.c
@@ -152,6 +152,7 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai,
int slots, int slot_width)
{
struct snd_soc_component *component = dai->component;
+ struct tas5720_data *tas5720 = snd_soc_component_get_drvdata(component);
unsigned int first_slot;
int ret;
@@ -185,6 +186,20 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai,
if (ret < 0)
goto error_snd_soc_component_update_bits;
+ /* Configure TDM slot width. This is only applicable to TAS5722. */
+ switch (tas5720->devtype) {
+ case TAS5722:
+ ret = snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG,
+ TAS5722_TDM_SLOT_16B,
+ slot_width == 16 ?
+ TAS5722_TDM_SLOT_16B : 0);
+ if (ret < 0)
+ goto error_snd_soc_component_update_bits;
+ break;
+ default:
+ break;
+ }
+
return 0;
error_snd_soc_component_update_bits:
@@ -485,15 +500,56 @@ static const DECLARE_TLV_DB_RANGE(dac_analog_tlv,
);
/*
- * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that
- * setting the gain below -100 dB (register value <0x7) is effectively a MUTE
- * as per device datasheet.
+ * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB or 0.25 dB steps
+ * depending on the device. Note that setting the gain below -100 dB
+ * (register value <0x7) is effectively a MUTE as per device datasheet.
+ *
+ * Note that for the TAS5722 the digital volume controls are actually split
+ * over two registers, so we need custom getters/setters for access.
*/
-static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas5720_dac_tlv, -10350, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas5722_dac_tlv, -10350, 25, 0);
+
+static int tas5722_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+ unsigned int val;
+
+ snd_soc_component_read(component, TAS5720_VOLUME_CTRL_REG, &val);
+ ucontrol->value.integer.value[0] = val << 1;
+
+ snd_soc_component_read(component, TAS5722_DIGITAL_CTRL2_REG, &val);
+ ucontrol->value.integer.value[0] |= val & TAS5722_VOL_CONTROL_LSB;
+
+ return 0;
+}
+
+static int tas5722_volume_set(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+ unsigned int sel = ucontrol->value.integer.value[0];
+
+ snd_soc_component_write(component, TAS5720_VOLUME_CTRL_REG, sel >> 1);
+ snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG,
+ TAS5722_VOL_CONTROL_LSB, sel);
+
+ return 0;
+}
static const struct snd_kcontrol_new tas5720_snd_controls[] = {
SOC_SINGLE_TLV("Speaker Driver Playback Volume",
- TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv),
+ TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, tas5720_dac_tlv),
+ SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG,
+ TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv),
+};
+
+static const struct snd_kcontrol_new tas5722_snd_controls[] = {
+ SOC_SINGLE_EXT_TLV("Speaker Driver Playback Volume",
+ 0, 0, 511, 0,
+ tas5722_volume_get, tas5722_volume_set,
+ tas5722_dac_tlv),
SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG,
TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv),
};
@@ -527,6 +583,23 @@ static const struct snd_soc_component_driver soc_component_dev_tas5720 = {
.non_legacy_dai_naming = 1,
};
+static const struct snd_soc_component_driver soc_component_dev_tas5722 = {
+ .probe = tas5720_codec_probe,
+ .remove = tas5720_codec_remove,
+ .suspend = tas5720_suspend,
+ .resume = tas5720_resume,
+ .controls = tas5722_snd_controls,
+ .num_controls = ARRAY_SIZE(tas5722_snd_controls),
+ .dapm_widgets = tas5720_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets),
+ .dapm_routes = tas5720_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map),
+ .idle_bias_on = 1,
+ .use_pmdown_time = 1,
+ .endianness = 1,
+ .non_legacy_dai_naming = 1,
+};
+
/* PCM rates supported by the TAS5720 driver */
#define TAS5720_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
@@ -613,9 +686,23 @@ static int tas5720_probe(struct i2c_client *client,
dev_set_drvdata(dev, data);
- ret = devm_snd_soc_register_component(&client->dev,
- &soc_component_dev_tas5720,
- tas5720_dai, ARRAY_SIZE(tas5720_dai));
+ switch (id->driver_data) {
+ case TAS5720:
+ ret = devm_snd_soc_register_component(&client->dev,
+ &soc_component_dev_tas5720,
+ tas5720_dai,
+ ARRAY_SIZE(tas5720_dai));
+ break;
+ case TAS5722:
+ ret = devm_snd_soc_register_component(&client->dev,
+ &soc_component_dev_tas5722,
+ tas5720_dai,
+ ARRAY_SIZE(tas5720_dai));
+ break;
+ default:
+ dev_err(dev, "unexpected private driver data\n");
+ return -EINVAL;
+ }
if (ret < 0) {
dev_err(dev, "failed to register component: %d\n", ret);
return ret;
diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c
index 0d6145549a98..36aebdb8f55c 100644
--- a/sound/soc/codecs/tas6424.c
+++ b/sound/soc/codecs/tas6424.c
@@ -41,6 +41,7 @@ struct tas6424_data {
struct regmap *regmap;
struct regulator_bulk_data supplies[TAS6424_NUM_SUPPLIES];
struct delayed_work fault_check_work;
+ unsigned int last_cfault;
unsigned int last_fault1;
unsigned int last_fault2;
unsigned int last_warn;
@@ -406,9 +407,54 @@ static void tas6424_fault_check_work(struct work_struct *work)
unsigned int reg;
int ret;
+ ret = regmap_read(tas6424->regmap, TAS6424_CHANNEL_FAULT, &reg);
+ if (ret < 0) {
+ dev_err(dev, "failed to read CHANNEL_FAULT register: %d\n", ret);
+ goto out;
+ }
+
+ if (!reg) {
+ tas6424->last_cfault = reg;
+ goto check_global_fault1_reg;
+ }
+
+ /*
+ * Only flag errors once for a given occurrence. This is needed as
+ * the TAS6424 will take time clearing the fault condition internally
+ * during which we don't want to bombard the system with the same
+ * error message over and over.
+ */
+ if ((reg & TAS6424_FAULT_OC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH1))
+ dev_crit(dev, "experienced a channel 1 overcurrent fault\n");
+
+ if ((reg & TAS6424_FAULT_OC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH2))
+ dev_crit(dev, "experienced a channel 2 overcurrent fault\n");
+
+ if ((reg & TAS6424_FAULT_OC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH3))
+ dev_crit(dev, "experienced a channel 3 overcurrent fault\n");
+
+ if ((reg & TAS6424_FAULT_OC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH4))
+ dev_crit(dev, "experienced a channel 4 overcurrent fault\n");
+
+ if ((reg & TAS6424_FAULT_DC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH1))
+ dev_crit(dev, "experienced a channel 1 DC fault\n");
+
+ if ((reg & TAS6424_FAULT_DC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH2))
+ dev_crit(dev, "experienced a channel 2 DC fault\n");
+
+ if ((reg & TAS6424_FAULT_DC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH3))
+ dev_crit(dev, "experienced a channel 3 DC fault\n");
+
+ if ((reg & TAS6424_FAULT_DC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH4))
+ dev_crit(dev, "experienced a channel 4 DC fault\n");
+
+ /* Store current fault1 value so we can detect any changes next time */
+ tas6424->last_cfault = reg;
+
+check_global_fault1_reg:
ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT1, &reg);
if (ret < 0) {
- dev_err(dev, "failed to read FAULT1 register: %d\n", ret);
+ dev_err(dev, "failed to read GLOB_FAULT1 register: %d\n", ret);
goto out;
}
@@ -429,12 +475,6 @@ static void tas6424_fault_check_work(struct work_struct *work)
goto check_global_fault2_reg;
}
- /*
- * Only flag errors once for a given occurrence. This is needed as
- * the TAS6424 will take time clearing the fault condition internally
- * during which we don't want to bombard the system with the same
- * error message over and over.
- */
if ((reg & TAS6424_FAULT_PVDD_OV) && !(tas6424->last_fault1 & TAS6424_FAULT_PVDD_OV))
dev_crit(dev, "experienced a PVDD overvoltage fault\n");
@@ -453,7 +493,7 @@ static void tas6424_fault_check_work(struct work_struct *work)
check_global_fault2_reg:
ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT2, &reg);
if (ret < 0) {
- dev_err(dev, "failed to read FAULT2 register: %d\n", ret);
+ dev_err(dev, "failed to read GLOB_FAULT2 register: %d\n", ret);
goto out;
}
@@ -530,7 +570,7 @@ check_warn_reg:
/* Store current warn value so we can detect any changes next time */
tas6424->last_warn = reg;
- /* Clear any faults by toggling the CLEAR_FAULT control bit */
+ /* Clear any warnings by toggling the CLEAR_FAULT control bit */
ret = regmap_write_bits(tas6424->regmap, TAS6424_MISC_CTRL3,
TAS6424_CLEAR_FAULT, TAS6424_CLEAR_FAULT);
if (ret < 0)
diff --git a/sound/soc/codecs/tas6424.h b/sound/soc/codecs/tas6424.h
index b5958c45ed0e..c67a7835ca66 100644
--- a/sound/soc/codecs/tas6424.h
+++ b/sound/soc/codecs/tas6424.h
@@ -116,6 +116,16 @@
#define TAS6424_LDGBYPASS_MASK BIT(TAS6424_LDGBYPASS_SHIFT)
/* TAS6424_GLOB_FAULT1_REG */
+#define TAS6424_FAULT_OC_CH1 BIT(7)
+#define TAS6424_FAULT_OC_CH2 BIT(6)
+#define TAS6424_FAULT_OC_CH3 BIT(5)
+#define TAS6424_FAULT_OC_CH4 BIT(4)
+#define TAS6424_FAULT_DC_CH1 BIT(3)
+#define TAS6424_FAULT_DC_CH2 BIT(2)
+#define TAS6424_FAULT_DC_CH3 BIT(1)
+#define TAS6424_FAULT_DC_CH4 BIT(0)
+
+/* TAS6424_GLOB_FAULT1_REG */
#define TAS6424_FAULT_CLOCK BIT(4)
#define TAS6424_FAULT_PVDD_OV BIT(3)
#define TAS6424_FAULT_VBAT_OV BIT(2)
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index bf92d36b8f8a..608ad49ad978 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -167,6 +167,7 @@ struct aic31xx_priv {
u8 p_div;
int rate_div_line;
bool master_dapm_route_applied;
+ int irq;
};
struct aic31xx_rate_divs {
@@ -1391,6 +1392,69 @@ static const struct acpi_device_id aic31xx_acpi_match[] = {
MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match);
#endif
+static irqreturn_t aic31xx_irq(int irq, void *data)
+{
+ struct aic31xx_priv *aic31xx = data;
+ struct device *dev = aic31xx->dev;
+ unsigned int value;
+ bool handled = false;
+ int ret;
+
+ ret = regmap_read(aic31xx->regmap, AIC31XX_INTRDACFLAG, &value);
+ if (ret) {
+ dev_err(dev, "Failed to read interrupt mask: %d\n", ret);
+ goto exit;
+ }
+
+ if (value)
+ handled = true;
+ else
+ goto read_overflow;
+
+ if (value & AIC31XX_HPLSCDETECT)
+ dev_err(dev, "Short circuit on Left output is detected\n");
+ if (value & AIC31XX_HPRSCDETECT)
+ dev_err(dev, "Short circuit on Right output is detected\n");
+ if (value & ~(AIC31XX_HPLSCDETECT |
+ AIC31XX_HPRSCDETECT))
+ dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value);
+
+read_overflow:
+ ret = regmap_read(aic31xx->regmap, AIC31XX_OFFLAG, &value);
+ if (ret) {
+ dev_err(dev, "Failed to read overflow flag: %d\n", ret);
+ goto exit;
+ }
+
+ if (value)
+ handled = true;
+ else
+ goto exit;
+
+ if (value & AIC31XX_DAC_OF_LEFT)
+ dev_warn(dev, "Left-channel DAC overflow has occurred\n");
+ if (value & AIC31XX_DAC_OF_RIGHT)
+ dev_warn(dev, "Right-channel DAC overflow has occurred\n");
+ if (value & AIC31XX_DAC_OF_SHIFTER)
+ dev_warn(dev, "DAC barrel shifter overflow has occurred\n");
+ if (value & AIC31XX_ADC_OF)
+ dev_warn(dev, "ADC overflow has occurred\n");
+ if (value & AIC31XX_ADC_OF_SHIFTER)
+ dev_warn(dev, "ADC barrel shifter overflow has occurred\n");
+ if (value & ~(AIC31XX_DAC_OF_LEFT |
+ AIC31XX_DAC_OF_RIGHT |
+ AIC31XX_DAC_OF_SHIFTER |
+ AIC31XX_ADC_OF |
+ AIC31XX_ADC_OF_SHIFTER))
+ dev_warn(dev, "Unknown overflow interrupt flags: 0x%08x\n", value);
+
+exit:
+ if (handled)
+ return IRQ_HANDLED;
+ else
+ return IRQ_NONE;
+}
+
static int aic31xx_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -1413,6 +1477,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
return ret;
}
aic31xx->dev = &i2c->dev;
+ aic31xx->irq = i2c->irq;
aic31xx->codec_type = id->driver_data;
@@ -1456,6 +1521,26 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
return ret;
}
+ if (aic31xx->irq > 0) {
+ regmap_update_bits(aic31xx->regmap, AIC31XX_GPIO1,
+ AIC31XX_GPIO1_FUNC_MASK,
+ AIC31XX_GPIO1_INT1 <<
+ AIC31XX_GPIO1_FUNC_SHIFT);
+
+ regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL,
+ AIC31XX_SC |
+ AIC31XX_ENGINE);
+
+ ret = devm_request_threaded_irq(aic31xx->dev, aic31xx->irq,
+ NULL, aic31xx_irq,
+ IRQF_ONESHOT, "aic31xx-irq",
+ aic31xx);
+ if (ret) {
+ dev_err(aic31xx->dev, "Unable to request IRQ\n");
+ return ret;
+ }
+ }
+
if (aic31xx->codec_type & DAC31XX_BIT)
return devm_snd_soc_register_component(&i2c->dev,
&soc_codec_driver_aic31xx,
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
index 0b587585b38b..2636f2c6bc79 100644
--- a/sound/soc/codecs/tlv320aic31xx.h
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -173,6 +173,13 @@ struct aic31xx_pdata {
#define AIC31XX_HPRDRVPWRSTATUS_MASK BIT(1)
#define AIC31XX_SPRDRVPWRSTATUS_MASK BIT(0)
+/* AIC31XX_OFFLAG */
+#define AIC31XX_DAC_OF_LEFT BIT(7)
+#define AIC31XX_DAC_OF_RIGHT BIT(6)
+#define AIC31XX_DAC_OF_SHIFTER BIT(5)
+#define AIC31XX_ADC_OF BIT(3)
+#define AIC31XX_ADC_OF_SHIFTER BIT(1)
+
/* AIC31XX_INTRDACFLAG */
#define AIC31XX_HPLSCDETECT BIT(7)
#define AIC31XX_HPRSCDETECT BIT(6)
@@ -191,6 +198,22 @@ struct aic31xx_pdata {
#define AIC31XX_SC BIT(3)
#define AIC31XX_ENGINE BIT(2)
+/* AIC31XX_GPIO1 */
+#define AIC31XX_GPIO1_FUNC_MASK GENMASK(5, 2)
+#define AIC31XX_GPIO1_FUNC_SHIFT 2
+#define AIC31XX_GPIO1_DISABLED 0x00
+#define AIC31XX_GPIO1_INPUT 0x01
+#define AIC31XX_GPIO1_GPI 0x02
+#define AIC31XX_GPIO1_GPO 0x03
+#define AIC31XX_GPIO1_CLKOUT 0x04
+#define AIC31XX_GPIO1_INT1 0x05
+#define AIC31XX_GPIO1_INT2 0x06
+#define AIC31XX_GPIO1_ADC_WCLK 0x07
+#define AIC31XX_GPIO1_SBCLK 0x08
+#define AIC31XX_GPIO1_SWCLK 0x09
+#define AIC31XX_GPIO1_ADC_MOD_CLK 0x10
+#define AIC31XX_GPIO1_SDOUT 0x11
+
/* AIC31XX_DACSETUP */
#define AIC31XX_SOFTSTEP_MASK GENMASK(1, 0)
diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c
index ff85a0bf6170..93d84e5ae2d5 100644
--- a/sound/soc/codecs/tscs454.c
+++ b/sound/soc/codecs/tscs454.c
@@ -3459,7 +3459,7 @@ static int tscs454_i2c_probe(struct i2c_client *i2c,
/* Sync pg sel reg with cache */
regmap_write(tscs454->regmap, R_PAGESEL, 0x00);
- ret = snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454,
+ ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454,
tscs454_dais, ARRAY_SIZE(tscs454_dais));
if (ret) {
dev_err(&i2c->dev, "Failed to register component (%d)\n", ret);
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index c5ae07234a00..bba330e30162 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -88,19 +88,6 @@ static int wm2000_write(struct i2c_client *i2c, unsigned int reg,
return regmap_write(wm2000->regmap, reg, value);
}
-static unsigned int wm2000_read(struct i2c_client *i2c, unsigned int r)
-{
- struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c);
- unsigned int val;
- int ret;
-
- ret = regmap_read(wm2000->regmap, r, &val);
- if (ret < 0)
- return -1;
-
- return val;
-}
-
static void wm2000_reset(struct wm2000_priv *wm2000)
{
struct i2c_client *i2c = wm2000->i2c;
@@ -115,14 +102,15 @@ static void wm2000_reset(struct wm2000_priv *wm2000)
static int wm2000_poll_bit(struct i2c_client *i2c,
unsigned int reg, u8 mask)
{
+ struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c);
int timeout = 4000;
- int val;
+ unsigned int val;
- val = wm2000_read(i2c, reg);
+ regmap_read(wm2000->regmap, reg, &val);
while (!(val & mask) && --timeout) {
msleep(1);
- val = wm2000_read(i2c, reg);
+ regmap_read(wm2000->regmap, reg, &val);
}
if (timeout == 0)
@@ -135,6 +123,7 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
{
struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
unsigned long rate;
+ unsigned int val;
int ret;
if (WARN_ON(wm2000->anc_mode != ANC_OFF))
@@ -213,12 +202,17 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
WM2000_MODE_THERMAL_ENABLE);
}
- ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY);
+ ret = regmap_read(wm2000->regmap, WM2000_REG_SPEECH_CLARITY, &val);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Unable to read Speech Clarity: %d\n", ret);
+ regulator_bulk_disable(WM2000_NUM_SUPPLIES, wm2000->supplies);
+ return ret;
+ }
if (wm2000->speech_clarity)
- ret |= WM2000_SPEECH_CLARITY;
+ val |= WM2000_SPEECH_CLARITY;
else
- ret &= ~WM2000_SPEECH_CLARITY;
- wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret);
+ val &= ~WM2000_SPEECH_CLARITY;
+ wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, val);
wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33);
wm2000_write(i2c, WM2000_REG_SYS_START1, 0x02);
@@ -824,7 +818,7 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
const char *filename;
const struct firmware *fw = NULL;
int ret, i;
- int reg;
+ unsigned int reg;
u16 id;
wm2000 = devm_kzalloc(&i2c->dev, sizeof(*wm2000), GFP_KERNEL);
@@ -860,9 +854,17 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
}
/* Verify that this is a WM2000 */
- reg = wm2000_read(i2c, WM2000_REG_ID1);
+ ret = regmap_read(wm2000->regmap, WM2000_REG_ID1, &reg);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Unable to read ID1: %d\n", ret);
+ return ret;
+ }
id = reg << 8;
- reg = wm2000_read(i2c, WM2000_REG_ID2);
+ ret = regmap_read(wm2000->regmap, WM2000_REG_ID2, &reg);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Unable to read ID2: %d\n", ret);
+ return ret;
+ }
id |= reg & 0xff;
if (id != 0x2000) {
@@ -871,7 +873,11 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
goto err_supplies;
}
- reg = wm2000_read(i2c, WM2000_REG_REVISON);
+ ret = regmap_read(wm2000->regmap, WM2000_REG_REVISON, &reg);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Unable to read Revision: %d\n", ret);
+ return ret;
+ }
dev_info(&i2c->dev, "revision %c\n", reg + 'A');
wm2000->mclk = devm_clk_get(&i2c->dev, "MCLK");
diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c
index 317db9a149a7..cf2cdbece122 100644
--- a/sound/soc/codecs/wm8782.c
+++ b/sound/soc/codecs/wm8782.c
@@ -20,6 +20,7 @@
#include <linux/module.h>
#include <linux/kernel.h>
#include <linux/device.h>
+#include <linux/regulator/consumer.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
@@ -50,7 +51,51 @@ static struct snd_soc_dai_driver wm8782_dai = {
},
};
+/* regulator power supply names */
+static const char *supply_names[] = {
+ "Vdda", /* analog supply, 2.7V - 3.6V */
+ "Vdd", /* digital supply, 2.7V - 5.5V */
+};
+
+struct wm8782_priv {
+ struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
+};
+
+static int wm8782_soc_probe(struct snd_soc_component *component)
+{
+ struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+ return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+
+static void wm8782_soc_remove(struct snd_soc_component *component)
+{
+ struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+ regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+
+#ifdef CONFIG_PM
+static int wm8782_soc_suspend(struct snd_soc_component *component)
+{
+ struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+ regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+ return 0;
+}
+
+static int wm8782_soc_resume(struct snd_soc_component *component)
+{
+ struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+ return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+#else
+#define wm8782_soc_suspend NULL
+#define wm8782_soc_resume NULL
+#endif /* CONFIG_PM */
+
static const struct snd_soc_component_driver soc_component_dev_wm8782 = {
+ .probe = wm8782_soc_probe,
+ .remove = wm8782_soc_remove,
+ .suspend = wm8782_soc_suspend,
+ .resume = wm8782_soc_resume,
.dapm_widgets = wm8782_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm8782_dapm_widgets),
.dapm_routes = wm8782_dapm_routes,
@@ -63,6 +108,24 @@ static const struct snd_soc_component_driver soc_component_dev_wm8782 = {
static int wm8782_probe(struct platform_device *pdev)
{
+ struct device *dev = &pdev->dev;
+ struct wm8782_priv *priv;
+ int ret, i;
+
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ 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;
+
return devm_snd_soc_register_component(&pdev->dev,
&soc_component_dev_wm8782, &wm8782_dai, 1);
}
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 1965635ec07c..2a3e5fbd04e4 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -13,7 +13,6 @@
#include <linux/clk.h>
#include <linux/module.h>
-#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 43edaf8cd276..593a11960888 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -11,7 +11,6 @@
*/
#include <linux/module.h>
-#include <linux/moduleparam.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index ade34c26ad2f..e873baa9e778 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -638,13 +638,14 @@ static int wm9712_soc_probe(struct snd_soc_component *component)
{
struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component);
struct regmap *regmap;
- int ret;
if (wm9712->mfd_pdata) {
wm9712->ac97 = wm9712->mfd_pdata->ac97;
regmap = wm9712->mfd_pdata->regmap;
} else {
#ifdef CONFIG_SND_SOC_AC97_BUS
+ int ret;
+
wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID,
WM9712_VENDOR_ID_MASK);
if (IS_ERR(wm9712->ac97)) {
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index f61656070225..a53dc174bbf0 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -311,12 +311,12 @@ struct wm_adsp_alg_xm_struct {
};
struct wm_adsp_buffer {
- __be32 X_buf_base; /* XM base addr of first X area */
- __be32 X_buf_size; /* Size of 1st X area in words */
- __be32 X_buf_base2; /* XM base addr of 2nd X area */
- __be32 X_buf_brk; /* Total X size in words */
- __be32 Y_buf_base; /* YM base addr of Y area */
- __be32 wrap; /* Total size X and Y in words */
+ __be32 buf1_base; /* Base addr of first buffer area */
+ __be32 buf1_size; /* Size of buf1 area in DSP words */
+ __be32 buf2_base; /* Base addr of 2nd buffer area */
+ __be32 buf1_buf2_size; /* Size of buf1+buf2 in DSP words */
+ __be32 buf3_base; /* Base addr of buf3 area */
+ __be32 buf_total_size; /* Size of buf1+buf2+buf3 in DSP words */
__be32 high_water_mark; /* Point at which IRQ is asserted */
__be32 irq_count; /* bits 1-31 count IRQ assertions */
__be32 irq_ack; /* acked IRQ count, bit 0 enables IRQ */
@@ -393,18 +393,18 @@ struct wm_adsp_buffer_region_def {
static const struct wm_adsp_buffer_region_def default_regions[] = {
{
.mem_type = WMFW_ADSP2_XM,
- .base_offset = HOST_BUFFER_FIELD(X_buf_base),
- .size_offset = HOST_BUFFER_FIELD(X_buf_size),
+ .base_offset = HOST_BUFFER_FIELD(buf1_base),
+ .size_offset = HOST_BUFFER_FIELD(buf1_size),
},
{
.mem_type = WMFW_ADSP2_XM,
- .base_offset = HOST_BUFFER_FIELD(X_buf_base2),
- .size_offset = HOST_BUFFER_FIELD(X_buf_brk),
+ .base_offset = HOST_BUFFER_FIELD(buf2_base),
+ .size_offset = HOST_BUFFER_FIELD(buf1_buf2_size),
},
{
.mem_type = WMFW_ADSP2_YM,
- .base_offset = HOST_BUFFER_FIELD(Y_buf_base),
- .size_offset = HOST_BUFFER_FIELD(wrap),
+ .base_offset = HOST_BUFFER_FIELD(buf3_base),
+ .size_offset = HOST_BUFFER_FIELD(buf_total_size),
},
};
@@ -3345,7 +3345,7 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf)
region->cumulative_size = offset;
adsp_dbg(buf->dsp,
- "region=%d type=%d base=%04x off=%04x size=%04x\n",
+ "region=%d type=%d base=%08x off=%08x size=%08x\n",
i, region->mem_type, region->base_addr,
region->offset, region->cumulative_size);
}
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index f70db8412c7c..267aee776b2d 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -1041,6 +1041,42 @@ static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp,
return error_ppm;
}
+static inline u32 davinci_mcasp_tx_delay(struct davinci_mcasp *mcasp)
+{
+ if (!mcasp->txnumevt)
+ return 0;
+
+ return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_WFIFOSTS_OFFSET);
+}
+
+static inline u32 davinci_mcasp_rx_delay(struct davinci_mcasp *mcasp)
+{
+ if (!mcasp->rxnumevt)
+ return 0;
+
+ return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_RFIFOSTS_OFFSET);
+}
+
+static snd_pcm_sframes_t davinci_mcasp_delay(
+ struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+ u32 fifo_use;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ fifo_use = davinci_mcasp_tx_delay(mcasp);
+ else
+ fifo_use = davinci_mcasp_rx_delay(mcasp);
+
+ /*
+ * Divide the used locations with the channel count to get the
+ * FIFO usage in samples (don't care about partial samples in the
+ * buffer).
+ */
+ return fifo_use / substream->runtime->channels;
+}
+
static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *cpu_dai)
@@ -1365,6 +1401,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
.startup = davinci_mcasp_startup,
.shutdown = davinci_mcasp_shutdown,
.trigger = davinci_mcasp_trigger,
+ .delay = davinci_mcasp_delay,
.hw_params = davinci_mcasp_hw_params,
.set_fmt = davinci_mcasp_set_dai_fmt,
.set_clkdiv = davinci_mcasp_set_clkdiv,
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 1033ac6631b0..01052a0808b0 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -151,7 +151,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream,
int ret;
/* Fetch the Back-End dma_data from DPCM */
- list_for_each_entry(dpcm, &rtd->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(rtd, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_pcm_substream *substream_be;
struct snd_soc_dai *dai = be->cpu_dai;
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index c1d1d06783e5..57b484768a58 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -807,7 +807,7 @@ static int fsl_esai_probe(struct platform_device *pdev)
return -ENOMEM;
esai_priv->pdev = pdev;
- strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1);
+ snprintf(esai_priv->name, sizeof(esai_priv->name), "%pOFn", np);
/* Get the addresses and IRQ */
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
index 7f0fa4b52223..9981668ab590 100644
--- a/sound/soc/fsl/fsl_utils.c
+++ b/sound/soc/fsl/fsl_utils.c
@@ -57,8 +57,8 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np,
of_node_put(dma_channel_np);
return ret;
}
- snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
- (unsigned long long) res.start, dma_channel_np->name);
+ snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%pOFn",
+ (unsigned long long) res.start, dma_channel_np);
iprop = of_get_property(dma_channel_np, "cell-index", NULL);
if (!iprop) {
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index ec731223cab3..e339f36cea95 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -57,6 +57,7 @@ static int pcm030_fabric_probe(struct platform_device *op)
struct device_node *platform_np;
struct snd_soc_card *card = &pcm030_card;
struct pcm030_audio_data *pdata;
+ struct snd_soc_dai_link *dai_link;
int ret;
int i;
@@ -78,8 +79,8 @@ static int pcm030_fabric_probe(struct platform_device *op)
return -ENODEV;
}
- for (i = 0; i < card->num_links; i++)
- card->dai_link[i].platform_of_node = platform_np;
+ for_each_card_prelinks(card, i, dai_link)
+ dai_link->platform_of_node = platform_np;
ret = request_module("snd-soc-wm9712");
if (ret)
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 2094d2c8919f..25c819e402e1 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -25,6 +25,8 @@ struct graph_card_data {
struct graph_dai_props {
struct asoc_simple_dai cpu_dai;
struct asoc_simple_dai codec_dai;
+ struct snd_soc_dai_link_component codecs; /* single codec */
+ struct snd_soc_dai_link_component platform;
unsigned int mclk_fs;
} *dai_props;
unsigned int mclk_fs;
@@ -180,7 +182,8 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port,
if (ret < 0)
goto dai_link_of_err;
- of_property_read_u32(rcpu_ep, "mclk-fs", &dai_props->mclk_fs);
+ of_property_read_u32(cpu_ep, "mclk-fs", &dai_props->mclk_fs);
+ of_property_read_u32(codec_ep, "mclk-fs", &dai_props->mclk_fs);
ret = asoc_simple_card_parse_graph_cpu(cpu_ep, dai_link);
if (ret < 0)
@@ -213,7 +216,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port,
ret = asoc_simple_card_set_dailink_name(dev, dai_link,
"%s-%s",
dai_link->cpu_dai_name,
- dai_link->codec_dai_name);
+ dai_link->codecs->dai_name);
if (ret < 0)
goto dai_link_of_err;
@@ -299,7 +302,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
struct graph_dai_props *dai_props;
struct device *dev = &pdev->dev;
struct snd_soc_card *card;
- int num, ret;
+ int num, ret, i;
/* Allocate the private data and the DAI link array */
priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -315,6 +318,18 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
if (!dai_props || !dai_link)
return -ENOMEM;
+ /*
+ * Use snd_soc_dai_link_component instead of legacy style
+ * It is codec only. but cpu/platform will be supported in the future.
+ * see
+ * soc-core.c :: snd_soc_init_multicodec()
+ */
+ for (i = 0; i < num; i++) {
+ dai_link[i].codecs = &dai_props[i].codecs;
+ dai_link[i].num_codecs = 1;
+ dai_link[i].platform = &dai_props[i].platform;
+ }
+
priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW);
if (IS_ERR(priv->pa_gpio)) {
ret = PTR_ERR(priv->pa_gpio);
diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c
index 92882e392d6c..b83bb31021a9 100644
--- a/sound/soc/generic/audio-graph-scu-card.c
+++ b/sound/soc/generic/audio-graph-scu-card.c
@@ -25,7 +25,11 @@
struct graph_card_data {
struct snd_soc_card snd_card;
struct snd_soc_codec_conf codec_conf;
- struct asoc_simple_dai *dai_props;
+ struct graph_dai_props {
+ struct asoc_simple_dai dai;
+ struct snd_soc_dai_link_component codecs;
+ struct snd_soc_dai_link_component platform;
+ } *dai_props;
struct snd_soc_dai_link *dai_link;
struct asoc_simple_card_data adata;
};
@@ -39,18 +43,18 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
- struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num);
+ struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num);
- return asoc_simple_card_clk_enable(dai_props);
+ return asoc_simple_card_clk_enable(&dai_props->dai);
}
static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
- struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num);
+ struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num);
- asoc_simple_card_clk_disable(dai_props);
+ asoc_simple_card_clk_disable(&dai_props->dai);
}
static const struct snd_soc_ops asoc_graph_card_ops = {
@@ -63,7 +67,7 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd)
struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai;
struct snd_soc_dai_link *dai_link;
- struct asoc_simple_dai *dai_props;
+ struct graph_dai_props *dai_props;
int num = rtd->num;
dai_link = graph_priv_to_link(priv, num);
@@ -72,7 +76,7 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd)
rtd->cpu_dai :
rtd->codec_dai;
- return asoc_simple_card_init_dai(dai, dai_props);
+ return asoc_simple_card_init_dai(dai, &dai_props->dai);
}
static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
@@ -92,15 +96,18 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep,
{
struct device *dev = graph_priv_to_dev(priv);
struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, idx);
- struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, idx);
+ struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx);
struct snd_soc_card *card = graph_priv_to_card(priv);
int ret;
if (is_fe) {
+ struct snd_soc_dai_link_component *codecs;
+
/* BE is dummy */
- dai_link->codec_of_node = NULL;
- dai_link->codec_dai_name = "snd-soc-dummy-dai";
- dai_link->codec_name = "snd-soc-dummy";
+ codecs = dai_link->codecs;
+ codecs->of_node = NULL;
+ codecs->dai_name = "snd-soc-dummy-dai";
+ codecs->name = "snd-soc-dummy";
/* FE settings */
dai_link->dynamic = 1;
@@ -110,7 +117,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep,
if (ret)
return ret;
- ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai_props);
+ ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, &dai_props->dai);
if (ret < 0)
return ret;
@@ -137,23 +144,23 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep,
if (ret < 0)
return ret;
- ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai_props);
+ ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, &dai_props->dai);
if (ret < 0)
return ret;
ret = asoc_simple_card_set_dailink_name(dev, dai_link,
"be.%s",
- dai_link->codec_dai_name);
+ dai_link->codecs->dai_name);
if (ret < 0)
return ret;
snd_soc_of_parse_audio_prefix(card,
&priv->codec_conf,
- dai_link->codec_of_node,
+ dai_link->codecs->of_node,
"prefix");
}
- ret = asoc_simple_card_of_parse_tdm(ep, dai_props);
+ ret = asoc_simple_card_of_parse_tdm(ep, &dai_props->dai);
if (ret)
return ret;
@@ -331,10 +338,10 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
{
struct graph_card_data *priv;
struct snd_soc_dai_link *dai_link;
- struct asoc_simple_dai *dai_props;
+ struct graph_dai_props *dai_props;
struct device *dev = &pdev->dev;
struct snd_soc_card *card;
- int num, ret;
+ int num, ret, i;
/* Allocate the private data and the DAI link array */
priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -350,6 +357,18 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
if (!dai_props || !dai_link)
return -ENOMEM;
+ /*
+ * Use snd_soc_dai_link_component instead of legacy style
+ * It is codec only. but cpu/platform will be supported in the future.
+ * see
+ * soc-core.c :: snd_soc_init_multicodec()
+ */
+ for (i = 0; i < num; i++) {
+ dai_link[i].codecs = &dai_props[i].codecs;
+ dai_link[i].num_codecs = 1;
+ dai_link[i].platform = &dai_props[i].platform;
+ }
+
priv->dai_props = dai_props;
priv->dai_link = dai_link;
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index d3f3f0fec74c..f34cc6cddfa2 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -173,12 +173,24 @@ int asoc_simple_card_parse_clk(struct device *dev,
struct device_node *node,
struct device_node *dai_of_node,
struct asoc_simple_dai *simple_dai,
- const char *name)
+ const char *dai_name,
+ struct snd_soc_dai_link_component *dlc)
{
struct clk *clk;
u32 val;
/*
+ * Use snd_soc_dai_link_component instead of legacy style.
+ * It is only for codec, but cpu will be supported in the future.
+ * see
+ * soc-core.c :: snd_soc_init_multicodec()
+ */
+ if (dlc) {
+ dai_of_node = dlc->of_node;
+ dai_name = dlc->dai_name;
+ }
+
+ /*
* Parse dai->sysclk come from "clocks = <&xxx>"
* (if system has common clock)
* or "system-clock-frequency = <xxx>"
@@ -200,7 +212,7 @@ int asoc_simple_card_parse_clk(struct device *dev,
if (of_property_read_bool(node, "system-clock-direction-out"))
simple_dai->clk_direction = SND_SOC_CLOCK_OUT;
- dev_dbg(dev, "%s : sysclk = %d, direction %d\n", name,
+ dev_dbg(dev, "%s : sysclk = %d, direction %d\n", dai_name,
simple_dai->sysclk, simple_dai->clk_direction);
return 0;
@@ -208,6 +220,7 @@ int asoc_simple_card_parse_clk(struct device *dev,
EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk);
int asoc_simple_card_parse_dai(struct device_node *node,
+ struct snd_soc_dai_link_component *dlc,
struct device_node **dai_of_node,
const char **dai_name,
const char *list_name,
@@ -221,6 +234,17 @@ int asoc_simple_card_parse_dai(struct device_node *node,
return 0;
/*
+ * Use snd_soc_dai_link_component instead of legacy style.
+ * It is only for codec, but cpu will be supported in the future.
+ * see
+ * soc-core.c :: snd_soc_init_multicodec()
+ */
+ if (dlc) {
+ dai_name = &dlc->dai_name;
+ dai_of_node = &dlc->of_node;
+ }
+
+ /*
* Get node via "sound-dai = <&phandle port>"
* it will be used as xxx_of_node on soc_bind_dai_link()
*/
@@ -278,6 +302,7 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep)
}
int asoc_simple_card_parse_graph_dai(struct device_node *ep,
+ struct snd_soc_dai_link_component *dlc,
struct device_node **dai_of_node,
const char **dai_name)
{
@@ -285,6 +310,17 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep,
struct of_phandle_args args;
int ret;
+ /*
+ * Use snd_soc_dai_link_component instead of legacy style.
+ * It is only for codec, but cpu will be supported in the future.
+ * see
+ * soc-core.c :: snd_soc_init_multicodec()
+ */
+ if (dlc) {
+ dai_name = &dlc->dai_name;
+ dai_of_node = &dlc->of_node;
+ }
+
if (!ep)
return 0;
if (!dai_name)
@@ -340,10 +376,11 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai);
int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link)
{
/* Assumes platform == cpu */
- if (!dai_link->platform_of_node)
- dai_link->platform_of_node = dai_link->cpu_of_node;
+ if (!dai_link->platform->of_node)
+ dai_link->platform->of_node = dai_link->cpu_of_node;
return 0;
+
}
EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_dailink);
@@ -367,13 +404,11 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_cpu);
int asoc_simple_card_clean_reference(struct snd_soc_card *card)
{
struct snd_soc_dai_link *dai_link;
- int num_links;
+ int i;
- for (num_links = 0, dai_link = card->dai_link;
- num_links < card->num_links;
- num_links++, dai_link++) {
+ for_each_card_prelinks(card, i, dai_link) {
of_node_put(dai_link->cpu_of_node);
- of_node_put(dai_link->codec_of_node);
+ of_node_put(dai_link->codecs->of_node);
}
return 0;
}
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 64bf3560c1d1..5a3f59aa4ba5 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -20,6 +20,8 @@ struct simple_card_data {
struct simple_dai_props {
struct asoc_simple_dai cpu_dai;
struct asoc_simple_dai codec_dai;
+ struct snd_soc_dai_link_component codecs; /* single codec */
+ struct snd_soc_dai_link_component platform;
unsigned int mclk_fs;
} *dai_props;
unsigned int mclk_fs;
@@ -234,7 +236,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
ret = asoc_simple_card_set_dailink_name(dev, dai_link,
"%s-%s",
dai_link->cpu_dai_name,
- dai_link->codec_dai_name);
+ dai_link->codecs->dai_name);
if (ret < 0)
goto dai_link_of_err;
@@ -363,7 +365,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
struct device *dev = &pdev->dev;
struct device_node *np = dev->of_node;
struct snd_soc_card *card;
- int num, ret;
+ int num, ret, i;
/* Get the number of DAI links */
if (np && of_get_child_by_name(np, PREFIX "dai-link"))
@@ -381,6 +383,18 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
if (!dai_props || !dai_link)
return -ENOMEM;
+ /*
+ * Use snd_soc_dai_link_component instead of legacy style
+ * It is codec only. but cpu/platform will be supported in the future.
+ * see
+ * soc-core.c :: snd_soc_init_multicodec()
+ */
+ for (i = 0; i < num; i++) {
+ dai_link[i].codecs = &dai_props[i].codecs;
+ dai_link[i].num_codecs = 1;
+ dai_link[i].platform = &dai_props[i].platform;
+ }
+
priv->dai_props = dai_props;
priv->dai_link = dai_link;
@@ -403,6 +417,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
} else {
struct asoc_simple_card_info *cinfo;
+ struct snd_soc_dai_link_component *codecs;
+ struct snd_soc_dai_link_component *platform;
cinfo = dev->platform_data;
if (!cinfo) {
@@ -419,13 +435,17 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
return -EINVAL;
}
+ codecs = dai_link->codecs;
+ codecs->name = cinfo->codec;
+ codecs->dai_name = cinfo->codec_dai.name;
+
+ platform = dai_link->platform;
+ platform->name = cinfo->platform;
+
card->name = (cinfo->card) ? cinfo->card : cinfo->name;
dai_link->name = cinfo->name;
dai_link->stream_name = cinfo->name;
- dai_link->platform_name = cinfo->platform;
- dai_link->codec_name = cinfo->codec;
dai_link->cpu_dai_name = cinfo->cpu_dai.name;
- dai_link->codec_dai_name = cinfo->codec_dai.name;
dai_link->dai_fmt = cinfo->daifmt;
dai_link->init = asoc_simple_card_dai_init;
memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai,
diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c
index 16a83bc51e0e..85b46f0eae0f 100644
--- a/sound/soc/generic/simple-scu-card.c
+++ b/sound/soc/generic/simple-scu-card.c
@@ -22,7 +22,11 @@
struct simple_card_data {
struct snd_soc_card snd_card;
struct snd_soc_codec_conf codec_conf;
- struct asoc_simple_dai *dai_props;
+ struct simple_dai_props {
+ struct asoc_simple_dai dai;
+ struct snd_soc_dai_link_component codecs;
+ struct snd_soc_dai_link_component platform;
+ } *dai_props;
struct snd_soc_dai_link *dai_link;
struct asoc_simple_card_data adata;
};
@@ -40,20 +44,20 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
- struct asoc_simple_dai *dai_props =
+ struct simple_dai_props *dai_props =
simple_priv_to_props(priv, rtd->num);
- return asoc_simple_card_clk_enable(dai_props);
+ return asoc_simple_card_clk_enable(&dai_props->dai);
}
static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
- struct asoc_simple_dai *dai_props =
+ struct simple_dai_props *dai_props =
simple_priv_to_props(priv, rtd->num);
- asoc_simple_card_clk_disable(dai_props);
+ asoc_simple_card_clk_disable(&dai_props->dai);
}
static const struct snd_soc_ops asoc_simple_card_ops = {
@@ -66,7 +70,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai;
struct snd_soc_dai_link *dai_link;
- struct asoc_simple_dai *dai_props;
+ struct simple_dai_props *dai_props;
int num = rtd->num;
dai_link = simple_priv_to_link(priv, num);
@@ -75,7 +79,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
rtd->cpu_dai :
rtd->codec_dai;
- return asoc_simple_card_init_dai(dai, dai_props);
+ return asoc_simple_card_init_dai(dai, &dai_props->dai);
}
static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
@@ -95,17 +99,19 @@ static int asoc_simple_card_dai_link_of(struct device_node *np,
{
struct device *dev = simple_priv_to_dev(priv);
struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx);
- struct asoc_simple_dai *dai_props = simple_priv_to_props(priv, idx);
+ struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx);
struct snd_soc_card *card = simple_priv_to_card(priv);
int ret;
if (is_fe) {
int is_single_links = 0;
+ struct snd_soc_dai_link_component *codecs;
/* BE is dummy */
- dai_link->codec_of_node = NULL;
- dai_link->codec_dai_name = "snd-soc-dummy-dai";
- dai_link->codec_name = "snd-soc-dummy";
+ codecs = dai_link->codecs;
+ codecs->of_node = NULL;
+ codecs->dai_name = "snd-soc-dummy-dai";
+ codecs->name = "snd-soc-dummy";
/* FE settings */
dai_link->dynamic = 1;
@@ -116,7 +122,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np,
if (ret)
return ret;
- ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai_props);
+ ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, &dai_props->dai);
if (ret < 0)
return ret;
@@ -141,23 +147,23 @@ static int asoc_simple_card_dai_link_of(struct device_node *np,
if (ret < 0)
return ret;
- ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai_props);
+ ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, &dai_props->dai);
if (ret < 0)
return ret;
ret = asoc_simple_card_set_dailink_name(dev, dai_link,
"be.%s",
- dai_link->codec_dai_name);
+ dai_link->codecs->dai_name);
if (ret < 0)
return ret;
snd_soc_of_parse_audio_prefix(card,
&priv->codec_conf,
- dai_link->codec_of_node,
+ dai_link->codecs->of_node,
PREFIX "prefix");
}
- ret = asoc_simple_card_of_parse_tdm(np, dai_props);
+ ret = asoc_simple_card_of_parse_tdm(np, &dai_props->dai);
if (ret)
return ret;
@@ -230,11 +236,11 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
{
struct simple_card_data *priv;
struct snd_soc_dai_link *dai_link;
- struct asoc_simple_dai *dai_props;
+ struct simple_dai_props *dai_props;
struct snd_soc_card *card;
struct device *dev = &pdev->dev;
struct device_node *np = dev->of_node;
- int num, ret;
+ int num, ret, i;
/* Allocate the private data */
priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -248,6 +254,18 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
if (!dai_props || !dai_link)
return -ENOMEM;
+ /*
+ * Use snd_soc_dai_link_component instead of legacy style
+ * It is codec only. but cpu/platform will be supported in the future.
+ * see
+ * soc-core.c :: snd_soc_init_multicodec()
+ */
+ for (i = 0; i < num; i++) {
+ dai_link[i].codecs = &dai_props[i].codecs;
+ dai_link[i].num_codecs = 1;
+ dai_link[i].platform = &dai_props[i].platform;
+ }
+
priv->dai_props = dai_props;
priv->dai_link = dai_link;
diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c
index 53344a3b7a60..a69e5b11b3da 100644
--- a/sound/soc/hisilicon/hi6210-i2s.c
+++ b/sound/soc/hisilicon/hi6210-i2s.c
@@ -269,13 +269,13 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream,
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_U16_LE:
signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
- /* fallthru */
+ /* fall through */
case SNDRV_PCM_FORMAT_S16_LE:
bits = HII2S_BITS_16;
break;
case SNDRV_PCM_FORMAT_U24_LE:
signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
- /* fallthru */
+ /* fall through */
case SNDRV_PCM_FORMAT_S24_LE:
bits = HII2S_BITS_24;
break;
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 6c36da560877..afc559866095 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -765,7 +765,7 @@ static int sst_soc_prepare(struct device *dev)
snd_soc_poweroff(drv->soc_card->dev);
/* set the SSPs to idle */
- list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) {
+ for_each_card_rtds(drv->soc_card, rtd) {
struct snd_soc_dai *dai = rtd->cpu_dai;
if (dai->active) {
@@ -786,7 +786,7 @@ static void sst_soc_complete(struct device *dev)
return;
/* restart SSPs */
- list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) {
+ for_each_card_rtds(drv->soc_card, rtd) {
struct snd_soc_dai *dai = rtd->cpu_dai;
if (dai->active) {
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index cccda87f4b34..73ca1350aa31 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -279,6 +279,28 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH
This adds support for ASoC Onboard Codec I2S machine driver. This will
create an alsa sound card for DA7219 + MAX98357A I2S audio codec.
Say Y if you have such a device.
+
+config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH
+ tristate "KBL with DA7219 and MAX98927 in I2S Mode"
+ depends on MFD_INTEL_LPSS && I2C && ACPI
+ select SND_SOC_DA7219
+ select SND_SOC_MAX98927
+ select SND_SOC_DMIC
+ select SND_SOC_HDAC_HDMI
+ help
+ This adds support for ASoC Onboard Codec I2S machine driver. This will
+ create an alsa sound card for DA7219 + MAX98927 I2S audio codec.
+ Say Y if you have such a device.
+ If unsure select "N".
+
+config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
+ tristate "SKL/KBL/BXT/APL with HDA Codecs"
+ select SND_SOC_HDAC_HDMI
+ select SND_SOC_HDAC_HDA
+ help
+ This adds support for ASoC machine driver for Intel platforms
+ SKL/KBL/BXT/APL with iDisp, HDA audio codecs.
+ Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 87ef8b4058e5..5381e27df9cc 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -17,9 +17,11 @@ snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o
snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o
snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o
snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o
+snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o
snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o
snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o
snd-soc-skl_rt286-objs := skl_rt286.o
+snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o
snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o
snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o
@@ -41,8 +43,10 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o
obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o
+obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o
obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o
obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt5663_rt5514_max98927.o
obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o
obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o
obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o
+obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 7b0ee67b4fc8..68e6543e6cb0 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -223,7 +223,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
static int broadwell_suspend(struct snd_soc_card *card){
struct snd_soc_component *component;
- list_for_each_entry(component, &card->component_dev_list, card_list) {
+ for_each_card_components(card, component) {
if (!strcmp(component->name, "i2c-INT343A:00")) {
dev_dbg(component->dev, "disabling jack detect before going to suspend.\n");
@@ -237,7 +237,7 @@ static int broadwell_suspend(struct snd_soc_card *card){
static int broadwell_resume(struct snd_soc_card *card){
struct snd_soc_component *component;
- list_for_each_entry(component, &card->component_dev_list, card_list) {
+ for_each_card_components(card, component) {
if (!strcmp(component->name, "i2c-INT343A:00")) {
dev_dbg(component->dev, "enabling jack detect for resume.\n");
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index b6dc524830b2..8587bd3d1cc1 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -1048,7 +1048,7 @@ static int byt_rt5640_suspend(struct snd_soc_card *card)
if (!BYT_RT5640_JDSRC(byt_rt5640_quirk))
return 0;
- list_for_each_entry(component, &card->component_dev_list, card_list) {
+ for_each_card_components(card, component) {
if (!strcmp(component->name, byt_rt5640_codec_name)) {
dev_dbg(component->dev, "disabling jack detect before suspend\n");
snd_soc_component_set_jack(component, NULL, NULL);
@@ -1067,7 +1067,7 @@ static int byt_rt5640_resume(struct snd_soc_card *card)
if (!BYT_RT5640_JDSRC(byt_rt5640_quirk))
return 0;
- list_for_each_entry(component, &card->component_dev_list, card_list) {
+ for_each_card_components(card, component) {
if (!strcmp(component->name, byt_rt5640_codec_name)) {
dev_dbg(component->dev, "re-enabling jack detect after resume\n");
snd_soc_component_set_jack(component, &priv->jack, NULL);
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 6af02bf879ac..c44298130720 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -742,7 +742,7 @@ static int byt_rt5651_suspend(struct snd_soc_card *card)
if (!BYT_RT5651_JDSRC(byt_rt5651_quirk))
return 0;
- list_for_each_entry(component, &card->component_dev_list, card_list) {
+ for_each_card_components(card, component) {
if (!strcmp(component->name, byt_rt5651_codec_name)) {
dev_dbg(component->dev, "disabling jack detect before suspend\n");
snd_soc_component_set_jack(component, NULL, NULL);
@@ -761,7 +761,7 @@ static int byt_rt5651_resume(struct snd_soc_card *card)
if (!BYT_RT5651_JDSRC(byt_rt5651_quirk))
return 0;
- list_for_each_entry(component, &card->component_dev_list, card_list) {
+ for_each_card_components(card, component) {
if (!strcmp(component->name, byt_rt5651_codec_name)) {
dev_dbg(component->dev, "re-enabling jack detect after resume\n");
snd_soc_component_set_jack(component, &priv->jack, NULL);
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
index e5aa13058dd7..51f0d45d6f8f 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5672.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -16,6 +16,7 @@
* General Public License for more details.
*/
+#include <linux/input.h>
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
@@ -212,6 +213,10 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
if (ret)
return ret;
+ snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+ snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+ snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+
rt5670_set_jack_detect(component, &ctx->headset);
if (ctx->mclk) {
/*
@@ -342,7 +347,7 @@ static int cht_suspend_pre(struct snd_soc_card *card)
struct snd_soc_component *component;
struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card);
- list_for_each_entry(component, &card->component_dev_list, card_list) {
+ for_each_card_components(card, component) {
if (!strncmp(component->name,
ctx->codec_name, sizeof(ctx->codec_name))) {
@@ -359,7 +364,7 @@ static int cht_resume_post(struct snd_soc_card *card)
struct snd_soc_component *component;
struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card);
- list_for_each_entry(component, &card->component_dev_list, card_list) {
+ for_each_card_components(card, component) {
if (!strncmp(component->name,
ctx->codec_name, sizeof(ctx->codec_name))) {
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
new file mode 100644
index 000000000000..3fa1c3ca6d37
--- /dev/null
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -0,0 +1,983 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2018 Intel Corporation.
+
+/*
+ * Intel Kabylake I2S Machine Driver with MAX98927 & DA7219 Codecs
+ *
+ * Modified from:
+ * Intel Kabylake I2S Machine driver supporting MAX98927 and
+ * RT5663 codecs
+ */
+
+#include <linux/input.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/da7219.h"
+#include "../../codecs/hdac_hdmi.h"
+#include "../skylake/skl.h"
+#include "../../codecs/da7219-aad.h"
+
+#define KBL_DIALOG_CODEC_DAI "da7219-hifi"
+#define MAX98927_CODEC_DAI "max98927-aif1"
+#define MAXIM_DEV0_NAME "i2c-MX98927:00"
+#define MAXIM_DEV1_NAME "i2c-MX98927:01"
+#define DUAL_CHANNEL 2
+#define QUAD_CHANNEL 4
+#define NAME_SIZE 32
+
+static struct snd_soc_card *kabylake_audio_card;
+static struct snd_soc_jack kabylake_hdmi[3];
+
+struct kbl_hdmi_pcm {
+ struct list_head head;
+ struct snd_soc_dai *codec_dai;
+ int device;
+};
+
+struct kbl_codec_private {
+ struct snd_soc_jack kabylake_headset;
+ struct list_head hdmi_pcm_list;
+};
+
+enum {
+ KBL_DPCM_AUDIO_PB = 0,
+ KBL_DPCM_AUDIO_CP,
+ KBL_DPCM_AUDIO_ECHO_REF_CP,
+ KBL_DPCM_AUDIO_REF_CP,
+ KBL_DPCM_AUDIO_DMIC_CP,
+ KBL_DPCM_AUDIO_HDMI1_PB,
+ KBL_DPCM_AUDIO_HDMI2_PB,
+ KBL_DPCM_AUDIO_HDMI3_PB,
+ KBL_DPCM_AUDIO_HS_PB,
+};
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ struct snd_soc_dapm_context *dapm = w->dapm;
+ struct snd_soc_card *card = dapm->card;
+ struct snd_soc_dai *codec_dai;
+ int ret = 0;
+
+ codec_dai = snd_soc_card_get_codec_dai(card, KBL_DIALOG_CODEC_DAI);
+ if (!codec_dai) {
+ dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n");
+ return -EIO;
+ }
+
+ /* Configure sysclk for codec */
+ ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000,
+ SND_SOC_CLOCK_IN);
+ if (ret) {
+ dev_err(card->dev, "can't set codec sysclk configuration\n");
+ return ret;
+ }
+
+ if (SND_SOC_DAPM_EVENT_OFF(event)) {
+ ret = snd_soc_dai_set_pll(codec_dai, 0,
+ DA7219_SYSCLK_MCLK, 0, 0);
+ if (ret)
+ dev_err(card->dev, "failed to stop PLL: %d\n", ret);
+ } else if (SND_SOC_DAPM_EVENT_ON(event)) {
+ ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM,
+ 0, DA7219_PLL_FREQ_OUT_98304);
+ if (ret)
+ dev_err(card->dev, "failed to start PLL: %d\n", ret);
+ }
+
+ return ret;
+}
+
+static const struct snd_kcontrol_new kabylake_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Left Spk"),
+ SOC_DAPM_PIN_SWITCH("Right Spk"),
+};
+
+static const struct snd_soc_dapm_widget kabylake_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_SPK("Left Spk", NULL),
+ SND_SOC_DAPM_SPK("Right Spk", NULL),
+ SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+ SND_SOC_DAPM_SPK("DP", NULL),
+ SND_SOC_DAPM_SPK("HDMI", NULL),
+ SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+ platform_clock_control, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route kabylake_map[] = {
+ /* speaker */
+ { "Left Spk", NULL, "Left BE_OUT" },
+ { "Right Spk", NULL, "Right BE_OUT" },
+
+ /* other jacks */
+ { "DMic", NULL, "SoC DMIC" },
+
+ { "HDMI", NULL, "hif5 Output" },
+ { "DP", NULL, "hif6 Output" },
+
+ /* CODEC BE connections */
+ { "Left HiFi Playback", NULL, "ssp0 Tx" },
+ { "Right HiFi Playback", NULL, "ssp0 Tx" },
+ { "ssp0 Tx", NULL, "spk_out" },
+
+ /* IV feedback path */
+ { "codec0_fb_in", NULL, "ssp0 Rx"},
+ { "ssp0 Rx", NULL, "Left HiFi Capture" },
+ { "ssp0 Rx", NULL, "Right HiFi Capture" },
+
+ /* AEC capture path */
+ { "echo_ref_out", NULL, "ssp0 Rx" },
+
+ /* DMIC */
+ { "dmic01_hifi", NULL, "DMIC01 Rx" },
+ { "DMIC01 Rx", NULL, "DMIC AIF" },
+
+ { "hifi1", NULL, "iDisp1 Tx" },
+ { "iDisp1 Tx", NULL, "iDisp1_out" },
+ { "hifi2", NULL, "iDisp2 Tx" },
+ { "iDisp2 Tx", NULL, "iDisp2_out" },
+ { "hifi3", NULL, "iDisp3 Tx"},
+ { "iDisp3 Tx", NULL, "iDisp3_out"},
+};
+
+static const struct snd_soc_dapm_route kabylake_ssp1_map[] = {
+ { "Headphone Jack", NULL, "HPL" },
+ { "Headphone Jack", NULL, "HPR" },
+
+ /* other jacks */
+ { "MIC", NULL, "Headset Mic" },
+
+ /* CODEC BE connections */
+ { "Playback", NULL, "ssp1 Tx" },
+ { "ssp1 Tx", NULL, "codec1_out" },
+
+ { "hs_in", NULL, "ssp1 Rx" },
+ { "ssp1 Rx", NULL, "Capture" },
+
+ { "Headphone Jack", NULL, "Platform Clock" },
+ { "Headset Mic", NULL, "Platform Clock" },
+};
+
+static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *runtime = substream->private_data;
+ int ret = 0, j;
+
+ for (j = 0; j < runtime->num_codecs; j++) {
+ struct snd_soc_dai *codec_dai = runtime->codec_dais[j];
+
+ if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16);
+ if (ret < 0) {
+ dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret);
+ return ret;
+ }
+ }
+ if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) {
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16);
+ if (ret < 0) {
+ dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret);
+ return ret;
+ }
+ }
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops kabylake_ssp0_ops = {
+ .hw_params = kabylake_ssp0_hw_params,
+};
+
+static int kabylake_ssp_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);
+ struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ struct snd_soc_dpcm *dpcm = container_of(
+ params, struct snd_soc_dpcm, hw_params);
+ struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link;
+ struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link;
+
+ /*
+ * The ADSP will convert the FE rate to 48k, stereo, 24 bit
+ */
+ if (!strcmp(fe_dai_link->name, "Kbl Audio Port") ||
+ !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
+ !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+ snd_mask_none(fmt);
+ snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+ }
+
+ /*
+ * The speaker on the SSP0 supports S16_LE and not S24_LE.
+ * thus changing the mask here
+ */
+ if (!strcmp(be_dai_link->name, "SSP0-Codec"))
+ snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE);
+
+ return 0;
+}
+
+static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_jack *jack;
+ struct snd_soc_card *card = rtd->card;
+ int ret;
+
+
+ ret = snd_soc_dapm_add_routes(&card->dapm,
+ kabylake_ssp1_map,
+ ARRAY_SIZE(kabylake_ssp1_map));
+
+ /*
+ * Headset buttons map to the google Reference headset.
+ * These can be configured by userspace.
+ */
+ ret = snd_soc_card_jack_new(kabylake_audio_card, "Headset Jack",
+ SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+ SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT,
+ &ctx->kabylake_headset, NULL, 0);
+ if (ret) {
+ dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
+ return ret;
+ }
+
+ jack = &ctx->kabylake_headset;
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
+
+ da7219_aad_jack_det(component, &ctx->kabylake_headset);
+
+ ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC");
+ if (ret)
+ dev_err(rtd->dev, "SoC DMIC - Ignore suspend failed %d\n", ret);
+
+ return ret;
+}
+
+static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
+{
+ struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_dai *dai = rtd->codec_dai;
+ struct kbl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
+
+ pcm->device = device;
+ pcm->codec_dai = dai;
+
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
+}
+
+static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
+{
+ return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB);
+}
+
+static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
+{
+ return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB);
+}
+
+static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
+{
+ return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB);
+}
+
+static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dapm_context *dapm;
+ struct snd_soc_component *component = rtd->cpu_dai->component;
+
+ dapm = snd_soc_component_get_dapm(component);
+ snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
+
+ return 0;
+}
+
+static const unsigned int rates[] = {
+ 48000,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static const unsigned int channels[] = {
+ DUAL_CHANNEL,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_channels = {
+ .count = ARRAY_SIZE(channels),
+ .list = channels,
+ .mask = 0,
+};
+
+static unsigned int channels_quad[] = {
+ QUAD_CHANNEL,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_channels_quad = {
+ .count = ARRAY_SIZE(channels_quad),
+ .list = channels_quad,
+ .mask = 0,
+};
+
+static int kbl_fe_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ /*
+ * On this platform for PCM device we support,
+ * 48Khz
+ * stereo
+ * 16 bit audio
+ */
+
+ runtime->hw.channels_max = DUAL_CHANNEL;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_channels);
+
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+ snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16);
+
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+
+ return 0;
+}
+
+static const struct snd_soc_ops kabylake_da7219_fe_ops = {
+ .startup = kbl_fe_startup,
+};
+
+static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /*
+ * set BE channel constraint as user FE channels
+ */
+
+ if (params_channels(params) == 2)
+ channels->min = channels->max = 2;
+ else
+ channels->min = channels->max = 4;
+
+ return 0;
+}
+
+static int kabylake_dmic_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_channels_quad);
+
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+}
+
+static struct snd_soc_ops kabylake_dmic_ops = {
+ .startup = kabylake_dmic_startup,
+};
+
+static const unsigned int rates_16000[] = {
+ 16000,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_16000 = {
+ .count = ARRAY_SIZE(rates_16000),
+ .list = rates_16000,
+};
+
+static const unsigned int ch_mono[] = {
+ 1,
+};
+static const struct snd_pcm_hw_constraint_list constraints_refcap = {
+ .count = ARRAY_SIZE(ch_mono),
+ .list = ch_mono,
+};
+
+static int kabylake_refcap_startup(struct snd_pcm_substream *substream)
+{
+ substream->runtime->hw.channels_max = 1;
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_refcap);
+
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_16000);
+}
+
+
+static struct snd_soc_ops skylaye_refcap_ops = {
+ .startup = kabylake_refcap_startup,
+};
+
+static struct snd_soc_codec_conf max98927_codec_conf[] = {
+
+ {
+ .dev_name = MAXIM_DEV0_NAME,
+ .name_prefix = "Right",
+ },
+
+ {
+ .dev_name = MAXIM_DEV1_NAME,
+ .name_prefix = "Left",
+ },
+};
+
+static struct snd_soc_dai_link_component ssp0_codec_components[] = {
+ { /* Left */
+ .name = MAXIM_DEV0_NAME,
+ .dai_name = MAX98927_CODEC_DAI,
+ },
+
+ { /* For Right */
+ .name = MAXIM_DEV1_NAME,
+ .dai_name = MAX98927_CODEC_DAI,
+ },
+
+};
+
+/* kabylake digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link kabylake_dais[] = {
+ /* Front End DAI links */
+ [KBL_DPCM_AUDIO_PB] = {
+ .name = "Kbl Audio Port",
+ .stream_name = "Audio",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "0000:00:1f.3",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .nonatomic = 1,
+ .init = kabylake_da7219_fe_init,
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ .ops = &kabylake_da7219_fe_ops,
+ },
+ [KBL_DPCM_AUDIO_CP] = {
+ .name = "Kbl Audio Capture Port",
+ .stream_name = "Audio Record",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "0000:00:1f.3",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .nonatomic = 1,
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ .ops = &kabylake_da7219_fe_ops,
+ },
+ [KBL_DPCM_AUDIO_ECHO_REF_CP] = {
+ .name = "Kbl Audio Echo Reference cap",
+ .stream_name = "Echoreference Capture",
+ .cpu_dai_name = "Echoref Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:1f.3",
+ .init = NULL,
+ .capture_only = 1,
+ .nonatomic = 1,
+ },
+ [KBL_DPCM_AUDIO_REF_CP] = {
+ .name = "Kbl Audio Reference cap",
+ .stream_name = "Wake on Voice",
+ .cpu_dai_name = "Reference Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:1f.3",
+ .init = NULL,
+ .dpcm_capture = 1,
+ .nonatomic = 1,
+ .dynamic = 1,
+ .ops = &skylaye_refcap_ops,
+ },
+ [KBL_DPCM_AUDIO_DMIC_CP] = {
+ .name = "Kbl Audio DMIC cap",
+ .stream_name = "dmiccap",
+ .cpu_dai_name = "DMIC Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:1f.3",
+ .init = NULL,
+ .dpcm_capture = 1,
+ .nonatomic = 1,
+ .dynamic = 1,
+ .ops = &kabylake_dmic_ops,
+ },
+ [KBL_DPCM_AUDIO_HDMI1_PB] = {
+ .name = "Kbl HDMI Port1",
+ .stream_name = "Hdmi1",
+ .cpu_dai_name = "HDMI1 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:1f.3",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [KBL_DPCM_AUDIO_HDMI2_PB] = {
+ .name = "Kbl HDMI Port2",
+ .stream_name = "Hdmi2",
+ .cpu_dai_name = "HDMI2 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:1f.3",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [KBL_DPCM_AUDIO_HDMI3_PB] = {
+ .name = "Kbl HDMI Port3",
+ .stream_name = "Hdmi3",
+ .cpu_dai_name = "HDMI3 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:1f.3",
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ .init = NULL,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [KBL_DPCM_AUDIO_HS_PB] = {
+ .name = "Kbl Audio Headset Playback",
+ .stream_name = "Headset Audio",
+ .cpu_dai_name = "System Pin2",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:1f.3",
+ .dpcm_playback = 1,
+ .nonatomic = 1,
+ .dynamic = 1,
+ .init = kabylake_da7219_fe_init,
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .ops = &kabylake_da7219_fe_ops,
+
+ },
+
+ /* Back End DAI links */
+ {
+ /* SSP0 - Codec */
+ .name = "SSP0-Codec",
+ .id = 0,
+ .cpu_dai_name = "SSP0 Pin",
+ .platform_name = "0000:00:1f.3",
+ .no_pcm = 1,
+ .codecs = ssp0_codec_components,
+ .num_codecs = ARRAY_SIZE(ssp0_codec_components),
+ .dai_fmt = SND_SOC_DAIFMT_DSP_B |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = kabylake_ssp_fixup,
+ .ops = &kabylake_ssp0_ops,
+ },
+ {
+ /* SSP1 - Codec */
+ .name = "SSP1-Codec",
+ .id = 1,
+ .cpu_dai_name = "SSP1 Pin",
+ .platform_name = "0000:00:1f.3",
+ .no_pcm = 1,
+ .codec_name = "i2c-DLGS7219:00",
+ .codec_dai_name = KBL_DIALOG_CODEC_DAI,
+ .init = kabylake_da7219_codec_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = kabylake_ssp_fixup,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+ {
+ .name = "dmic01",
+ .id = 2,
+ .cpu_dai_name = "DMIC01 Pin",
+ .codec_name = "dmic-codec",
+ .codec_dai_name = "dmic-hifi",
+ .platform_name = "0000:00:1f.3",
+ .be_hw_params_fixup = kabylake_dmic_fixup,
+ .ignore_suspend = 1,
+ .dpcm_capture = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp1",
+ .id = 3,
+ .cpu_dai_name = "iDisp1 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi1",
+ .platform_name = "0000:00:1f.3",
+ .dpcm_playback = 1,
+ .init = kabylake_hdmi1_init,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp2",
+ .id = 4,
+ .cpu_dai_name = "iDisp2 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi2",
+ .platform_name = "0000:00:1f.3",
+ .init = kabylake_hdmi2_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp3",
+ .id = 5,
+ .cpu_dai_name = "iDisp3 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi3",
+ .platform_name = "0000:00:1f.3",
+ .init = kabylake_hdmi3_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+};
+
+/* kabylake digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link kabylake_max98927_dais[] = {
+ /* Front End DAI links */
+ [KBL_DPCM_AUDIO_PB] = {
+ .name = "Kbl Audio Port",
+ .stream_name = "Audio",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "0000:00:1f.3",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .nonatomic = 1,
+ .init = kabylake_da7219_fe_init,
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ .ops = &kabylake_da7219_fe_ops,
+ },
+ [KBL_DPCM_AUDIO_CP] = {
+ .name = "Kbl Audio Capture Port",
+ .stream_name = "Audio Record",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "0000:00:1f.3",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .nonatomic = 1,
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ .ops = &kabylake_da7219_fe_ops,
+ },
+ [KBL_DPCM_AUDIO_ECHO_REF_CP] = {
+ .name = "Kbl Audio Echo Reference cap",
+ .stream_name = "Echoreference Capture",
+ .cpu_dai_name = "Echoref Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:1f.3",
+ .init = NULL,
+ .capture_only = 1,
+ .nonatomic = 1,
+ },
+ [KBL_DPCM_AUDIO_REF_CP] = {
+ .name = "Kbl Audio Reference cap",
+ .stream_name = "Wake on Voice",
+ .cpu_dai_name = "Reference Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:1f.3",
+ .init = NULL,
+ .dpcm_capture = 1,
+ .nonatomic = 1,
+ .dynamic = 1,
+ .ops = &skylaye_refcap_ops,
+ },
+ [KBL_DPCM_AUDIO_DMIC_CP] = {
+ .name = "Kbl Audio DMIC cap",
+ .stream_name = "dmiccap",
+ .cpu_dai_name = "DMIC Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:1f.3",
+ .init = NULL,
+ .dpcm_capture = 1,
+ .nonatomic = 1,
+ .dynamic = 1,
+ .ops = &kabylake_dmic_ops,
+ },
+ [KBL_DPCM_AUDIO_HDMI1_PB] = {
+ .name = "Kbl HDMI Port1",
+ .stream_name = "Hdmi1",
+ .cpu_dai_name = "HDMI1 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:1f.3",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [KBL_DPCM_AUDIO_HDMI2_PB] = {
+ .name = "Kbl HDMI Port2",
+ .stream_name = "Hdmi2",
+ .cpu_dai_name = "HDMI2 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:1f.3",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [KBL_DPCM_AUDIO_HDMI3_PB] = {
+ .name = "Kbl HDMI Port3",
+ .stream_name = "Hdmi3",
+ .cpu_dai_name = "HDMI3 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:1f.3",
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ .init = NULL,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+
+ /* Back End DAI links */
+ {
+ /* SSP0 - Codec */
+ .name = "SSP0-Codec",
+ .id = 0,
+ .cpu_dai_name = "SSP0 Pin",
+ .platform_name = "0000:00:1f.3",
+ .no_pcm = 1,
+ .codecs = ssp0_codec_components,
+ .num_codecs = ARRAY_SIZE(ssp0_codec_components),
+ .dai_fmt = SND_SOC_DAIFMT_DSP_B |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = kabylake_ssp_fixup,
+ .ops = &kabylake_ssp0_ops,
+ },
+ {
+ .name = "dmic01",
+ .id = 1,
+ .cpu_dai_name = "DMIC01 Pin",
+ .codec_name = "dmic-codec",
+ .codec_dai_name = "dmic-hifi",
+ .platform_name = "0000:00:1f.3",
+ .be_hw_params_fixup = kabylake_dmic_fixup,
+ .ignore_suspend = 1,
+ .dpcm_capture = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp1",
+ .id = 2,
+ .cpu_dai_name = "iDisp1 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi1",
+ .platform_name = "0000:00:1f.3",
+ .dpcm_playback = 1,
+ .init = kabylake_hdmi1_init,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp2",
+ .id = 3,
+ .cpu_dai_name = "iDisp2 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi2",
+ .platform_name = "0000:00:1f.3",
+ .init = kabylake_hdmi2_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp3",
+ .id = 4,
+ .cpu_dai_name = "iDisp3 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi3",
+ .platform_name = "0000:00:1f.3",
+ .init = kabylake_hdmi3_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+};
+
+static int kabylake_card_late_probe(struct snd_soc_card *card)
+{
+ struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card);
+ struct kbl_hdmi_pcm *pcm;
+ struct snd_soc_component *component = NULL;
+ int err, i = 0;
+ char jack_name[NAME_SIZE];
+
+ list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+ component = pcm->codec_dai->component;
+ snprintf(jack_name, sizeof(jack_name),
+ "HDMI/DP, pcm=%d Jack", pcm->device);
+ err = snd_soc_card_jack_new(card, jack_name,
+ SND_JACK_AVOUT, &kabylake_hdmi[i],
+ NULL, 0);
+
+ if (err)
+ return err;
+
+ err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
+ &kabylake_hdmi[i]);
+ if (err < 0)
+ return err;
+
+ i++;
+ }
+
+ if (!component)
+ return -EINVAL;
+
+ return hdac_hdmi_jack_port_init(component, &card->dapm);
+
+ return 0;
+}
+
+/* kabylake audio machine driver for SPT + DA7219 */
+static struct snd_soc_card kbl_audio_card_da7219_m98927 = {
+ .name = "kblda7219m98927",
+ .owner = THIS_MODULE,
+ .dai_link = kabylake_dais,
+ .num_links = ARRAY_SIZE(kabylake_dais),
+ .controls = kabylake_controls,
+ .num_controls = ARRAY_SIZE(kabylake_controls),
+ .dapm_widgets = kabylake_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets),
+ .dapm_routes = kabylake_map,
+ .num_dapm_routes = ARRAY_SIZE(kabylake_map),
+ .codec_conf = max98927_codec_conf,
+ .num_configs = ARRAY_SIZE(max98927_codec_conf),
+ .fully_routed = true,
+ .late_probe = kabylake_card_late_probe,
+};
+
+/* kabylake audio machine driver for Maxim98927 */
+static struct snd_soc_card kbl_audio_card_max98927 = {
+ .name = "kblmax98927",
+ .owner = THIS_MODULE,
+ .dai_link = kabylake_max98927_dais,
+ .num_links = ARRAY_SIZE(kabylake_max98927_dais),
+ .controls = kabylake_controls,
+ .num_controls = ARRAY_SIZE(kabylake_controls),
+ .dapm_widgets = kabylake_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets),
+ .dapm_routes = kabylake_map,
+ .num_dapm_routes = ARRAY_SIZE(kabylake_map),
+ .codec_conf = max98927_codec_conf,
+ .num_configs = ARRAY_SIZE(max98927_codec_conf),
+ .fully_routed = true,
+ .late_probe = kabylake_card_late_probe,
+};
+
+static int kabylake_audio_probe(struct platform_device *pdev)
+{
+ struct kbl_codec_private *ctx;
+
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+ if (!ctx)
+ return -ENOMEM;
+
+ INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
+ kabylake_audio_card =
+ (struct snd_soc_card *)pdev->id_entry->driver_data;
+
+ kabylake_audio_card->dev = &pdev->dev;
+ snd_soc_card_set_drvdata(kabylake_audio_card, ctx);
+
+ return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card);
+}
+
+static const struct platform_device_id kbl_board_ids[] = {
+ {
+ .name = "kbl_da7219_max98927",
+ .driver_data =
+ (kernel_ulong_t)&kbl_audio_card_da7219_m98927,
+ },
+ {
+ .name = "kbl_max98927",
+ .driver_data =
+ (kernel_ulong_t)&kbl_audio_card_max98927,
+ },
+ { }
+};
+
+static struct platform_driver kabylake_audio = {
+ .probe = kabylake_audio_probe,
+ .driver = {
+ .name = "kbl_da7219_max98927",
+ .pm = &snd_soc_pm_ops,
+ },
+ .id_table = kbl_board_ids,
+};
+
+module_platform_driver(kabylake_audio)
+
+/* Module information */
+MODULE_DESCRIPTION("Audio KabyLake Machine driver for MAX98927 & DA7219");
+MODULE_AUTHOR("Mac Chiang <mac.chiang@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:kbl_da7219_max98927");
+MODULE_ALIAS("platform:kbl_max98927");
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 21a6490746a6..99e1320c485f 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -488,11 +488,10 @@ static int kabylake_ssp0_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;
int ret = 0, j;
- for (j = 0; j < rtd->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
-
+ for_each_rtd_codec_dai(rtd, j, codec_dai) {
if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
/*
* Use channel 4 and 5 for the first amp
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index a892b37eab7c..a737c915d46a 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -353,11 +353,10 @@ static int kabylake_ssp0_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;
int ret = 0, j;
- for (j = 0; j < rtd->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
-
+ for_each_rtd_codec_dai(rtd, j, codec_dai) {
if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) {
ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16);
if (ret < 0) {
diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c
new file mode 100644
index 000000000000..3fdbf239da74
--- /dev/null
+++ b/sound/soc/intel/boards/skl_hda_dsp_common.c
@@ -0,0 +1,127 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2015-18 Intel Corporation.
+
+/*
+ * Common functions used in different Intel machine drivers
+ */
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/hdac_hdmi.h"
+#include "../skylake/skl.h"
+#include "skl_hda_dsp_common.h"
+
+#define NAME_SIZE 32
+
+int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device)
+{
+ struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
+ struct skl_hda_hdmi_pcm *pcm;
+ char dai_name[NAME_SIZE];
+
+ pcm = devm_kzalloc(card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
+
+ snprintf(dai_name, sizeof(dai_name), "intel-hdmi-hifi%d",
+ ctx->dai_index);
+ pcm->codec_dai = snd_soc_card_get_codec_dai(card, dai_name);
+ if (!pcm->codec_dai)
+ return -EINVAL;
+
+ pcm->device = device;
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
+}
+
+/* skl_hda_digital audio interface glue - connects codec <--> CPU */
+struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = {
+ /* Back End DAI links */
+ {
+ .name = "iDisp1",
+ .id = 1,
+ .cpu_dai_name = "iDisp1 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi1",
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp2",
+ .id = 2,
+ .cpu_dai_name = "iDisp2 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi2",
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp3",
+ .id = 3,
+ .cpu_dai_name = "iDisp3 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi3",
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "Analog Playback and Capture",
+ .id = 4,
+ .cpu_dai_name = "Analog CPU DAI",
+ .codec_name = "ehdaudio0D0",
+ .codec_dai_name = "Analog Codec DAI",
+ .platform_name = "0000:00:1f.3",
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .init = NULL,
+ .no_pcm = 1,
+ },
+ {
+ .name = "Digital Playback and Capture",
+ .id = 5,
+ .cpu_dai_name = "Digital CPU DAI",
+ .codec_name = "ehdaudio0D0",
+ .codec_dai_name = "Digital Codec DAI",
+ .platform_name = "0000:00:1f.3",
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .init = NULL,
+ .no_pcm = 1,
+ },
+};
+
+int skl_hda_hdmi_jack_init(struct snd_soc_card *card)
+{
+ struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
+ struct snd_soc_component *component = NULL;
+ struct skl_hda_hdmi_pcm *pcm;
+ char jack_name[NAME_SIZE];
+ int err;
+
+ list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+ component = pcm->codec_dai->component;
+ snprintf(jack_name, sizeof(jack_name),
+ "HDMI/DP, pcm=%d Jack", pcm->device);
+ err = snd_soc_card_jack_new(card, jack_name,
+ SND_JACK_AVOUT, &pcm->hdmi_jack,
+ NULL, 0);
+
+ if (err)
+ return err;
+
+ err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
+ &pcm->hdmi_jack);
+ if (err < 0)
+ return err;
+ }
+
+ if (!component)
+ return -EINVAL;
+
+ return hdac_hdmi_jack_port_init(component, &card->dapm);
+}
diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h
new file mode 100644
index 000000000000..87c50aff56cd
--- /dev/null
+++ b/sound/soc/intel/boards/skl_hda_dsp_common.h
@@ -0,0 +1,38 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright(c) 2015-18 Intel Corporation.
+ */
+
+/*
+ * This file defines data structures used in Machine Driver for Intel
+ * platforms with HDA Codecs.
+ */
+
+#ifndef __SOUND_SOC_HDA_DSP_COMMON_H
+#define __SOUND_SOC_HDA_DSP_COMMON_H
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+
+#define HDA_DSP_MAX_BE_DAI_LINKS 5
+
+struct skl_hda_hdmi_pcm {
+ struct list_head head;
+ struct snd_soc_dai *codec_dai;
+ struct snd_soc_jack hdmi_jack;
+ int device;
+};
+
+struct skl_hda_private {
+ struct list_head hdmi_pcm_list;
+ int pcm_count;
+ int dai_index;
+ const char *platform_name;
+};
+
+extern struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS];
+int skl_hda_hdmi_jack_init(struct snd_soc_card *card);
+int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device);
+
+#endif /* __SOUND_SOC_HDA_DSP_COMMON_H */
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
new file mode 100644
index 000000000000..b415dd4c85f5
--- /dev/null
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -0,0 +1,183 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2015-18 Intel Corporation.
+
+/*
+ * Machine Driver for SKL+ platforms with DSP and iDisp, HDA Codecs
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/hdac_hdmi.h"
+#include "../skylake/skl.h"
+#include "skl_hda_dsp_common.h"
+
+static const struct snd_soc_dapm_widget skl_hda_widgets[] = {
+ SND_SOC_DAPM_HP("Analog Out", NULL),
+ SND_SOC_DAPM_MIC("Analog In", NULL),
+ SND_SOC_DAPM_HP("Alt Analog Out", NULL),
+ SND_SOC_DAPM_MIC("Alt Analog In", NULL),
+ SND_SOC_DAPM_SPK("Digital Out", NULL),
+ SND_SOC_DAPM_MIC("Digital In", NULL),
+};
+
+static const struct snd_soc_dapm_route skl_hda_map[] = {
+ { "hifi3", NULL, "iDisp3 Tx"},
+ { "iDisp3 Tx", NULL, "iDisp3_out"},
+ { "hifi2", NULL, "iDisp2 Tx"},
+ { "iDisp2 Tx", NULL, "iDisp2_out"},
+ { "hifi1", NULL, "iDisp1 Tx"},
+ { "iDisp1 Tx", NULL, "iDisp1_out"},
+
+ { "Analog Out", NULL, "Codec Output Pin1" },
+ { "Digital Out", NULL, "Codec Output Pin2" },
+ { "Alt Analog Out", NULL, "Codec Output Pin3" },
+
+ { "Codec Input Pin1", NULL, "Analog In" },
+ { "Codec Input Pin2", NULL, "Digital In" },
+ { "Codec Input Pin3", NULL, "Alt Analog In" },
+
+ /* CODEC BE connections */
+ { "Analog Codec Playback", NULL, "Analog CPU Playback" },
+ { "Analog CPU Playback", NULL, "codec0_out" },
+ { "Digital Codec Playback", NULL, "Digital CPU Playback" },
+ { "Digital CPU Playback", NULL, "codec1_out" },
+ { "Alt Analog Codec Playback", NULL, "Alt Analog CPU Playback" },
+ { "Alt Analog CPU Playback", NULL, "codec2_out" },
+
+ { "codec0_in", NULL, "Analog CPU Capture" },
+ { "Analog CPU Capture", NULL, "Analog Codec Capture" },
+ { "codec1_in", NULL, "Digital CPU Capture" },
+ { "Digital CPU Capture", NULL, "Digital Codec Capture" },
+ { "codec2_in", NULL, "Alt Analog CPU Capture" },
+ { "Alt Analog CPU Capture", NULL, "Alt Analog Codec Capture" },
+};
+
+static int skl_hda_card_late_probe(struct snd_soc_card *card)
+{
+ return skl_hda_hdmi_jack_init(card);
+}
+
+static int
+skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link)
+{
+ struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
+ int ret = 0;
+
+ dev_dbg(card->dev, "%s: dai link name - %s\n", __func__, link->name);
+ link->platform_name = ctx->platform_name;
+ link->nonatomic = 1;
+
+ if (strstr(link->name, "HDMI")) {
+ ret = skl_hda_hdmi_add_pcm(card, ctx->pcm_count);
+
+ if (ret < 0)
+ return ret;
+
+ ctx->dai_index++;
+ }
+
+ ctx->pcm_count++;
+ return ret;
+}
+
+static struct snd_soc_card hda_soc_card = {
+ .name = "skl_hda_card",
+ .owner = THIS_MODULE,
+ .dai_link = skl_hda_be_dai_links,
+ .dapm_widgets = skl_hda_widgets,
+ .dapm_routes = skl_hda_map,
+ .add_dai_link = skl_hda_add_dai_link,
+ .fully_routed = true,
+ .late_probe = skl_hda_card_late_probe,
+};
+
+#define IDISP_DAI_COUNT 3
+/* there are two routes per iDisp output */
+#define IDISP_ROUTE_COUNT (IDISP_DAI_COUNT * 2)
+#define IDISP_CODEC_MASK 0x4
+
+static int skl_hda_fill_card_info(struct skl_machine_pdata *pdata)
+{
+ struct snd_soc_card *card = &hda_soc_card;
+ struct snd_soc_dai_link *dai_link;
+ u32 codec_count, codec_mask;
+ int i, num_links, num_route;
+
+ codec_mask = pdata->codec_mask;
+ codec_count = hweight_long(codec_mask);
+
+ if (codec_count == 1 && pdata->codec_mask & IDISP_CODEC_MASK) {
+ num_links = IDISP_DAI_COUNT;
+ num_route = IDISP_ROUTE_COUNT;
+ } else if (codec_count == 2 && codec_mask & IDISP_CODEC_MASK) {
+ num_links = ARRAY_SIZE(skl_hda_be_dai_links);
+ num_route = ARRAY_SIZE(skl_hda_map),
+ card->dapm_widgets = skl_hda_widgets;
+ card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets);
+ } else {
+ return -EINVAL;
+ }
+
+ card->num_links = num_links;
+ card->num_dapm_routes = num_route;
+
+ for_each_card_prelinks(card, i, dai_link)
+ dai_link->platform_name = pdata->platform;
+
+ return 0;
+}
+
+static int skl_hda_audio_probe(struct platform_device *pdev)
+{
+ struct skl_machine_pdata *pdata;
+ struct skl_hda_private *ctx;
+ int ret;
+
+ dev_dbg(&pdev->dev, "%s: entry\n", __func__);
+
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+ if (!ctx)
+ return -ENOMEM;
+
+ INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
+ pdata = dev_get_drvdata(&pdev->dev);
+ if (!pdata)
+ return -EINVAL;
+
+ ret = skl_hda_fill_card_info(pdata);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n");
+ return ret;
+ }
+
+ ctx->pcm_count = hda_soc_card.num_links;
+ ctx->dai_index = 1; /* hdmi codec dai name starts from index 1 */
+ ctx->platform_name = pdata->platform;
+
+ hda_soc_card.dev = &pdev->dev;
+ snd_soc_card_set_drvdata(&hda_soc_card, ctx);
+
+ return devm_snd_soc_register_card(&pdev->dev, &hda_soc_card);
+}
+
+static struct platform_driver skl_hda_audio = {
+ .probe = skl_hda_audio_probe,
+ .driver = {
+ .name = "skl_hda_dsp_generic",
+ .pm = &snd_soc_pm_ops,
+ },
+};
+
+module_platform_driver(skl_hda_audio)
+
+/* Module information */
+MODULE_DESCRIPTION("SKL/KBL/BXT/APL HDA Generic Machine driver");
+MODULE_AUTHOR("Rakesh Ughreja <rakesh.a.ughreja@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:skl_hda_dsp_generic");
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index 915a34cdc8ac..c1f50a079d34 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -7,7 +7,8 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m
soc-acpi-intel-hsw-bdw-match.o \
soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \
soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \
- soc-acpi-intel-cnl-match.o
+ soc-acpi-intel-cnl-match.o \
+ soc-acpi-intel-hda-match.o
obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o
diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
index 4daa8a4f0c0c..097dc06377ba 100644
--- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
@@ -34,6 +34,13 @@ static const struct dmi_system_id byt_table[] = {
.callback = byt_thinkpad10_quirk_cb,
.matches = {
DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+ DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 8"),
+ },
+ },
+ {
+ .callback = byt_thinkpad10_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 10"),
},
},
diff --git a/sound/soc/intel/common/soc-acpi-intel-hda-match.c b/sound/soc/intel/common/soc-acpi-intel-hda-match.c
new file mode 100644
index 000000000000..533c1064f84b
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-hda-match.c
@@ -0,0 +1,40 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2018, Intel Corporation.
+
+/*
+ * soc-apci-intel-hda-match.c - tables and support for HDA+ACPI enumeration.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+#include "../skylake/skl.h"
+
+static struct skl_machine_pdata hda_pdata = {
+ .use_tplg_pcm = true,
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[] = {
+ {
+ /* .id is not used in this file */
+ .drv_name = "skl_hda_dsp_generic",
+
+ /* .fw_filename is dynamically set in skylake driver */
+
+ /* .sof_fw_filename is dynamically set in sof/intel driver */
+
+ .sof_tplg_filename = "intel/sof-hda-generic.tplg",
+
+ /*
+ * .machine_quirk and .quirk_data are not used here but
+ * can be used if we need a more complicated machine driver
+ * combining HDA+other device (e.g. DMIC).
+ */
+ .pdata = &hda_pdata,
+ },
+ {},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_hda_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
index 0ee173ca437d..a317b7790fce 100644
--- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
@@ -32,6 +32,11 @@ static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = {
.codecs = {"MX98357A"}
};
+static struct snd_soc_acpi_codecs kbl_7219_98927_codecs = {
+ .num_codecs = 1,
+ .codecs = {"MX98927"}
+};
+
struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = {
{
.id = "INT343A",
@@ -83,6 +88,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = {
.quirk_data = &kbl_7219_98357_codecs,
.pdata = &skl_dmic_data,
},
+ {
+ .id = "DLGS7219",
+ .drv_name = "kbl_da7219_max98927",
+ .fw_filename = "intel/dsp_fw_kbl.bin",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &kbl_7219_98927_codecs,
+ .pdata = &skl_dmic_data
+ },
{},
};
EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines);
diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c
index 11041aedea31..1e067504b604 100644
--- a/sound/soc/intel/common/sst-firmware.c
+++ b/sound/soc/intel/common/sst-firmware.c
@@ -355,7 +355,7 @@ struct sst_fw *sst_fw_new(struct sst_dsp *dsp,
/* allocate DMA buffer to store FW data */
sst_fw->dma_buf = dma_alloc_coherent(dsp->dma_dev, sst_fw->size,
- &sst_fw->dmable_fw_paddr, GFP_DMA | GFP_KERNEL);
+ &sst_fw->dmable_fw_paddr, GFP_KERNEL);
if (!sst_fw->dma_buf) {
dev_err(dsp->dev, "error: DMA alloc failed\n");
kfree(sst_fw);
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 823e39103edd..557f80c0bfe5 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -32,6 +32,7 @@
#define HDA_MONO 1
#define HDA_STEREO 2
#define HDA_QUAD 4
+#define HDA_MAX 8
static const struct snd_pcm_hardware azx_pcm_hw = {
.info = (SNDRV_PCM_INFO_MMAP |
@@ -494,6 +495,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
stream->lpib);
snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
}
+ /* fall through */
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
@@ -569,7 +571,10 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream,
stream_tag = hdac_stream(link_dev)->stream_tag;
/* set the stream tag in the codec dai dma params */
- snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0);
+ else
+ snd_soc_dai_set_tdm_slot(codec_dai, 0, stream_tag, 0, 0);
p_params.s_fmt = snd_pcm_format_width(params_format(params));
p_params.ch = params_channels(params);
@@ -995,21 +1000,63 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
},
},
{
- .name = "HD-Codec Pin",
+ .name = "Analog CPU DAI",
.ops = &skl_link_dai_ops,
.playback = {
- .stream_name = "HD-Codec Tx",
- .channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .stream_name = "Analog CPU Playback",
+ .channels_min = HDA_MONO,
+ .channels_max = HDA_MAX,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
},
.capture = {
- .stream_name = "HD-Codec Rx",
- .channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .stream_name = "Analog CPU Capture",
+ .channels_min = HDA_MONO,
+ .channels_max = HDA_MAX,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ },
+},
+{
+ .name = "Alt Analog CPU DAI",
+ .ops = &skl_link_dai_ops,
+ .playback = {
+ .stream_name = "Alt Analog CPU Playback",
+ .channels_min = HDA_MONO,
+ .channels_max = HDA_MAX,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .capture = {
+ .stream_name = "Alt Analog CPU Capture",
+ .channels_min = HDA_MONO,
+ .channels_max = HDA_MAX,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ },
+},
+{
+ .name = "Digital CPU DAI",
+ .ops = &skl_link_dai_ops,
+ .playback = {
+ .stream_name = "Digital CPU Playback",
+ .channels_min = HDA_MONO,
+ .channels_max = HDA_MAX,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .capture = {
+ .stream_name = "Digital CPU Capture",
+ .channels_min = HDA_MONO,
+ .channels_max = HDA_MAX,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
},
},
};
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 2620d77729c5..cf8848b779dc 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -898,11 +898,10 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w,
bc = (struct skl_algo_data *)sb->dobj.private;
if (bc->set_params == SKL_PARAM_BIND) {
- params = kzalloc(bc->max, GFP_KERNEL);
+ params = kmemdup(bc->params, bc->max, GFP_KERNEL);
if (!params)
return -ENOMEM;
- memcpy(params, bc->params, bc->max);
skl_fill_sink_instance_id(ctx, params, bc->max,
mconfig);
@@ -2461,6 +2460,7 @@ static int skl_tplg_get_token(struct device *dev,
case SKL_TKN_U8_CORE_ID:
mconfig->core_id = tkn_elem->value;
+ break;
case SKL_TKN_U8_MOD_TYPE:
mconfig->m_type = tkn_elem->value;
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 1d17be0f78a0..29225623b4b4 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -33,9 +33,11 @@
#include <sound/hda_register.h>
#include <sound/hdaudio.h>
#include <sound/hda_i915.h>
+#include <sound/hda_codec.h>
#include "skl.h"
#include "skl-sst-dsp.h"
#include "skl-sst-ipc.h"
+#include "../../../soc/codecs/hdac_hda.h"
/*
* initialize the PCI registers
@@ -472,6 +474,25 @@ static struct skl_ssp_clk skl_ssp_clks[] = {
{.name = "ssp5_sclkfs"},
};
+static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl *skl,
+ struct snd_soc_acpi_mach *machines)
+{
+ struct hdac_bus *bus = skl_to_bus(skl);
+ struct snd_soc_acpi_mach *mach;
+
+ /* check if we have any codecs detected on bus */
+ if (bus->codec_mask == 0)
+ return NULL;
+
+ /* point to common table */
+ mach = snd_soc_acpi_intel_hda_machines;
+
+ /* all entries in the machine table use the same firmware */
+ mach->fw_filename = machines->fw_filename;
+
+ return mach;
+}
+
static int skl_find_machine(struct skl *skl, void *driver_data)
{
struct hdac_bus *bus = skl_to_bus(skl);
@@ -479,9 +500,13 @@ static int skl_find_machine(struct skl *skl, void *driver_data)
struct skl_machine_pdata *pdata;
mach = snd_soc_acpi_find_machine(mach);
- if (mach == NULL) {
- dev_err(bus->dev, "No matching machine driver found\n");
- return -ENODEV;
+ if (!mach) {
+ dev_dbg(bus->dev, "No matching I2S machine driver found\n");
+ mach = skl_find_hda_machine(skl, driver_data);
+ if (!mach) {
+ dev_err(bus->dev, "No matching machine driver found\n");
+ return -ENODEV;
+ }
}
skl->mach = mach;
@@ -498,8 +523,9 @@ static int skl_find_machine(struct skl *skl, void *driver_data)
static int skl_machine_device_register(struct skl *skl)
{
- struct hdac_bus *bus = skl_to_bus(skl);
struct snd_soc_acpi_mach *mach = skl->mach;
+ struct hdac_bus *bus = skl_to_bus(skl);
+ struct skl_machine_pdata *pdata;
struct platform_device *pdev;
int ret;
@@ -516,8 +542,12 @@ static int skl_machine_device_register(struct skl *skl)
return -EIO;
}
- if (mach->pdata)
+ if (mach->pdata) {
+ pdata = (struct skl_machine_pdata *)mach->pdata;
+ pdata->platform = dev_name(bus->dev);
+ pdata->codec_mask = bus->codec_mask;
dev_set_drvdata(&pdev->dev, mach->pdata);
+ }
skl->i2s_dev = pdev;
@@ -628,6 +658,24 @@ static void skl_clock_device_unregister(struct skl *skl)
platform_device_unregister(skl->clk_dev);
}
+#define IDISP_INTEL_VENDOR_ID 0x80860000
+
+/*
+ * load the legacy codec driver
+ */
+static void load_codec_module(struct hda_codec *codec)
+{
+#ifdef MODULE
+ char modalias[MODULE_NAME_LEN];
+ const char *mod = NULL;
+
+ snd_hdac_codec_modalias(&codec->core, modalias, sizeof(modalias));
+ mod = modalias;
+ dev_dbg(&codec->core.dev, "loading %s codec module\n", mod);
+ request_module(mod);
+#endif
+}
+
/*
* Probe the given codec address
*/
@@ -637,7 +685,9 @@ static int probe_codec(struct hdac_bus *bus, int addr)
(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
unsigned int res = -1;
struct skl *skl = bus_to_skl(bus);
+ struct hdac_hda_priv *hda_codec;
struct hdac_device *hdev;
+ int err;
mutex_lock(&bus->cmd_mutex);
snd_hdac_bus_send_cmd(bus, cmd);
@@ -645,13 +695,26 @@ static int probe_codec(struct hdac_bus *bus, int addr)
mutex_unlock(&bus->cmd_mutex);
if (res == -1)
return -EIO;
- dev_dbg(bus->dev, "codec #%d probed OK\n", addr);
+ dev_dbg(bus->dev, "codec #%d probed OK: %x\n", addr, res);
- hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL);
- if (!hdev)
+ hda_codec = devm_kzalloc(&skl->pci->dev, sizeof(*hda_codec),
+ GFP_KERNEL);
+ if (!hda_codec)
return -ENOMEM;
- return snd_hdac_ext_bus_device_init(bus, addr, hdev);
+ hda_codec->codec.bus = skl_to_hbus(skl);
+ hdev = &hda_codec->codec.core;
+
+ err = snd_hdac_ext_bus_device_init(bus, addr, hdev);
+ if (err < 0)
+ return err;
+
+ /* use legacy bus only for HDA codecs, idisp uses ext bus */
+ if ((res & 0xFFFF0000) != IDISP_INTEL_VENDOR_ID) {
+ hdev->type = HDA_DEV_LEGACY;
+ load_codec_module(&hda_codec->codec);
+ }
+ return 0;
}
/* Codec initialization */
@@ -786,9 +849,10 @@ static int skl_create(struct pci_dev *pci,
const struct hdac_io_ops *io_ops,
struct skl **rskl)
{
+ struct hdac_ext_bus_ops *ext_ops = NULL;
struct skl *skl;
struct hdac_bus *bus;
-
+ struct hda_bus *hbus;
int err;
*rskl = NULL;
@@ -803,13 +867,23 @@ static int skl_create(struct pci_dev *pci,
return -ENOMEM;
}
+ hbus = skl_to_hbus(skl);
bus = skl_to_bus(skl);
- snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, NULL);
+
+#if IS_ENABLED(CONFIG_SND_SOC_HDAC_HDA)
+ ext_ops = snd_soc_hdac_hda_get_ops();
+#endif
+ snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops);
bus->use_posbuf = 1;
skl->pci = pci;
INIT_WORK(&skl->probe_work, skl_probe_work);
bus->bdl_pos_adj = 0;
+ mutex_init(&hbus->prepare_mutex);
+ hbus->pci = pci;
+ hbus->mixer_assigned = -1;
+ hbus->modelname = "sklbus";
+
*rskl = skl;
return 0;
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index 78aa8bdcb619..8d48cd7c56c8 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -23,6 +23,7 @@
#include <sound/hda_register.h>
#include <sound/hdaudio_ext.h>
+#include <sound/hda_codec.h>
#include <sound/soc.h>
#include "skl-nhlt.h"
#include "skl-ssp-clk.h"
@@ -71,7 +72,7 @@ struct skl_fw_config {
};
struct skl {
- struct hdac_bus hbus;
+ struct hda_bus hbus;
struct pci_dev *pci;
unsigned int init_done:1; /* delayed init status */
@@ -105,8 +106,11 @@ struct skl {
struct snd_soc_acpi_mach *mach;
};
-#define skl_to_bus(s) (&(s)->hbus)
-#define bus_to_skl(bus) container_of(bus, struct skl, hbus)
+#define skl_to_bus(s) (&(s)->hbus.core)
+#define bus_to_skl(bus) container_of(bus, struct skl, hbus.core)
+
+#define skl_to_hbus(s) (&(s)->hbus)
+#define hbus_to_skl(hbus) container_of((hbus), struct skl, (hbus))
/* to pass dai dma data */
struct skl_dma_params {
@@ -117,6 +121,8 @@ struct skl_dma_params {
struct skl_machine_pdata {
u32 dmic_num;
bool use_tplg_pcm; /* use dais and dai links from topology */
+ const char *platform;
+ u32 codec_mask;
};
struct skl_dsp_ops {
diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c
index 666282b865a8..97f9f38ce6b3 100644
--- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c
+++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c
@@ -299,6 +299,7 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev)
devm_kzalloc(&pdev->dev, sizeof(struct mt2701_cs42448_private),
GFP_KERNEL);
struct device *dev = &pdev->dev;
+ struct snd_soc_dai_link *dai_link;
if (!priv)
return -ENOMEM;
@@ -309,10 +310,10 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
return -EINVAL;
}
- for (i = 0; i < card->num_links; i++) {
- if (mt2701_cs42448_dai_links[i].platform_name)
+ for_each_card_prelinks(card, i, dai_link) {
+ if (dai_link->platform_name)
continue;
- mt2701_cs42448_dai_links[i].platform_of_node = platform_node;
+ dai_link->platform_of_node = platform_node;
}
card->dev = dev;
@@ -324,10 +325,10 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev)
"Property 'audio-codec' missing or invalid\n");
return -EINVAL;
}
- for (i = 0; i < card->num_links; i++) {
- if (mt2701_cs42448_dai_links[i].codec_name)
+ for_each_card_prelinks(card, i, dai_link) {
+ if (dai_link->codec_name)
continue;
- mt2701_cs42448_dai_links[i].codec_of_node = codec_node;
+ dai_link->codec_of_node = codec_node;
}
codec_node_bt_mrg = of_parse_phandle(pdev->dev.of_node,
diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c
index 89f34efd9747..6bc1d3d58e64 100644
--- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c
+++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c
@@ -97,6 +97,7 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &mt2701_wm8960_card;
struct device_node *platform_node, *codec_node;
+ struct snd_soc_dai_link *dai_link;
int ret, i;
platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -105,10 +106,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
return -EINVAL;
}
- for (i = 0; i < card->num_links; i++) {
- if (mt2701_wm8960_dai_links[i].platform_name)
+ for_each_card_prelinks(card, i, dai_link) {
+ if (dai_link->platform_name)
continue;
- mt2701_wm8960_dai_links[i].platform_of_node = platform_node;
+ dai_link->platform_of_node = platform_node;
}
card->dev = &pdev->dev;
@@ -120,10 +121,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
"Property 'audio-codec' missing or invalid\n");
return -EINVAL;
}
- for (i = 0; i < card->num_links; i++) {
- if (mt2701_wm8960_dai_links[i].codec_name)
+ for_each_card_prelinks(card, i, dai_link) {
+ if (dai_link->codec_name)
continue;
- mt2701_wm8960_dai_links[i].codec_of_node = codec_node;
+ dai_link->codec_of_node = codec_node;
}
ret = snd_soc_of_parse_audio_routing(card, "audio-routing");
@@ -150,7 +151,6 @@ static const struct of_device_id mt2701_wm8960_machine_dt_match[] = {
static struct platform_driver mt2701_wm8960_machine = {
.driver = {
.name = "mt2701-wm8960",
- .owner = THIS_MODULE,
#ifdef CONFIG_OF
.of_match_table = mt2701_wm8960_machine_dt_match,
#endif
diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c
index b1558c57b9ca..cc41eb531653 100644
--- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c
+++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c
@@ -158,6 +158,7 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &mt6797_mt6351_card;
struct device_node *platform_node, *codec_node;
+ struct snd_soc_dai_link *dai_link;
int ret, i;
card->dev = &pdev->dev;
@@ -168,10 +169,10 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
return -EINVAL;
}
- for (i = 0; i < card->num_links; i++) {
- if (mt6797_mt6351_dai_links[i].platform_name)
+ for_each_card_prelinks(card, i, dai_link) {
+ if (dai_link->platform_name)
continue;
- mt6797_mt6351_dai_links[i].platform_of_node = platform_node;
+ dai_link->platform_of_node = platform_node;
}
codec_node = of_parse_phandle(pdev->dev.of_node,
@@ -181,10 +182,10 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev)
"Property 'audio-codec' missing or invalid\n");
return -EINVAL;
}
- for (i = 0; i < card->num_links; i++) {
- if (mt6797_mt6351_dai_links[i].codec_name)
+ for_each_card_prelinks(card, i, dai_link) {
+ if (dai_link->codec_name)
continue;
- mt6797_mt6351_dai_links[i].codec_of_node = codec_node;
+ dai_link->codec_of_node = codec_node;
}
ret = devm_snd_soc_register_card(&pdev->dev, card);
@@ -205,7 +206,6 @@ static const struct of_device_id mt6797_mt6351_dt_match[] = {
static struct platform_driver mt6797_mt6351_driver = {
.driver = {
.name = "mt6797-mt6351",
- .owner = THIS_MODULE,
#ifdef CONFIG_OF
.of_match_table = mt6797_mt6351_dt_match,
#endif
diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c
index 902d111016d6..4d6596d5cb07 100644
--- a/sound/soc/mediatek/mt8173/mt8173-max98090.c
+++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c
@@ -137,6 +137,7 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &mt8173_max98090_card;
struct device_node *codec_node, *platform_node;
+ struct snd_soc_dai_link *dai_link;
int ret, i;
platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -145,10 +146,10 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
return -EINVAL;
}
- for (i = 0; i < card->num_links; i++) {
- if (mt8173_max98090_dais[i].platform_name)
+ for_each_card_prelinks(card, i, dai_link) {
+ if (dai_link->platform_name)
continue;
- mt8173_max98090_dais[i].platform_of_node = platform_node;
+ dai_link->platform_of_node = platform_node;
}
codec_node = of_parse_phandle(pdev->dev.of_node,
@@ -158,10 +159,10 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
"Property 'audio-codec' missing or invalid\n");
return -EINVAL;
}
- for (i = 0; i < card->num_links; i++) {
- if (mt8173_max98090_dais[i].codec_name)
+ for_each_card_prelinks(card, i, dai_link) {
+ if (dai_link->codec_name)
continue;
- mt8173_max98090_dais[i].codec_of_node = codec_node;
+ dai_link->codec_of_node = codec_node;
}
card->dev = &pdev->dev;
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
index 582174d98c6c..da5b58ce791b 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
@@ -44,11 +44,10 @@ static int mt8173_rt5650_rt5514_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;
int i, ret;
- for (i = 0; i < rtd->num_codecs; i++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
-
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
/* pll from mclk 12.288M */
ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
params_rate(params) * 512);
@@ -179,6 +178,7 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &mt8173_rt5650_rt5514_card;
struct device_node *platform_node;
+ struct snd_soc_dai_link *dai_link;
int i, ret;
platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -188,10 +188,10 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev)
return -EINVAL;
}
- for (i = 0; i < card->num_links; i++) {
- if (mt8173_rt5650_rt5514_dais[i].platform_name)
+ for_each_card_prelinks(card, i, dai_link) {
+ if (dai_link->platform_name)
continue;
- mt8173_rt5650_rt5514_dais[i].platform_of_node = platform_node;
+ dai_link->platform_of_node = platform_node;
}
mt8173_rt5650_rt5514_codecs[0].of_node =
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
index b3670c8a5b8d..d83cd039b413 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
@@ -48,11 +48,10 @@ static int mt8173_rt5650_rt5676_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;
int i, ret;
- for (i = 0; i < rtd->num_codecs; i++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
-
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
/* pll from mclk 12.288M */
ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
params_rate(params) * 512);
@@ -225,6 +224,7 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &mt8173_rt5650_rt5676_card;
struct device_node *platform_node;
+ struct snd_soc_dai_link *dai_link;
int i, ret;
platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -234,10 +234,10 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
return -EINVAL;
}
- for (i = 0; i < card->num_links; i++) {
- if (mt8173_rt5650_rt5676_dais[i].platform_name)
+ for_each_card_prelinks(card, i, dai_link) {
+ if (dai_link->platform_name)
continue;
- mt8173_rt5650_rt5676_dais[i].platform_of_node = platform_node;
+ dai_link->platform_of_node = platform_node;
}
mt8173_rt5650_rt5676_codecs[0].of_node =
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
index 7a89b4aad182..7edf250c8fb1 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
@@ -59,6 +59,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
unsigned int mclk_clock;
+ struct snd_soc_dai *codec_dai;
int i, ret;
switch (mt8173_rt5650_priv.pll_from) {
@@ -76,9 +77,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream,
break;
}
- for (i = 0; i < rtd->num_codecs; i++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
-
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
/* pll from mclk */
ret = snd_soc_dai_set_pll(codec_dai, 0, 0, mclk_clock,
params_rate(params) * 512);
@@ -240,6 +239,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
struct device_node *platform_node;
struct device_node *np;
const char *codec_capture_dai;
+ struct snd_soc_dai_link *dai_link;
int i, ret;
platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -249,10 +249,10 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
return -EINVAL;
}
- for (i = 0; i < card->num_links; i++) {
- if (mt8173_rt5650_dais[i].platform_name)
+ for_each_card_prelinks(card, i, dai_link) {
+ if (dai_link->platform_name)
continue;
- mt8173_rt5650_dais[i].platform_of_node = platform_node;
+ dai_link->platform_of_node = platform_node;
}
mt8173_rt5650_codecs[0].of_node =
diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig
index 8af8bc358a90..8b8426ed2363 100644
--- a/sound/soc/meson/Kconfig
+++ b/sound/soc/meson/Kconfig
@@ -4,6 +4,8 @@ menu "ASoC support for Amlogic platforms"
config SND_MESON_AXG_FIFO
tristate
select REGMAP_MMIO
+ imply COMMON_CLK_AXG_AUDIO
+ imply RESET_MESON_AUDIO_ARB
config SND_MESON_AXG_FRDDR
tristate "Amlogic AXG Playback FIFO support"
@@ -22,6 +24,7 @@ config SND_MESON_AXG_TODDR
config SND_MESON_AXG_TDM_FORMATTER
tristate
select REGMAP_MMIO
+ imply COMMON_CLK_AXG_AUDIO
config SND_MESON_AXG_TDM_INTERFACE
tristate
@@ -51,6 +54,7 @@ config SND_MESON_AXG_SOUND_CARD
imply SND_MESON_AXG_TDMIN
imply SND_MESON_AXG_TDMOUT
imply SND_MESON_AXG_SPDIFOUT
+ imply SND_MESON_AXG_PDM
help
Select Y or M to add support for the AXG SoC sound card
@@ -58,8 +62,17 @@ config SND_MESON_AXG_SPDIFOUT
tristate "Amlogic AXG SPDIF Output Support"
select SND_PCM_IEC958
imply SND_SOC_SPDIF
+ imply COMMON_CLK_AXG_AUDIO
help
Select Y or M to add support for SPDIF output serializer embedded
in the Amlogic AXG SoC family
+config SND_MESON_AXG_PDM
+ tristate "Amlogic AXG PDM Input Support"
+ imply SND_SOC_DMIC
+ imply COMMON_CLK_AXG_AUDIO
+ help
+ Select Y or M to add support for PDM input embedded
+ in the Amlogic AXG SoC family
+
endmenu
diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile
index c5e003b093db..4cd25104029d 100644
--- a/sound/soc/meson/Makefile
+++ b/sound/soc/meson/Makefile
@@ -9,6 +9,7 @@ snd-soc-meson-axg-tdmin-objs := axg-tdmin.o
snd-soc-meson-axg-tdmout-objs := axg-tdmout.o
snd-soc-meson-axg-sound-card-objs := axg-card.o
snd-soc-meson-axg-spdifout-objs := axg-spdifout.o
+snd-soc-meson-axg-pdm-objs := axg-pdm.o
obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o
obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o
@@ -19,3 +20,4 @@ obj-$(CONFIG_SND_MESON_AXG_TDMIN) += snd-soc-meson-axg-tdmin.o
obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o
obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o
obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o
+obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 2914ba0d965b..aa54d2c612c9 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -97,14 +97,14 @@ static void axg_card_clean_references(struct axg_card *priv)
{
struct snd_soc_card *card = &priv->card;
struct snd_soc_dai_link *link;
+ struct snd_soc_dai_link_component *codec;
int i, j;
if (card->dai_link) {
- for (i = 0; i < card->num_links; i++) {
- link = &card->dai_link[i];
+ for_each_card_prelinks(card, i, link) {
of_node_put(link->cpu_of_node);
- for (j = 0; j < link->num_codecs; j++)
- of_node_put(link->codecs[j].of_node);
+ for_each_link_codecs(link, j, codec)
+ of_node_put(codec->of_node);
}
}
@@ -167,8 +167,7 @@ static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream,
if (be->mclk_fs) {
mclk = params_rate(params) * be->mclk_fs;
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
SND_SOC_CLOCK_IN);
if (ret && ret != -ENOTSUPP)
@@ -196,8 +195,7 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_dai *codec_dai;
int ret, i;
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
ret = snd_soc_dai_set_tdm_slot(codec_dai,
be->codec_masks[i].tx,
be->codec_masks[i].rx,
@@ -478,7 +476,7 @@ static int axg_card_set_be_link(struct snd_soc_card *card,
ret = axg_card_set_link_name(card, link, "be");
if (ret)
- dev_err(card->dev, "error setting %s link name\n", np->name);
+ dev_err(card->dev, "error setting %pOFn link name\n", np);
return ret;
}
diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
index 30262550e37b..0e4f65e654c4 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -203,6 +203,8 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss)
ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0,
dev_name(dev), ss);
+ if (ret)
+ return ret;
/* Enable pclk to access registers and clock the fifo ip */
ret = clk_prepare_enable(fifo->pclk);
diff --git a/sound/soc/meson/axg-pdm.c b/sound/soc/meson/axg-pdm.c
new file mode 100644
index 000000000000..9d5684493ffc
--- /dev/null
+++ b/sound/soc/meson/axg-pdm.c
@@ -0,0 +1,654 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_irq.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/pcm_params.h>
+
+#define PDM_CTRL 0x00
+#define PDM_CTRL_EN BIT(31)
+#define PDM_CTRL_OUT_MODE BIT(29)
+#define PDM_CTRL_BYPASS_MODE BIT(28)
+#define PDM_CTRL_RST_FIFO BIT(16)
+#define PDM_CTRL_CHAN_RSTN_MASK GENMASK(15, 8)
+#define PDM_CTRL_CHAN_RSTN(x) ((x) << 8)
+#define PDM_CTRL_CHAN_EN_MASK GENMASK(7, 0)
+#define PDM_CTRL_CHAN_EN(x) ((x) << 0)
+#define PDM_HCIC_CTRL1 0x04
+#define PDM_FILTER_EN BIT(31)
+#define PDM_HCIC_CTRL1_GAIN_SFT_MASK GENMASK(29, 24)
+#define PDM_HCIC_CTRL1_GAIN_SFT(x) ((x) << 24)
+#define PDM_HCIC_CTRL1_GAIN_MULT_MASK GENMASK(23, 16)
+#define PDM_HCIC_CTRL1_GAIN_MULT(x) ((x) << 16)
+#define PDM_HCIC_CTRL1_DSR_MASK GENMASK(8, 4)
+#define PDM_HCIC_CTRL1_DSR(x) ((x) << 4)
+#define PDM_HCIC_CTRL1_STAGE_NUM_MASK GENMASK(3, 0)
+#define PDM_HCIC_CTRL1_STAGE_NUM(x) ((x) << 0)
+#define PDM_HCIC_CTRL2 0x08
+#define PDM_F1_CTRL 0x0c
+#define PDM_LPF_ROUND_MODE_MASK GENMASK(17, 16)
+#define PDM_LPF_ROUND_MODE(x) ((x) << 16)
+#define PDM_LPF_DSR_MASK GENMASK(15, 12)
+#define PDM_LPF_DSR(x) ((x) << 12)
+#define PDM_LPF_STAGE_NUM_MASK GENMASK(8, 0)
+#define PDM_LPF_STAGE_NUM(x) ((x) << 0)
+#define PDM_LPF_MAX_STAGE 336
+#define PDM_LPF_NUM 3
+#define PDM_F2_CTRL 0x10
+#define PDM_F3_CTRL 0x14
+#define PDM_HPF_CTRL 0x18
+#define PDM_HPF_SFT_STEPS_MASK GENMASK(20, 16)
+#define PDM_HPF_SFT_STEPS(x) ((x) << 16)
+#define PDM_HPF_OUT_FACTOR_MASK GENMASK(15, 0)
+#define PDM_HPF_OUT_FACTOR(x) ((x) << 0)
+#define PDM_CHAN_CTRL 0x1c
+#define PDM_CHAN_CTRL_POINTER_WIDTH 8
+#define PDM_CHAN_CTRL_POINTER_MAX ((1 << PDM_CHAN_CTRL_POINTER_WIDTH) - 1)
+#define PDM_CHAN_CTRL_NUM 4
+#define PDM_CHAN_CTRL1 0x20
+#define PDM_COEFF_ADDR 0x24
+#define PDM_COEFF_DATA 0x28
+#define PDM_CLKG_CTRL 0x2c
+#define PDM_STS 0x30
+
+struct axg_pdm_lpf {
+ unsigned int ds;
+ unsigned int round_mode;
+ const unsigned int *tap;
+ unsigned int tap_num;
+};
+
+struct axg_pdm_hcic {
+ unsigned int shift;
+ unsigned int mult;
+ unsigned int steps;
+ unsigned int ds;
+};
+
+struct axg_pdm_hpf {
+ unsigned int out_factor;
+ unsigned int steps;
+};
+
+struct axg_pdm_filters {
+ struct axg_pdm_hcic hcic;
+ struct axg_pdm_hpf hpf;
+ struct axg_pdm_lpf lpf[PDM_LPF_NUM];
+};
+
+struct axg_pdm_cfg {
+ const struct axg_pdm_filters *filters;
+ unsigned int sys_rate;
+};
+
+struct axg_pdm {
+ const struct axg_pdm_cfg *cfg;
+ struct regmap *map;
+ struct clk *dclk;
+ struct clk *sysclk;
+ struct clk *pclk;
+};
+
+static void axg_pdm_enable(struct regmap *map)
+{
+ /* Reset AFIFO */
+ regmap_update_bits(map, PDM_CTRL, PDM_CTRL_RST_FIFO, PDM_CTRL_RST_FIFO);
+ regmap_update_bits(map, PDM_CTRL, PDM_CTRL_RST_FIFO, 0);
+
+ /* Enable PDM */
+ regmap_update_bits(map, PDM_CTRL, PDM_CTRL_EN, PDM_CTRL_EN);
+}
+
+static void axg_pdm_disable(struct regmap *map)
+{
+ regmap_update_bits(map, PDM_CTRL, PDM_CTRL_EN, 0);
+}
+
+static void axg_pdm_filters_enable(struct regmap *map, bool enable)
+{
+ unsigned int val = enable ? PDM_FILTER_EN : 0;
+
+ regmap_update_bits(map, PDM_HCIC_CTRL1, PDM_FILTER_EN, val);
+ regmap_update_bits(map, PDM_F1_CTRL, PDM_FILTER_EN, val);
+ regmap_update_bits(map, PDM_F2_CTRL, PDM_FILTER_EN, val);
+ regmap_update_bits(map, PDM_F3_CTRL, PDM_FILTER_EN, val);
+ regmap_update_bits(map, PDM_HPF_CTRL, PDM_FILTER_EN, val);
+}
+
+static int axg_pdm_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ axg_pdm_enable(priv->map);
+ return 0;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ axg_pdm_disable(priv->map);
+ return 0;
+
+ default:
+ return -EINVAL;
+ }
+}
+
+static unsigned int axg_pdm_get_os(struct axg_pdm *priv)
+{
+ const struct axg_pdm_filters *filters = priv->cfg->filters;
+ unsigned int os = filters->hcic.ds;
+ int i;
+
+ /*
+ * The global oversampling factor is defined by the down sampling
+ * factor applied by each filter (HCIC and LPFs)
+ */
+
+ for (i = 0; i < PDM_LPF_NUM; i++)
+ os *= filters->lpf[i].ds;
+
+ return os;
+}
+
+static int axg_pdm_set_sysclk(struct axg_pdm *priv, unsigned int os,
+ unsigned int rate)
+{
+ unsigned int sys_rate = os * 2 * rate * PDM_CHAN_CTRL_POINTER_MAX;
+
+ /*
+ * Set the default system clock rate unless it is too fast for
+ * for the requested sample rate. In this case, the sample pointer
+ * counter could overflow so set a lower system clock rate
+ */
+ if (sys_rate < priv->cfg->sys_rate)
+ return clk_set_rate(priv->sysclk, sys_rate);
+
+ return clk_set_rate(priv->sysclk, priv->cfg->sys_rate);
+}
+
+static int axg_pdm_set_sample_pointer(struct axg_pdm *priv)
+{
+ unsigned int spmax, sp, val;
+ int i;
+
+ /* Max sample counter value per half period of dclk */
+ spmax = DIV_ROUND_UP_ULL((u64)clk_get_rate(priv->sysclk),
+ clk_get_rate(priv->dclk) * 2);
+
+ /* Check if sysclk is not too fast - should not happen */
+ if (WARN_ON(spmax > PDM_CHAN_CTRL_POINTER_MAX))
+ return -EINVAL;
+
+ /* Capture the data when we are at 75% of the half period */
+ sp = spmax * 3 / 4;
+
+ for (i = 0, val = 0; i < PDM_CHAN_CTRL_NUM; i++)
+ val |= sp << (PDM_CHAN_CTRL_POINTER_WIDTH * i);
+
+ regmap_write(priv->map, PDM_CHAN_CTRL, val);
+ regmap_write(priv->map, PDM_CHAN_CTRL1, val);
+
+ return 0;
+}
+
+static void axg_pdm_set_channel_mask(struct axg_pdm *priv,
+ unsigned int channels)
+{
+ unsigned int mask = GENMASK(channels - 1, 0);
+
+ /* Put all channel in reset */
+ regmap_update_bits(priv->map, PDM_CTRL,
+ PDM_CTRL_CHAN_RSTN_MASK, 0);
+
+ /* Take the necessary channels out of reset and enable them */
+ regmap_update_bits(priv->map, PDM_CTRL,
+ PDM_CTRL_CHAN_RSTN_MASK |
+ PDM_CTRL_CHAN_EN_MASK,
+ PDM_CTRL_CHAN_RSTN(mask) |
+ PDM_CTRL_CHAN_EN(mask));
+}
+
+static int axg_pdm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+ unsigned int os = axg_pdm_get_os(priv);
+ unsigned int rate = params_rate(params);
+ unsigned int val;
+ int ret;
+
+ switch (params_width(params)) {
+ case 24:
+ val = PDM_CTRL_OUT_MODE;
+ break;
+ case 32:
+ val = 0;
+ break;
+ default:
+ dev_err(dai->dev, "unsupported sample width\n");
+ return -EINVAL;
+ }
+
+ regmap_update_bits(priv->map, PDM_CTRL, PDM_CTRL_OUT_MODE, val);
+
+ ret = axg_pdm_set_sysclk(priv, os, rate);
+ if (ret) {
+ dev_err(dai->dev, "failed to set system clock\n");
+ return ret;
+ }
+
+ ret = clk_set_rate(priv->dclk, rate * os);
+ if (ret) {
+ dev_err(dai->dev, "failed to set dclk\n");
+ return ret;
+ }
+
+ ret = axg_pdm_set_sample_pointer(priv);
+ if (ret) {
+ dev_err(dai->dev, "invalid clock setting\n");
+ return ret;
+ }
+
+ axg_pdm_set_channel_mask(priv, params_channels(params));
+
+ return 0;
+}
+
+static int axg_pdm_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+ int ret;
+
+ ret = clk_prepare_enable(priv->dclk);
+ if (ret) {
+ dev_err(dai->dev, "enabling dclk failed\n");
+ return ret;
+ }
+
+ /* Enable the filters */
+ axg_pdm_filters_enable(priv->map, true);
+
+ return ret;
+}
+
+static void axg_pdm_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+
+ axg_pdm_filters_enable(priv->map, false);
+ clk_disable_unprepare(priv->dclk);
+}
+
+static const struct snd_soc_dai_ops axg_pdm_dai_ops = {
+ .trigger = axg_pdm_trigger,
+ .hw_params = axg_pdm_hw_params,
+ .startup = axg_pdm_startup,
+ .shutdown = axg_pdm_shutdown,
+};
+
+static void axg_pdm_set_hcic_ctrl(struct axg_pdm *priv)
+{
+ const struct axg_pdm_hcic *hcic = &priv->cfg->filters->hcic;
+ unsigned int val;
+
+ val = PDM_HCIC_CTRL1_STAGE_NUM(hcic->steps);
+ val |= PDM_HCIC_CTRL1_DSR(hcic->ds);
+ val |= PDM_HCIC_CTRL1_GAIN_MULT(hcic->mult);
+ val |= PDM_HCIC_CTRL1_GAIN_SFT(hcic->shift);
+
+ regmap_update_bits(priv->map, PDM_HCIC_CTRL1,
+ PDM_HCIC_CTRL1_STAGE_NUM_MASK |
+ PDM_HCIC_CTRL1_DSR_MASK |
+ PDM_HCIC_CTRL1_GAIN_MULT_MASK |
+ PDM_HCIC_CTRL1_GAIN_SFT_MASK,
+ val);
+}
+
+static void axg_pdm_set_lpf_ctrl(struct axg_pdm *priv, unsigned int index)
+{
+ const struct axg_pdm_lpf *lpf = &priv->cfg->filters->lpf[index];
+ unsigned int offset = index * regmap_get_reg_stride(priv->map)
+ + PDM_F1_CTRL;
+ unsigned int val;
+
+ val = PDM_LPF_STAGE_NUM(lpf->tap_num);
+ val |= PDM_LPF_DSR(lpf->ds);
+ val |= PDM_LPF_ROUND_MODE(lpf->round_mode);
+
+ regmap_update_bits(priv->map, offset,
+ PDM_LPF_STAGE_NUM_MASK |
+ PDM_LPF_DSR_MASK |
+ PDM_LPF_ROUND_MODE_MASK,
+ val);
+}
+
+static void axg_pdm_set_hpf_ctrl(struct axg_pdm *priv)
+{
+ const struct axg_pdm_hpf *hpf = &priv->cfg->filters->hpf;
+ unsigned int val;
+
+ val = PDM_HPF_OUT_FACTOR(hpf->out_factor);
+ val |= PDM_HPF_SFT_STEPS(hpf->steps);
+
+ regmap_update_bits(priv->map, PDM_HPF_CTRL,
+ PDM_HPF_OUT_FACTOR_MASK |
+ PDM_HPF_SFT_STEPS_MASK,
+ val);
+}
+
+static int axg_pdm_set_lpf_filters(struct axg_pdm *priv)
+{
+ const struct axg_pdm_lpf *lpf = priv->cfg->filters->lpf;
+ unsigned int count = 0;
+ int i, j;
+
+ for (i = 0; i < PDM_LPF_NUM; i++)
+ count += lpf[i].tap_num;
+
+ /* Make sure the coeffs fit in the memory */
+ if (count >= PDM_LPF_MAX_STAGE)
+ return -EINVAL;
+
+ /* Set the initial APB bus register address */
+ regmap_write(priv->map, PDM_COEFF_ADDR, 0);
+
+ /* Set the tap filter values of all 3 filters */
+ for (i = 0; i < PDM_LPF_NUM; i++) {
+ axg_pdm_set_lpf_ctrl(priv, i);
+
+ for (j = 0; j < lpf[i].tap_num; j++)
+ regmap_write(priv->map, PDM_COEFF_DATA, lpf[i].tap[j]);
+ }
+
+ return 0;
+}
+
+static int axg_pdm_dai_probe(struct snd_soc_dai *dai)
+{
+ struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+ int ret;
+
+ ret = clk_prepare_enable(priv->pclk);
+ if (ret) {
+ dev_err(dai->dev, "enabling pclk failed\n");
+ return ret;
+ }
+
+ /*
+ * sysclk must be set and enabled as well to access the pdm registers
+ * Accessing the register w/o it will give a bus error.
+ */
+ ret = clk_set_rate(priv->sysclk, priv->cfg->sys_rate);
+ if (ret) {
+ dev_err(dai->dev, "setting sysclk failed\n");
+ goto err_pclk;
+ }
+
+ ret = clk_prepare_enable(priv->sysclk);
+ if (ret) {
+ dev_err(dai->dev, "enabling sysclk failed\n");
+ goto err_pclk;
+ }
+
+ /* Make sure the device is initially disabled */
+ axg_pdm_disable(priv->map);
+
+ /* Make sure filter bypass is disabled */
+ regmap_update_bits(priv->map, PDM_CTRL, PDM_CTRL_BYPASS_MODE, 0);
+
+ /* Load filter settings */
+ axg_pdm_set_hcic_ctrl(priv);
+ axg_pdm_set_hpf_ctrl(priv);
+
+ ret = axg_pdm_set_lpf_filters(priv);
+ if (ret) {
+ dev_err(dai->dev, "invalid filter configuration\n");
+ goto err_sysclk;
+ }
+
+ return 0;
+
+err_sysclk:
+ clk_disable_unprepare(priv->sysclk);
+err_pclk:
+ clk_disable_unprepare(priv->pclk);
+ return ret;
+}
+
+static int axg_pdm_dai_remove(struct snd_soc_dai *dai)
+{
+ struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+
+ clk_disable_unprepare(priv->sysclk);
+ clk_disable_unprepare(priv->pclk);
+
+ return 0;
+}
+
+static struct snd_soc_dai_driver axg_pdm_dai_drv = {
+ .name = "PDM",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 5512,
+ .rate_max = 48000,
+ .formats = (SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE),
+ },
+ .ops = &axg_pdm_dai_ops,
+ .probe = axg_pdm_dai_probe,
+ .remove = axg_pdm_dai_remove,
+};
+
+static const struct snd_soc_component_driver axg_pdm_component_drv = {};
+
+static const struct regmap_config axg_pdm_regmap_cfg = {
+ .reg_bits = 32,
+ .val_bits = 32,
+ .reg_stride = 4,
+ .max_register = PDM_STS,
+};
+
+static const unsigned int lpf1_default_tap[] = {
+ 0x000014, 0xffffb2, 0xfffed9, 0xfffdce, 0xfffd45,
+ 0xfffe32, 0x000147, 0x000645, 0x000b86, 0x000e21,
+ 0x000ae3, 0x000000, 0xffeece, 0xffdca8, 0xffd212,
+ 0xffd7d1, 0xfff2a7, 0x001f4c, 0x0050c2, 0x0072aa,
+ 0x006ff1, 0x003c32, 0xffdc4e, 0xff6a18, 0xff0fef,
+ 0xfefbaf, 0xff4c40, 0x000000, 0x00ebc8, 0x01c077,
+ 0x02209e, 0x01c1a4, 0x008e60, 0xfebe52, 0xfcd690,
+ 0xfb8fa5, 0xfba498, 0xfd9812, 0x0181ce, 0x06f5f3,
+ 0x0d112f, 0x12a958, 0x169686, 0x18000e, 0x169686,
+ 0x12a958, 0x0d112f, 0x06f5f3, 0x0181ce, 0xfd9812,
+ 0xfba498, 0xfb8fa5, 0xfcd690, 0xfebe52, 0x008e60,
+ 0x01c1a4, 0x02209e, 0x01c077, 0x00ebc8, 0x000000,
+ 0xff4c40, 0xfefbaf, 0xff0fef, 0xff6a18, 0xffdc4e,
+ 0x003c32, 0x006ff1, 0x0072aa, 0x0050c2, 0x001f4c,
+ 0xfff2a7, 0xffd7d1, 0xffd212, 0xffdca8, 0xffeece,
+ 0x000000, 0x000ae3, 0x000e21, 0x000b86, 0x000645,
+ 0x000147, 0xfffe32, 0xfffd45, 0xfffdce, 0xfffed9,
+ 0xffffb2, 0x000014,
+};
+
+static const unsigned int lpf2_default_tap[] = {
+ 0x00050a, 0xfff004, 0x0002c1, 0x003c12, 0xffa818,
+ 0xffc87d, 0x010aef, 0xff5223, 0xfebd93, 0x028f41,
+ 0xff5c0e, 0xfc63f8, 0x055f81, 0x000000, 0xf478a0,
+ 0x11c5e3, 0x2ea74d, 0x11c5e3, 0xf478a0, 0x000000,
+ 0x055f81, 0xfc63f8, 0xff5c0e, 0x028f41, 0xfebd93,
+ 0xff5223, 0x010aef, 0xffc87d, 0xffa818, 0x003c12,
+ 0x0002c1, 0xfff004, 0x00050a,
+};
+
+static const unsigned int lpf3_default_tap[] = {
+ 0x000000, 0x000081, 0x000000, 0xfffedb, 0x000000,
+ 0x00022d, 0x000000, 0xfffc46, 0x000000, 0x0005f7,
+ 0x000000, 0xfff6eb, 0x000000, 0x000d4e, 0x000000,
+ 0xffed1e, 0x000000, 0x001a1c, 0x000000, 0xffdcb0,
+ 0x000000, 0x002ede, 0x000000, 0xffc2d1, 0x000000,
+ 0x004ebe, 0x000000, 0xff9beb, 0x000000, 0x007dd7,
+ 0x000000, 0xff633a, 0x000000, 0x00c1d2, 0x000000,
+ 0xff11d5, 0x000000, 0x012368, 0x000000, 0xfe9c45,
+ 0x000000, 0x01b252, 0x000000, 0xfdebf6, 0x000000,
+ 0x0290b8, 0x000000, 0xfcca0d, 0x000000, 0x041d7c,
+ 0x000000, 0xfa8152, 0x000000, 0x07e9c6, 0x000000,
+ 0xf28fb5, 0x000000, 0x28b216, 0x3fffde, 0x28b216,
+ 0x000000, 0xf28fb5, 0x000000, 0x07e9c6, 0x000000,
+ 0xfa8152, 0x000000, 0x041d7c, 0x000000, 0xfcca0d,
+ 0x000000, 0x0290b8, 0x000000, 0xfdebf6, 0x000000,
+ 0x01b252, 0x000000, 0xfe9c45, 0x000000, 0x012368,
+ 0x000000, 0xff11d5, 0x000000, 0x00c1d2, 0x000000,
+ 0xff633a, 0x000000, 0x007dd7, 0x000000, 0xff9beb,
+ 0x000000, 0x004ebe, 0x000000, 0xffc2d1, 0x000000,
+ 0x002ede, 0x000000, 0xffdcb0, 0x000000, 0x001a1c,
+ 0x000000, 0xffed1e, 0x000000, 0x000d4e, 0x000000,
+ 0xfff6eb, 0x000000, 0x0005f7, 0x000000, 0xfffc46,
+ 0x000000, 0x00022d, 0x000000, 0xfffedb, 0x000000,
+ 0x000081, 0x000000,
+};
+
+/*
+ * These values are sane defaults for the axg platform:
+ * - OS = 64
+ * - Latency = 38700 (?)
+ *
+ * TODO: There is a lot of different HCIC, LPFs and HPF configurations possible.
+ * the configuration may depend on the dmic used by the platform, the
+ * expected tradeoff between latency and quality, etc ... If/When other
+ * settings are required, we should add a fw interface to this driver to
+ * load new filter settings.
+ */
+static const struct axg_pdm_filters axg_default_filters = {
+ .hcic = {
+ .shift = 0x15,
+ .mult = 0x80,
+ .steps = 7,
+ .ds = 8,
+ },
+ .hpf = {
+ .out_factor = 0x8000,
+ .steps = 13,
+ },
+ .lpf = {
+ [0] = {
+ .ds = 2,
+ .round_mode = 1,
+ .tap = lpf1_default_tap,
+ .tap_num = ARRAY_SIZE(lpf1_default_tap),
+ },
+ [1] = {
+ .ds = 2,
+ .round_mode = 0,
+ .tap = lpf2_default_tap,
+ .tap_num = ARRAY_SIZE(lpf2_default_tap),
+ },
+ [2] = {
+ .ds = 2,
+ .round_mode = 1,
+ .tap = lpf3_default_tap,
+ .tap_num = ARRAY_SIZE(lpf3_default_tap)
+ },
+ },
+};
+
+static const struct axg_pdm_cfg axg_pdm_config = {
+ .filters = &axg_default_filters,
+ .sys_rate = 250000000,
+};
+
+static const struct of_device_id axg_pdm_of_match[] = {
+ {
+ .compatible = "amlogic,axg-pdm",
+ .data = &axg_pdm_config,
+ }, {}
+};
+MODULE_DEVICE_TABLE(of, axg_pdm_of_match);
+
+static int axg_pdm_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct axg_pdm *priv;
+ struct resource *res;
+ void __iomem *regs;
+ int ret;
+
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+ platform_set_drvdata(pdev, priv);
+
+ priv->cfg = of_device_get_match_data(dev);
+ if (!priv->cfg) {
+ dev_err(dev, "failed to match device\n");
+ return -ENODEV;
+ }
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ regs = devm_ioremap_resource(dev, res);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ priv->map = devm_regmap_init_mmio(dev, regs, &axg_pdm_regmap_cfg);
+ if (IS_ERR(priv->map)) {
+ dev_err(dev, "failed to init regmap: %ld\n",
+ PTR_ERR(priv->map));
+ return PTR_ERR(priv->map);
+ }
+
+ priv->pclk = devm_clk_get(dev, "pclk");
+ if (IS_ERR(priv->pclk)) {
+ ret = PTR_ERR(priv->pclk);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "failed to get pclk: %d\n", ret);
+ return ret;
+ }
+
+ priv->dclk = devm_clk_get(dev, "dclk");
+ if (IS_ERR(priv->dclk)) {
+ ret = PTR_ERR(priv->dclk);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "failed to get dclk: %d\n", ret);
+ return ret;
+ }
+
+ priv->sysclk = devm_clk_get(dev, "sysclk");
+ if (IS_ERR(priv->sysclk)) {
+ ret = PTR_ERR(priv->sysclk);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "failed to get dclk: %d\n", ret);
+ return ret;
+ }
+
+ return devm_snd_soc_register_component(dev, &axg_pdm_component_drv,
+ &axg_pdm_dai_drv, 1);
+}
+
+static struct platform_driver axg_pdm_pdrv = {
+ .probe = axg_pdm_probe,
+ .driver = {
+ .name = "axg-pdm",
+ .of_match_table = axg_pdm_of_match,
+ },
+};
+module_platform_driver(axg_pdm_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG PDM Input driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index 7b8baf46d968..585ce030b79b 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -42,6 +42,7 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
struct axg_tdm_stream *rx = (struct axg_tdm_stream *)
dai->capture_dma_data;
unsigned int tx_slots, rx_slots;
+ unsigned int fmt = 0;
tx_slots = axg_tdm_slots_total(tx_mask);
rx_slots = axg_tdm_slots_total(rx_mask);
@@ -52,38 +53,45 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
return -EINVAL;
}
- /*
- * Amend the dai driver channel number and let dpcm channel merge do
- * its job
- */
- if (tx) {
- tx->mask = tx_mask;
- dai->driver->playback.channels_max = tx_slots;
- }
-
- if (rx) {
- rx->mask = rx_mask;
- dai->driver->capture.channels_max = rx_slots;
- }
-
iface->slots = slots;
switch (slot_width) {
case 0:
- /* defaults width to 32 if not provided */
- iface->slot_width = 32;
- break;
- case 8:
- case 16:
- case 24:
+ slot_width = 32;
+ /* Fall-through */
case 32:
- iface->slot_width = slot_width;
+ fmt |= SNDRV_PCM_FMTBIT_S32_LE;
+ /* Fall-through */
+ case 24:
+ fmt |= SNDRV_PCM_FMTBIT_S24_LE;
+ fmt |= SNDRV_PCM_FMTBIT_S20_LE;
+ /* Fall-through */
+ case 16:
+ fmt |= SNDRV_PCM_FMTBIT_S16_LE;
+ /* Fall-through */
+ case 8:
+ fmt |= SNDRV_PCM_FMTBIT_S8;
break;
default:
dev_err(dai->dev, "unsupported slot width: %d\n", slot_width);
return -EINVAL;
}
+ iface->slot_width = slot_width;
+
+ /* Amend the dai driver and let dpcm merge do its job */
+ if (tx) {
+ tx->mask = tx_mask;
+ dai->driver->playback.channels_max = tx_slots;
+ dai->driver->playback.formats = fmt;
+ }
+
+ if (rx) {
+ rx->mask = rx_mask;
+ dai->driver->capture.channels_max = rx_slots;
+ dai->driver->capture.formats = fmt;
+ }
+
return 0;
}
EXPORT_SYMBOL_GPL(axg_tdm_set_tdm_slots);
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index 81b09d740ed9..6384bb6dacfd 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -356,7 +356,7 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev)
if (ret)
goto out;
- ret = snd_soc_register_component(&pdev->dev, &nuc900_ac97_component,
+ ret = devm_snd_soc_register_component(&pdev->dev, &nuc900_ac97_component,
&nuc900_ac97_dai, 1);
if (ret)
goto out;
@@ -373,8 +373,6 @@ out:
static int nuc900_ac97_drvremove(struct platform_device *pdev)
{
- snd_soc_unregister_component(&pdev->dev);
-
nuc900_ac97_data = NULL;
snd_soc_set_ac97_ops(NULL);
diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c
index 8a99a8837dc9..673a9eb153b2 100644
--- a/sound/soc/omap/omap-hdmi-audio.c
+++ b/sound/soc/omap/omap-hdmi-audio.c
@@ -348,7 +348,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
default:
return -EINVAL;
}
- ret = snd_soc_register_component(ad->dssdev, &omap_hdmi_component,
+ ret = devm_snd_soc_register_component(ad->dssdev, &omap_hdmi_component,
dai_drv, 1);
if (ret)
return ret;
@@ -383,7 +383,6 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
ret = snd_soc_register_card(card);
if (ret) {
dev_err(dev, "snd_soc_register_card failed (%d)\n", ret);
- snd_soc_unregister_component(ad->dssdev);
return ret;
}
@@ -400,7 +399,6 @@ static int omap_hdmi_audio_remove(struct platform_device *pdev)
struct hdmi_audio_data *ad = platform_get_drvdata(pdev);
snd_soc_unregister_card(ad->card);
- snd_soc_unregister_component(ad->dssdev);
return 0;
}
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 776e148b0aa2..943b44de1464 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -19,14 +19,13 @@ config SND_MMP_SOC
config SND_PXA2XX_AC97
tristate
- select SND_AC97_CODEC
config SND_PXA2XX_SOC_AC97
tristate
- select AC97_BUS
+ select AC97_BUS_NEW
select SND_PXA2XX_LIB
select SND_PXA2XX_LIB_AC97
- select SND_SOC_AC97_BUS
+ select SND_SOC_AC97_BUS_NEW
config SND_PXA2XX_SOC_I2S
select SND_PXA2XX_LIB
@@ -80,6 +79,7 @@ config SND_PXA2XX_SOC_TOSA
tristate "SoC AC97 Audio support for Tosa"
depends on SND_PXA2XX_SOC && MACH_TOSA
depends on MFD_TC6393XB
+ depends on !AC97_BUS
select SND_PXA2XX_SOC_AC97
select SND_SOC_WM9712
help
@@ -89,6 +89,7 @@ config SND_PXA2XX_SOC_TOSA
config SND_PXA2XX_SOC_E740
tristate "SoC AC97 Audio support for e740"
depends on SND_PXA2XX_SOC && MACH_E740
+ depends on !AC97_BUS
select SND_SOC_WM9705
select SND_PXA2XX_SOC_AC97
help
@@ -98,6 +99,7 @@ config SND_PXA2XX_SOC_E740
config SND_PXA2XX_SOC_E750
tristate "SoC AC97 Audio support for e750"
depends on SND_PXA2XX_SOC && MACH_E750
+ depends on !AC97_BUS
select SND_SOC_WM9705
select SND_PXA2XX_SOC_AC97
help
@@ -107,6 +109,7 @@ config SND_PXA2XX_SOC_E750
config SND_PXA2XX_SOC_E800
tristate "SoC AC97 Audio support for e800"
depends on SND_PXA2XX_SOC && MACH_E800
+ depends on !AC97_BUS
select SND_SOC_WM9712
select SND_PXA2XX_SOC_AC97
help
@@ -117,6 +120,7 @@ config SND_PXA2XX_SOC_EM_X270
tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300"
depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \
MACH_CM_X300)
+ depends on !AC97_BUS
select SND_PXA2XX_SOC_AC97
select SND_SOC_WM9712
help
@@ -127,6 +131,7 @@ config SND_PXA2XX_SOC_PALM27X
bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive"
depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \
MACH_PALMT5 || MACH_PALMTE2)
+ depends on !AC97_BUS
select SND_PXA2XX_SOC_AC97
select SND_SOC_WM9712
help
@@ -156,6 +161,7 @@ config SND_SOC_TTC_DKB
config SND_SOC_ZYLONITE
tristate "SoC Audio support for Marvell Zylonite"
depends on SND_PXA2XX_SOC && MACH_ZYLONITE
+ depends on !AC97_BUS
select SND_PXA2XX_SOC_AC97
select SND_PXA_SOC_SSP
select SND_SOC_WM9713
@@ -195,6 +201,7 @@ config SND_PXA2XX_SOC_MAGICIAN
config SND_PXA2XX_SOC_MIOA701
tristate "SoC Audio support for MIO A701"
depends on SND_PXA2XX_SOC && MACH_MIOA701
+ depends on !AC97_BUS
select SND_PXA2XX_SOC_AC97
select SND_SOC_WM9713
help
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 69033e1a84e6..adcf8ba9d287 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -103,6 +103,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
pxa_ssp_disable(ssp);
}
+ if (priv->extclk)
+ clk_prepare_enable(priv->extclk);
+
dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL);
if (!dma)
return -ENOMEM;
@@ -125,6 +128,9 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
clk_disable_unprepare(ssp->clk);
}
+ if (priv->extclk)
+ clk_disable_unprepare(priv->extclk);
+
kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
}
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 9f779657bc86..f8a3aa6c6d4e 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -17,6 +17,7 @@
#include <linux/dmaengine.h>
#include <linux/dma/pxa-dma.h>
+#include <sound/ac97/controller.h>
#include <sound/core.h>
#include <sound/ac97_codec.h>
#include <sound/soc.h>
@@ -27,43 +28,35 @@
#include <mach/regs-ac97.h>
#include <mach/audio.h>
-static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
+static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv)
{
pxa2xx_ac97_try_warm_reset();
pxa2xx_ac97_finish_reset();
}
-static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97)
+static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv)
{
pxa2xx_ac97_try_cold_reset();
pxa2xx_ac97_finish_reset();
}
-static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97,
- unsigned short reg)
+static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot,
+ unsigned short reg)
{
- int ret;
-
- ret = pxa2xx_ac97_read(ac97->num, reg);
- if (ret < 0)
- return 0;
- else
- return (unsigned short)(ret & 0xffff);
+ return pxa2xx_ac97_read(slot, reg);
}
-static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97,
- unsigned short reg, unsigned short val)
+static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot,
+ unsigned short reg, unsigned short val)
{
- int ret;
-
- ret = pxa2xx_ac97_write(ac97->num, reg, val);
+ return pxa2xx_ac97_write(slot, reg, val);
}
-static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
- .read = pxa2xx_ac97_legacy_read,
- .write = pxa2xx_ac97_legacy_write,
+static struct ac97_controller_ops pxa2xx_ac97_ops = {
+ .read = pxa2xx_ac97_read_actrl,
+ .write = pxa2xx_ac97_write_actrl,
.warm_reset = pxa2xx_ac97_warm_reset,
.reset = pxa2xx_ac97_cold_reset,
};
@@ -233,6 +226,9 @@ MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids);
static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
{
int ret;
+ struct ac97_controller *ctrl;
+ pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data;
+ void **codecs_pdata;
if (pdev->id != -1) {
dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n");
@@ -245,10 +241,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
return ret;
}
- ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops);
- if (ret != 0)
- return ret;
+ codecs_pdata = pdata ? pdata->codec_pdata : NULL;
+ ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev,
+ AC97_SLOTS_AVAILABLE_ALL,
+ codecs_pdata);
+ if (IS_ERR(ctrl))
+ return PTR_ERR(ctrl);
+ platform_set_drvdata(pdev, ctrl);
/* Punt most of the init to the SoC probe; we may need the machine
* driver to do interesting things with the clocking to get us up
* and running.
@@ -259,8 +259,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
static int pxa2xx_ac97_dev_remove(struct platform_device *pdev)
{
+ struct ac97_controller *ctrl = platform_get_drvdata(pdev);
+
snd_soc_unregister_component(&pdev->dev);
- snd_soc_set_ac97_ops(NULL);
+ snd_ac97_controller_unregister(ctrl);
pxa2xx_ac97_hw_remove(pdev);
return 0;
}
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 1543e85629f8..fb45f396ab4a 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -25,13 +25,12 @@ static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
static void apq8096_add_be_ops(struct snd_soc_card *card)
{
- struct snd_soc_dai_link *link = card->dai_link;
- int i, num_links = card->num_links;
+ struct snd_soc_dai_link *link;
+ int i;
- for (i = 0; i < num_links; i++) {
+ for_each_card_prelinks(card, i, link) {
if (link->no_pcm == 1)
link->be_hw_params_fixup = apq8096_be_hw_params_fixup;
- link++;
}
}
diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c
index 932c3ebfd252..da242515e146 100644
--- a/sound/soc/qcom/qdsp6/q6adm.c
+++ b/sound/soc/qcom/qdsp6/q6adm.c
@@ -2,25 +2,24 @@
// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
// Copyright (c) 2018, Linaro Limited
-#include <linux/slab.h>
-#include <linux/wait.h>
-#include <linux/kernel.h>
#include <linux/device.h>
-#include <linux/module.h>
-#include <linux/sched.h>
#include <linux/jiffies.h>
+#include <linux/kernel.h>
+#include <linux/kref.h>
+#include <linux/module.h>
#include <linux/of.h>
#include <linux/of_platform.h>
-#include <linux/kref.h>
-#include <linux/wait.h>
-#include <linux/soc/qcom/apr.h>
#include <linux/platform_device.h>
+#include <linux/sched.h>
+#include <linux/slab.h>
+#include <linux/soc/qcom/apr.h>
+#include <linux/wait.h>
#include <sound/asound.h>
#include "q6adm.h"
#include "q6afe.h"
#include "q6core.h"
-#include "q6dsp-errno.h"
#include "q6dsp-common.h"
+#include "q6dsp-errno.h"
#define ADM_CMD_DEVICE_OPEN_V5 0x00010326
#define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 9db9a2944ef2..a16c71c03058 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -8,7 +8,6 @@
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/soc.h>
-#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/pcm.h>
#include <asm/dma.h>
@@ -319,10 +318,11 @@ static int q6asm_dai_open(struct snd_pcm_substream *substream)
prtd->audio_client = q6asm_audio_client_alloc(dev,
(q6asm_cb)event_handler, prtd, stream_id,
LEGACY_PCM_MODE);
- if (!prtd->audio_client) {
+ if (IS_ERR(prtd->audio_client)) {
pr_info("%s: Could not allocate memory\n", __func__);
+ ret = PTR_ERR(prtd->audio_client);
kfree(prtd);
- return -ENOMEM;
+ return ret;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -493,7 +493,7 @@ static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)
}
}
- return ret;
+ return 0;
}
static void q6asm_dai_pcm_free(struct snd_pcm *pcm)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 2b2c7233bb5f..e1cfa846a1dc 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -11,7 +11,6 @@
#include <linux/spinlock.h>
#include <linux/kref.h>
#include <linux/of.h>
-#include <linux/of_platform.h>
#include <uapi/sound/asound.h>
#include <linux/delay.h>
#include <linux/slab.h>
diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c
index 06f03a5fe9bd..cdfc8ab6cfc0 100644
--- a/sound/soc/qcom/qdsp6/q6core.c
+++ b/sound/soc/qcom/qdsp6/q6core.c
@@ -10,7 +10,6 @@
#include <linux/of.h>
#include <linux/of_platform.h>
#include <linux/jiffies.h>
-#include <linux/wait.h>
#include <linux/soc/qcom/apr.h>
#include "q6core.h"
#include "q6dsp-errno.h"
@@ -105,12 +104,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data)
bytes = sizeof(*fwk) + fwk->num_services *
sizeof(fwk->svc_api_info[0]);
- core->fwk_version = kzalloc(bytes, GFP_ATOMIC);
+ core->fwk_version = kmemdup(data->payload, bytes, GFP_ATOMIC);
if (!core->fwk_version)
return -ENOMEM;
- memcpy(core->fwk_version, data->payload, bytes);
-
core->fwk_version_supported = true;
core->resp_received = true;
@@ -124,12 +121,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data)
len = sizeof(*v) + v->num_services * sizeof(v->svc_api_info[0]);
- core->svc_version = kzalloc(len, GFP_ATOMIC);
+ core->svc_version = kmemdup(data->payload, len, GFP_ATOMIC);
if (!core->svc_version)
return -ENOMEM;
- memcpy(core->svc_version, data->payload, len);
-
core->get_version_supported = true;
core->resp_received = true;
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 2a781d87ee65..9effbecc571f 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -195,15 +195,14 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
static void sdm845_add_be_ops(struct snd_soc_card *card)
{
- struct snd_soc_dai_link *link = card->dai_link;
- int i, num_links = card->num_links;
+ struct snd_soc_dai_link *link;
+ int i;
- for (i = 0; i < num_links; i++) {
+ for_each_card_prelinks(card, i, link) {
if (link->no_pcm == 1) {
link->ops = &sdm845_be_ops;
link->be_hw_params_fixup = sdm845_be_hw_params_fixup;
}
- link++;
}
}
diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c
index 929b3fe289b0..a472d5eb2950 100644
--- a/sound/soc/rockchip/rk3288_hdmi_analog.c
+++ b/sound/soc/rockchip/rk3288_hdmi_analog.c
@@ -286,7 +286,6 @@ static struct platform_driver rockchip_sound_driver = {
.probe = snd_rk_mc_probe,
.driver = {
.name = DRV_NAME,
- .owner = THIS_MODULE,
.pm = &snd_soc_pm_ops,
.of_match_table = rockchip_sound_of_match,
},
diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c
index f77538319221..9e7b5fa4cf59 100644
--- a/sound/soc/rockchip/rockchip_pcm.c
+++ b/sound/soc/rockchip/rockchip_pcm.c
@@ -21,7 +21,8 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_RESUME,
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_INTERLEAVED,
.period_bytes_min = 32,
.period_bytes_max = 8192,
.periods_min = 1,
diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c
index 43332c32d7e9..dc93941e01c3 100644
--- a/sound/soc/samsung/tm2_wm5110.c
+++ b/sound/soc/samsung/tm2_wm5110.c
@@ -491,6 +491,7 @@ static int tm2_probe(struct platform_device *pdev)
struct snd_soc_card *card = &tm2_card;
struct tm2_machine_priv *priv;
struct of_phandle_args args;
+ struct snd_soc_dai_link *dai_link;
int num_codecs, ret, i;
priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -558,18 +559,18 @@ static int tm2_probe(struct platform_device *pdev)
}
/* Initialize WM5110 - I2S and HDMI - I2S1 DAI links */
- for (i = 0; i < card->num_links; i++) {
+ for_each_card_prelinks(card, i, dai_link) {
unsigned int dai_index = 0; /* WM5110 */
- card->dai_link[i].cpu_name = NULL;
- card->dai_link[i].platform_name = NULL;
+ dai_link->cpu_name = NULL;
+ dai_link->platform_name = NULL;
if (num_codecs > 1 && i == card->num_links - 1)
dai_index = 1; /* HDMI */
- card->dai_link[i].codec_of_node = codec_dai_node[dai_index];
- card->dai_link[i].cpu_of_node = cpu_dai_node[dai_index];
- card->dai_link[i].platform_of_node = cpu_dai_node[dai_index];
+ dai_link->codec_of_node = codec_dai_node[dai_index];
+ dai_link->cpu_of_node = cpu_dai_node[dai_index];
+ dai_link->platform_of_node = cpu_dai_node[dai_index];
}
if (num_codecs > 1) {
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index c2b496398e6b..17622ceb98c0 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -319,13 +319,12 @@ static int hac_soc_platform_probe(struct platform_device *pdev)
if (ret != 0)
return ret;
- return snd_soc_register_component(&pdev->dev, &sh4_hac_component,
+ return devm_snd_soc_register_component(&pdev->dev, &sh4_hac_component,
sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
}
static int hac_soc_platform_remove(struct platform_device *pdev)
{
- snd_soc_unregister_component(&pdev->dev);
snd_soc_set_ac97_ops(NULL);
return 0;
}
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 051f96405346..28327dd2c6cb 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -582,7 +582,7 @@ static void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct rsnd_adg *adg)
int i;
for_each_rsnd_clk(clk, adg, i)
- dev_dbg(dev, "%s : %p : %ld\n",
+ dev_dbg(dev, "%s : %pa : %ld\n",
clk_name[i], clk, clk_get_rate(clk));
dev_dbg(dev, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n",
@@ -595,7 +595,7 @@ static void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct rsnd_adg *adg)
* by BRGCKR::BRGCKR_31
*/
for_each_rsnd_clkout(clk, adg, i)
- dev_dbg(dev, "clkout %d : %p : %ld\n", i,
+ dev_dbg(dev, "clkout %d : %pa : %ld\n", i,
clk, clk_get_rate(clk));
}
#else
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index d23c2bbff0cf..f930f51b686f 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -102,7 +102,9 @@
#include "rsnd.h"
#define RSND_RATES SNDRV_PCM_RATE_8000_192000
-#define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
+#define RSND_FMTS (SNDRV_PCM_FMTBIT_S8 |\
+ SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
static const struct of_device_id rsnd_of_match[] = {
{ .compatible = "renesas,rcar_sound-gen1", .data = (void *)RSND_GEN1 },
@@ -280,6 +282,8 @@ u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io)
struct device *dev = rsnd_priv_to_dev(priv);
switch (snd_pcm_format_width(runtime->format)) {
+ case 8:
+ return 16 << 16;
case 16:
return 8 << 16;
case 24:
@@ -331,7 +335,7 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io)
target = cmd ? cmd : ssiu;
}
- /* Non target mod or 24bit data needs normal DALIGN */
+ /* Non target mod or non 16bit needs normal DALIGN */
if ((snd_pcm_format_width(runtime->format) != 16) ||
(mod != target))
return 0x76543210;
@@ -367,7 +371,7 @@ u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod)
* HW 24bit data is located as 0x******00
*
*/
- if (snd_pcm_format_width(runtime->format) == 16)
+ if (snd_pcm_format_width(runtime->format) != 24)
return 0;
for (i = 0; i < ARRAY_SIZE(playback_mods); i++) {
@@ -540,6 +544,14 @@ int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai,
return rdai->ssi_lane;
}
+int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width)
+{
+ if (width > 0)
+ rdai->chan_width = width;
+
+ return rdai->chan_width;
+}
+
struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id)
{
if ((id < 0) || (id >= rsnd_rdai_nr(priv)))
@@ -681,6 +693,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
rdai->frm_clk_inv = 0;
break;
case SND_SOC_DAIFMT_LEFT_J:
+ case SND_SOC_DAIFMT_DSP_B:
rdai->sys_delay = 1;
rdai->data_alignment = 0;
rdai->frm_clk_inv = 1;
@@ -690,6 +703,11 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
rdai->data_alignment = 1;
rdai->frm_clk_inv = 1;
break;
+ case SND_SOC_DAIFMT_DSP_A:
+ rdai->sys_delay = 0;
+ rdai->data_alignment = 0;
+ rdai->frm_clk_inv = 1;
+ break;
}
/* set clock inversion */
@@ -720,6 +738,16 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai,
struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
struct device *dev = rsnd_priv_to_dev(priv);
+ switch (slot_width) {
+ case 16:
+ case 24:
+ case 32:
+ break;
+ default:
+ /* use default */
+ slot_width = 32;
+ }
+
switch (slots) {
case 2:
case 6:
@@ -727,6 +755,7 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai,
/* TDM Extend Mode */
rsnd_rdai_channels_set(rdai, slots);
rsnd_rdai_ssi_lane_set(rdai, 1);
+ rsnd_rdai_width_set(rdai, slot_width);
break;
default:
dev_err(dev, "unsupported TDM slots (%d)\n", slots);
@@ -755,7 +784,7 @@ static unsigned int rsnd_soc_hw_rate_list[] = {
192000,
};
-static int rsnd_soc_hw_rule(struct rsnd_priv *priv,
+static int rsnd_soc_hw_rule(struct rsnd_dai *rdai,
unsigned int *list, int list_num,
struct snd_interval *baseline, struct snd_interval *iv)
{
@@ -772,14 +801,14 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv,
if (!snd_interval_test(iv, list[i]))
continue;
- rate = rsnd_ssi_clk_query(priv,
+ rate = rsnd_ssi_clk_query(rdai,
baseline->min, list[i], NULL);
if (rate > 0) {
p.min = min(p.min, list[i]);
p.max = max(p.max, list[i]);
}
- rate = rsnd_ssi_clk_query(priv,
+ rate = rsnd_ssi_clk_query(rdai,
baseline->max, list[i], NULL);
if (rate > 0) {
p.min = min(p.min, list[i]);
@@ -790,17 +819,14 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv,
return snd_interval_refine(iv, &p);
}
-static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params,
- struct snd_pcm_hw_rule *rule,
- int is_play)
+static int rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
{
struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval ic;
- struct snd_soc_dai *dai = rule->private;
- struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
- struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
- struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture;
+ struct rsnd_dai_stream *io = rule->private;
+ struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
/*
* possible sampling rate limitation is same as
@@ -811,34 +837,19 @@ static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params,
ic.min =
ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params);
- return rsnd_soc_hw_rule(priv, rsnd_soc_hw_rate_list,
+ return rsnd_soc_hw_rule(rdai, rsnd_soc_hw_rate_list,
ARRAY_SIZE(rsnd_soc_hw_rate_list),
&ic, ir);
}
-static int rsnd_soc_hw_rule_rate_playback(struct snd_pcm_hw_params *params,
- struct snd_pcm_hw_rule *rule)
-{
- return __rsnd_soc_hw_rule_rate(params, rule, 1);
-}
-
-static int rsnd_soc_hw_rule_rate_capture(struct snd_pcm_hw_params *params,
- struct snd_pcm_hw_rule *rule)
-{
- return __rsnd_soc_hw_rule_rate(params, rule, 0);
-}
-
-static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params,
- struct snd_pcm_hw_rule *rule,
- int is_play)
+static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
{
struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval ic;
- struct snd_soc_dai *dai = rule->private;
- struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
- struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
- struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture;
+ struct rsnd_dai_stream *io = rule->private;
+ struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
/*
* possible sampling rate limitation is same as
@@ -849,23 +860,11 @@ static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params,
ic.min =
ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params);
- return rsnd_soc_hw_rule(priv, rsnd_soc_hw_channels_list,
+ return rsnd_soc_hw_rule(rdai, rsnd_soc_hw_channels_list,
ARRAY_SIZE(rsnd_soc_hw_channels_list),
ir, &ic);
}
-static int rsnd_soc_hw_rule_channels_playback(struct snd_pcm_hw_params *params,
- struct snd_pcm_hw_rule *rule)
-{
- return __rsnd_soc_hw_rule_channels(params, rule, 1);
-}
-
-static int rsnd_soc_hw_rule_channels_capture(struct snd_pcm_hw_params *params,
- struct snd_pcm_hw_rule *rule)
-{
- return __rsnd_soc_hw_rule_channels(params, rule, 0);
-}
-
static const struct snd_pcm_hardware rsnd_pcm_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_MMAP |
@@ -882,12 +881,10 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
- struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream);
struct snd_pcm_hw_constraint_list *constraint = &rdai->constraint;
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned int max_channels = rsnd_rdai_channels_get(rdai);
- int ret;
int i;
rsnd_dai_stream_init(io, substream);
@@ -922,25 +919,16 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream,
int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
- is_play ? rsnd_soc_hw_rule_rate_playback :
- rsnd_soc_hw_rule_rate_capture,
- dai,
+ rsnd_soc_hw_rule_rate,
+ is_play ? &rdai->playback : &rdai->capture,
SNDRV_PCM_HW_PARAM_CHANNELS, -1);
snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
- is_play ? rsnd_soc_hw_rule_channels_playback :
- rsnd_soc_hw_rule_channels_capture,
- dai,
+ rsnd_soc_hw_rule_channels,
+ is_play ? &rdai->playback : &rdai->capture,
SNDRV_PCM_HW_PARAM_RATE, -1);
}
- /*
- * call rsnd_dai_call without spinlock
- */
- ret = rsnd_dai_call(nolock_start, io, priv);
- if (ret < 0)
- rsnd_dai_call(nolock_stop, io, priv);
-
- return ret;
+ return 0;
}
static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream,
@@ -953,7 +941,7 @@ static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream,
/*
* call rsnd_dai_call without spinlock
*/
- rsnd_dai_call(nolock_stop, io, priv);
+ rsnd_dai_call(cleanup, io, priv);
rsnd_dai_stream_quit(io);
}
@@ -1083,6 +1071,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv,
rdai->capture.rdai = rdai;
rsnd_rdai_channels_set(rdai, 2); /* default 2ch */
rsnd_rdai_ssi_lane_set(rdai, 1); /* default 1lane */
+ rsnd_rdai_width_set(rdai, 32); /* default 32bit width */
for (io_i = 0;; io_i++) {
playback = of_parse_phandle(dai_np, "playback", io_i);
@@ -1274,8 +1263,15 @@ int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io)
int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io)
{
struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+ struct rsnd_priv *priv = rsnd_io_to_priv(io);
+ struct device *dev = rsnd_priv_to_dev(priv);
+
+ if (!runtime) {
+ dev_warn(dev, "Can't update kctrl when idle\n");
+ return 0;
+ }
- return !!runtime;
+ return 1;
}
struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg)
diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c
index 6a55aa753003..ad702377a6c3 100644
--- a/sound/soc/sh/rcar/ctu.c
+++ b/sound/soc/sh/rcar/ctu.c
@@ -258,7 +258,7 @@ static int rsnd_ctu_hw_params(struct rsnd_mod *mod,
struct snd_pcm_hw_params *be_params;
int stream = substream->stream;
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
be_params = &dpcm->hw_params;
if (params_channels(fe_params) != params_channels(be_params))
ctu->channels = params_channels(be_params);
diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index d65ea7bc4dac..6d1947515dc8 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -106,9 +106,9 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod,
return 0;
}
-static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod,
- struct rsnd_dai_stream *io,
- struct rsnd_priv *priv)
+static int rsnd_dmaen_cleanup(struct rsnd_mod *mod,
+ struct rsnd_dai_stream *io,
+ struct rsnd_priv *priv)
{
struct rsnd_dma *dma = rsnd_mod_to_dma(mod);
struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma);
@@ -116,7 +116,7 @@ static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod,
/*
* DMAEngine release uses mutex lock.
* Thus, it shouldn't be called under spinlock.
- * Let's call it under nolock_start
+ * Let's call it under prepare
*/
if (dmaen->chan)
dma_release_channel(dmaen->chan);
@@ -126,23 +126,22 @@ static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod,
return 0;
}
-static int rsnd_dmaen_nolock_start(struct rsnd_mod *mod,
- struct rsnd_dai_stream *io,
- struct rsnd_priv *priv)
+static int rsnd_dmaen_prepare(struct rsnd_mod *mod,
+ struct rsnd_dai_stream *io,
+ struct rsnd_priv *priv)
{
struct rsnd_dma *dma = rsnd_mod_to_dma(mod);
struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma);
struct device *dev = rsnd_priv_to_dev(priv);
- if (dmaen->chan) {
- dev_err(dev, "it already has dma channel\n");
- return -EIO;
- }
+ /* maybe suspended */
+ if (dmaen->chan)
+ return 0;
/*
* DMAEngine request uses mutex lock.
* Thus, it shouldn't be called under spinlock.
- * Let's call it under nolock_start
+ * Let's call it under prepare
*/
dmaen->chan = rsnd_dmaen_request_channel(io,
dma->mod_from,
@@ -291,8 +290,8 @@ static int rsnd_dmaen_pointer(struct rsnd_mod *mod,
static struct rsnd_mod_ops rsnd_dmaen_ops = {
.name = "audmac",
- .nolock_start = rsnd_dmaen_nolock_start,
- .nolock_stop = rsnd_dmaen_nolock_stop,
+ .prepare = rsnd_dmaen_prepare,
+ .cleanup = rsnd_dmaen_cleanup,
.start = rsnd_dmaen_start,
.stop = rsnd_dmaen_stop,
.pointer= rsnd_dmaen_pointer,
@@ -302,16 +301,26 @@ static struct rsnd_mod_ops rsnd_dmaen_ops = {
* Audio DMAC peri peri
*/
static const u8 gen2_id_table_ssiu[] = {
- 0x00, /* SSI00 */
- 0x04, /* SSI10 */
- 0x08, /* SSI20 */
- 0x0c, /* SSI3 */
- 0x0d, /* SSI4 */
- 0x0e, /* SSI5 */
- 0x0f, /* SSI6 */
- 0x10, /* SSI7 */
- 0x11, /* SSI8 */
- 0x12, /* SSI90 */
+ /* SSI00 ~ SSI07 */
+ 0x00, 0x01, 0x02, 0x03, 0x39, 0x3a, 0x3b, 0x3c,
+ /* SSI10 ~ SSI17 */
+ 0x04, 0x05, 0x06, 0x07, 0x3d, 0x3e, 0x3f, 0x40,
+ /* SSI20 ~ SSI27 */
+ 0x08, 0x09, 0x0a, 0x0b, 0x41, 0x42, 0x43, 0x44,
+ /* SSI30 ~ SSI37 */
+ 0x0c, 0x45, 0x46, 0x47, 0x48, 0x49, 0x4a, 0x4b,
+ /* SSI40 ~ SSI47 */
+ 0x0d, 0x4c, 0x4d, 0x4e, 0x4f, 0x50, 0x51, 0x52,
+ /* SSI5 */
+ 0x0e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ /* SSI6 */
+ 0x0f, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ /* SSI7 */
+ 0x10, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ /* SSI8 */
+ 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ /* SSI90 ~ SSI97 */
+ 0x12, 0x13, 0x14, 0x15, 0x53, 0x54, 0x55, 0x56,
};
static const u8 gen2_id_table_scu[] = {
0x2d, /* SCU_SRCI0 */
@@ -337,18 +346,23 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io,
struct rsnd_mod *src = rsnd_io_to_mod_src(io);
struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io);
const u8 *entry = NULL;
- int id = rsnd_mod_id(mod);
+ int id = 255;
int size = 0;
if (mod == ssi) {
+ int busif = rsnd_ssi_get_busif(io);
+
entry = gen2_id_table_ssiu;
size = ARRAY_SIZE(gen2_id_table_ssiu);
+ id = (rsnd_mod_id(mod) * 8) + busif;
} else if (mod == src) {
entry = gen2_id_table_scu;
size = ARRAY_SIZE(gen2_id_table_scu);
+ id = rsnd_mod_id(mod);
} else if (mod == dvc) {
entry = gen2_id_table_cmd;
size = ARRAY_SIZE(gen2_id_table_cmd);
+ id = rsnd_mod_id(mod);
}
if ((!entry) || (size <= id)) {
@@ -382,7 +396,7 @@ static void rsnd_dmapp_write(struct rsnd_dma *dma, u32 data, u32 reg)
struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv);
struct device *dev = rsnd_priv_to_dev(priv);
- dev_dbg(dev, "w %p : %08x\n", rsnd_dmapp_addr(dmac, dma, reg), data);
+ dev_dbg(dev, "w 0x%px : %08x\n", rsnd_dmapp_addr(dmac, dma, reg), data);
iowrite32(data, rsnd_dmapp_addr(dmac, dma, reg));
}
@@ -491,11 +505,11 @@ static struct rsnd_mod_ops rsnd_dmapp_ops = {
#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_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_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400))
+#define RDMA_SSIU_O_N(addr, i, j) RDMA_SSIU_I_N(addr, i, j)
-#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_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400))
+#define RDMA_SSIU_O_P(addr, i, j) RDMA_SSIU_I_P(addr, i, j)
#define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i))
#define RDMA_SRC_O_N(addr, i) (addr ##_reg - 0x004fc000 + (0x400 * i))
@@ -521,6 +535,7 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io,
!!rsnd_io_to_mod_mix(io) ||
!!rsnd_io_to_mod_ctu(io);
int id = rsnd_mod_id(mod);
+ int busif = rsnd_ssi_get_busif(io);
struct dma_addr {
dma_addr_t out_addr;
dma_addr_t in_addr;
@@ -537,25 +552,35 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io,
},
/* SSI */
/* Capture */
- {{{ RDMA_SSI_O_N(ssi, id), 0 },
- { RDMA_SSIU_O_P(ssi, id), 0 },
- { RDMA_SSIU_O_P(ssi, id), 0 } },
+ {{{ RDMA_SSI_O_N(ssi, id), 0 },
+ { RDMA_SSIU_O_P(ssi, id, busif), 0 },
+ { RDMA_SSIU_O_P(ssi, id, busif), 0 } },
/* Playback */
- {{ 0, RDMA_SSI_I_N(ssi, id) },
- { 0, RDMA_SSIU_I_P(ssi, id) },
- { 0, RDMA_SSIU_I_P(ssi, id) } }
+ {{ 0, RDMA_SSI_I_N(ssi, id) },
+ { 0, RDMA_SSIU_I_P(ssi, id, busif) },
+ { 0, RDMA_SSIU_I_P(ssi, id, busif) } }
},
/* SSIU */
/* Capture */
- {{{ RDMA_SSIU_O_N(ssi, id), 0 },
- { RDMA_SSIU_O_P(ssi, id), 0 },
- { RDMA_SSIU_O_P(ssi, id), 0 } },
+ {{{ RDMA_SSIU_O_N(ssi, id, busif), 0 },
+ { RDMA_SSIU_O_P(ssi, id, busif), 0 },
+ { RDMA_SSIU_O_P(ssi, id, busif), 0 } },
/* Playback */
- {{ 0, RDMA_SSIU_I_N(ssi, id) },
- { 0, RDMA_SSIU_I_P(ssi, id) },
- { 0, RDMA_SSIU_I_P(ssi, id) } } },
+ {{ 0, RDMA_SSIU_I_N(ssi, id, busif) },
+ { 0, RDMA_SSIU_I_P(ssi, id, busif) },
+ { 0, RDMA_SSIU_I_P(ssi, id, busif) } } },
};
+ /*
+ * FIXME
+ *
+ * We can't support SSI9-4/5/6/7, because its address is
+ * out of calculation rule
+ */
+ if ((id == 9) && (busif >= 4))
+ dev_err(dev, "This driver doesn't support SSI%d-%d, so far",
+ id, busif);
+
/* it shouldn't happen */
if (use_cmd && !use_src)
dev_err(dev, "DVC is selected without SRC\n");
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 0230301fe078..1f7881cc16b2 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -219,12 +219,33 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv)
RSND_GEN_S_REG(HDMI1_SEL, 0x9e4),
/* 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(SSI_BUSIF_DALIGN,0x8, 0x80),
- RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80),
- RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80),
- RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF0_MODE, 0x0, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF0_ADINR, 0x4, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF0_DALIGN, 0x8, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF1_MODE, 0x20, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF1_ADINR, 0x24, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF1_DALIGN, 0x28, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF2_MODE, 0x40, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF2_ADINR, 0x44, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF2_DALIGN, 0x48, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF3_MODE, 0x60, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF3_ADINR, 0x64, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF3_DALIGN, 0x68, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF4_ADINR, 0x504, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF5_MODE, 0x520, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF5_ADINR, 0x524, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF5_DALIGN, 0x528, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF6_MODE, 0x540, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF6_ADINR, 0x544, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF6_DALIGN, 0x548, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF7_MODE, 0x560, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF7_ADINR, 0x564, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF7_DALIGN, 0x568, 0x80),
+ RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80),
+ RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80),
+ RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80),
};
static const struct rsnd_regmap_field_conf conf_scu[] = {
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 8f7a0abfa751..4464d1d0a042 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -156,9 +156,30 @@ enum rsnd_reg {
RSND_REG_SSI_MODE2,
RSND_REG_SSI_CONTROL,
RSND_REG_SSI_CTRL,
- RSND_REG_SSI_BUSIF_MODE,
- RSND_REG_SSI_BUSIF_ADINR,
- RSND_REG_SSI_BUSIF_DALIGN,
+ RSND_REG_SSI_BUSIF0_MODE,
+ RSND_REG_SSI_BUSIF0_ADINR,
+ RSND_REG_SSI_BUSIF0_DALIGN,
+ RSND_REG_SSI_BUSIF1_MODE,
+ RSND_REG_SSI_BUSIF1_ADINR,
+ RSND_REG_SSI_BUSIF1_DALIGN,
+ RSND_REG_SSI_BUSIF2_MODE,
+ RSND_REG_SSI_BUSIF2_ADINR,
+ RSND_REG_SSI_BUSIF2_DALIGN,
+ RSND_REG_SSI_BUSIF3_MODE,
+ RSND_REG_SSI_BUSIF3_ADINR,
+ RSND_REG_SSI_BUSIF3_DALIGN,
+ RSND_REG_SSI_BUSIF4_MODE,
+ RSND_REG_SSI_BUSIF4_ADINR,
+ RSND_REG_SSI_BUSIF4_DALIGN,
+ RSND_REG_SSI_BUSIF5_MODE,
+ RSND_REG_SSI_BUSIF5_ADINR,
+ RSND_REG_SSI_BUSIF5_DALIGN,
+ RSND_REG_SSI_BUSIF6_MODE,
+ RSND_REG_SSI_BUSIF6_ADINR,
+ RSND_REG_SSI_BUSIF6_DALIGN,
+ RSND_REG_SSI_BUSIF7_MODE,
+ RSND_REG_SSI_BUSIF7_ADINR,
+ RSND_REG_SSI_BUSIF7_DALIGN,
RSND_REG_SSI_INT_ENABLE,
RSND_REG_SSI_SYS_STATUS0,
RSND_REG_SSI_SYS_STATUS1,
@@ -274,15 +295,12 @@ struct rsnd_mod_ops {
int (*fallback)(struct rsnd_mod *mod,
struct rsnd_dai_stream *io,
struct rsnd_priv *priv);
- int (*nolock_start)(struct rsnd_mod *mod,
- struct rsnd_dai_stream *io,
- struct rsnd_priv *priv);
- int (*nolock_stop)(struct rsnd_mod *mod,
- struct rsnd_dai_stream *io,
- struct rsnd_priv *priv);
int (*prepare)(struct rsnd_mod *mod,
struct rsnd_dai_stream *io,
struct rsnd_priv *priv);
+ int (*cleanup)(struct rsnd_mod *mod,
+ struct rsnd_dai_stream *io,
+ struct rsnd_priv *priv);
};
struct rsnd_dai_stream;
@@ -300,9 +318,8 @@ struct rsnd_mod {
/*
* status
*
- * 0xH0000CBA
+ * 0xH0000CB0
*
- * A 0: nolock_start 1: nolock_stop
* B 0: init 1: quit
* C 0: start 1: stop
*
@@ -313,9 +330,8 @@ struct rsnd_mod {
* H 0: hw_params
* H 0: pointer
* H 0: prepare
+ * H 0: cleanup
*/
-#define __rsnd_mod_shift_nolock_start 0
-#define __rsnd_mod_shift_nolock_stop 0
#define __rsnd_mod_shift_init 4
#define __rsnd_mod_shift_quit 4
#define __rsnd_mod_shift_start 8
@@ -328,11 +344,12 @@ struct rsnd_mod {
#define __rsnd_mod_shift_hw_params 28 /* always called */
#define __rsnd_mod_shift_pointer 28 /* always called */
#define __rsnd_mod_shift_prepare 28 /* always called */
+#define __rsnd_mod_shift_cleanup 28 /* always called */
#define __rsnd_mod_add_probe 0
#define __rsnd_mod_add_remove 0
-#define __rsnd_mod_add_nolock_start 1
-#define __rsnd_mod_add_nolock_stop -1
+#define __rsnd_mod_add_prepare 0
+#define __rsnd_mod_add_cleanup 0
#define __rsnd_mod_add_init 1
#define __rsnd_mod_add_quit -1
#define __rsnd_mod_add_start 1
@@ -342,10 +359,11 @@ struct rsnd_mod {
#define __rsnd_mod_add_fallback 0
#define __rsnd_mod_add_hw_params 0
#define __rsnd_mod_add_pointer 0
-#define __rsnd_mod_add_prepare 0
#define __rsnd_mod_call_probe 0
#define __rsnd_mod_call_remove 0
+#define __rsnd_mod_call_prepare 0
+#define __rsnd_mod_call_cleanup 0
#define __rsnd_mod_call_init 0
#define __rsnd_mod_call_quit 1
#define __rsnd_mod_call_start 0
@@ -355,9 +373,6 @@ struct rsnd_mod {
#define __rsnd_mod_call_fallback 0
#define __rsnd_mod_call_hw_params 0
#define __rsnd_mod_call_pointer 0
-#define __rsnd_mod_call_nolock_start 0
-#define __rsnd_mod_call_nolock_stop 1
-#define __rsnd_mod_call_prepare 0
#define rsnd_mod_to_priv(mod) ((mod)->priv)
#define rsnd_mod_name(mod) ((mod)->ops->name)
@@ -438,6 +453,7 @@ struct rsnd_dai_stream {
char name[RSND_DAI_NAME_SIZE];
struct snd_pcm_substream *substream;
struct rsnd_mod *mod[RSND_MOD_MAX];
+ struct rsnd_mod *dma;
struct rsnd_dai *rdai;
struct device *dmac_dev; /* for IPMMU */
u32 parent_ssi_status;
@@ -467,6 +483,7 @@ struct rsnd_dai {
int max_channels; /* 2ch - 16ch */
int ssi_lane; /* 1lane - 4lane */
+ int chan_width; /* 16/24/32 bit width */
unsigned int clk_master:1;
unsigned int bit_clk_inv:1;
@@ -500,6 +517,11 @@ int rsnd_rdai_channels_ctrl(struct rsnd_dai *rdai,
int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai,
int ssi_lane);
+#define rsnd_rdai_width_set(rdai, width) \
+ rsnd_rdai_width_ctrl(rdai, width)
+#define rsnd_rdai_width_get(rdai) \
+ rsnd_rdai_width_ctrl(rdai, 0)
+int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width);
void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io);
int rsnd_dai_connect(struct rsnd_mod *mod,
struct rsnd_dai_stream *io,
@@ -692,6 +714,7 @@ void rsnd_ssi_remove(struct rsnd_priv *priv);
struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id);
int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod);
int rsnd_ssi_use_busif(struct rsnd_dai_stream *io);
+int rsnd_ssi_get_busif(struct rsnd_dai_stream *io);
u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io);
#define RSND_SSI_HDMI_PORT0 0xf0
@@ -709,7 +732,7 @@ int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod);
void rsnd_parse_connect_ssi(struct rsnd_dai *rdai,
struct device_node *playback,
struct device_node *capture);
-unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv,
+unsigned int rsnd_ssi_clk_query(struct rsnd_dai *rdai,
int param1, int param2, int *idx);
/*
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index beccfbac7581..cd38a43b976f 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -158,7 +158,7 @@ static int rsnd_src_hw_params(struct rsnd_mod *mod,
struct snd_soc_dpcm *dpcm;
struct snd_pcm_hw_params *be_params;
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
be_params = &dpcm->hw_params;
if (params_rate(fe_params) != params_rate(be_params))
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 3f880ec66459..fcb4df23248c 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -42,7 +42,13 @@
#define DWL_24 (5 << 19) /* Data Word Length */
#define DWL_32 (6 << 19) /* Data Word Length */
+/*
+ * System word length
+ */
+#define SWL_16 (1 << 16) /* R/W System Word Length */
+#define SWL_24 (2 << 16) /* R/W System Word Length */
#define SWL_32 (3 << 16) /* R/W System Word Length */
+
#define SCKD (1 << 15) /* Serial Bit Clock Direction */
#define SWSD (1 << 14) /* Serial WS Direction */
#define SCKP (1 << 13) /* Serial Bit Clock Polarity */
@@ -72,7 +78,6 @@
struct rsnd_ssi {
struct rsnd_mod mod;
- struct rsnd_mod *dma;
u32 flags;
u32 cr_own;
@@ -145,6 +150,11 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io)
return use_busif;
}
+int rsnd_ssi_get_busif(struct rsnd_dai_stream *io)
+{
+ return 0; /* BUSIF0 only for now */
+}
+
static void rsnd_ssi_status_clear(struct rsnd_mod *mod)
{
rsnd_mod_write(mod, SSISR, 0);
@@ -220,14 +230,32 @@ u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io)
return 0;
}
-unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv,
+static u32 rsnd_rdai_width_to_swl(struct rsnd_dai *rdai)
+{
+ struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ int width = rsnd_rdai_width_get(rdai);
+
+ switch (width) {
+ case 32: return SWL_32;
+ case 24: return SWL_24;
+ case 16: return SWL_16;
+ }
+
+ dev_err(dev, "unsupported slot width value: %d\n", width);
+ return 0;
+}
+
+unsigned int rsnd_ssi_clk_query(struct rsnd_dai *rdai,
int param1, int param2, int *idx)
{
+ struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
int ssi_clk_mul_table[] = {
1, 2, 4, 8, 16, 6, 12,
};
int j, ret;
unsigned int main_rate;
+ int width = rsnd_rdai_width_get(rdai);
for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) {
@@ -240,12 +268,7 @@ unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv,
if (j == 0)
continue;
- /*
- * this driver is assuming that
- * system word is 32bit x chan
- * see rsnd_ssi_init()
- */
- main_rate = 32 * param1 * param2 * ssi_clk_mul_table[j];
+ main_rate = width * param1 * param2 * ssi_clk_mul_table[j];
ret = rsnd_adg_clk_query(priv, main_rate);
if (ret < 0)
@@ -289,10 +312,15 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
return -EINVAL;
}
+ if (ssi->chan != chan) {
+ dev_err(dev, "SSI parent/child should use same chan\n");
+ return -EINVAL;
+ }
+
return 0;
}
- main_rate = rsnd_ssi_clk_query(priv, rate, chan, &idx);
+ main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx);
if (!main_rate) {
dev_err(dev, "unsupported clock rate\n");
return -EIO;
@@ -312,9 +340,11 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
* SSICR : FORCE, SCKD, SWSD
* SSIWSR : CONT
*/
- ssi->cr_clk = FORCE | SWL_32 | SCKD | SWSD | CKDV(idx);
+ ssi->cr_clk = FORCE | rsnd_rdai_width_to_swl(rdai) |
+ SCKD | SWSD | CKDV(idx);
ssi->wsr = CONT;
ssi->rate = rate;
+ ssi->chan = chan;
dev_dbg(dev, "%s[%d] outputs %u Hz\n",
rsnd_mod_name(mod),
@@ -340,6 +370,7 @@ static void rsnd_ssi_master_clk_stop(struct rsnd_mod *mod,
ssi->cr_clk = 0;
ssi->rate = 0;
+ ssi->chan = 0;
rsnd_adg_ssi_clk_stop(mod);
}
@@ -357,15 +388,11 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
is_tdm = rsnd_runtime_is_ssi_tdm(io);
- /*
- * always use 32bit system word.
- * see also rsnd_ssi_master_clk_enable()
- */
- cr_own |= FORCE | SWL_32;
+ cr_own |= FORCE | rsnd_rdai_width_to_swl(rdai);
if (rdai->bit_clk_inv)
cr_own |= SCKP;
- if (rdai->frm_clk_inv ^ is_tdm)
+ if (rdai->frm_clk_inv && !is_tdm)
cr_own |= SWSP;
if (rdai->data_alignment)
cr_own |= SDTA;
@@ -373,6 +400,17 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
cr_own |= DEL;
/*
+ * TDM Mode
+ * see
+ * rsnd_ssiu_init_gen2()
+ */
+ wsr = ssi->wsr;
+ if (is_tdm) {
+ wsr |= WS_MODE;
+ cr_own |= CHNL_8;
+ }
+
+ /*
* We shouldn't exchange SWSP after running.
* This means, parent needs to care it.
*/
@@ -384,6 +422,9 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
cr_own &= ~DWL_MASK;
switch (snd_pcm_format_width(runtime->format)) {
+ case 8:
+ cr_own |= DWL_8;
+ break;
case 16:
cr_own |= DWL_16;
break;
@@ -399,16 +440,6 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
cr_mode = DIEN; /* PIO : enable Data interrupt */
}
- /*
- * TDM Extend Mode
- * see
- * rsnd_ssiu_init_gen2()
- */
- wsr = ssi->wsr;
- if (is_tdm) {
- wsr |= WS_MODE;
- cr_own |= CHNL_8;
- }
init_end:
ssi->cr_own = cr_own;
ssi->cr_mode = cr_mode;
@@ -488,26 +519,16 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod,
struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
- int chan = params_channels(params);
+ struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
+ unsigned int fmt_width = snd_pcm_format_width(params_format(params));
- /*
- * snd_pcm_ops::hw_params will be called *before*
- * snd_soc_dai_ops::trigger. Thus, ssi->usrcnt is 0
- * in 1st call.
- */
- if (ssi->usrcnt) {
- /*
- * Already working.
- * It will happen if SSI has parent/child connection.
- * it is error if child <-> parent SSI uses
- * different channels.
- */
- if (ssi->chan != chan)
- return -EIO;
- }
+ if (fmt_width > rdai->chan_width) {
+ struct rsnd_priv *priv = rsnd_io_to_priv(io);
+ struct device *dev = rsnd_priv_to_dev(priv);
- ssi->chan = chan;
+ dev_err(dev, "invalid combination of slot-width and format-data-width\n");
+ return -EINVAL;
+ }
return 0;
}
@@ -873,7 +894,6 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
struct rsnd_dai_stream *io,
struct rsnd_priv *priv)
{
- struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
int ret;
/*
@@ -888,7 +908,7 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
return ret;
/* SSI probe might be called many times in MUX multi path */
- ret = rsnd_dma_attach(io, mod, &ssi->dma);
+ ret = rsnd_dma_attach(io, mod, &io->dma);
return ret;
}
diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index 016fbf5ac242..39b67643b5dc 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -10,9 +10,12 @@
struct rsnd_ssiu {
struct rsnd_mod mod;
+ u32 busif_status[8]; /* for BUSIF0 - BUSIF7 */
+ unsigned int usrcnt;
};
#define rsnd_ssiu_nr(priv) ((priv)->ssiu_nr)
+#define rsnd_mod_to_ssiu(_mod) container_of((_mod), struct rsnd_ssiu, mod)
#define for_each_rsnd_ssiu(pos, priv, i) \
for (i = 0; \
(i < rsnd_ssiu_nr(priv)) && \
@@ -120,6 +123,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
struct rsnd_dai_stream *io,
struct rsnd_priv *priv)
{
+ struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod);
int hdmi = rsnd_ssi_hdmi_port(io);
int ret;
u32 mode = 0;
@@ -128,6 +132,8 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
if (ret < 0)
return ret;
+ ssiu->usrcnt++;
+
if (rsnd_runtime_is_ssi_tdm(io)) {
/*
* TDM Extend Mode
@@ -140,15 +146,59 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
rsnd_mod_write(mod, SSI_MODE, mode);
if (rsnd_ssi_use_busif(io)) {
- rsnd_mod_write(mod, SSI_BUSIF_ADINR,
- rsnd_get_adinr_bit(mod, io) |
- (rsnd_io_is_play(io) ?
- rsnd_runtime_channel_after_ctu(io) :
- rsnd_runtime_channel_original(io)));
- rsnd_mod_write(mod, SSI_BUSIF_MODE,
- rsnd_get_busif_shift(io, mod) | 1);
- rsnd_mod_write(mod, SSI_BUSIF_DALIGN,
- rsnd_get_dalign(mod, io));
+ int id = rsnd_mod_id(mod);
+ int busif = rsnd_ssi_get_busif(io);
+
+ /*
+ * FIXME
+ *
+ * We can't support SSI9-4/5/6/7, because its address is
+ * out of calculation rule
+ */
+ if ((id == 9) && (busif >= 4)) {
+ struct device *dev = rsnd_priv_to_dev(priv);
+
+ dev_err(dev, "This driver doesn't support SSI%d-%d, so far",
+ id, busif);
+ }
+
+#define RSND_WRITE_BUSIF(i) \
+ rsnd_mod_write(mod, SSI_BUSIF##i##_ADINR, \
+ rsnd_get_adinr_bit(mod, io) | \
+ (rsnd_io_is_play(io) ? \
+ rsnd_runtime_channel_after_ctu(io) : \
+ rsnd_runtime_channel_original(io))); \
+ rsnd_mod_write(mod, SSI_BUSIF##i##_MODE, \
+ rsnd_get_busif_shift(io, mod) | 1); \
+ rsnd_mod_write(mod, SSI_BUSIF##i##_DALIGN, \
+ rsnd_get_dalign(mod, io))
+
+ switch (busif) {
+ case 0:
+ RSND_WRITE_BUSIF(0);
+ break;
+ case 1:
+ RSND_WRITE_BUSIF(1);
+ break;
+ case 2:
+ RSND_WRITE_BUSIF(2);
+ break;
+ case 3:
+ RSND_WRITE_BUSIF(3);
+ break;
+ case 4:
+ RSND_WRITE_BUSIF(4);
+ break;
+ case 5:
+ RSND_WRITE_BUSIF(5);
+ break;
+ case 6:
+ RSND_WRITE_BUSIF(6);
+ break;
+ case 7:
+ RSND_WRITE_BUSIF(7);
+ break;
+ }
}
if (hdmi) {
@@ -194,10 +244,12 @@ static int rsnd_ssiu_start_gen2(struct rsnd_mod *mod,
struct rsnd_dai_stream *io,
struct rsnd_priv *priv)
{
+ int busif = rsnd_ssi_get_busif(io);
+
if (!rsnd_ssi_use_busif(io))
return 0;
- rsnd_mod_write(mod, SSI_CTRL, 0x1);
+ rsnd_mod_bset(mod, SSI_CTRL, 1 << (busif * 4), 1 << (busif * 4));
if (rsnd_ssi_multi_slaves_runtime(io))
rsnd_mod_write(mod, SSI_CONTROL, 0x1);
@@ -209,10 +261,16 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod,
struct rsnd_dai_stream *io,
struct rsnd_priv *priv)
{
+ struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod);
+ int busif = rsnd_ssi_get_busif(io);
+
if (!rsnd_ssi_use_busif(io))
return 0;
- rsnd_mod_write(mod, SSI_CTRL, 0);
+ rsnd_mod_bset(mod, SSI_CTRL, 1 << (busif * 4), 0);
+
+ if (--ssiu->usrcnt)
+ return 0;
if (rsnd_ssi_multi_slaves_runtime(io))
rsnd_mod_write(mod, SSI_CONTROL, 0);
@@ -246,6 +304,16 @@ int rsnd_ssiu_attach(struct rsnd_dai_stream *io,
return rsnd_dai_connect(mod, io, mod->type);
}
+static u32 *rsnd_ssiu_get_status(struct rsnd_dai_stream *io,
+ struct rsnd_mod *mod,
+ enum rsnd_mod_type type)
+{
+ struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod);
+ int busif = rsnd_ssi_get_busif(io);
+
+ return &ssiu->busif_status[busif];
+}
+
int rsnd_ssiu_probe(struct rsnd_priv *priv)
{
struct device *dev = rsnd_priv_to_dev(priv);
@@ -269,7 +337,7 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv)
for_each_rsnd_ssiu(ssiu, priv, i) {
ret = rsnd_mod_init(priv, rsnd_mod_get(ssiu),
- ops, NULL, rsnd_mod_get_status,
+ ops, NULL, rsnd_ssiu_get_status,
RSND_MOD_SSIU, i);
if (ret)
return ret;
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 409d082e80d1..699397a09167 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -157,7 +157,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
ret = dpcm_be_dai_startup(fe, stream);
if (ret < 0) {
/* clean up all links */
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+ for_each_dpcm_be(fe, stream, dpcm)
dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
dpcm_be_disconnect(fe, stream);
@@ -321,7 +321,7 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream)
ret = dpcm_be_dai_shutdown(fe, stream);
/* mark FE's links ready to prune */
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+ for_each_dpcm_be(fe, stream, dpcm)
dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 473eefe8658e..6ddcf12bc030 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -52,6 +52,10 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root);
static DEFINE_MUTEX(client_mutex);
static LIST_HEAD(component_list);
+static LIST_HEAD(unbind_card_list);
+
+#define for_each_component(component) \
+ list_for_each_entry(component, &component_list, list)
/*
* This is a timeout to do a DAPM powerdown after a stream is closed().
@@ -62,8 +66,9 @@ static int pmdown_time = 5000;
module_param(pmdown_time, int, 0);
MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
-/* If a DMI filed contain strings in this blacklist (e.g.
- * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken
+/*
+ * If a DMI filed contain strings in this blacklist (e.g.
+ * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken
* as invalid and dropped when setting the card long name from DMI info.
*/
static const char * const dmi_blacklist[] = {
@@ -175,8 +180,8 @@ static int dai_list_show(struct seq_file *m, void *v)
mutex_lock(&client_mutex);
- list_for_each_entry(component, &component_list, list)
- list_for_each_entry(dai, &component->dai_list, list)
+ for_each_component(component)
+ for_each_component_dais(component, dai)
seq_printf(m, "%s\n", dai->name);
mutex_unlock(&client_mutex);
@@ -191,7 +196,7 @@ static int component_list_show(struct seq_file *m, void *v)
mutex_lock(&client_mutex);
- list_for_each_entry(component, &component_list, list)
+ for_each_component(component)
seq_printf(m, "%s\n", component->name);
mutex_unlock(&client_mutex);
@@ -218,7 +223,7 @@ static void soc_init_card_debugfs(struct snd_soc_card *card)
&card->pop_time);
if (!card->debugfs_pop_time)
dev_warn(card->dev,
- "ASoC: Failed to create pop time debugfs file\n");
+ "ASoC: Failed to create pop time debugfs file\n");
}
static void soc_cleanup_card_debugfs(struct snd_soc_card *card)
@@ -341,7 +346,7 @@ struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
{
struct snd_soc_pcm_runtime *rtd;
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
if (rtd->dai_link->no_pcm &&
!strcmp(rtd->dai_link->name, dai_link))
return rtd->pcm->streams[stream].substream;
@@ -398,7 +403,7 @@ static void soc_remove_pcm_runtimes(struct snd_soc_card *card)
{
struct snd_soc_pcm_runtime *rtd, *_rtd;
- list_for_each_entry_safe(rtd, _rtd, &card->rtd_list, list) {
+ for_each_card_rtds_safe(card, rtd, _rtd) {
list_del(&rtd->list);
soc_free_pcm_runtime(rtd);
}
@@ -411,7 +416,7 @@ struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card,
{
struct snd_soc_pcm_runtime *rtd;
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
if (!strcmp(rtd->dai_link->name, dai_link))
return rtd;
}
@@ -422,7 +427,8 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime);
static void codec2codec_close_delayed_work(struct work_struct *work)
{
- /* Currently nothing to do for c2c links
+ /*
+ * Currently nothing to do for c2c links
* Since c2c links are internal nodes in the DAPM graph and
* don't interface with the outside world or application layer
* we don't have to do any special handling on close.
@@ -442,8 +448,9 @@ int snd_soc_suspend(struct device *dev)
if (!card->instantiated)
return 0;
- /* Due to the resume being scheduled into a workqueue we could
- * suspend before that's finished - wait for it to complete.
+ /*
+ * Due to the resume being scheduled into a workqueue we could
+ * suspend before that's finished - wait for it to complete.
*/
snd_power_wait(card->snd_card, SNDRV_CTL_POWER_D0);
@@ -451,13 +458,13 @@ int snd_soc_suspend(struct device *dev)
snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot);
/* mute any active DACs */
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
+ struct snd_soc_dai *dai;
if (rtd->dai_link->ignore_suspend)
continue;
- for (i = 0; i < rtd->num_codecs; i++) {
- struct snd_soc_dai *dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, dai) {
struct snd_soc_dai_driver *drv = dai->driver;
if (drv->ops->digital_mute && dai->playback_active)
@@ -466,7 +473,7 @@ int snd_soc_suspend(struct device *dev)
}
/* suspend all pcms */
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
if (rtd->dai_link->ignore_suspend)
continue;
@@ -476,7 +483,7 @@ int snd_soc_suspend(struct device *dev)
if (card->suspend_pre)
card->suspend_pre(card);
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
if (rtd->dai_link->ignore_suspend)
@@ -487,10 +494,10 @@ int snd_soc_suspend(struct device *dev)
}
/* close any waiting streams */
- list_for_each_entry(rtd, &card->rtd_list, list)
+ for_each_card_rtds(card, rtd)
flush_delayed_work(&rtd->delayed_work);
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
if (rtd->dai_link->ignore_suspend)
continue;
@@ -509,11 +516,14 @@ int snd_soc_suspend(struct device *dev)
snd_soc_dapm_sync(&card->dapm);
/* suspend all COMPONENTs */
- list_for_each_entry(component, &card->component_dev_list, card_list) {
- struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+ for_each_card_components(card, component) {
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_component_get_dapm(component);
- /* If there are paths active then the COMPONENT will be held with
- * bias _ON and should not be suspended. */
+ /*
+ * If there are paths active then the COMPONENT will be held
+ * with bias _ON and should not be suspended.
+ */
if (!component->suspended) {
switch (snd_soc_dapm_get_bias_level(dapm)) {
case SND_SOC_BIAS_STANDBY:
@@ -547,7 +557,7 @@ int snd_soc_suspend(struct device *dev)
}
}
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
if (rtd->dai_link->ignore_suspend)
@@ -567,18 +577,21 @@ int snd_soc_suspend(struct device *dev)
}
EXPORT_SYMBOL_GPL(snd_soc_suspend);
-/* deferred resume work, so resume can complete before we finished
+/*
+ * deferred resume work, so resume can complete before we finished
* setting our codec back up, which can be very slow on I2C
*/
static void soc_resume_deferred(struct work_struct *work)
{
struct snd_soc_card *card =
- container_of(work, struct snd_soc_card, deferred_resume_work);
+ container_of(work, struct snd_soc_card,
+ deferred_resume_work);
struct snd_soc_pcm_runtime *rtd;
struct snd_soc_component *component;
int i;
- /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
+ /*
+ * our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
* so userspace apps are blocked from touching us
*/
@@ -591,7 +604,7 @@ static void soc_resume_deferred(struct work_struct *work)
card->resume_pre(card);
/* resume control bus DAIs */
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
if (rtd->dai_link->ignore_suspend)
@@ -601,7 +614,7 @@ static void soc_resume_deferred(struct work_struct *work)
cpu_dai->driver->resume(cpu_dai);
}
- list_for_each_entry(component, &card->component_dev_list, card_list) {
+ for_each_card_components(card, component) {
if (component->suspended) {
if (component->driver->resume)
component->driver->resume(component);
@@ -609,7 +622,7 @@ static void soc_resume_deferred(struct work_struct *work)
}
}
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
if (rtd->dai_link->ignore_suspend)
continue;
@@ -624,13 +637,13 @@ static void soc_resume_deferred(struct work_struct *work)
}
/* unmute any active DACs */
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
+ struct snd_soc_dai *dai;
if (rtd->dai_link->ignore_suspend)
continue;
- for (i = 0; i < rtd->num_codecs; i++) {
- struct snd_soc_dai *dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, dai) {
struct snd_soc_dai_driver *drv = dai->driver;
if (drv->ops->digital_mute && dai->playback_active)
@@ -638,7 +651,7 @@ static void soc_resume_deferred(struct work_struct *work)
}
}
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
if (rtd->dai_link->ignore_suspend)
@@ -673,16 +686,15 @@ int snd_soc_resume(struct device *dev)
return 0;
/* activate pins from sleep state */
- list_for_each_entry(rtd, &card->rtd_list, list) {
- struct snd_soc_dai **codec_dais = rtd->codec_dais;
+ for_each_card_rtds(card, rtd) {
+ struct snd_soc_dai *codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int j;
if (cpu_dai->active)
pinctrl_pm_select_default_state(cpu_dai->dev);
- for (j = 0; j < rtd->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = codec_dais[j];
+ for_each_rtd_codec_dai(rtd, j, codec_dai) {
if (codec_dai->active)
pinctrl_pm_select_default_state(codec_dai->dev);
}
@@ -694,8 +706,9 @@ int snd_soc_resume(struct device *dev)
* have that problem and may take a substantial amount of time to resume
* due to I/O costs and anti-pop so handle them out of line.
*/
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
bus_control |= cpu_dai->driver->bus_control;
}
if (bus_control) {
@@ -725,7 +738,7 @@ static struct snd_soc_component *soc_find_component(
lockdep_assert_held(&client_mutex);
- list_for_each_entry(component, &component_list, list) {
+ for_each_component(component) {
if (of_node) {
if (component->dev->of_node == of_node)
return component;
@@ -737,6 +750,24 @@ static struct snd_soc_component *soc_find_component(
return NULL;
}
+static int snd_soc_is_matching_component(
+ const struct snd_soc_dai_link_component *dlc,
+ struct snd_soc_component *component)
+{
+ struct device_node *component_of_node;
+
+ component_of_node = component->dev->of_node;
+ if (!component_of_node && component->dev->parent)
+ component_of_node = component->dev->parent->of_node;
+
+ if (dlc->of_node && component_of_node != dlc->of_node)
+ return 0;
+ if (dlc->name && strcmp(component->name, dlc->name))
+ return 0;
+
+ return 1;
+}
+
/**
* snd_soc_find_dai - Find a registered DAI
*
@@ -753,21 +784,14 @@ struct snd_soc_dai *snd_soc_find_dai(
{
struct snd_soc_component *component;
struct snd_soc_dai *dai;
- struct device_node *component_of_node;
lockdep_assert_held(&client_mutex);
- /* Find CPU DAI from registered DAIs*/
- list_for_each_entry(component, &component_list, list) {
- component_of_node = component->dev->of_node;
- if (!component_of_node && component->dev->parent)
- component_of_node = component->dev->parent->of_node;
-
- if (dlc->of_node && component_of_node != dlc->of_node)
- continue;
- if (dlc->name && strcmp(component->name, dlc->name))
+ /* Find CPU DAI from registered DAIs */
+ for_each_component(component) {
+ if (!snd_soc_is_matching_component(dlc, component))
continue;
- list_for_each_entry(dai, &component->dai_list, list) {
+ for_each_component_dais(component, dai) {
if (dlc->dai_name && strcmp(dai->name, dlc->dai_name)
&& (!dai->driver->name
|| strcmp(dai->driver->name, dlc->dai_name)))
@@ -781,7 +805,6 @@ struct snd_soc_dai *snd_soc_find_dai(
}
EXPORT_SYMBOL_GPL(snd_soc_find_dai);
-
/**
* snd_soc_find_dai_link - Find a DAI link
*
@@ -805,7 +828,7 @@ struct snd_soc_dai_link *snd_soc_find_dai_link(struct snd_soc_card *card,
lockdep_assert_held(&client_mutex);
- list_for_each_entry_safe(link, _link, &card->dai_link_list, list) {
+ for_each_card_links_safe(card, link, _link) {
if (link->id != id)
continue;
@@ -828,7 +851,7 @@ static bool soc_is_dai_link_bound(struct snd_soc_card *card,
{
struct snd_soc_pcm_runtime *rtd;
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
if (rtd->dai_link == dai_link)
return true;
}
@@ -844,8 +867,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
struct snd_soc_dai_link_component cpu_dai_component;
struct snd_soc_component *component;
struct snd_soc_dai **codec_dais;
- struct device_node *platform_of_node;
- const char *platform_name;
int i;
if (dai_link->ignore)
@@ -877,6 +898,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
rtd->num_codecs = dai_link->num_codecs;
/* Find CODEC from registered CODECs */
+ /* we can use for_each_rtd_codec_dai() after this */
codec_dais = rtd->codec_dais;
for (i = 0; i < rtd->num_codecs; i++) {
codec_dais[i] = snd_soc_find_dai(&codecs[i]);
@@ -891,24 +913,11 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
/* Single codec links expect codec and codec_dai in runtime data */
rtd->codec_dai = codec_dais[0];
- /* 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)
- platform_name = "snd-soc-dummy";
-
/* find one from the set of registered platforms */
- list_for_each_entry(component, &component_list, list) {
- platform_of_node = component->dev->of_node;
- if (!platform_of_node && component->dev->parent->of_node)
- platform_of_node = component->dev->parent->of_node;
-
- if (dai_link->platform_of_node) {
- if (platform_of_node != dai_link->platform_of_node)
- continue;
- } else {
- if (strcmp(component->name, platform_name))
- continue;
- }
+ for_each_component(component) {
+ if (!snd_soc_is_matching_component(dai_link->platform,
+ component))
+ continue;
snd_soc_rtdcom_add(rtd, component);
}
@@ -918,7 +927,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
_err_defer:
soc_free_pcm_runtime(rtd);
- return -EPROBE_DEFER;
+ return -EPROBE_DEFER;
}
static void soc_remove_component(struct snd_soc_component *component)
@@ -942,23 +951,25 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order)
{
int err;
- if (dai && dai->probed &&
- dai->driver->remove_order == order) {
- if (dai->driver->remove) {
- err = dai->driver->remove(dai);
- if (err < 0)
- dev_err(dai->dev,
- "ASoC: failed to remove %s: %d\n",
- dai->name, err);
- }
- dai->probed = 0;
+ if (!dai || !dai->probed ||
+ dai->driver->remove_order != order)
+ return;
+
+ if (dai->driver->remove) {
+ err = dai->driver->remove(dai);
+ if (err < 0)
+ dev_err(dai->dev,
+ "ASoC: failed to remove %s: %d\n",
+ dai->name, err);
}
+ dai->probed = 0;
}
static void soc_remove_link_dais(struct snd_soc_card *card,
struct snd_soc_pcm_runtime *rtd, int order)
{
int i;
+ struct snd_soc_dai *codec_dai;
/* unregister the rtd device */
if (rtd->dev_registered) {
@@ -967,8 +978,8 @@ static void soc_remove_link_dais(struct snd_soc_card *card,
}
/* remove the CODEC DAI */
- for (i = 0; i < rtd->num_codecs; i++)
- soc_remove_dai(rtd->codec_dais[i], order);
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ soc_remove_dai(codec_dai, order);
soc_remove_dai(rtd->cpu_dai, order);
}
@@ -993,28 +1004,57 @@ static void soc_remove_dai_links(struct snd_soc_card *card)
struct snd_soc_pcm_runtime *rtd;
struct snd_soc_dai_link *link, *_link;
- for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
- order++) {
- list_for_each_entry(rtd, &card->rtd_list, list)
+ for_each_comp_order(order) {
+ for_each_card_rtds(card, rtd)
soc_remove_link_dais(card, rtd, order);
}
- for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
- order++) {
- list_for_each_entry(rtd, &card->rtd_list, list)
+ for_each_comp_order(order) {
+ for_each_card_rtds(card, rtd)
soc_remove_link_components(card, rtd, order);
}
- list_for_each_entry_safe(link, _link, &card->dai_link_list, list) {
+ for_each_card_links_safe(card, link, _link) {
if (link->dobj.type == SND_SOC_DOBJ_DAI_LINK)
dev_warn(card->dev, "Topology forgot to remove link %s?\n",
link->name);
list_del(&link->list);
- card->num_dai_links--;
}
}
+static int snd_soc_init_platform(struct snd_soc_card *card,
+ struct snd_soc_dai_link *dai_link)
+{
+ struct snd_soc_dai_link_component *platform = dai_link->platform;
+
+ /*
+ * FIXME
+ *
+ * this function should be removed in the future
+ */
+ /* convert Legacy platform link */
+ if (!platform) {
+ platform = devm_kzalloc(card->dev,
+ sizeof(struct snd_soc_dai_link_component),
+ GFP_KERNEL);
+ if (!platform)
+ return -ENOMEM;
+
+ dai_link->platform = platform;
+ platform->name = dai_link->platform_name;
+ platform->of_node = dai_link->platform_of_node;
+ platform->dai_name = NULL;
+ }
+
+ /* if there's no platform we match on the empty platform */
+ if (!platform->name &&
+ !platform->of_node)
+ platform->name = "snd-soc-dummy";
+
+ return 0;
+}
+
static int snd_soc_init_multicodec(struct snd_soc_card *card,
struct snd_soc_dai_link *dai_link)
{
@@ -1043,9 +1083,16 @@ static int snd_soc_init_multicodec(struct snd_soc_card *card,
}
static int soc_init_dai_link(struct snd_soc_card *card,
- struct snd_soc_dai_link *link)
+ struct snd_soc_dai_link *link)
{
int i, ret;
+ struct snd_soc_dai_link_component *codec;
+
+ ret = snd_soc_init_platform(card, link);
+ if (ret) {
+ dev_err(card->dev, "ASoC: failed to init multiplatform\n");
+ return ret;
+ }
ret = snd_soc_init_multicodec(card, link);
if (ret) {
@@ -1053,19 +1100,19 @@ static int soc_init_dai_link(struct snd_soc_card *card,
return ret;
}
- for (i = 0; i < link->num_codecs; i++) {
+ for_each_link_codecs(link, i, codec) {
/*
* Codec must be specified by 1 of name or OF node,
* not both or neither.
*/
- if (!!link->codecs[i].name ==
- !!link->codecs[i].of_node) {
+ if (!!codec->name ==
+ !!codec->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[i].dai_name) {
+ if (!codec->dai_name) {
dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n",
link->name);
return -EINVAL;
@@ -1076,13 +1123,12 @@ static int soc_init_dai_link(struct snd_soc_card *card,
* Platform may be specified by either name or OF node, but
* can be left unspecified, and a dummy platform will be used.
*/
- if (link->platform_name && link->platform_of_node) {
+ if (link->platform->name && link->platform->of_node) {
dev_err(card->dev,
"ASoC: Both platform name/of_node are set for %s\n",
link->name);
return -EINVAL;
}
-
/*
* CPU device may be specified by either name or OF node, but
* can be left unspecified, and will be matched based on DAI
@@ -1111,7 +1157,8 @@ static int soc_init_dai_link(struct snd_soc_card *card,
void snd_soc_disconnect_sync(struct device *dev)
{
- struct snd_soc_component *component = snd_soc_lookup_component(dev, NULL);
+ struct snd_soc_component *component =
+ snd_soc_lookup_component(dev, NULL);
if (!component || !component->card)
return;
@@ -1142,14 +1189,14 @@ int snd_soc_add_dai_link(struct snd_soc_card *card,
}
lockdep_assert_held(&client_mutex);
- /* Notify the machine driver for extra initialization
+ /*
+ * Notify the machine driver for extra initialization
* on the link created by topology.
*/
if (dai_link->dobj.type && card->add_dai_link)
card->add_dai_link(card, dai_link);
list_add_tail(&dai_link->list, &card->dai_link_list);
- card->num_dai_links++;
return 0;
}
@@ -1178,16 +1225,16 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card,
}
lockdep_assert_held(&client_mutex);
- /* Notify the machine driver for extra destruction
+ /*
+ * Notify the machine driver for extra destruction
* on the link created by topology.
*/
if (dai_link->dobj.type && card->remove_dai_link)
card->remove_dai_link(card, dai_link);
- list_for_each_entry_safe(link, _link, &card->dai_link_list, list) {
+ for_each_card_links_safe(card, link, _link) {
if (link == dai_link) {
list_del(&link->list);
- card->num_dai_links--;
return;
}
}
@@ -1239,7 +1286,8 @@ static void soc_set_name_prefix(struct snd_soc_card *card,
static int soc_probe_component(struct snd_soc_card *card,
struct snd_soc_component *component)
{
- struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_component_get_dapm(component);
struct snd_soc_dai *dai;
int ret;
@@ -1277,7 +1325,7 @@ static int soc_probe_component(struct snd_soc_card *card,
}
}
- list_for_each_entry(dai, &component->dai_list, list) {
+ for_each_component_dais(component, dai) {
ret = snd_soc_dapm_new_dai_widgets(dapm, dai);
if (ret != 0) {
dev_err(component->dev,
@@ -1320,6 +1368,7 @@ static int soc_probe_component(struct snd_soc_card *card,
component->driver->num_dapm_routes);
list_add(&dapm->list, &card->dapm_list);
+ /* see for_each_card_components */
list_add(&component->card_list, &card->component_dev_list);
return 0;
@@ -1370,8 +1419,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
}
static int soc_probe_link_components(struct snd_soc_card *card,
- struct snd_soc_pcm_runtime *rtd,
- int order)
+ struct snd_soc_pcm_runtime *rtd, int order)
{
struct snd_soc_component *component;
struct snd_soc_rtdcom_list *rtdcom;
@@ -1398,6 +1446,7 @@ static int soc_probe_dai(struct snd_soc_dai *dai, int order)
if (dai->driver->probe) {
int ret = dai->driver->probe(dai);
+
if (ret < 0) {
dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n",
dai->name, ret);
@@ -1431,48 +1480,6 @@ static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais,
return 0;
}
-static int soc_link_dai_widgets(struct snd_soc_card *card,
- struct snd_soc_dai_link *dai_link,
- 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 *sink, *source;
- int ret;
-
- if (rtd->num_codecs > 1)
- dev_warn(card->dev, "ASoC: Multiple codecs not supported yet\n");
-
- /* link the DAI widgets */
- sink = codec_dai->playback_widget;
- source = cpu_dai->capture_widget;
- if (sink && source) {
- ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params,
- dai_link->num_params,
- source, sink);
- if (ret != 0) {
- dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n",
- sink->name, source->name, ret);
- return ret;
- }
- }
-
- sink = cpu_dai->playback_widget;
- source = codec_dai->capture_widget;
- if (sink && source) {
- ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params,
- dai_link->num_params,
- source, sink);
- if (ret != 0) {
- dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n",
- sink->name, source->name, ret);
- return ret;
- }
- }
-
- return 0;
-}
-
static int soc_probe_link_dais(struct snd_soc_card *card,
struct snd_soc_pcm_runtime *rtd, int order)
{
@@ -1480,6 +1487,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_rtdcom_list *rtdcom;
struct snd_soc_component *component;
+ struct snd_soc_dai *codec_dai;
int i, ret, num;
dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n",
@@ -1493,8 +1501,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
return ret;
/* probe the CODEC DAI */
- for (i = 0; i < rtd->num_codecs; i++) {
- ret = soc_probe_dai(rtd->codec_dais[i], order);
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ ret = soc_probe_dai(codec_dai, order);
if (ret)
return ret;
}
@@ -1546,7 +1554,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
}
if (cpu_dai->driver->compress_new) {
- /*create compress_device"*/
+ /* create compress_device" */
ret = cpu_dai->driver->compress_new(rtd, num);
if (ret < 0) {
dev_err(card->dev, "ASoC: can't create compress %s\n",
@@ -1560,7 +1568,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
ret = soc_new_pcm(rtd, num);
if (ret < 0) {
dev_err(card->dev, "ASoC: can't create pcm %s :%d\n",
- dai_link->stream_name, ret);
+ dai_link->stream_name, ret);
return ret;
}
ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd);
@@ -1573,11 +1581,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
} else {
INIT_DELAYED_WORK(&rtd->delayed_work,
codec2codec_close_delayed_work);
-
- /* link the DAI widgets */
- ret = soc_link_dai_widgets(card, dai_link, rtd);
- if (ret)
- return ret;
}
}
@@ -1628,8 +1631,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card)
int order;
int ret;
- for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
- order++) {
+ for_each_comp_order(order) {
list_for_each_entry(comp, &card->aux_comp_list, card_aux_list) {
if (comp->driver->probe_order == order) {
ret = soc_probe_component(card, comp);
@@ -1651,8 +1653,7 @@ static void soc_remove_aux_devices(struct snd_soc_card *card)
struct snd_soc_component *comp, *_comp;
int order;
- for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
- order++) {
+ for_each_comp_order(order) {
list_for_each_entry_safe(comp, _comp,
&card->aux_comp_list, card_aux_list) {
@@ -1681,14 +1682,12 @@ static void soc_remove_aux_devices(struct snd_soc_card *card)
int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
unsigned int dai_fmt)
{
- struct snd_soc_dai **codec_dais = rtd->codec_dais;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai;
unsigned int i;
int ret;
- for (i = 0; i < rtd->num_codecs; i++) {
- struct snd_soc_dai *codec_dai = codec_dais[i];
-
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt);
if (ret != 0 && ret != -ENOTSUPP) {
dev_warn(codec_dai->dev,
@@ -1697,8 +1696,10 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
}
}
- /* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */
- /* the component which has non_legacy_dai_naming is Codec */
+ /*
+ * Flip the polarity for the "CPU" end of a CODEC<->CODEC link
+ * the component which has non_legacy_dai_naming is Codec
+ */
if (cpu_dai->component->driver->non_legacy_dai_naming) {
unsigned int inv_dai_fmt;
@@ -1732,9 +1733,9 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
}
EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt);
-
#ifdef CONFIG_DMI
-/* Trim special characters, and replace '-' with '_' since '-' is used to
+/*
+ * Trim special characters, and replace '-' with '_' since '-' is used to
* separate different DMI fields in the card long name. Only number and
* alphabet characters and a few separator characters are kept.
*/
@@ -1753,7 +1754,8 @@ static void cleanup_dmi_name(char *name)
name[j] = '\0';
}
-/* Check if a DMI field is valid, i.e. not containing any string
+/*
+ * Check if a DMI field is valid, i.e. not containing any string
* in the black list.
*/
static int is_dmi_valid(const char *field)
@@ -1816,7 +1818,6 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
return 0;
}
-
snprintf(card->dmi_longname, sizeof(card->snd_card->longname),
"%s", vendor);
cleanup_dmi_name(card->dmi_longname);
@@ -1832,7 +1833,8 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
if (len < longname_buf_size)
cleanup_dmi_name(card->dmi_longname + len);
- /* some vendors like Lenovo may only put a self-explanatory
+ /*
+ * some vendors like Lenovo may only put a self-explanatory
* name in the product version field
*/
product_version = dmi_get_system_info(DMI_PRODUCT_VERSION);
@@ -1891,7 +1893,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
struct snd_soc_dai_link *dai_link;
int i;
- list_for_each_entry(component, &component_list, list) {
+ for_each_component(component) {
/* does this component override FEs ? */
if (!component->driver->ignore_machine)
@@ -1903,9 +1905,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
continue;
/* machine matches, so override the rtd data */
- for (i = 0; i < card->num_links; i++) {
-
- dai_link = &card->dai_link[i];
+ for_each_card_prelinks(card, i, dai_link) {
/* ignore this FE */
if (dai_link->dynamic) {
@@ -1917,7 +1917,11 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
card->dai_link[i].name);
/* override platform component */
- dai_link->platform_name = component->name;
+ if (snd_soc_init_platform(card, dai_link) < 0) {
+ dev_err(card->dev, "init platform error");
+ continue;
+ }
+ dai_link->platform->name = component->name;
/* convert non BE into BE */
dai_link->no_pcm = 1;
@@ -1926,7 +1930,8 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
dai_link->be_hw_params_fixup =
component->driver->be_hw_params_fixup;
- /* most BE links don't set stream name, so set it to
+ /*
+ * most BE links don't set stream name, so set it to
* dai link name if it's NULL to help bind widgets.
*/
if (!dai_link->stream_name)
@@ -1936,7 +1941,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
/* Inform userspace we are using alternate topology */
if (component->driver->topology_name_prefix) {
- /* topology shortname created ? */
+ /* topology shortname created? */
if (!card->topology_shortname_created) {
comp_drv = component->driver;
@@ -1965,8 +1970,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
soc_check_tplg_fes(card);
/* bind DAIs */
- for (i = 0; i < card->num_links; i++) {
- ret = soc_bind_dai_link(card, &card->dai_link[i]);
+ for_each_card_prelinks(card, i, dai_link) {
+ ret = soc_bind_dai_link(card, dai_link);
if (ret != 0)
goto base_error;
}
@@ -1979,8 +1984,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
}
/* add predefined DAI links to the list */
- for (i = 0; i < card->num_links; i++)
- snd_soc_add_dai_link(card, card->dai_link+i);
+ for_each_card_prelinks(card, i, dai_link)
+ snd_soc_add_dai_link(card, dai_link);
/* card bind complete so register a sound card */
ret = snd_card_new(card->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
@@ -2024,9 +2029,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
}
/* probe all components used by DAI links on this card */
- for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
- order++) {
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_comp_order(order) {
+ for_each_card_rtds(card, rtd) {
ret = soc_probe_link_components(card, rtd, order);
if (ret < 0) {
dev_err(card->dev,
@@ -2042,10 +2046,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
if (ret < 0)
goto probe_dai_err;
- /* Find new DAI links added during probing components and bind them.
+ /*
+ * Find new DAI links added during probing components and bind them.
* Components with topology may bring new DAIs and DAI links.
*/
- list_for_each_entry(dai_link, &card->dai_link_list, list) {
+ for_each_card_links(card, dai_link) {
if (soc_is_dai_link_bound(card, dai_link))
continue;
@@ -2058,9 +2063,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
}
/* probe all DAI links on this card */
- for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
- order++) {
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_comp_order(order) {
+ for_each_card_rtds(card, rtd) {
ret = soc_probe_link_dais(card, rtd, order);
if (ret < 0) {
dev_err(card->dev,
@@ -2075,7 +2079,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
snd_soc_dapm_connect_dai_link_widgets(card);
if (card->controls)
- snd_soc_add_card_controls(card, card->controls, card->num_controls);
+ snd_soc_add_card_controls(card, card->controls,
+ card->num_controls);
if (card->dapm_routes)
snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
@@ -2181,7 +2186,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card)
struct snd_soc_pcm_runtime *rtd;
/* make sure any delayed work runs */
- list_for_each_entry(rtd, &card->rtd_list, list)
+ for_each_card_rtds(card, rtd)
flush_delayed_work(&rtd->delayed_work);
/* free the ALSA card at first; this syncs with pending operations */
@@ -2221,21 +2226,23 @@ int snd_soc_poweroff(struct device *dev)
if (!card->instantiated)
return 0;
- /* Flush out pmdown_time work - we actually do want to run it
- * now, we're shutting down so no imminent restart. */
- list_for_each_entry(rtd, &card->rtd_list, list)
+ /*
+ * Flush out pmdown_time work - we actually do want to run it
+ * now, we're shutting down so no imminent restart.
+ */
+ for_each_card_rtds(card, rtd)
flush_delayed_work(&rtd->delayed_work);
snd_soc_dapm_shutdown(card);
/* deactivate pins to sleep state */
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai;
int i;
pinctrl_pm_select_sleep_state(cpu_dai->dev);
- for (i = 0; i < rtd->num_codecs; i++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
pinctrl_pm_select_sleep_state(codec_dai->dev);
}
}
@@ -2315,6 +2322,7 @@ static int snd_soc_add_controls(struct snd_card *card, struct device *dev,
for (i = 0; i < num_controls; i++) {
const struct snd_kcontrol_new *control = &controls[i];
+
err = snd_ctl_add(card, snd_soc_cnew(control, data,
control->name, prefix));
if (err < 0) {
@@ -2432,8 +2440,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
*
* Configures the CODEC master (MCLK) or system (SYSCLK) clocking.
*/
-int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id,
- int source, unsigned int freq, int dir)
+int snd_soc_component_set_sysclk(struct snd_soc_component *component,
+ int clk_id, int source, unsigned int freq,
+ int dir)
{
if (component->driver->set_sysclk)
return component->driver->set_sysclk(component, clk_id, source,
@@ -2501,7 +2510,7 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id,
{
if (component->driver->set_pll)
return component->driver->set_pll(component, pll_id, source,
- freq_in, freq_out);
+ freq_in, freq_out);
return -EINVAL;
}
@@ -2532,8 +2541,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio);
*/
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
- if (dai->driver == NULL)
- return -EINVAL;
if (dai->driver->ops->set_fmt == NULL)
return -ENOTSUPP;
return dai->driver->ops->set_fmt(dai, fmt);
@@ -2549,8 +2556,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
* Generates the TDM tx and rx slot default masks for DAI.
*/
static int snd_soc_xlate_tdm_slot_mask(unsigned int slots,
- unsigned int *tx_mask,
- unsigned int *rx_mask)
+ unsigned int *tx_mask,
+ unsigned int *rx_mask)
{
if (*tx_mask || *rx_mask)
return 0;
@@ -2680,9 +2687,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
int direction)
{
- if (!dai->driver)
- return -ENOTSUPP;
-
if (dai->driver->ops->mute_stream)
return dai->driver->ops->mute_stream(dai, mute, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
@@ -2693,6 +2697,33 @@ 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_bind_card(struct snd_soc_card *card)
+{
+ struct snd_soc_pcm_runtime *rtd;
+ int ret;
+
+ ret = snd_soc_instantiate_card(card);
+ if (ret != 0)
+ return ret;
+
+ /* deactivate pins to sleep state */
+ for_each_card_rtds(card, rtd) {
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai;
+ int j;
+
+ for_each_rtd_codec_dai(rtd, j, codec_dai) {
+ if (!codec_dai->active)
+ pinctrl_pm_select_sleep_state(codec_dai->dev);
+ }
+
+ if (!cpu_dai->active)
+ pinctrl_pm_select_sleep_state(cpu_dai->dev);
+ }
+
+ return ret;
+}
+
/**
* snd_soc_register_card - Register a card with the ASoC core
*
@@ -2702,13 +2733,12 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
int snd_soc_register_card(struct snd_soc_card *card)
{
int i, ret;
- struct snd_soc_pcm_runtime *rtd;
+ struct snd_soc_dai_link *link;
if (!card->name || !card->dev)
return -EINVAL;
- for (i = 0; i < card->num_links; i++) {
- struct snd_soc_dai_link *link = &card->dai_link[i];
+ for_each_card_prelinks(card, i, link) {
ret = soc_init_dai_link(card, link);
if (ret) {
@@ -2723,7 +2753,6 @@ int snd_soc_register_card(struct snd_soc_card *card)
snd_soc_initialize_card_lists(card);
INIT_LIST_HEAD(&card->dai_link_list);
- card->num_dai_links = 0;
INIT_LIST_HEAD(&card->rtd_list);
card->num_rtd = 0;
@@ -2734,28 +2763,23 @@ int snd_soc_register_card(struct snd_soc_card *card)
mutex_init(&card->mutex);
mutex_init(&card->dapm_mutex);
- ret = snd_soc_instantiate_card(card);
- if (ret != 0)
- return ret;
-
- /* deactivate pins to sleep state */
- list_for_each_entry(rtd, &card->rtd_list, list) {
- 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);
- }
+ return snd_soc_bind_card(card);
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_card);
- if (!cpu_dai->active)
- pinctrl_pm_select_sleep_state(cpu_dai->dev);
+static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister)
+{
+ if (card->instantiated) {
+ card->instantiated = false;
+ snd_soc_dapm_shutdown(card);
+ soc_cleanup_card_resources(card);
+ if (!unregister)
+ list_add(&card->list, &unbind_card_list);
+ } else {
+ if (unregister)
+ list_del(&card->list);
}
-
- return ret;
}
-EXPORT_SYMBOL_GPL(snd_soc_register_card);
/**
* snd_soc_unregister_card - Unregister a card with the ASoC core
@@ -2765,12 +2789,8 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card);
*/
int snd_soc_unregister_card(struct snd_soc_card *card)
{
- if (card->instantiated) {
- card->instantiated = false;
- snd_soc_dapm_shutdown(card);
- soc_cleanup_card_resources(card);
- dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
- }
+ snd_soc_unbind_card(card, true);
+ dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
return 0;
}
@@ -2802,7 +2822,7 @@ static char *fmt_single_name(struct device *dev, int *id)
}
} else {
- /* I2C component devices are named "bus-addr" */
+ /* I2C component devices are named "bus-addr" */
if (sscanf(name, "%x-%x", &id1, &id2) == 2) {
char tmp[NAME_SIZE];
@@ -2810,7 +2830,8 @@ static char *fmt_single_name(struct device *dev, int *id)
*id = ((id1 & 0xffff) << 16) + id2;
/* sanitize component name for DAI link creation */
- snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name, name);
+ snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name,
+ name);
strlcpy(name, tmp, NAME_SIZE);
} else
*id = 0;
@@ -2845,7 +2866,7 @@ static void snd_soc_unregister_dais(struct snd_soc_component *component)
{
struct snd_soc_dai *dai, *_dai;
- list_for_each_entry_safe(dai, _dai, &component->dai_list, list) {
+ for_each_component_dais_safe(component, dai, _dai) {
dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n",
dai->name);
list_del(&dai->list);
@@ -2877,7 +2898,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component,
* component-less anymore.
*/
if (legacy_dai_naming &&
- (dai_drv->id == 0 || dai_drv->name == NULL)) {
+ (dai_drv->id == 0 || dai_drv->name == NULL)) {
dai->name = fmt_single_name(dev, &dai->id);
} else {
dai->name = fmt_multiple_name(dev, dai_drv);
@@ -2897,6 +2918,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component,
if (!dai->driver->ops)
dai->driver->ops = &null_dai_ops;
+ /* see for_each_component_dais */
list_add_tail(&dai->list, &component->dai_list);
component->num_dai++;
@@ -2910,11 +2932,10 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component,
* @component: The component the DAIs are registered for
* @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_dai_driver *dai_drv, size_t count)
+ struct snd_soc_dai_driver *dai_drv,
+ size_t count)
{
struct device *dev = component->dev;
struct snd_soc_dai *dai;
@@ -2925,8 +2946,8 @@ static int snd_soc_register_dais(struct snd_soc_component *component,
for (i = 0; i < count; i++) {
- dai = soc_add_dai(component, dai_drv + i,
- count == 1 && !component->driver->non_legacy_dai_naming);
+ dai = soc_add_dai(component, dai_drv + i, count == 1 &&
+ !component->driver->non_legacy_dai_naming);
if (dai == NULL) {
ret = -ENOMEM;
goto err;
@@ -2970,7 +2991,8 @@ int snd_soc_register_dai(struct snd_soc_component *component,
if (!dai)
return -ENOMEM;
- /* Create the DAI widgets here. After adding DAIs, topology may
+ /*
+ * Create the DAI widgets here. After adding DAIs, topology may
* also add routes that need these widgets as source or sink.
*/
ret = snd_soc_dapm_new_dai_widgets(dapm, dai);
@@ -3052,7 +3074,8 @@ static void snd_soc_component_setup_regmap(struct snd_soc_component *component)
#ifdef CONFIG_REGMAP
/**
- * snd_soc_component_init_regmap() - Initialize regmap instance for the component
+ * snd_soc_component_init_regmap() - Initialize regmap instance for the
+ * component
* @component: The component for which to initialize the regmap instance
* @regmap: The regmap instance that should be used by the component
*
@@ -3070,7 +3093,8 @@ void snd_soc_component_init_regmap(struct snd_soc_component *component,
EXPORT_SYMBOL_GPL(snd_soc_component_init_regmap);
/**
- * snd_soc_component_exit_regmap() - De-initialize regmap instance for the component
+ * snd_soc_component_exit_regmap() - De-initialize regmap instance for the
+ * component
* @component: The component for which to de-initialize the regmap instance
*
* Calls regmap_exit() on the regmap instance associated to the component and
@@ -3094,11 +3118,13 @@ static void snd_soc_component_add(struct snd_soc_component *component)
if (!component->driver->write && !component->driver->read) {
if (!component->regmap)
- component->regmap = dev_get_regmap(component->dev, NULL);
+ component->regmap = dev_get_regmap(component->dev,
+ NULL);
if (component->regmap)
snd_soc_component_setup_regmap(component);
}
+ /* see for_each_component */
list_add(&component->list, &component_list);
INIT_LIST_HEAD(&component->dobj_list);
@@ -3116,7 +3142,7 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component)
struct snd_soc_card *card = component->card;
if (card)
- snd_soc_unregister_card(card);
+ snd_soc_unbind_card(card, false);
list_del(&component->list);
}
@@ -3156,6 +3182,18 @@ static void convert_endianness_formats(struct snd_soc_pcm_stream *stream)
stream->formats |= endianness_format_map[i];
}
+static void snd_soc_try_rebind_card(void)
+{
+ struct snd_soc_card *card, *c;
+
+ if (!list_empty(&unbind_card_list)) {
+ list_for_each_entry_safe(card, c, &unbind_card_list, list) {
+ if (!snd_soc_bind_card(card))
+ list_del(&card->list);
+ }
+ }
+}
+
int snd_soc_add_component(struct device *dev,
struct snd_soc_component *component,
const struct snd_soc_component_driver *component_driver,
@@ -3183,6 +3221,7 @@ int snd_soc_add_component(struct device *dev,
}
snd_soc_component_add(component);
+ snd_soc_try_rebind_card();
return 0;
@@ -3221,27 +3260,28 @@ static int __snd_soc_unregister_component(struct device *dev)
int found = 0;
mutex_lock(&client_mutex);
- list_for_each_entry(component, &component_list, list) {
+ for_each_component(component) {
if (dev != component->dev)
continue;
- snd_soc_tplg_component_remove(component, SND_SOC_TPLG_INDEX_ALL);
+ snd_soc_tplg_component_remove(component,
+ SND_SOC_TPLG_INDEX_ALL);
snd_soc_component_del_unlocked(component);
found = 1;
break;
}
mutex_unlock(&client_mutex);
- if (found) {
+ if (found)
snd_soc_component_cleanup(component);
- }
return found;
}
void snd_soc_unregister_component(struct device *dev)
{
- while (__snd_soc_unregister_component(dev));
+ while (__snd_soc_unregister_component(dev))
+ ;
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_component);
@@ -3253,7 +3293,7 @@ struct snd_soc_component *snd_soc_lookup_component(struct device *dev,
ret = NULL;
mutex_lock(&client_mutex);
- list_for_each_entry(component, &component_list, list) {
+ for_each_component(component) {
if (dev != component->dev)
continue;
@@ -3653,7 +3693,7 @@ int snd_soc_get_dai_id(struct device_node *ep)
*/
ret = -ENOTSUPP;
mutex_lock(&client_mutex);
- list_for_each_entry(pos, &component_list, list) {
+ for_each_component(pos) {
struct device_node *component_of_node = pos->dev->of_node;
if (!component_of_node && pos->dev->parent)
@@ -3683,7 +3723,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args,
int ret = -EPROBE_DEFER;
mutex_lock(&client_mutex);
- list_for_each_entry(pos, &component_list, list) {
+ for_each_component(pos) {
component_of_node = pos->dev->of_node;
if (!component_of_node && pos->dev->parent)
component_of_node = pos->dev->parent->of_node;
@@ -3719,7 +3759,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args,
ret = 0;
/* find target DAI */
- list_for_each_entry(dai, &pos->dai_list, list) {
+ for_each_component_dais(pos, dai) {
if (id == 0)
break;
id--;
@@ -3764,10 +3804,10 @@ EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name);
*/
void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link)
{
- struct snd_soc_dai_link_component *component = dai_link->codecs;
+ struct snd_soc_dai_link_component *component;
int index;
- for (index = 0; index < dai_link->num_codecs; index++, component++) {
+ for_each_link_codecs(dai_link, index, component) {
if (!component->of_node)
break;
of_node_put(component->of_node);
@@ -3819,12 +3859,10 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev,
dai_link->num_codecs = num_codecs;
/* Parse the list */
- for (index = 0, component = dai_link->codecs;
- index < dai_link->num_codecs;
- index++, component++) {
+ for_each_link_codecs(dai_link, index, component) {
ret = of_parse_phandle_with_args(of_node, name,
"#sound-dai-cells",
- index, &args);
+ index, &args);
if (ret)
goto err;
component->of_node = args.np;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 461d951917c0..a5178845065b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -18,7 +18,6 @@
// device reopen.
#include <linux/module.h>
-#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/async.h>
#include <linux/delay.h>
@@ -364,10 +363,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
ret = PTR_ERR(data->widget);
goto err_data;
}
- if (!data->widget) {
- ret = -ENOMEM;
- goto err_data;
- }
}
break;
case snd_soc_dapm_demux:
@@ -402,10 +397,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
ret = PTR_ERR(data->widget);
goto err_data;
}
- if (!data->widget) {
- ret = -ENOMEM;
- goto err_data;
- }
snd_soc_dapm_add_path(widget->dapm, data->widget,
widget, NULL, NULL);
@@ -1026,9 +1017,10 @@ static int dapm_new_dai_link(struct snd_soc_dapm_widget *w)
struct snd_kcontrol *kcontrol;
struct snd_soc_dapm_context *dapm = w->dapm;
struct snd_card *card = dapm->card->snd_card;
+ struct snd_soc_pcm_runtime *rtd = w->priv;
/* create control for links with > 1 config */
- if (w->num_params <= 1)
+ if (rtd->dai_link->num_params <= 1)
return 0;
/* add kcontrol */
@@ -1320,14 +1312,13 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w,
soc_dapm_async_complete(w->dapm);
-#ifdef CONFIG_HAVE_CLK
if (SND_SOC_DAPM_EVENT_ON(event)) {
return clk_prepare_enable(w->clk);
} else {
clk_disable_unprepare(w->clk);
return 0;
}
-#endif
+
return 0;
}
EXPORT_SYMBOL_GPL(dapm_clock_event);
@@ -1953,7 +1944,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
dapm_pre_sequence_async(&card->dapm, 0);
/* Run other bias changes in parallel */
list_for_each_entry(d, &card->dapm_list, list) {
- if (d != &card->dapm)
+ if (d != &card->dapm && d->bias_level != d->target_bias_level)
async_schedule_domain(dapm_pre_sequence_async, d,
&async_domain);
}
@@ -1977,7 +1968,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
/* Run all the bias changes in parallel */
list_for_each_entry(d, &card->dapm_list, list) {
- if (d != &card->dapm)
+ if (d != &card->dapm && d->bias_level != d->target_bias_level)
async_schedule_domain(dapm_post_sequence_async, d,
&async_domain);
}
@@ -2371,12 +2362,13 @@ 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);
+ struct snd_soc_dai *codec_dai;
int i, count = 0;
mutex_lock(&rtd->card->dapm_mutex);
- for (i = 0; i < rtd->num_codecs; i++) {
- struct snd_soc_component *cmpnt = rtd->codec_dais[i]->component;
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ struct snd_soc_component *cmpnt = codec_dai->component;
count += dapm_widget_show_component(cmpnt, buf + count);
}
@@ -3426,35 +3418,6 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch);
struct snd_soc_dapm_widget *
-snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
- const struct snd_soc_dapm_widget *widget)
-{
- struct snd_soc_dapm_widget *w;
-
- mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
- w = snd_soc_dapm_new_control_unlocked(dapm, widget);
- /* Do not nag about probe deferrals */
- if (IS_ERR(w)) {
- int ret = PTR_ERR(w);
-
- if (ret != -EPROBE_DEFER)
- dev_err(dapm->dev,
- "ASoC: Failed to create DAPM control %s (%d)\n",
- widget->name, ret);
- goto out_unlock;
- }
- if (!w)
- dev_err(dapm->dev,
- "ASoC: Failed to create DAPM control %s\n",
- widget->name);
-
-out_unlock:
- mutex_unlock(&dapm->card->dapm_mutex);
- return w;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
-
-struct snd_soc_dapm_widget *
snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_widget *widget)
{
@@ -3464,53 +3427,37 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
int ret;
if ((w = dapm_cnew_widget(widget)) == NULL)
- return NULL;
+ return ERR_PTR(-ENOMEM);
switch (w->id) {
case snd_soc_dapm_regulator_supply:
w->regulator = devm_regulator_get(dapm->dev, w->name);
if (IS_ERR(w->regulator)) {
ret = PTR_ERR(w->regulator);
- if (ret == -EPROBE_DEFER)
- return ERR_PTR(ret);
- dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
- w->name, ret);
- return NULL;
+ goto request_failed;
}
if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) {
ret = regulator_allow_bypass(w->regulator, true);
if (ret != 0)
- dev_warn(w->dapm->dev,
+ dev_warn(dapm->dev,
"ASoC: Failed to bypass %s: %d\n",
w->name, ret);
}
break;
case snd_soc_dapm_pinctrl:
w->pinctrl = devm_pinctrl_get(dapm->dev);
- if (IS_ERR_OR_NULL(w->pinctrl)) {
+ if (IS_ERR(w->pinctrl)) {
ret = PTR_ERR(w->pinctrl);
- if (ret == -EPROBE_DEFER)
- return ERR_PTR(ret);
- dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
- w->name, ret);
- return NULL;
+ goto request_failed;
}
break;
case snd_soc_dapm_clock_supply:
-#ifdef CONFIG_CLKDEV_LOOKUP
w->clk = devm_clk_get(dapm->dev, w->name);
if (IS_ERR(w->clk)) {
ret = PTR_ERR(w->clk);
- if (ret == -EPROBE_DEFER)
- return ERR_PTR(ret);
- dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
- w->name, ret);
- return NULL;
+ goto request_failed;
}
-#else
- return NULL;
-#endif
break;
default:
break;
@@ -3523,7 +3470,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
w->name = kstrdup_const(widget->name, GFP_KERNEL);
if (w->name == NULL) {
kfree(w);
- return NULL;
+ return ERR_PTR(-ENOMEM);
}
switch (w->id) {
@@ -3600,7 +3547,37 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
/* machine layer sets up unconnected pins and insertions */
w->connected = 1;
return w;
+
+request_failed:
+ if (ret != -EPROBE_DEFER)
+ dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
+ w->name, ret);
+
+ return ERR_PTR(ret);
+}
+
+/**
+ * snd_soc_dapm_new_control - create new dapm control
+ * @dapm: DAPM context
+ * @widget: widget template
+ *
+ * Creates new DAPM control based upon a template.
+ *
+ * Returns a widget pointer on success or an error pointer on failure
+ */
+struct snd_soc_dapm_widget *
+snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_widget *widget)
+{
+ struct snd_soc_dapm_widget *w;
+
+ mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+ w = snd_soc_dapm_new_control_unlocked(dapm, widget);
+ mutex_unlock(&dapm->card->dapm_mutex);
+
+ return w;
}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
/**
* snd_soc_dapm_new_controls - create new dapm controls
@@ -3625,19 +3602,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
w = snd_soc_dapm_new_control_unlocked(dapm, widget);
if (IS_ERR(w)) {
ret = PTR_ERR(w);
- /* Do not nag about probe deferrals */
- if (ret == -EPROBE_DEFER)
- break;
- dev_err(dapm->dev,
- "ASoC: Failed to create DAPM control %s (%d)\n",
- widget->name, ret);
- break;
- }
- if (!w) {
- dev_err(dapm->dev,
- "ASoC: Failed to create DAPM control %s\n",
- widget->name);
- ret = -ENOMEM;
break;
}
widget++;
@@ -3650,32 +3614,23 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);
static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_dapm_path *source_p, *sink_p;
+ struct snd_soc_dapm_path *path;
struct snd_soc_dai *source, *sink;
struct snd_soc_pcm_runtime *rtd = w->priv;
- const struct snd_soc_pcm_stream *config = w->params + w->params_select;
+ const struct snd_soc_pcm_stream *config;
struct snd_pcm_substream substream;
struct snd_pcm_hw_params *params = NULL;
struct snd_pcm_runtime *runtime = NULL;
unsigned int fmt;
- int ret;
+ int ret = 0;
+
+ config = rtd->dai_link->params + rtd->params_select;
if (WARN_ON(!config) ||
WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) ||
list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])))
return -EINVAL;
- /* We only support a single source and sink, pick the first */
- source_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_OUT],
- struct snd_soc_dapm_path,
- list_node[SND_SOC_DAPM_DIR_OUT]);
- sink_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_IN],
- struct snd_soc_dapm_path,
- list_node[SND_SOC_DAPM_DIR_IN]);
-
- source = source_p->source->priv;
- sink = sink_p->sink->priv;
-
/* Be a little careful as we don't want to overflow the mask array */
if (config->formats) {
fmt = ffs(config->formats) - 1;
@@ -3717,59 +3672,95 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
substream.stream = SNDRV_PCM_STREAM_CAPTURE;
- if (source->driver->ops->startup) {
- ret = source->driver->ops->startup(&substream, source);
- if (ret < 0) {
- dev_err(source->dev,
- "ASoC: startup() failed: %d\n", ret);
- goto out;
+ snd_soc_dapm_widget_for_each_source_path(w, path) {
+ source = path->source->priv;
+
+ if (source->driver->ops->startup) {
+ ret = source->driver->ops->startup(&substream,
+ source);
+ if (ret < 0) {
+ dev_err(source->dev,
+ "ASoC: startup() failed: %d\n",
+ ret);
+ goto out;
+ }
+ source->active++;
}
- source->active++;
+ ret = soc_dai_hw_params(&substream, params, source);
+ if (ret < 0)
+ goto out;
}
- ret = soc_dai_hw_params(&substream, params, source);
- if (ret < 0)
- goto out;
substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
- if (sink->driver->ops->startup) {
- ret = sink->driver->ops->startup(&substream, sink);
- if (ret < 0) {
- dev_err(sink->dev,
- "ASoC: startup() failed: %d\n", ret);
- goto out;
+ snd_soc_dapm_widget_for_each_sink_path(w, path) {
+ sink = path->sink->priv;
+
+ if (sink->driver->ops->startup) {
+ ret = sink->driver->ops->startup(&substream,
+ sink);
+ if (ret < 0) {
+ dev_err(sink->dev,
+ "ASoC: startup() failed: %d\n",
+ ret);
+ goto out;
+ }
+ sink->active++;
}
- sink->active++;
+ ret = soc_dai_hw_params(&substream, params, sink);
+ if (ret < 0)
+ goto out;
}
- ret = soc_dai_hw_params(&substream, params, sink);
- if (ret < 0)
- goto out;
break;
case SND_SOC_DAPM_POST_PMU:
- ret = snd_soc_dai_digital_mute(sink, 0,
- SNDRV_PCM_STREAM_PLAYBACK);
- if (ret != 0 && ret != -ENOTSUPP)
- dev_warn(sink->dev, "ASoC: Failed to unmute: %d\n", ret);
- ret = 0;
+ snd_soc_dapm_widget_for_each_sink_path(w, path) {
+ sink = path->sink->priv;
+
+ ret = snd_soc_dai_digital_mute(sink, 0,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret != 0 && ret != -ENOTSUPP)
+ dev_warn(sink->dev,
+ "ASoC: Failed to unmute: %d\n", ret);
+ ret = 0;
+ }
break;
case SND_SOC_DAPM_PRE_PMD:
- ret = snd_soc_dai_digital_mute(sink, 1,
- SNDRV_PCM_STREAM_PLAYBACK);
- if (ret != 0 && ret != -ENOTSUPP)
- dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret);
- ret = 0;
+ snd_soc_dapm_widget_for_each_sink_path(w, path) {
+ sink = path->sink->priv;
+
+ ret = snd_soc_dai_digital_mute(sink, 1,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret != 0 && ret != -ENOTSUPP)
+ dev_warn(sink->dev,
+ "ASoC: Failed to mute: %d\n", ret);
+ ret = 0;
+ }
- source->active--;
- if (source->driver->ops->shutdown) {
- substream.stream = SNDRV_PCM_STREAM_CAPTURE;
- source->driver->ops->shutdown(&substream, source);
+ substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+ snd_soc_dapm_widget_for_each_source_path(w, path) {
+ source = path->source->priv;
+
+ if (source->driver->ops->hw_free)
+ source->driver->ops->hw_free(&substream,
+ source);
+
+ source->active--;
+ if (source->driver->ops->shutdown)
+ source->driver->ops->shutdown(&substream,
+ source);
}
- sink->active--;
- if (sink->driver->ops->shutdown) {
- substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
- sink->driver->ops->shutdown(&substream, sink);
+ substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+ snd_soc_dapm_widget_for_each_sink_path(w, path) {
+ sink = path->sink->priv;
+
+ if (sink->driver->ops->hw_free)
+ sink->driver->ops->hw_free(&substream, sink);
+
+ sink->active--;
+ if (sink->driver->ops->shutdown)
+ sink->driver->ops->shutdown(&substream, sink);
}
break;
@@ -3788,8 +3779,9 @@ static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_pcm_runtime *rtd = w->priv;
- ucontrol->value.enumerated.item[0] = w->params_select;
+ ucontrol->value.enumerated.item[0] = rtd->params_select;
return 0;
}
@@ -3798,18 +3790,19 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_pcm_runtime *rtd = w->priv;
/* Can't change the config when widget is already powered */
if (w->power)
return -EBUSY;
- if (ucontrol->value.enumerated.item[0] == w->params_select)
+ if (ucontrol->value.enumerated.item[0] == rtd->params_select)
return 0;
- if (ucontrol->value.enumerated.item[0] >= w->num_params)
+ if (ucontrol->value.enumerated.item[0] >= rtd->dai_link->num_params)
return -EINVAL;
- w->params_select = ucontrol->value.enumerated.item[0];
+ rtd->params_select = ucontrol->value.enumerated.item[0];
return 0;
}
@@ -3896,12 +3889,10 @@ outfree_w_param:
return NULL;
}
-int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
- struct snd_soc_pcm_runtime *rtd,
- const struct snd_soc_pcm_stream *params,
- unsigned int num_params,
- struct snd_soc_dapm_widget *source,
- struct snd_soc_dapm_widget *sink)
+static struct snd_soc_dapm_widget *
+snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
+ struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
{
struct snd_soc_dapm_widget template;
struct snd_soc_dapm_widget *w;
@@ -3913,7 +3904,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s",
source->name, sink->name);
if (!link_name)
- return -ENOMEM;
+ return ERR_PTR(-ENOMEM);
memset(&template, 0, sizeof(template));
template.reg = SND_SOC_NOPM;
@@ -3925,9 +3916,10 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
template.kcontrol_news = NULL;
/* allocate memory for control, only in case of multiple configs */
- if (num_params > 1) {
- w_param_text = devm_kcalloc(card->dev, num_params,
- sizeof(char *), GFP_KERNEL);
+ if (rtd->dai_link->num_params > 1) {
+ w_param_text = devm_kcalloc(card->dev,
+ rtd->dai_link->num_params,
+ sizeof(char *), GFP_KERNEL);
if (!w_param_text) {
ret = -ENOMEM;
goto param_fail;
@@ -3936,7 +3928,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
template.num_kcontrols = 1;
template.kcontrol_news =
snd_soc_dapm_alloc_kcontrol(card,
- link_name, params, num_params,
+ link_name,
+ rtd->dai_link->params,
+ rtd->dai_link->num_params,
w_param_text, &private_value);
if (!template.kcontrol_news) {
ret = -ENOMEM;
@@ -3950,37 +3944,20 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template);
if (IS_ERR(w)) {
ret = PTR_ERR(w);
- /* Do not nag about probe deferrals */
- if (ret != -EPROBE_DEFER)
- dev_err(card->dev,
- "ASoC: Failed to create %s widget (%d)\n",
- link_name, ret);
- goto outfree_kcontrol_news;
- }
- if (!w) {
- dev_err(card->dev, "ASoC: Failed to create %s widget\n",
- link_name);
- ret = -ENOMEM;
goto outfree_kcontrol_news;
}
- w->params = params;
- w->num_params = num_params;
w->priv = rtd;
- ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL);
- if (ret)
- goto outfree_w;
- return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL);
+ return w;
-outfree_w:
- devm_kfree(card->dev, w);
outfree_kcontrol_news:
devm_kfree(card->dev, (void *)template.kcontrol_news);
- snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text);
+ snd_soc_dapm_free_kcontrol(card, &private_value,
+ rtd->dai_link->num_params, w_param_text);
param_fail:
devm_kfree(card->dev, link_name);
- return ret;
+ return ERR_PTR(ret);
}
int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
@@ -4003,21 +3980,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
template.name);
w = snd_soc_dapm_new_control_unlocked(dapm, &template);
- if (IS_ERR(w)) {
- int ret = PTR_ERR(w);
-
- /* Do not nag about probe deferrals */
- if (ret != -EPROBE_DEFER)
- dev_err(dapm->dev,
- "ASoC: Failed to create %s widget (%d)\n",
- dai->driver->playback.stream_name, ret);
- return ret;
- }
- if (!w) {
- dev_err(dapm->dev, "ASoC: Failed to create %s widget\n",
- dai->driver->playback.stream_name);
- return -ENOMEM;
- }
+ if (IS_ERR(w))
+ return PTR_ERR(w);
w->priv = dai;
dai->playback_widget = w;
@@ -4032,21 +3996,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
template.name);
w = snd_soc_dapm_new_control_unlocked(dapm, &template);
- if (IS_ERR(w)) {
- int ret = PTR_ERR(w);
-
- /* Do not nag about probe deferrals */
- if (ret != -EPROBE_DEFER)
- dev_err(dapm->dev,
- "ASoC: Failed to create %s widget (%d)\n",
- dai->driver->playback.stream_name, ret);
- return ret;
- }
- if (!w) {
- dev_err(dapm->dev, "ASoC: Failed to create %s widget\n",
- dai->driver->capture.stream_name);
- return -ENOMEM;
- }
+ if (IS_ERR(w))
+ return PTR_ERR(w);
w->priv = dai;
dai->capture_widget = w;
@@ -4115,34 +4066,79 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dapm_widget *sink, *source;
+ struct snd_soc_dai *codec_dai;
+ struct snd_soc_dapm_widget *playback = NULL, *capture = NULL;
+ struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu;
int i;
- for (i = 0; i < rtd->num_codecs; i++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+ if (rtd->dai_link->params) {
+ playback_cpu = cpu_dai->capture_widget;
+ capture_cpu = cpu_dai->playback_widget;
+ } else {
+ playback = cpu_dai->playback_widget;
+ capture = cpu_dai->capture_widget;
+ playback_cpu = playback;
+ capture_cpu = capture;
+ }
+
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
/* connect BE DAI playback if widgets are valid */
- if (codec_dai->playback_widget && cpu_dai->playback_widget) {
- source = cpu_dai->playback_widget;
- sink = codec_dai->playback_widget;
+ codec = codec_dai->playback_widget;
+
+ if (playback_cpu && codec) {
+ if (!playback) {
+ playback = snd_soc_dapm_new_dai(card, rtd,
+ playback_cpu,
+ codec);
+ if (IS_ERR(playback)) {
+ dev_err(rtd->dev,
+ "ASoC: Failed to create DAI %s: %ld\n",
+ codec_dai->name,
+ PTR_ERR(playback));
+ continue;
+ }
+
+ snd_soc_dapm_add_path(&card->dapm, playback_cpu,
+ playback, NULL, NULL);
+ }
+
dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
- cpu_dai->component->name, source->name,
- codec_dai->component->name, sink->name);
+ cpu_dai->component->name, playback_cpu->name,
+ codec_dai->component->name, codec->name);
- snd_soc_dapm_add_path(&card->dapm, source, sink,
- NULL, NULL);
+ snd_soc_dapm_add_path(&card->dapm, playback, codec,
+ NULL, NULL);
}
+ }
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
/* connect BE DAI capture if widgets are valid */
- if (codec_dai->capture_widget && cpu_dai->capture_widget) {
- source = codec_dai->capture_widget;
- sink = cpu_dai->capture_widget;
+ codec = codec_dai->capture_widget;
+
+ if (codec && capture_cpu) {
+ if (!capture) {
+ capture = snd_soc_dapm_new_dai(card, rtd,
+ codec,
+ capture_cpu);
+ if (IS_ERR(capture)) {
+ dev_err(rtd->dev,
+ "ASoC: Failed to create DAI %s: %ld\n",
+ codec_dai->name,
+ PTR_ERR(capture));
+ continue;
+ }
+
+ snd_soc_dapm_add_path(&card->dapm, capture,
+ capture_cpu, NULL, NULL);
+ }
+
dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
- codec_dai->component->name, source->name,
- cpu_dai->component->name, sink->name);
+ codec_dai->component->name, codec->name,
+ cpu_dai->component->name, capture_cpu->name);
- snd_soc_dapm_add_path(&card->dapm, source, sink,
- NULL, NULL);
+ snd_soc_dapm_add_path(&card->dapm, codec, capture,
+ NULL, NULL);
}
}
}
@@ -4192,12 +4188,12 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
struct snd_soc_pcm_runtime *rtd;
/* for each BE DAI link... */
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
/*
* dynamic FE links have no fixed DAI mapping.
* CODEC<->CODEC links have no direct connection.
*/
- if (rtd->dai_link->dynamic || rtd->dai_link->params)
+ if (rtd->dai_link->dynamic)
continue;
dapm_connect_dai_link_widgets(card, rtd);
@@ -4207,11 +4203,12 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
int event)
{
+ struct snd_soc_dai *codec_dai;
int i;
soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event);
- for (i = 0; i < rtd->num_codecs; i++)
- soc_dapm_dai_stream_event(rtd->codec_dais[i], stream, event);
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ soc_dapm_dai_stream_event(codec_dai, stream, event);
dapm_power_widgets(rtd->card, event);
}
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 592efb370c44..f4dc3d445aae 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -373,7 +373,7 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol,
unsigned int rshift = mc->rshift;
int max = mc->max;
int min = mc->min;
- unsigned int mask = (1 << (fls(min + max) - 1)) - 1;
+ unsigned int mask = (1U << (fls(min + max) - 1)) - 1;
unsigned int val;
int ret;
@@ -418,7 +418,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol,
unsigned int rshift = mc->rshift;
int max = mc->max;
int min = mc->min;
- unsigned int mask = (1 << (fls(min + max) - 1)) - 1;
+ unsigned int mask = (1U << (fls(min + max) - 1)) - 1;
int err = 0;
unsigned int val, val_mask, val2 = 0;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e8b98bfd4cf1..03f36e534050 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -59,25 +59,26 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream)
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;
int i;
lockdep_assert_held(&rtd->pcm_mutex);
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
cpu_dai->playback_active++;
- for (i = 0; i < rtd->num_codecs; i++)
- rtd->codec_dais[i]->playback_active++;
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ codec_dai->playback_active++;
} else {
cpu_dai->capture_active++;
- for (i = 0; i < rtd->num_codecs; i++)
- rtd->codec_dais[i]->capture_active++;
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ codec_dai->capture_active++;
}
cpu_dai->active++;
cpu_dai->component->active++;
- for (i = 0; i < rtd->num_codecs; i++) {
- rtd->codec_dais[i]->active++;
- rtd->codec_dais[i]->component->active++;
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ codec_dai->active++;
+ codec_dai->component->active++;
}
}
@@ -94,25 +95,26 @@ 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;
int i;
lockdep_assert_held(&rtd->pcm_mutex);
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
cpu_dai->playback_active--;
- for (i = 0; i < rtd->num_codecs; i++)
- rtd->codec_dais[i]->playback_active--;
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ codec_dai->playback_active--;
} else {
cpu_dai->capture_active--;
- for (i = 0; i < rtd->num_codecs; i++)
- rtd->codec_dais[i]->capture_active--;
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ codec_dai->capture_active--;
}
cpu_dai->active--;
cpu_dai->component->active--;
- for (i = 0; i < rtd->num_codecs; i++) {
- rtd->codec_dais[i]->component->active--;
- rtd->codec_dais[i]->active--;
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ codec_dai->component->active--;
+ codec_dai->active--;
}
}
@@ -172,7 +174,7 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir,
{
struct snd_soc_dpcm *dpcm;
- list_for_each_entry(dpcm, &fe->dpcm[dir].be_clients, list_be) {
+ for_each_dpcm_be(fe, dir, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
@@ -253,6 +255,7 @@ 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;
unsigned int rate, channels, sample_bits, symmetry, i;
rate = params_rate(params);
@@ -263,8 +266,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
symmetry = cpu_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;
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ symmetry |= codec_dai->driver->symmetric_rates;
if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) {
dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n",
@@ -275,8 +278,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
symmetry = cpu_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;
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ symmetry |= codec_dai->driver->symmetric_channels;
if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) {
dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n",
@@ -287,8 +290,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
symmetry = cpu_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;
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ symmetry |= codec_dai->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",
@@ -304,17 +307,18 @@ 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_link *link = rtd->dai_link;
+ struct snd_soc_dai *codec_dai;
unsigned int symmetry, i;
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++)
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
symmetry = symmetry ||
- rtd->codec_dais[i]->driver->symmetric_rates ||
- rtd->codec_dais[i]->driver->symmetric_channels ||
- rtd->codec_dais[i]->driver->symmetric_samplebits;
+ codec_dai->driver->symmetric_rates ||
+ codec_dai->driver->symmetric_channels ||
+ codec_dai->driver->symmetric_samplebits;
return symmetry;
}
@@ -342,8 +346,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream)
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];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
if (codec_dai->driver->playback.sig_bits == 0) {
bits = 0;
break;
@@ -352,8 +355,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream)
}
cpu_bits = cpu_dai->driver->playback.sig_bits;
} else {
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
if (codec_dai->driver->capture.sig_bits == 0) {
bits = 0;
break;
@@ -372,6 +374,7 @@ 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 *codec_dai;
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;
@@ -388,7 +391,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
cpu_stream = &cpu_dai_drv->capture;
/* first calculate min/max only for CODECs in the DAI link */
- for (i = 0; i < rtd->num_codecs; i++) {
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
/*
* Skip CODECs which don't support the current stream type.
@@ -399,11 +402,11 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
* bailed out on a higher level, since there would be no
* CODEC to support the transfer direction in that case.
*/
- if (!snd_soc_dai_stream_valid(rtd->codec_dais[i],
+ if (!snd_soc_dai_stream_valid(codec_dai,
substream->stream))
continue;
- codec_dai_drv = rtd->codec_dais[i]->driver;
+ codec_dai_drv = codec_dai->driver;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
codec_stream = &codec_dai_drv->playback;
else
@@ -482,8 +485,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
int i, ret = 0;
pinctrl_pm_select_default_state(cpu_dai->dev);
- for (i = 0; i < rtd->num_codecs; i++)
- pinctrl_pm_select_default_state(rtd->codec_dais[i]->dev);
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ pinctrl_pm_select_default_state(codec_dai->dev);
for_each_rtdcom(rtd, rtdcom) {
component = rtdcom->component;
@@ -520,8 +523,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
}
component = NULL;
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
if (codec_dai->driver->ops->startup) {
ret = codec_dai->driver->ops->startup(substream,
codec_dai);
@@ -588,10 +590,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
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]);
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ if (codec_dai->active) {
+ ret = soc_pcm_apply_symmetry(substream, codec_dai);
if (ret != 0)
goto config_err;
}
@@ -620,8 +621,7 @@ machine_err:
i = rtd->num_codecs;
codec_dai_err:
- while (--i >= 0) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) {
if (codec_dai->driver->ops->shutdown)
codec_dai->driver->ops->shutdown(substream, codec_dai);
}
@@ -641,9 +641,9 @@ out:
pm_runtime_put_autosuspend(component->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);
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ if (!codec_dai->active)
+ pinctrl_pm_select_sleep_state(codec_dai->dev);
}
if (!cpu_dai->active)
pinctrl_pm_select_sleep_state(cpu_dai->dev);
@@ -701,8 +701,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
if (!cpu_dai->active)
cpu_dai->rate = 0;
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
if (!codec_dai->active)
codec_dai->rate = 0;
}
@@ -712,8 +711,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
if (cpu_dai->driver->ops->shutdown)
cpu_dai->driver->ops->shutdown(substream, cpu_dai);
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
if (codec_dai->driver->ops->shutdown)
codec_dai->driver->ops->shutdown(substream, codec_dai);
}
@@ -751,9 +749,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
pm_runtime_put_autosuspend(component->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);
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ if (!codec_dai->active)
+ pinctrl_pm_select_sleep_state(codec_dai->dev);
}
if (!cpu_dai->active)
pinctrl_pm_select_sleep_state(cpu_dai->dev);
@@ -801,8 +799,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
}
}
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
if (codec_dai->driver->ops->prepare) {
ret = codec_dai->driver->ops->prepare(substream,
codec_dai);
@@ -834,8 +831,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
snd_soc_dapm_stream_event(rtd, substream->stream,
SND_SOC_DAPM_STREAM_START);
- for (i = 0; i < rtd->num_codecs; i++)
- snd_soc_dai_digital_mute(rtd->codec_dais[i], 0,
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ snd_soc_dai_digital_mute(codec_dai, 0,
substream->stream);
snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream);
@@ -920,6 +917,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_component *component;
struct snd_soc_rtdcom_list *rtdcom;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai;
int i, ret = 0;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
@@ -932,8 +930,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
}
}
- for (i = 0; i < rtd->num_codecs; i++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
struct snd_pcm_hw_params codec_params;
/*
@@ -1018,8 +1015,7 @@ interface_err:
i = rtd->num_codecs;
codec_err:
- while (--i >= 0) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) {
if (codec_dai->driver->ops->hw_free)
codec_dai->driver->ops->hw_free(substream, codec_dai);
codec_dai->rate = 0;
@@ -1052,8 +1048,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
cpu_dai->sample_bits = 0;
}
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
if (codec_dai->active == 1) {
codec_dai->rate = 0;
codec_dai->channels = 0;
@@ -1062,10 +1057,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
}
/* apply codec digital mute */
- 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,
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ if ((playback && codec_dai->playback_active == 1) ||
+ (!playback && codec_dai->capture_active == 1))
+ snd_soc_dai_digital_mute(codec_dai, 1,
substream->stream);
}
@@ -1077,8 +1072,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
soc_pcm_components_hw_free(substream, NULL);
/* now free hw params for the DAIs */
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
if (codec_dai->driver->ops->hw_free)
codec_dai->driver->ops->hw_free(substream, codec_dai);
}
@@ -1099,8 +1093,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
struct snd_soc_dai *codec_dai;
int i, ret;
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
if (codec_dai->driver->ops->trigger) {
ret = codec_dai->driver->ops->trigger(substream,
cmd, codec_dai);
@@ -1144,8 +1137,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai;
int i, ret;
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
if (codec_dai->driver->ops->bespoke_trigger) {
ret = codec_dai->driver->ops->bespoke_trigger(substream,
cmd, codec_dai);
@@ -1199,8 +1191,7 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
if (cpu_dai->driver->ops->delay)
delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
if (codec_dai->driver->ops->delay)
codec_delay = max(codec_delay,
codec_dai->driver->ops->delay(substream,
@@ -1220,7 +1211,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe,
struct snd_soc_dpcm *dpcm;
/* only add new dpcms */
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
if (dpcm->be == be && dpcm->fe == fe)
return 0;
}
@@ -1261,7 +1252,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe,
be_substream = snd_soc_dpcm_get_substream(be, stream);
- list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+ for_each_dpcm_fe(be, stream, dpcm) {
if (dpcm->fe == fe)
continue;
@@ -1281,7 +1272,7 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream)
{
struct snd_soc_dpcm *dpcm, *d;
- list_for_each_entry_safe(dpcm, d, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be_safe(fe, stream, dpcm, d) {
dev_dbg(fe->dev, "ASoC: BE %s disconnect check for %s\n",
stream ? "capture" : "playback",
dpcm->be->dai_link->name);
@@ -1310,12 +1301,13 @@ 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;
+ struct snd_soc_dai *dai;
int i;
dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name);
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- list_for_each_entry(be, &card->rtd_list, list) {
+ for_each_card_rtds(card, be) {
if (!be->dai_link->no_pcm)
continue;
@@ -1327,15 +1319,14 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
if (be->cpu_dai->playback_widget == widget)
return be;
- for (i = 0; i < be->num_codecs; i++) {
- struct snd_soc_dai *dai = be->codec_dais[i];
+ for_each_rtd_codec_dai(be, i, dai) {
if (dai->playback_widget == widget)
return be;
}
}
} else {
- list_for_each_entry(be, &card->rtd_list, list) {
+ for_each_card_rtds(card, be) {
if (!be->dai_link->no_pcm)
continue;
@@ -1347,8 +1338,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
if (be->cpu_dai->capture_widget == widget)
return be;
- for (i = 0; i < be->num_codecs; i++) {
- struct snd_soc_dai *dai = be->codec_dais[i];
+ for_each_rtd_codec_dai(be, i, dai) {
if (dai->capture_widget == widget)
return be;
}
@@ -1388,32 +1378,31 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget,
{
struct snd_soc_card *card = widget->dapm->card;
struct snd_soc_pcm_runtime *rtd;
+ struct snd_soc_dai *dai;
int i;
if (dir == SND_SOC_DAPM_DIR_OUT) {
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
if (!rtd->dai_link->no_pcm)
continue;
if (rtd->cpu_dai->playback_widget == widget)
return true;
- for (i = 0; i < rtd->num_codecs; ++i) {
- struct snd_soc_dai *dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, dai) {
if (dai->playback_widget == widget)
return true;
}
}
} else { /* SND_SOC_DAPM_DIR_IN */
- list_for_each_entry(rtd, &card->rtd_list, list) {
+ for_each_card_rtds(card, rtd) {
if (!rtd->dai_link->no_pcm)
continue;
if (rtd->cpu_dai->capture_widget == widget)
return true;
- for (i = 0; i < rtd->num_codecs; ++i) {
- struct snd_soc_dai *dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, dai) {
if (dai->capture_widget == widget)
return true;
}
@@ -1445,10 +1434,11 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
struct snd_soc_dpcm *dpcm;
struct snd_soc_dapm_widget_list *list = *list_;
struct snd_soc_dapm_widget *widget;
+ struct snd_soc_dai *dai;
int prune = 0;
/* Destroy any old FE <--> BE connections */
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
unsigned int i;
/* is there a valid CPU DAI widget for this BE */
@@ -1459,8 +1449,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
continue;
/* is there a valid CODEC DAI widget for this BE */
- for (i = 0; i < dpcm->be->num_codecs; i++) {
- struct snd_soc_dai *dai = dpcm->be->codec_dais[i];
+ for_each_rtd_codec_dai(dpcm->be, i, dai) {
widget = dai_get_widget(dai, stream);
/* prune the BE if it's no longer in our active list */
@@ -1555,7 +1544,7 @@ void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream)
{
struct snd_soc_dpcm *dpcm;
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+ for_each_dpcm_be(fe, stream, dpcm)
dpcm->be->dpcm[stream].runtime_update =
SND_SOC_DPCM_UPDATE_NO;
}
@@ -1566,7 +1555,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe,
struct snd_soc_dpcm *dpcm;
/* disable any enabled and non active backends */
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_pcm_substream *be_substream =
@@ -1595,7 +1584,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream)
int err, count = 0;
/* only startup BE DAIs that are either sinks or sources to this FE DAI */
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_pcm_substream *be_substream =
@@ -1649,7 +1638,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream)
unwind:
/* disable any enabled and non active backends */
- list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be_rollback(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_pcm_substream *be_substream =
snd_soc_dpcm_get_substream(be, stream);
@@ -1680,7 +1669,7 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
struct snd_soc_pcm_stream *stream)
{
runtime->hw.rate_min = stream->rate_min;
- runtime->hw.rate_max = stream->rate_max;
+ runtime->hw.rate_max = min_not_zero(stream->rate_max, UINT_MAX);
runtime->hw.channels_min = stream->channels_min;
runtime->hw.channels_max = stream->channels_max;
if (runtime->hw.formats)
@@ -1695,6 +1684,7 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *fe = substream->private_data;
struct snd_soc_dpcm *dpcm;
+ struct snd_soc_dai *dai;
int stream = substream->stream;
if (!fe->dai_link->dpcm_merged_format)
@@ -1705,22 +1695,21 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream,
* if FE want to use it (= dpcm_merged_format)
*/
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_soc_dai_driver *codec_dai_drv;
struct snd_soc_pcm_stream *codec_stream;
int i;
- for (i = 0; i < be->num_codecs; i++) {
+ for_each_rtd_codec_dai(be, i, dai) {
/*
* Skip CODECs which don't support the current stream
* type. See soc_pcm_init_runtime_hw() for more details
*/
- if (!snd_soc_dai_stream_valid(be->codec_dais[i],
- stream))
+ if (!snd_soc_dai_stream_valid(dai, stream))
continue;
- codec_dai_drv = be->codec_dais[i]->driver;
+ codec_dai_drv = dai->driver;
if (stream == SNDRV_PCM_STREAM_PLAYBACK)
codec_stream = &codec_dai_drv->playback;
else
@@ -1747,7 +1736,7 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream,
* if FE want to use it (= dpcm_merged_chan)
*/
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver;
struct snd_soc_dai_driver *codec_dai_drv;
@@ -1799,12 +1788,13 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream,
* if FE want to use it (= dpcm_merged_chan)
*/
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_soc_dai_driver *cpu_dai_drv = be->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;
+ struct snd_soc_dai *dai;
int i;
if (stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -1816,16 +1806,15 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream,
*rate_max = min_not_zero(*rate_max, cpu_stream->rate_max);
*rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates);
- for (i = 0; i < be->num_codecs; i++) {
+ for_each_rtd_codec_dai(be, i, dai) {
/*
* Skip CODECs which don't support the current stream
* type. See soc_pcm_init_runtime_hw() for more details
*/
- if (!snd_soc_dai_stream_valid(be->codec_dais[i],
- stream))
+ if (!snd_soc_dai_stream_valid(dai, stream))
continue;
- codec_dai_drv = be->codec_dais[i]->driver;
+ codec_dai_drv = dai->driver;
if (stream == SNDRV_PCM_STREAM_PLAYBACK)
codec_stream = &codec_dai_drv->playback;
else
@@ -1902,11 +1891,12 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream,
}
/* apply symmetry for BE */
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_pcm_substream *be_substream =
snd_soc_dpcm_get_substream(be, stream);
struct snd_soc_pcm_runtime *rtd = be_substream->private_data;
+ struct snd_soc_dai *codec_dai;
int i;
if (rtd->dai_link->be_hw_params_fixup)
@@ -1923,10 +1913,10 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream,
return err;
}
- for (i = 0; i < rtd->num_codecs; i++) {
- if (rtd->codec_dais[i]->active) {
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ if (codec_dai->active) {
err = soc_pcm_apply_symmetry(fe_substream,
- rtd->codec_dais[i]);
+ codec_dai);
if (err < 0)
return err;
}
@@ -1986,7 +1976,7 @@ int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream)
struct snd_soc_dpcm *dpcm;
/* only shutdown BEs that are either sinks or sources to this FE DAI */
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_pcm_substream *be_substream =
@@ -2050,7 +2040,7 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
/* only hw_params backends that are either sinks or sources
* to this frontend DAI */
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_pcm_substream *be_substream =
@@ -2119,7 +2109,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream)
struct snd_soc_dpcm *dpcm;
int ret;
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_pcm_substream *be_substream =
@@ -2170,7 +2160,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream)
unwind:
/* disable any enabled and non active backends */
- list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be_rollback(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_pcm_substream *be_substream =
snd_soc_dpcm_get_substream(be, stream);
@@ -2250,7 +2240,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
struct snd_soc_dpcm *dpcm;
int ret = 0;
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_pcm_substream *be_substream =
@@ -2436,7 +2426,7 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream)
struct snd_soc_dpcm *dpcm;
int ret = 0;
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_pcm_substream *be_substream =
@@ -2646,7 +2636,7 @@ close:
dpcm_be_dai_shutdown(fe, stream);
disconnect:
/* disconnect any non started BEs */
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
@@ -2771,14 +2761,14 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
/* shutdown all old paths first */
- list_for_each_entry(fe, &card->rtd_list, list) {
+ for_each_card_rtds(card, fe) {
ret = soc_dpcm_fe_runtime_update(fe, 0);
if (ret)
goto out;
}
/* bring new paths up */
- list_for_each_entry(fe, &card->rtd_list, list) {
+ for_each_card_rtds(card, fe) {
ret = soc_dpcm_fe_runtime_update(fe, 1);
if (ret)
goto out;
@@ -2791,10 +2781,9 @@ out:
int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
{
struct snd_soc_dpcm *dpcm;
- struct list_head *clients =
- &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients;
+ struct snd_soc_dai *dai;
- list_for_each_entry(dpcm, clients, list_be) {
+ for_each_dpcm_be(fe, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
int i;
@@ -2802,8 +2791,7 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
if (be->dai_link->ignore_suspend)
continue;
- for (i = 0; i < be->num_codecs; i++) {
- struct snd_soc_dai *dai = be->codec_dais[i];
+ for_each_rtd_codec_dai(be, i, dai) {
struct snd_soc_dai_driver *drv = dai->driver;
dev_dbg(be->dev, "ASoC: BE digital mute %s\n",
@@ -2844,7 +2832,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
ret = dpcm_fe_dai_startup(fe_substream);
if (ret < 0) {
/* clean up all links */
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+ for_each_dpcm_be(fe, stream, dpcm)
dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
dpcm_be_disconnect(fe, stream);
@@ -2867,7 +2855,7 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
ret = dpcm_fe_dai_shutdown(fe_substream);
/* mark FE's links ready to prune */
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+ for_each_dpcm_be(fe, stream, dpcm)
dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
dpcm_be_disconnect(fe, stream);
@@ -3041,8 +3029,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
playback = rtd->dai_link->dpcm_playback;
capture = rtd->dai_link->dpcm_capture;
} else {
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
if (codec_dai->driver->playback.channels_min)
playback = 1;
if (codec_dai->driver->capture.channels_min)
@@ -3230,7 +3217,7 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
struct snd_soc_dpcm *dpcm;
int state;
- list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+ for_each_dpcm_fe(be, stream, dpcm) {
if (dpcm->fe == fe)
continue;
@@ -3257,7 +3244,7 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
struct snd_soc_dpcm *dpcm;
int state;
- list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+ for_each_dpcm_fe(be, stream, dpcm) {
if (dpcm->fe == fe)
continue;
@@ -3337,7 +3324,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe,
goto out;
}
- list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
params = &dpcm->hw_params;
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 66e77e020745..045ef136903d 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -993,7 +993,7 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count,
kfree(se);
continue;
}
- /* fall through and create texts */
+ /* fall through */
case SND_SOC_TPLG_CTL_ENUM:
case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
@@ -1310,7 +1310,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create(
ec->hdr.name);
goto err_se;
}
- /* fall through to create texts */
+ /* fall through */
case SND_SOC_TPLG_CTL_ENUM:
case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
@@ -1565,17 +1565,6 @@ widget:
widget = snd_soc_dapm_new_control_unlocked(dapm, &template);
if (IS_ERR(widget)) {
ret = PTR_ERR(widget);
- /* Do not nag about probe deferrals */
- if (ret != -EPROBE_DEFER)
- dev_err(tplg->dev,
- "ASoC: failed to create widget %s controls (%d)\n",
- w->name, ret);
- goto hdr_err;
- }
- if (widget == NULL) {
- dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n",
- w->name);
- ret = -ENOMEM;
goto hdr_err;
}
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index e0c93496c0cd..e3b9dd634c6d 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -273,13 +273,13 @@ static int dummy_dma_open(struct snd_pcm_substream *substream)
return 0;
}
-static const struct snd_pcm_ops dummy_dma_ops = {
+static const struct snd_pcm_ops snd_dummy_dma_ops = {
.open = dummy_dma_open,
.ioctl = snd_pcm_lib_ioctl,
};
static const struct snd_soc_component_driver dummy_platform = {
- .ops = &dummy_dma_ops,
+ .ops = &snd_dummy_dma_ops,
};
static const struct snd_soc_component_driver dummy_codec = {
diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig
index 9b2681397dba..c66ffa72057e 100644
--- a/sound/soc/stm/Kconfig
+++ b/sound/soc/stm/Kconfig
@@ -3,6 +3,7 @@ menu "STMicroelectronics STM32 SOC audio support"
config SND_SOC_STM32_SAI
tristate "STM32 SAI interface (Serial Audio Interface) support"
depends on (ARCH_STM32 && OF) || COMPILE_TEST
+ depends on COMMON_CLK
depends on SND_SOC
select SND_SOC_GENERIC_DMAENGINE_PCM
select REGMAP_MMIO
diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c
index f22654253c43..d597eba61992 100644
--- a/sound/soc/stm/stm32_sai.c
+++ b/sound/soc/stm/stm32_sai.c
@@ -104,7 +104,7 @@ static int stm32_sai_set_sync(struct stm32_sai_data *sai_client,
if (!pdev) {
dev_err(&sai_client->pdev->dev,
- "Device not found for node %s\n", np_provider->name);
+ "Device not found for node %pOFn\n", np_provider);
return -ENODEV;
}
diff --git a/sound/soc/stm/stm32_sai.h b/sound/soc/stm/stm32_sai.h
index f25422174909..08de899c766b 100644
--- a/sound/soc/stm/stm32_sai.h
+++ b/sound/soc/stm/stm32_sai.h
@@ -91,6 +91,9 @@
#define SAI_XCR1_OSR_SHIFT 26
#define SAI_XCR1_OSR BIT(SAI_XCR1_OSR_SHIFT)
+#define SAI_XCR1_MCKEN_SHIFT 27
+#define SAI_XCR1_MCKEN BIT(SAI_XCR1_MCKEN_SHIFT)
+
/******************* Bit definition for SAI_XCR2 register *******************/
#define SAI_XCR2_FTH_SHIFT 0
#define SAI_XCR2_FTH_MASK GENMASK(2, SAI_XCR2_FTH_SHIFT)
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index 06fba9650ac4..ea05cc91aa05 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -17,6 +17,7 @@
*/
#include <linux/clk.h>
+#include <linux/clk-provider.h>
#include <linux/kernel.h>
#include <linux/module.h>
#include <linux/of_irq.h>
@@ -68,6 +69,8 @@
#define SAI_IEC60958_BLOCK_FRAMES 192
#define SAI_IEC60958_STATUS_BYTES 24
+#define SAI_MCLK_NAME_LEN 32
+
/**
* struct stm32_sai_sub_data - private data of SAI sub block (block A or B)
* @pdev: device data pointer
@@ -80,6 +83,7 @@
* @pdata: SAI block parent data pointer
* @np_sync_provider: synchronization provider node
* @sai_ck: kernel clock feeding the SAI clock generator
+ * @sai_mclk: master clock from SAI mclk provider
* @phys_addr: SAI registers physical base address
* @mclk_rate: SAI block master clock frequency (Hz). set at init
* @id: SAI sub block id corresponding to sub-block A or B
@@ -110,6 +114,7 @@ struct stm32_sai_sub_data {
struct stm32_sai_data *pdata;
struct device_node *np_sync_provider;
struct clk *sai_ck;
+ struct clk *sai_mclk;
dma_addr_t phys_addr;
unsigned int mclk_rate;
unsigned int id;
@@ -251,6 +256,176 @@ static const struct snd_kcontrol_new iec958_ctls = {
.put = snd_pcm_iec958_put,
};
+struct stm32_sai_mclk_data {
+ struct clk_hw hw;
+ unsigned long freq;
+ struct stm32_sai_sub_data *sai_data;
+};
+
+#define to_mclk_data(_hw) container_of(_hw, struct stm32_sai_mclk_data, hw)
+#define STM32_SAI_MAX_CLKS 1
+
+static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai,
+ unsigned long input_rate,
+ unsigned long output_rate)
+{
+ int version = sai->pdata->conf->version;
+ int div;
+
+ div = DIV_ROUND_CLOSEST(input_rate, output_rate);
+ if (div > SAI_XCR1_MCKDIV_MAX(version)) {
+ dev_err(&sai->pdev->dev, "Divider %d out of range\n", div);
+ return -EINVAL;
+ }
+ dev_dbg(&sai->pdev->dev, "SAI divider %d\n", div);
+
+ if (input_rate % div)
+ dev_dbg(&sai->pdev->dev,
+ "Rate not accurate. requested (%ld), actual (%ld)\n",
+ output_rate, input_rate / div);
+
+ return div;
+}
+
+static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai,
+ unsigned int div)
+{
+ int version = sai->pdata->conf->version;
+ int ret, cr1, mask;
+
+ if (div > SAI_XCR1_MCKDIV_MAX(version)) {
+ dev_err(&sai->pdev->dev, "Divider %d out of range\n", div);
+ return -EINVAL;
+ }
+
+ mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version));
+ cr1 = SAI_XCR1_MCKDIV_SET(div);
+ ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1);
+ if (ret < 0)
+ dev_err(&sai->pdev->dev, "Failed to update CR1 register\n");
+
+ return ret;
+}
+
+static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate,
+ unsigned long *prate)
+{
+ struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+ struct stm32_sai_sub_data *sai = mclk->sai_data;
+ int div;
+
+ div = stm32_sai_get_clk_div(sai, *prate, rate);
+ if (div < 0)
+ return div;
+
+ mclk->freq = *prate / div;
+
+ return mclk->freq;
+}
+
+static unsigned long stm32_sai_mclk_recalc_rate(struct clk_hw *hw,
+ unsigned long parent_rate)
+{
+ struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+
+ return mclk->freq;
+}
+
+static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate,
+ unsigned long parent_rate)
+{
+ struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+ struct stm32_sai_sub_data *sai = mclk->sai_data;
+ unsigned int div;
+ int ret;
+
+ div = stm32_sai_get_clk_div(sai, parent_rate, rate);
+ if (div < 0)
+ return div;
+
+ ret = stm32_sai_set_clk_div(sai, div);
+ if (ret)
+ return ret;
+
+ mclk->freq = rate;
+
+ return 0;
+}
+
+static int stm32_sai_mclk_enable(struct clk_hw *hw)
+{
+ struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+ struct stm32_sai_sub_data *sai = mclk->sai_data;
+
+ dev_dbg(&sai->pdev->dev, "Enable master clock\n");
+
+ return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
+ SAI_XCR1_MCKEN, SAI_XCR1_MCKEN);
+}
+
+static void stm32_sai_mclk_disable(struct clk_hw *hw)
+{
+ struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+ struct stm32_sai_sub_data *sai = mclk->sai_data;
+
+ dev_dbg(&sai->pdev->dev, "Disable master clock\n");
+
+ regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_MCKEN, 0);
+}
+
+static const struct clk_ops mclk_ops = {
+ .enable = stm32_sai_mclk_enable,
+ .disable = stm32_sai_mclk_disable,
+ .recalc_rate = stm32_sai_mclk_recalc_rate,
+ .round_rate = stm32_sai_mclk_round_rate,
+ .set_rate = stm32_sai_mclk_set_rate,
+};
+
+static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai)
+{
+ struct clk_hw *hw;
+ struct stm32_sai_mclk_data *mclk;
+ struct device *dev = &sai->pdev->dev;
+ const char *pname = __clk_get_name(sai->sai_ck);
+ char *mclk_name, *p, *s = (char *)pname;
+ int ret, i = 0;
+
+ mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL);
+ if (!mclk)
+ return -ENOMEM;
+
+ mclk_name = devm_kcalloc(dev, sizeof(char),
+ SAI_MCLK_NAME_LEN, GFP_KERNEL);
+ if (!mclk_name)
+ return -ENOMEM;
+
+ /*
+ * Forge mclk clock name from parent clock name and suffix.
+ * String after "_" char is stripped in parent name.
+ */
+ p = mclk_name;
+ while (*s && *s != '_' && (i < (SAI_MCLK_NAME_LEN - 7))) {
+ *p++ = *s++;
+ i++;
+ }
+ STM_SAI_IS_SUB_A(sai) ? strcat(p, "a_mclk") : strcat(p, "b_mclk");
+
+ mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0);
+ mclk->sai_data = sai;
+ hw = &mclk->hw;
+
+ dev_dbg(dev, "Register master clock %s\n", mclk_name);
+ ret = devm_clk_hw_register(&sai->pdev->dev, hw);
+ if (ret) {
+ dev_err(dev, "mclk register returned %d\n", ret);
+ return ret;
+ }
+ sai->sai_mclk = hw->clk;
+
+ /* register mclk provider */
+ return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw);
+}
+
static irqreturn_t stm32_sai_isr(int irq, void *devid)
{
struct stm32_sai_sub_data *sai = (struct stm32_sai_sub_data *)devid;
@@ -312,15 +487,25 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai,
struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai);
int ret;
- if ((dir == SND_SOC_CLOCK_OUT) && sai->master) {
+ if (dir == SND_SOC_CLOCK_OUT) {
ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
SAI_XCR1_NODIV,
(unsigned int)~SAI_XCR1_NODIV);
if (ret < 0)
return ret;
- sai->mclk_rate = freq;
dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq);
+ sai->mclk_rate = freq;
+
+ if (sai->sai_mclk) {
+ ret = clk_set_rate_exclusive(sai->sai_mclk,
+ sai->mclk_rate);
+ if (ret) {
+ dev_err(cpu_dai->dev,
+ "Could not set mclk rate\n");
+ return ret;
+ }
+ }
}
return 0;
@@ -715,15 +900,9 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
{
struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai);
int cr1, mask, div = 0;
- int sai_clk_rate, mclk_ratio, den, ret;
- int version = sai->pdata->conf->version;
+ int sai_clk_rate, mclk_ratio, den;
unsigned int rate = params_rate(params);
- if (!sai->mclk_rate) {
- dev_err(cpu_dai->dev, "Mclk rate is null\n");
- return -EINVAL;
- }
-
if (!(rate % 11025))
clk_set_parent(sai->sai_ck, sai->pdata->clk_x11k);
else
@@ -731,14 +910,22 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
sai_clk_rate = clk_get_rate(sai->sai_ck);
if (STM_SAI_IS_F4(sai->pdata)) {
- /*
- * mclk_rate = 256 * fs
- * MCKDIV = 0 if sai_ck < 3/2 * mclk_rate
- * MCKDIV = sai_ck / (2 * mclk_rate) otherwise
+ /* mclk on (NODIV=0)
+ * mclk_rate = 256 * fs
+ * MCKDIV = 0 if sai_ck < 3/2 * mclk_rate
+ * MCKDIV = sai_ck / (2 * mclk_rate) otherwise
+ * mclk off (NODIV=1)
+ * MCKDIV ignored. sck = sai_ck
*/
- if (2 * sai_clk_rate >= 3 * sai->mclk_rate)
- div = DIV_ROUND_CLOSEST(sai_clk_rate,
- 2 * sai->mclk_rate);
+ if (!sai->mclk_rate)
+ return 0;
+
+ if (2 * sai_clk_rate >= 3 * sai->mclk_rate) {
+ div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+ 2 * sai->mclk_rate);
+ if (div < 0)
+ return div;
+ }
} else {
/*
* TDM mode :
@@ -750,8 +937,10 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
* Note: NOMCK/NODIV correspond to same bit.
*/
if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) {
- div = DIV_ROUND_CLOSEST(sai_clk_rate,
- (params_rate(params) * 128));
+ div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+ rate * 128);
+ if (div < 0)
+ return div;
} else {
if (sai->mclk_rate) {
mclk_ratio = sai->mclk_rate / rate;
@@ -764,31 +953,22 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
mclk_ratio);
return -EINVAL;
}
- div = DIV_ROUND_CLOSEST(sai_clk_rate,
- sai->mclk_rate);
+ div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+ sai->mclk_rate);
+ if (div < 0)
+ return div;
} else {
/* mclk-fs not set, master clock not active */
den = sai->fs_length * params_rate(params);
- div = DIV_ROUND_CLOSEST(sai_clk_rate, den);
+ div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+ den);
+ if (div < 0)
+ return div;
}
}
}
- if (div > SAI_XCR1_MCKDIV_MAX(version)) {
- dev_err(cpu_dai->dev, "Divider %d out of range\n", div);
- return -EINVAL;
- }
- dev_dbg(cpu_dai->dev, "SAI clock %d, divider %d\n", sai_clk_rate, div);
-
- mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version));
- cr1 = SAI_XCR1_MCKDIV_SET(div);
- ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1);
- if (ret < 0) {
- dev_err(cpu_dai->dev, "Failed to update CR1 register\n");
- return ret;
- }
-
- return 0;
+ return stm32_sai_set_clk_div(sai, div);
}
static int stm32_sai_hw_params(struct snd_pcm_substream *substream,
@@ -881,6 +1061,9 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream,
SAI_XCR1_NODIV);
clk_disable_unprepare(sai->sai_ck);
+
+ clk_rate_exclusive_put(sai->sai_mclk);
+
sai->substream = NULL;
}
@@ -903,6 +1086,8 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai)
struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev);
int cr1 = 0, cr1_mask;
+ sai->cpu_dai = cpu_dai;
+
sai->dma_params.addr = (dma_addr_t)(sai->phys_addr + STM_SAI_DR_REGX);
/*
* DMA supports 4, 8 or 16 burst sizes. Burst size 4 is the best choice,
@@ -1124,16 +1309,15 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev,
sai->sync = SAI_SYNC_NONE;
if (args.np) {
if (args.np == np) {
- dev_err(&pdev->dev, "%s sync own reference\n",
- np->name);
+ dev_err(&pdev->dev, "%pOFn sync own reference\n", np);
of_node_put(args.np);
return -EINVAL;
}
sai->np_sync_provider = of_get_parent(args.np);
if (!sai->np_sync_provider) {
- dev_err(&pdev->dev, "%s parent node not found\n",
- np->name);
+ dev_err(&pdev->dev, "%pOFn parent node not found\n",
+ np);
of_node_put(args.np);
return -ENODEV;
}
@@ -1182,6 +1366,23 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev,
return PTR_ERR(sai->sai_ck);
}
+ if (STM_SAI_IS_F4(sai->pdata))
+ return 0;
+
+ /* Register mclk provider if requested */
+ if (of_find_property(np, "#clock-cells", NULL)) {
+ ret = stm32_sai_add_mclk_provider(sai);
+ if (ret < 0)
+ return ret;
+ } else {
+ sai->sai_mclk = devm_clk_get(&pdev->dev, "MCLK");
+ if (IS_ERR(sai->sai_mclk)) {
+ if (PTR_ERR(sai->sai_mclk) != -ENOENT)
+ return PTR_ERR(sai->sai_mclk);
+ sai->sai_mclk = NULL;
+ }
+ }
+
return 0;
}
diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig
index 22408bc2d6ec..66aad0d3f9c7 100644
--- a/sound/soc/sunxi/Kconfig
+++ b/sound/soc/sunxi/Kconfig
@@ -12,7 +12,7 @@ config SND_SUN4I_CODEC
config SND_SUN8I_CODEC
tristate "Allwinner SUN8I audio codec"
depends on OF
- depends on MACH_SUN8I || COMPILE_TEST
+ depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST
select REGMAP_MMIO
help
This option enables the digital part of the internal audio codec for
@@ -23,11 +23,19 @@ config SND_SUN8I_CODEC
config SND_SUN8I_CODEC_ANALOG
tristate "Allwinner sun8i Codec Analog Controls Support"
depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST
- select REGMAP
+ select SND_SUN8I_ADDA_PR_REGMAP
help
Say Y or M if you want to add support for the analog controls for
the codec embedded in newer Allwinner SoCs.
+config SND_SUN50I_CODEC_ANALOG
+ tristate "Allwinner sun50i Codec Analog Controls Support"
+ depends on (ARM64 && ARCH_SUNXI) || COMPILE_TEST
+ select SND_SUNXI_ADDA_PR_REGMAP
+ help
+ Say Y or M if you want to add support for the analog controls for
+ the codec embedded in Allwinner A64 SoC.
+
config SND_SUN4I_I2S
tristate "Allwinner A10 I2S Support"
select SND_SOC_GENERIC_DMAENGINE_PCM
@@ -45,4 +53,9 @@ config SND_SUN4I_SPDIF
help
Say Y or M to add support for the S/PDIF audio block in the Allwinner
A10 and affiliated SoCs.
+
+config SND_SUN8I_ADDA_PR_REGMAP
+ tristate
+ select REGMAP
+
endmenu
diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile
index 4a9ef67386ca..a86be340a076 100644
--- a/sound/soc/sunxi/Makefile
+++ b/sound/soc/sunxi/Makefile
@@ -3,4 +3,6 @@ obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o
obj-$(CONFIG_SND_SUN4I_I2S) += sun4i-i2s.o
obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o
obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o
+obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o
obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o
+obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index a4aa931ebfae..d5ec1a20499d 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -644,40 +644,6 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
return 0;
}
-static int sun4i_i2s_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-
- /* Enable the whole hardware block */
- regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
- SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN);
-
- /* Enable the first output line */
- regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
- SUN4I_I2S_CTRL_SDO_EN_MASK,
- SUN4I_I2S_CTRL_SDO_EN(0));
-
-
- return clk_prepare_enable(i2s->mod_clk);
-}
-
-static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-
- clk_disable_unprepare(i2s->mod_clk);
-
- /* Disable our output lines */
- regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
- SUN4I_I2S_CTRL_SDO_EN_MASK, 0);
-
- /* Disable the whole hardware block */
- regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
- SUN4I_I2S_CTRL_GL_EN, 0);
-}
-
static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
@@ -695,8 +661,6 @@ static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = {
.hw_params = sun4i_i2s_hw_params,
.set_fmt = sun4i_i2s_set_fmt,
.set_sysclk = sun4i_i2s_set_sysclk,
- .shutdown = sun4i_i2s_shutdown,
- .startup = sun4i_i2s_startup,
.trigger = sun4i_i2s_trigger,
};
@@ -869,6 +833,21 @@ static int sun4i_i2s_runtime_resume(struct device *dev)
goto err_disable_clk;
}
+ /* Enable the whole hardware block */
+ regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+ SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN);
+
+ /* Enable the first output line */
+ regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+ SUN4I_I2S_CTRL_SDO_EN_MASK,
+ SUN4I_I2S_CTRL_SDO_EN(0));
+
+ ret = clk_prepare_enable(i2s->mod_clk);
+ if (ret) {
+ dev_err(dev, "Failed to enable module clock\n");
+ goto err_disable_clk;
+ }
+
return 0;
err_disable_clk:
@@ -880,6 +859,16 @@ static int sun4i_i2s_runtime_suspend(struct device *dev)
{
struct sun4i_i2s *i2s = dev_get_drvdata(dev);
+ clk_disable_unprepare(i2s->mod_clk);
+
+ /* Disable our output lines */
+ regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+ SUN4I_I2S_CTRL_SDO_EN_MASK, 0);
+
+ /* Disable the whole hardware block */
+ regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+ SUN4I_I2S_CTRL_GL_EN, 0);
+
regcache_cache_only(i2s->regmap, true);
clk_disable_unprepare(i2s->bus_clk);
@@ -961,6 +950,23 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
.field_rxchansel = REG_FIELD(SUN8I_I2S_RX_CHAN_SEL_REG, 0, 2),
};
+static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
+ .has_reset = true,
+ .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG,
+ .sun4i_i2s_regmap = &sun4i_i2s_regmap_config,
+ .has_slave_select_bit = true,
+ .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
+ .field_fmt_wss = REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
+ .field_fmt_sr = REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
+ .field_fmt_bclk = REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
+ .field_fmt_lrclk = REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+ .field_fmt_mode = REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
+ .field_txchanmap = REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
+ .field_rxchanmap = REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
+ .field_txchansel = REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
+ .field_rxchansel = REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
+};
+
static int sun4i_i2s_init_regmap_fields(struct device *dev,
struct sun4i_i2s *i2s)
{
@@ -1169,6 +1175,10 @@ static const struct of_device_id sun4i_i2s_match[] = {
.compatible = "allwinner,sun8i-h3-i2s",
.data = &sun8i_h3_i2s_quirks,
},
+ {
+ .compatible = "allwinner,sun50i-a64-codec-i2s",
+ .data = &sun50i_a64_codec_i2s_quirks,
+ },
{}
};
MODULE_DEVICE_TABLE(of, sun4i_i2s_match);
diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c
new file mode 100644
index 000000000000..8f5f999df631
--- /dev/null
+++ b/sound/soc/sunxi/sun50i-codec-analog.c
@@ -0,0 +1,444 @@
+// SPDX-License-Identifier: GPL-2.0+
+/*
+ * This driver supports the analog controls for the internal codec
+ * found in Allwinner's A64 SoC.
+ *
+ * Copyright (C) 2016 Chen-Yu Tsai <wens@csie.org>
+ * Copyright (C) 2017 Marcus Cooper <codekipper@gmail.com>
+ * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com>
+ *
+ * Based on sun8i-codec-analog.c
+ *
+ */
+
+#include <linux/io.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "sun8i-adda-pr-regmap.h"
+
+/* Codec analog control register offsets and bit fields */
+#define SUN50I_ADDA_HP_CTRL 0x00
+#define SUN50I_ADDA_HP_CTRL_PA_CLK_GATE 7
+#define SUN50I_ADDA_HP_CTRL_HPPA_EN 6
+#define SUN50I_ADDA_HP_CTRL_HPVOL 0
+
+#define SUN50I_ADDA_OL_MIX_CTRL 0x01
+#define SUN50I_ADDA_OL_MIX_CTRL_MIC1 6
+#define SUN50I_ADDA_OL_MIX_CTRL_MIC2 5
+#define SUN50I_ADDA_OL_MIX_CTRL_PHONE 4
+#define SUN50I_ADDA_OL_MIX_CTRL_PHONEN 3
+#define SUN50I_ADDA_OL_MIX_CTRL_LINEINL 2
+#define SUN50I_ADDA_OL_MIX_CTRL_DACL 1
+#define SUN50I_ADDA_OL_MIX_CTRL_DACR 0
+
+#define SUN50I_ADDA_OR_MIX_CTRL 0x02
+#define SUN50I_ADDA_OR_MIX_CTRL_MIC1 6
+#define SUN50I_ADDA_OR_MIX_CTRL_MIC2 5
+#define SUN50I_ADDA_OR_MIX_CTRL_PHONE 4
+#define SUN50I_ADDA_OR_MIX_CTRL_PHONEP 3
+#define SUN50I_ADDA_OR_MIX_CTRL_LINEINR 2
+#define SUN50I_ADDA_OR_MIX_CTRL_DACR 1
+#define SUN50I_ADDA_OR_MIX_CTRL_DACL 0
+
+#define SUN50I_ADDA_LINEOUT_CTRL0 0x05
+#define SUN50I_ADDA_LINEOUT_CTRL0_LEN 7
+#define SUN50I_ADDA_LINEOUT_CTRL0_REN 6
+#define SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL 5
+#define SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL 4
+
+#define SUN50I_ADDA_LINEOUT_CTRL1 0x06
+#define SUN50I_ADDA_LINEOUT_CTRL1_VOL 0
+
+#define SUN50I_ADDA_MIC1_CTRL 0x07
+#define SUN50I_ADDA_MIC1_CTRL_MIC1G 4
+#define SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN 3
+#define SUN50I_ADDA_MIC1_CTRL_MIC1BOOST 0
+
+#define SUN50I_ADDA_MIC2_CTRL 0x08
+#define SUN50I_ADDA_MIC2_CTRL_MIC2G 4
+#define SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN 3
+#define SUN50I_ADDA_MIC2_CTRL_MIC2BOOST 0
+
+#define SUN50I_ADDA_LINEIN_CTRL 0x09
+#define SUN50I_ADDA_LINEIN_CTRL_LINEING 0
+
+#define SUN50I_ADDA_MIX_DAC_CTRL 0x0a
+#define SUN50I_ADDA_MIX_DAC_CTRL_DACAREN 7
+#define SUN50I_ADDA_MIX_DAC_CTRL_DACALEN 6
+#define SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN 5
+#define SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN 4
+#define SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE 3
+#define SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE 2
+#define SUN50I_ADDA_MIX_DAC_CTRL_RHPIS 1
+#define SUN50I_ADDA_MIX_DAC_CTRL_LHPIS 0
+
+#define SUN50I_ADDA_L_ADCMIX_SRC 0x0b
+#define SUN50I_ADDA_L_ADCMIX_SRC_MIC1 6
+#define SUN50I_ADDA_L_ADCMIX_SRC_MIC2 5
+#define SUN50I_ADDA_L_ADCMIX_SRC_PHONE 4
+#define SUN50I_ADDA_L_ADCMIX_SRC_PHONEN 3
+#define SUN50I_ADDA_L_ADCMIX_SRC_LINEINL 2
+#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL 1
+#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR 0
+
+#define SUN50I_ADDA_R_ADCMIX_SRC 0x0c
+#define SUN50I_ADDA_R_ADCMIX_SRC_MIC1 6
+#define SUN50I_ADDA_R_ADCMIX_SRC_MIC2 5
+#define SUN50I_ADDA_R_ADCMIX_SRC_PHONE 4
+#define SUN50I_ADDA_R_ADCMIX_SRC_PHONEP 3
+#define SUN50I_ADDA_R_ADCMIX_SRC_LINEINR 2
+#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXR 1
+#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXL 0
+
+#define SUN50I_ADDA_ADC_CTRL 0x0d
+#define SUN50I_ADDA_ADC_CTRL_ADCREN 7
+#define SUN50I_ADDA_ADC_CTRL_ADCLEN 6
+#define SUN50I_ADDA_ADC_CTRL_ADCG 0
+
+#define SUN50I_ADDA_HS_MBIAS_CTRL 0x0e
+#define SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN 7
+
+#define SUN50I_ADDA_JACK_MIC_CTRL 0x1d
+#define SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN 5
+
+/* mixer controls */
+static const struct snd_kcontrol_new sun50i_a64_codec_mixer_controls[] = {
+ SOC_DAPM_DOUBLE_R("DAC Playback Switch",
+ SUN50I_ADDA_OL_MIX_CTRL,
+ SUN50I_ADDA_OR_MIX_CTRL,
+ SUN50I_ADDA_OL_MIX_CTRL_DACL, 1, 0),
+ SOC_DAPM_DOUBLE_R("DAC Reversed Playback Switch",
+ SUN50I_ADDA_OL_MIX_CTRL,
+ SUN50I_ADDA_OR_MIX_CTRL,
+ SUN50I_ADDA_OL_MIX_CTRL_DACR, 1, 0),
+ SOC_DAPM_DOUBLE_R("Line In Playback Switch",
+ SUN50I_ADDA_OL_MIX_CTRL,
+ SUN50I_ADDA_OR_MIX_CTRL,
+ SUN50I_ADDA_OL_MIX_CTRL_LINEINL, 1, 0),
+ SOC_DAPM_DOUBLE_R("Mic1 Playback Switch",
+ SUN50I_ADDA_OL_MIX_CTRL,
+ SUN50I_ADDA_OR_MIX_CTRL,
+ SUN50I_ADDA_OL_MIX_CTRL_MIC1, 1, 0),
+ SOC_DAPM_DOUBLE_R("Mic2 Playback Switch",
+ SUN50I_ADDA_OL_MIX_CTRL,
+ SUN50I_ADDA_OR_MIX_CTRL,
+ SUN50I_ADDA_OL_MIX_CTRL_MIC2, 1, 0),
+};
+
+/* ADC mixer controls */
+static const struct snd_kcontrol_new sun50i_codec_adc_mixer_controls[] = {
+ SOC_DAPM_DOUBLE_R("Mixer Capture Switch",
+ SUN50I_ADDA_L_ADCMIX_SRC,
+ SUN50I_ADDA_R_ADCMIX_SRC,
+ SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL, 1, 0),
+ SOC_DAPM_DOUBLE_R("Mixer Reversed Capture Switch",
+ SUN50I_ADDA_L_ADCMIX_SRC,
+ SUN50I_ADDA_R_ADCMIX_SRC,
+ SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR, 1, 0),
+ SOC_DAPM_DOUBLE_R("Line In Capture Switch",
+ SUN50I_ADDA_L_ADCMIX_SRC,
+ SUN50I_ADDA_R_ADCMIX_SRC,
+ SUN50I_ADDA_L_ADCMIX_SRC_LINEINL, 1, 0),
+ SOC_DAPM_DOUBLE_R("Mic1 Capture Switch",
+ SUN50I_ADDA_L_ADCMIX_SRC,
+ SUN50I_ADDA_R_ADCMIX_SRC,
+ SUN50I_ADDA_L_ADCMIX_SRC_MIC1, 1, 0),
+ SOC_DAPM_DOUBLE_R("Mic2 Capture Switch",
+ SUN50I_ADDA_L_ADCMIX_SRC,
+ SUN50I_ADDA_R_ADCMIX_SRC,
+ SUN50I_ADDA_L_ADCMIX_SRC_MIC2, 1, 0),
+};
+
+static const DECLARE_TLV_DB_SCALE(sun50i_codec_out_mixer_pregain_scale,
+ -450, 150, 0);
+static const DECLARE_TLV_DB_RANGE(sun50i_codec_mic_gain_scale,
+ 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+ 1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0),
+);
+
+static const DECLARE_TLV_DB_SCALE(sun50i_codec_hp_vol_scale, -6300, 100, 1);
+
+static const DECLARE_TLV_DB_RANGE(sun50i_codec_lineout_vol_scale,
+ 0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1),
+ 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0),
+);
+
+
+/* volume / mute controls */
+static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = {
+ SOC_SINGLE_TLV("Headphone Playback Volume",
+ SUN50I_ADDA_HP_CTRL,
+ SUN50I_ADDA_HP_CTRL_HPVOL, 0x3f, 0,
+ sun50i_codec_hp_vol_scale),
+
+ SOC_DOUBLE("Headphone Playback Switch",
+ SUN50I_ADDA_MIX_DAC_CTRL,
+ SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE,
+ SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE, 1, 0),
+
+ /* Mixer pre-gain */
+ SOC_SINGLE_TLV("Mic1 Playback Volume", SUN50I_ADDA_MIC1_CTRL,
+ SUN50I_ADDA_MIC1_CTRL_MIC1G,
+ 0x7, 0, sun50i_codec_out_mixer_pregain_scale),
+
+ /* Microphone Amp boost gain */
+ SOC_SINGLE_TLV("Mic1 Boost Volume", SUN50I_ADDA_MIC1_CTRL,
+ SUN50I_ADDA_MIC1_CTRL_MIC1BOOST, 0x7, 0,
+ sun50i_codec_mic_gain_scale),
+
+ /* Mixer pre-gain */
+ SOC_SINGLE_TLV("Mic2 Playback Volume",
+ SUN50I_ADDA_MIC2_CTRL, SUN50I_ADDA_MIC2_CTRL_MIC2G,
+ 0x7, 0, sun50i_codec_out_mixer_pregain_scale),
+
+ /* Microphone Amp boost gain */
+ SOC_SINGLE_TLV("Mic2 Boost Volume", SUN50I_ADDA_MIC2_CTRL,
+ SUN50I_ADDA_MIC2_CTRL_MIC2BOOST, 0x7, 0,
+ sun50i_codec_mic_gain_scale),
+
+ /* ADC */
+ SOC_SINGLE_TLV("ADC Gain Capture Volume", SUN50I_ADDA_ADC_CTRL,
+ SUN50I_ADDA_ADC_CTRL_ADCG, 0x7, 0,
+ sun50i_codec_out_mixer_pregain_scale),
+
+ /* Mixer pre-gain */
+ SOC_SINGLE_TLV("Line In Playback Volume", SUN50I_ADDA_LINEIN_CTRL,
+ SUN50I_ADDA_LINEIN_CTRL_LINEING,
+ 0x7, 0, sun50i_codec_out_mixer_pregain_scale),
+
+ SOC_SINGLE_TLV("Line Out Playback Volume",
+ SUN50I_ADDA_LINEOUT_CTRL1,
+ SUN50I_ADDA_LINEOUT_CTRL1_VOL, 0x1f, 0,
+ sun50i_codec_lineout_vol_scale),
+
+ SOC_DOUBLE("Line Out Playback Switch",
+ SUN50I_ADDA_LINEOUT_CTRL0,
+ SUN50I_ADDA_LINEOUT_CTRL0_LEN,
+ SUN50I_ADDA_LINEOUT_CTRL0_REN, 1, 0),
+
+};
+
+static const char * const sun50i_codec_hp_src_enum_text[] = {
+ "DAC", "Mixer",
+};
+
+static SOC_ENUM_DOUBLE_DECL(sun50i_codec_hp_src_enum,
+ SUN50I_ADDA_MIX_DAC_CTRL,
+ SUN50I_ADDA_MIX_DAC_CTRL_LHPIS,
+ SUN50I_ADDA_MIX_DAC_CTRL_RHPIS,
+ sun50i_codec_hp_src_enum_text);
+
+static const struct snd_kcontrol_new sun50i_codec_hp_src[] = {
+ SOC_DAPM_ENUM("Headphone Source Playback Route",
+ sun50i_codec_hp_src_enum),
+};
+
+static const char * const sun50i_codec_lineout_src_enum_text[] = {
+ "Stereo", "Mono Differential",
+};
+
+static SOC_ENUM_DOUBLE_DECL(sun50i_codec_lineout_src_enum,
+ SUN50I_ADDA_LINEOUT_CTRL0,
+ SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL,
+ SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL,
+ sun50i_codec_lineout_src_enum_text);
+
+static const struct snd_kcontrol_new sun50i_codec_lineout_src[] = {
+ SOC_DAPM_ENUM("Line Out Source Playback Route",
+ sun50i_codec_lineout_src_enum),
+};
+
+static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = {
+ /* DAC */
+ SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL,
+ SUN50I_ADDA_MIX_DAC_CTRL_DACALEN, 0),
+ SND_SOC_DAPM_DAC("Right DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL,
+ SUN50I_ADDA_MIX_DAC_CTRL_DACAREN, 0),
+ /* ADC */
+ SND_SOC_DAPM_ADC("Left ADC", NULL, SUN50I_ADDA_ADC_CTRL,
+ SUN50I_ADDA_ADC_CTRL_ADCLEN, 0),
+ SND_SOC_DAPM_ADC("Right ADC", NULL, SUN50I_ADDA_ADC_CTRL,
+ SUN50I_ADDA_ADC_CTRL_ADCREN, 0),
+ /*
+ * Due to this component and the codec belonging to separate DAPM
+ * contexts, we need to manually link the above widgets to their
+ * stream widgets at the card level.
+ */
+
+ SND_SOC_DAPM_MUX("Headphone Source Playback Route",
+ SND_SOC_NOPM, 0, 0, sun50i_codec_hp_src),
+ SND_SOC_DAPM_OUT_DRV("Headphone Amp", SUN50I_ADDA_HP_CTRL,
+ SUN50I_ADDA_HP_CTRL_HPPA_EN, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("HP"),
+
+ SND_SOC_DAPM_MUX("Line Out Source Playback Route",
+ SND_SOC_NOPM, 0, 0, sun50i_codec_lineout_src),
+ SND_SOC_DAPM_OUTPUT("LINEOUT"),
+
+ /* Microphone inputs */
+ SND_SOC_DAPM_INPUT("MIC1"),
+
+ /* Microphone Bias */
+ SND_SOC_DAPM_SUPPLY("MBIAS", SUN50I_ADDA_HS_MBIAS_CTRL,
+ SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN,
+ 0, NULL, 0),
+
+ /* Mic input path */
+ SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN50I_ADDA_MIC1_CTRL,
+ SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN, 0, NULL, 0),
+
+ /* Microphone input */
+ SND_SOC_DAPM_INPUT("MIC2"),
+
+ /* Microphone Bias */
+ SND_SOC_DAPM_SUPPLY("HBIAS", SUN50I_ADDA_JACK_MIC_CTRL,
+ SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN,
+ 0, NULL, 0),
+
+ /* Mic input path */
+ SND_SOC_DAPM_PGA("Mic2 Amplifier", SUN50I_ADDA_MIC2_CTRL,
+ SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN, 0, NULL, 0),
+
+ /* Line input */
+ SND_SOC_DAPM_INPUT("LINEIN"),
+
+ /* Mixers */
+ SND_SOC_DAPM_MIXER("Left Mixer", SUN50I_ADDA_MIX_DAC_CTRL,
+ SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN, 0,
+ sun50i_a64_codec_mixer_controls,
+ ARRAY_SIZE(sun50i_a64_codec_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Right Mixer", SUN50I_ADDA_MIX_DAC_CTRL,
+ SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN, 0,
+ sun50i_a64_codec_mixer_controls,
+ ARRAY_SIZE(sun50i_a64_codec_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Left ADC Mixer", SUN50I_ADDA_ADC_CTRL,
+ SUN50I_ADDA_ADC_CTRL_ADCLEN, 0,
+ sun50i_codec_adc_mixer_controls,
+ ARRAY_SIZE(sun50i_codec_adc_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Right ADC Mixer", SUN50I_ADDA_ADC_CTRL,
+ SUN50I_ADDA_ADC_CTRL_ADCREN, 0,
+ sun50i_codec_adc_mixer_controls,
+ ARRAY_SIZE(sun50i_codec_adc_mixer_controls)),
+};
+
+static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = {
+ /* Left Mixer Routes */
+ { "Left Mixer", "DAC Playback Switch", "Left DAC" },
+ { "Left Mixer", "DAC Reversed Playback Switch", "Right DAC" },
+ { "Left Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" },
+
+ /* Right Mixer Routes */
+ { "Right Mixer", "DAC Playback Switch", "Right DAC" },
+ { "Right Mixer", "DAC Reversed Playback Switch", "Left DAC" },
+ { "Right Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" },
+
+ /* Left ADC Mixer Routes */
+ { "Left ADC Mixer", "Mixer Capture Switch", "Left Mixer" },
+ { "Left ADC Mixer", "Mixer Reversed Capture Switch", "Right Mixer" },
+ { "Left ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" },
+
+ /* Right ADC Mixer Routes */
+ { "Right ADC Mixer", "Mixer Capture Switch", "Right Mixer" },
+ { "Right ADC Mixer", "Mixer Reversed Capture Switch", "Left Mixer" },
+ { "Right ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" },
+
+ /* ADC Routes */
+ { "Left ADC", NULL, "Left ADC Mixer" },
+ { "Right ADC", NULL, "Right ADC Mixer" },
+
+ /* Headphone Routes */
+ { "Headphone Source Playback Route", "DAC", "Left DAC" },
+ { "Headphone Source Playback Route", "DAC", "Right DAC" },
+ { "Headphone Source Playback Route", "Mixer", "Left Mixer" },
+ { "Headphone Source Playback Route", "Mixer", "Right Mixer" },
+ { "Headphone Amp", NULL, "Headphone Source Playback Route" },
+ { "HP", NULL, "Headphone Amp" },
+
+ /* Microphone Routes */
+ { "Mic1 Amplifier", NULL, "MIC1"},
+
+ /* Microphone Routes */
+ { "Mic2 Amplifier", NULL, "MIC2"},
+ { "Left Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" },
+ { "Right Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" },
+ { "Left ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" },
+ { "Right ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" },
+
+ /* Line-in Routes */
+ { "Left Mixer", "Line In Playback Switch", "LINEIN" },
+ { "Right Mixer", "Line In Playback Switch", "LINEIN" },
+ { "Left ADC Mixer", "Line In Capture Switch", "LINEIN" },
+ { "Right ADC Mixer", "Line In Capture Switch", "LINEIN" },
+
+ /* Line-out Routes */
+ { "Line Out Source Playback Route", "Stereo", "Left Mixer" },
+ { "Line Out Source Playback Route", "Stereo", "Right Mixer" },
+ { "Line Out Source Playback Route", "Mono Differential", "Left Mixer" },
+ { "Line Out Source Playback Route", "Mono Differential",
+ "Right Mixer" },
+ { "LINEOUT", NULL, "Line Out Source Playback Route" },
+};
+
+static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = {
+ .controls = sun50i_a64_codec_controls,
+ .num_controls = ARRAY_SIZE(sun50i_a64_codec_controls),
+ .dapm_widgets = sun50i_a64_codec_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(sun50i_a64_codec_widgets),
+ .dapm_routes = sun50i_a64_codec_routes,
+ .num_dapm_routes = ARRAY_SIZE(sun50i_a64_codec_routes),
+};
+
+static const struct of_device_id sun50i_codec_analog_of_match[] = {
+ {
+ .compatible = "allwinner,sun50i-a64-codec-analog",
+ },
+ {}
+};
+MODULE_DEVICE_TABLE(of, sun50i_codec_analog_of_match);
+
+static int sun50i_codec_analog_probe(struct platform_device *pdev)
+{
+ struct resource *res;
+ struct regmap *regmap;
+ void __iomem *base;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(base)) {
+ dev_err(&pdev->dev, "Failed to map the registers\n");
+ return PTR_ERR(base);
+ }
+
+ regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base);
+ if (IS_ERR(regmap)) {
+ dev_err(&pdev->dev, "Failed to create regmap\n");
+ return PTR_ERR(regmap);
+ }
+
+ return devm_snd_soc_register_component(&pdev->dev,
+ &sun50i_codec_analog_cmpnt_drv,
+ NULL, 0);
+}
+
+static struct platform_driver sun50i_codec_analog_driver = {
+ .driver = {
+ .name = "sun50i-codec-analog",
+ .of_match_table = sun50i_codec_analog_of_match,
+ },
+ .probe = sun50i_codec_analog_probe,
+};
+module_platform_driver(sun50i_codec_analog_driver);
+
+MODULE_DESCRIPTION("Allwinner internal codec analog controls driver for A64");
+MODULE_AUTHOR("Vasily Khoruzhick <anarsoul@gmail.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:sun50i-codec-analog");
diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.c b/sound/soc/sunxi/sun8i-adda-pr-regmap.c
new file mode 100644
index 000000000000..e68ce9d2884d
--- /dev/null
+++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.c
@@ -0,0 +1,102 @@
+// SPDX-License-Identifier: GPL-2.0+
+/*
+ * This driver provides regmap to access to analog part of audio codec
+ * found on Allwinner A23, A31s, A33, H3 and A64 Socs
+ *
+ * Copyright 2016 Chen-Yu Tsai <wens@csie.org>
+ * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com>
+ */
+
+#include <linux/io.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+
+#include "sun8i-adda-pr-regmap.h"
+
+/* Analog control register access bits */
+#define ADDA_PR 0x0 /* PRCM base + 0x1c0 */
+#define ADDA_PR_RESET BIT(28)
+#define ADDA_PR_WRITE BIT(24)
+#define ADDA_PR_ADDR_SHIFT 16
+#define ADDA_PR_ADDR_MASK GENMASK(4, 0)
+#define ADDA_PR_DATA_IN_SHIFT 8
+#define ADDA_PR_DATA_IN_MASK GENMASK(7, 0)
+#define ADDA_PR_DATA_OUT_SHIFT 0
+#define ADDA_PR_DATA_OUT_MASK GENMASK(7, 0)
+
+/* regmap access bits */
+static int adda_reg_read(void *context, unsigned int reg, unsigned int *val)
+{
+ void __iomem *base = (void __iomem *)context;
+ u32 tmp;
+
+ /* De-assert reset */
+ writel(readl(base) | ADDA_PR_RESET, base);
+
+ /* Clear write bit */
+ writel(readl(base) & ~ADDA_PR_WRITE, base);
+
+ /* Set register address */
+ tmp = readl(base);
+ tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
+ tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
+ writel(tmp, base);
+
+ /* Read back value */
+ *val = readl(base) & ADDA_PR_DATA_OUT_MASK;
+
+ return 0;
+}
+
+static int adda_reg_write(void *context, unsigned int reg, unsigned int val)
+{
+ void __iomem *base = (void __iomem *)context;
+ u32 tmp;
+
+ /* De-assert reset */
+ writel(readl(base) | ADDA_PR_RESET, base);
+
+ /* Set register address */
+ tmp = readl(base);
+ tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
+ tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
+ writel(tmp, base);
+
+ /* Set data to write */
+ tmp = readl(base);
+ tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT);
+ tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT;
+ writel(tmp, base);
+
+ /* Set write bit to signal a write */
+ writel(readl(base) | ADDA_PR_WRITE, base);
+
+ /* Clear write bit */
+ writel(readl(base) & ~ADDA_PR_WRITE, base);
+
+ return 0;
+}
+
+static const struct regmap_config adda_pr_regmap_cfg = {
+ .name = "adda-pr",
+ .reg_bits = 5,
+ .reg_stride = 1,
+ .val_bits = 8,
+ .reg_read = adda_reg_read,
+ .reg_write = adda_reg_write,
+ .fast_io = true,
+ .max_register = 31,
+};
+
+struct regmap *sun8i_adda_pr_regmap_init(struct device *dev,
+ void __iomem *base)
+{
+ return devm_regmap_init(dev, NULL, base, &adda_pr_regmap_cfg);
+}
+EXPORT_SYMBOL_GPL(sun8i_adda_pr_regmap_init);
+
+MODULE_DESCRIPTION("Allwinner analog audio codec regmap driver");
+MODULE_AUTHOR("Vasily Khoruzhick <anarsoul@gmail.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:sunxi-adda-pr");
diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.h b/sound/soc/sunxi/sun8i-adda-pr-regmap.h
new file mode 100644
index 000000000000..a5ae95dfebc1
--- /dev/null
+++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.h
@@ -0,0 +1,7 @@
+/* SPDX-License-Identifier: GPL-2.0+ */
+/*
+ * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com>
+ */
+
+struct regmap *sun8i_adda_pr_regmap_init(struct device *dev,
+ void __iomem *base);
diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c
index 485e79f292c4..916a46bbc1c8 100644
--- a/sound/soc/sunxi/sun8i-codec-analog.c
+++ b/sound/soc/sunxi/sun8i-codec-analog.c
@@ -27,6 +27,8 @@
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
+#include "sun8i-adda-pr-regmap.h"
+
/* Codec analog control register offsets and bit fields */
#define SUN8I_ADDA_HP_VOLC 0x00
#define SUN8I_ADDA_HP_VOLC_PA_CLK_GATE 7
@@ -120,81 +122,6 @@
#define SUN8I_ADDA_ADC_AP_EN_ADCLEN 6
#define SUN8I_ADDA_ADC_AP_EN_ADCG 0
-/* Analog control register access bits */
-#define ADDA_PR 0x0 /* PRCM base + 0x1c0 */
-#define ADDA_PR_RESET BIT(28)
-#define ADDA_PR_WRITE BIT(24)
-#define ADDA_PR_ADDR_SHIFT 16
-#define ADDA_PR_ADDR_MASK GENMASK(4, 0)
-#define ADDA_PR_DATA_IN_SHIFT 8
-#define ADDA_PR_DATA_IN_MASK GENMASK(7, 0)
-#define ADDA_PR_DATA_OUT_SHIFT 0
-#define ADDA_PR_DATA_OUT_MASK GENMASK(7, 0)
-
-/* regmap access bits */
-static int adda_reg_read(void *context, unsigned int reg, unsigned int *val)
-{
- void __iomem *base = (void __iomem *)context;
- u32 tmp;
-
- /* De-assert reset */
- writel(readl(base) | ADDA_PR_RESET, base);
-
- /* Clear write bit */
- writel(readl(base) & ~ADDA_PR_WRITE, base);
-
- /* Set register address */
- tmp = readl(base);
- tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
- tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
- writel(tmp, base);
-
- /* Read back value */
- *val = readl(base) & ADDA_PR_DATA_OUT_MASK;
-
- return 0;
-}
-
-static int adda_reg_write(void *context, unsigned int reg, unsigned int val)
-{
- void __iomem *base = (void __iomem *)context;
- u32 tmp;
-
- /* De-assert reset */
- writel(readl(base) | ADDA_PR_RESET, base);
-
- /* Set register address */
- tmp = readl(base);
- tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
- tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
- writel(tmp, base);
-
- /* Set data to write */
- tmp = readl(base);
- tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT);
- tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT;
- writel(tmp, base);
-
- /* Set write bit to signal a write */
- writel(readl(base) | ADDA_PR_WRITE, base);
-
- /* Clear write bit */
- writel(readl(base) & ~ADDA_PR_WRITE, base);
-
- return 0;
-}
-
-static const struct regmap_config adda_pr_regmap_cfg = {
- .name = "adda-pr",
- .reg_bits = 5,
- .reg_stride = 1,
- .val_bits = 8,
- .reg_read = adda_reg_read,
- .reg_write = adda_reg_write,
- .fast_io = true,
- .max_register = 24,
-};
-
/* mixer controls */
static const struct snd_kcontrol_new sun8i_codec_mixer_controls[] = {
SOC_DAPM_DOUBLE_R("DAC Playback Switch",
@@ -912,7 +839,7 @@ static int sun8i_codec_analog_probe(struct platform_device *pdev)
return PTR_ERR(base);
}
- regmap = devm_regmap_init(&pdev->dev, NULL, base, &adda_pr_regmap_cfg);
+ regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base);
if (IS_ERR(regmap)) {
dev_err(&pdev->dev, "Failed to create regmap\n");
return PTR_ERR(regmap);
diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c
index fb37dd927e33..522a72fde78d 100644
--- a/sound/soc/sunxi/sun8i-codec.c
+++ b/sound/soc/sunxi/sun8i-codec.c
@@ -24,6 +24,7 @@
#include <linux/io.h>
#include <linux/pm_runtime.h>
#include <linux/regmap.h>
+#include <linux/log2.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
@@ -52,7 +53,6 @@
#define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_INV 13
#define SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV 9
#define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV 6
-#define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_16 (1 << 6)
#define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ 4
#define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ_16 (1 << 4)
#define SUN8I_AIF1CLK_CTRL_AIF1_DATA_FMT 2
@@ -300,12 +300,23 @@ static u8 sun8i_codec_get_bclk_div(struct sun8i_codec *scodec,
return best_val;
}
+static int sun8i_codec_get_lrck_div(unsigned int channels,
+ unsigned int word_size)
+{
+ unsigned int div = word_size * channels;
+
+ if (div < 16 || div > 256)
+ return -EINVAL;
+
+ return ilog2(div) - 4;
+}
+
static int sun8i_codec_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct sun8i_codec *scodec = snd_soc_component_get_drvdata(dai->component);
- int sample_rate;
+ int sample_rate, lrck_div;
u8 bclk_div;
/*
@@ -321,9 +332,14 @@ static int sun8i_codec_hw_params(struct snd_pcm_substream *substream,
SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK,
bclk_div << SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV);
+ lrck_div = sun8i_codec_get_lrck_div(params_channels(params),
+ params_physical_width(params));
+ if (lrck_div < 0)
+ return lrck_div;
+
regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL,
SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_MASK,
- SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_16);
+ lrck_div << SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV);
sample_rate = sun8i_codec_get_hw_rate(params);
if (sample_rate < 0)
diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c
index 45a4aa9d2a47..901457da25ec 100644
--- a/sound/soc/tegra/tegra_sgtl5000.c
+++ b/sound/soc/tegra/tegra_sgtl5000.c
@@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev)
dev_err(&pdev->dev,
"Property 'nvidia,i2s-controller' missing/invalid\n");
ret = -EINVAL;
- goto err;
+ goto err_put_codec_of_node;
}
tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node;
ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
if (ret)
- goto err;
+ goto err_put_cpu_of_node;
ret = snd_soc_register_card(card);
if (ret) {
@@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev)
err_fini_utils:
tegra_asoc_utils_fini(&machine->util_data);
+err_put_cpu_of_node:
+ of_node_put(tegra_sgtl5000_dai.cpu_of_node);
+ tegra_sgtl5000_dai.cpu_of_node = NULL;
+ tegra_sgtl5000_dai.platform_of_node = NULL;
+err_put_codec_of_node:
+ of_node_put(tegra_sgtl5000_dai.codec_of_node);
+ tegra_sgtl5000_dai.codec_of_node = NULL;
err:
return ret;
}
@@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev)
tegra_asoc_utils_fini(&machine->util_data);
+ of_node_put(tegra_sgtl5000_dai.cpu_of_node);
+ tegra_sgtl5000_dai.cpu_of_node = NULL;
+ tegra_sgtl5000_dai.platform_of_node = NULL;
+ of_node_put(tegra_sgtl5000_dai.codec_of_node);
+ tegra_sgtl5000_dai.codec_of_node = NULL;
+
return ret;
}
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
index e2ad00e3cae1..1cfca698ae4b 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -208,13 +208,12 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev)
if (err < 0)
return err;
- return snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component,
+ return devm_snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component,
&txx9aclc_ac97_dai, 1);
}
static int txx9aclc_ac97_dev_remove(struct platform_device *pdev)
{
- snd_soc_unregister_component(&pdev->dev);
snd_soc_set_ac97_ops(NULL);
return 0;
}
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index d55ca48de3ea..f4a72e39ffa9 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -200,6 +200,7 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb)
break;
}
#ifdef CONFIG_SND_USB_CAIAQ_INPUT
+ /* fall through */
case EP1_CMD_READ_ERP:
case EP1_CMD_READ_ANALOG:
snd_usb_caiaq_input_dispatch(cdev, buf, urb->actual_length);
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index dcfc546d81b9..b737f0ec77d0 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1175,8 +1175,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream,
if (port->ep->umidi->disconnected) {
/* gobble up remaining bytes to prevent wait in
* snd_rawmidi_drain_output */
- while (!snd_rawmidi_transmit_empty(substream))
- snd_rawmidi_transmit_ack(substream, 1);
+ snd_rawmidi_proceed(substream);
return;
}
tasklet_schedule(&port->ep->tasklet);
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index cbfb48bdea51..85ae0ff2382a 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -29,6 +29,7 @@
#include <linux/hid.h>
#include <linux/init.h>
+#include <linux/math64.h>
#include <linux/slab.h>
#include <linux/usb.h>
#include <linux/usb/audio.h>
@@ -1817,6 +1818,380 @@ static int dell_dock_mixer_init(struct usb_mixer_interface *mixer)
return 0;
}
+/* RME Class Compliant device quirks */
+
+#define SND_RME_GET_STATUS1 23
+#define SND_RME_GET_CURRENT_FREQ 17
+#define SND_RME_CLK_SYSTEM_SHIFT 16
+#define SND_RME_CLK_SYSTEM_MASK 0x1f
+#define SND_RME_CLK_AES_SHIFT 8
+#define SND_RME_CLK_SPDIF_SHIFT 12
+#define SND_RME_CLK_AES_SPDIF_MASK 0xf
+#define SND_RME_CLK_SYNC_SHIFT 6
+#define SND_RME_CLK_SYNC_MASK 0x3
+#define SND_RME_CLK_FREQMUL_SHIFT 18
+#define SND_RME_CLK_FREQMUL_MASK 0x7
+#define SND_RME_CLK_SYSTEM(x) \
+ ((x >> SND_RME_CLK_SYSTEM_SHIFT) & SND_RME_CLK_SYSTEM_MASK)
+#define SND_RME_CLK_AES(x) \
+ ((x >> SND_RME_CLK_AES_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK)
+#define SND_RME_CLK_SPDIF(x) \
+ ((x >> SND_RME_CLK_SPDIF_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK)
+#define SND_RME_CLK_SYNC(x) \
+ ((x >> SND_RME_CLK_SYNC_SHIFT) & SND_RME_CLK_SYNC_MASK)
+#define SND_RME_CLK_FREQMUL(x) \
+ ((x >> SND_RME_CLK_FREQMUL_SHIFT) & SND_RME_CLK_FREQMUL_MASK)
+#define SND_RME_CLK_AES_LOCK 0x1
+#define SND_RME_CLK_AES_SYNC 0x4
+#define SND_RME_CLK_SPDIF_LOCK 0x2
+#define SND_RME_CLK_SPDIF_SYNC 0x8
+#define SND_RME_SPDIF_IF_SHIFT 4
+#define SND_RME_SPDIF_FORMAT_SHIFT 5
+#define SND_RME_BINARY_MASK 0x1
+#define SND_RME_SPDIF_IF(x) \
+ ((x >> SND_RME_SPDIF_IF_SHIFT) & SND_RME_BINARY_MASK)
+#define SND_RME_SPDIF_FORMAT(x) \
+ ((x >> SND_RME_SPDIF_FORMAT_SHIFT) & SND_RME_BINARY_MASK)
+
+static const u32 snd_rme_rate_table[] = {
+ 32000, 44100, 48000, 50000,
+ 64000, 88200, 96000, 100000,
+ 128000, 176400, 192000, 200000,
+ 256000, 352800, 384000, 400000,
+ 512000, 705600, 768000, 800000
+};
+/* maximum number of items for AES and S/PDIF rates for above table */
+#define SND_RME_RATE_IDX_AES_SPDIF_NUM 12
+
+enum snd_rme_domain {
+ SND_RME_DOMAIN_SYSTEM,
+ SND_RME_DOMAIN_AES,
+ SND_RME_DOMAIN_SPDIF
+};
+
+enum snd_rme_clock_status {
+ SND_RME_CLOCK_NOLOCK,
+ SND_RME_CLOCK_LOCK,
+ SND_RME_CLOCK_SYNC
+};
+
+static int snd_rme_read_value(struct snd_usb_audio *chip,
+ unsigned int item,
+ u32 *value)
+{
+ struct usb_device *dev = chip->dev;
+ int err;
+
+ err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
+ item,
+ USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE,
+ 0, 0,
+ value, sizeof(*value));
+ if (err < 0)
+ dev_err(&dev->dev,
+ "unable to issue vendor read request %d (ret = %d)",
+ item, err);
+ return err;
+}
+
+static int snd_rme_get_status1(struct snd_kcontrol *kcontrol,
+ u32 *status1)
+{
+ struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol);
+ struct snd_usb_audio *chip = list->mixer->chip;
+ int err;
+
+ err = snd_usb_lock_shutdown(chip);
+ if (err < 0)
+ return err;
+ err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, status1);
+ snd_usb_unlock_shutdown(chip);
+ return err;
+}
+
+static int snd_rme_rate_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u32 status1;
+ u32 rate = 0;
+ int idx;
+ int err;
+
+ err = snd_rme_get_status1(kcontrol, &status1);
+ if (err < 0)
+ return err;
+ switch (kcontrol->private_value) {
+ case SND_RME_DOMAIN_SYSTEM:
+ idx = SND_RME_CLK_SYSTEM(status1);
+ if (idx < ARRAY_SIZE(snd_rme_rate_table))
+ rate = snd_rme_rate_table[idx];
+ break;
+ case SND_RME_DOMAIN_AES:
+ idx = SND_RME_CLK_AES(status1);
+ if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM)
+ rate = snd_rme_rate_table[idx];
+ break;
+ case SND_RME_DOMAIN_SPDIF:
+ idx = SND_RME_CLK_SPDIF(status1);
+ if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM)
+ rate = snd_rme_rate_table[idx];
+ break;
+ default:
+ return -EINVAL;
+ }
+ ucontrol->value.integer.value[0] = rate;
+ return 0;
+}
+
+static int snd_rme_sync_state_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u32 status1;
+ int idx = SND_RME_CLOCK_NOLOCK;
+ int err;
+
+ err = snd_rme_get_status1(kcontrol, &status1);
+ if (err < 0)
+ return err;
+ switch (kcontrol->private_value) {
+ case SND_RME_DOMAIN_AES: /* AES */
+ if (status1 & SND_RME_CLK_AES_SYNC)
+ idx = SND_RME_CLOCK_SYNC;
+ else if (status1 & SND_RME_CLK_AES_LOCK)
+ idx = SND_RME_CLOCK_LOCK;
+ break;
+ case SND_RME_DOMAIN_SPDIF: /* SPDIF */
+ if (status1 & SND_RME_CLK_SPDIF_SYNC)
+ idx = SND_RME_CLOCK_SYNC;
+ else if (status1 & SND_RME_CLK_SPDIF_LOCK)
+ idx = SND_RME_CLOCK_LOCK;
+ break;
+ default:
+ return -EINVAL;
+ }
+ ucontrol->value.enumerated.item[0] = idx;
+ return 0;
+}
+
+static int snd_rme_spdif_if_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u32 status1;
+ int err;
+
+ err = snd_rme_get_status1(kcontrol, &status1);
+ if (err < 0)
+ return err;
+ ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_IF(status1);
+ return 0;
+}
+
+static int snd_rme_spdif_format_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u32 status1;
+ int err;
+
+ err = snd_rme_get_status1(kcontrol, &status1);
+ if (err < 0)
+ return err;
+ ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_FORMAT(status1);
+ return 0;
+}
+
+static int snd_rme_sync_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u32 status1;
+ int err;
+
+ err = snd_rme_get_status1(kcontrol, &status1);
+ if (err < 0)
+ return err;
+ ucontrol->value.enumerated.item[0] = SND_RME_CLK_SYNC(status1);
+ return 0;
+}
+
+static int snd_rme_current_freq_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol);
+ struct snd_usb_audio *chip = list->mixer->chip;
+ u32 status1;
+ const u64 num = 104857600000000ULL;
+ u32 den;
+ unsigned int freq;
+ int err;
+
+ err = snd_usb_lock_shutdown(chip);
+ if (err < 0)
+ return err;
+ err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, &status1);
+ if (err < 0)
+ goto end;
+ err = snd_rme_read_value(chip, SND_RME_GET_CURRENT_FREQ, &den);
+ if (err < 0)
+ goto end;
+ freq = (den == 0) ? 0 : div64_u64(num, den);
+ freq <<= SND_RME_CLK_FREQMUL(status1);
+ ucontrol->value.integer.value[0] = freq;
+
+end:
+ snd_usb_unlock_shutdown(chip);
+ return err;
+}
+
+static int snd_rme_rate_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ switch (kcontrol->private_value) {
+ case SND_RME_DOMAIN_SYSTEM:
+ uinfo->value.integer.min = 32000;
+ uinfo->value.integer.max = 800000;
+ break;
+ case SND_RME_DOMAIN_AES:
+ case SND_RME_DOMAIN_SPDIF:
+ default:
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 200000;
+ }
+ uinfo->value.integer.step = 0;
+ return 0;
+}
+
+static int snd_rme_sync_state_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char *const sync_states[] = {
+ "No Lock", "Lock", "Sync"
+ };
+
+ return snd_ctl_enum_info(uinfo, 1,
+ ARRAY_SIZE(sync_states), sync_states);
+}
+
+static int snd_rme_spdif_if_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char *const spdif_if[] = {
+ "Coaxial", "Optical"
+ };
+
+ return snd_ctl_enum_info(uinfo, 1,
+ ARRAY_SIZE(spdif_if), spdif_if);
+}
+
+static int snd_rme_spdif_format_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char *const optical_type[] = {
+ "Consumer", "Professional"
+ };
+
+ return snd_ctl_enum_info(uinfo, 1,
+ ARRAY_SIZE(optical_type), optical_type);
+}
+
+static int snd_rme_sync_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char *const sync_sources[] = {
+ "Internal", "AES", "SPDIF", "Internal"
+ };
+
+ return snd_ctl_enum_info(uinfo, 1,
+ ARRAY_SIZE(sync_sources), sync_sources);
+}
+
+static struct snd_kcontrol_new snd_rme_controls[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "AES Rate",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_rme_rate_info,
+ .get = snd_rme_rate_get,
+ .private_value = SND_RME_DOMAIN_AES
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "AES Sync",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_rme_sync_state_info,
+ .get = snd_rme_sync_state_get,
+ .private_value = SND_RME_DOMAIN_AES
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SPDIF Rate",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_rme_rate_info,
+ .get = snd_rme_rate_get,
+ .private_value = SND_RME_DOMAIN_SPDIF
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SPDIF Sync",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_rme_sync_state_info,
+ .get = snd_rme_sync_state_get,
+ .private_value = SND_RME_DOMAIN_SPDIF
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SPDIF Interface",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_rme_spdif_if_info,
+ .get = snd_rme_spdif_if_get,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SPDIF Format",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_rme_spdif_format_info,
+ .get = snd_rme_spdif_format_get,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Sync Source",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_rme_sync_source_info,
+ .get = snd_rme_sync_source_get
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "System Rate",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_rme_rate_info,
+ .get = snd_rme_rate_get,
+ .private_value = SND_RME_DOMAIN_SYSTEM
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Current Frequency",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_rme_rate_info,
+ .get = snd_rme_current_freq_get
+ }
+};
+
+static int snd_rme_controls_create(struct usb_mixer_interface *mixer)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(snd_rme_controls); ++i) {
+ err = add_single_ctl_with_resume(mixer, 0,
+ NULL,
+ &snd_rme_controls[i],
+ NULL);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
{
int err = 0;
@@ -1904,6 +2279,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */
err = dell_dock_mixer_init(mixer);
break;
+
+ case USB_ID(0x2a39, 0x3fd2): /* RME ADI-2 Pro */
+ case USB_ID(0x2a39, 0x3fd3): /* RME ADI-2 DAC */
+ case USB_ID(0x2a39, 0x3fd4): /* RME */
+ err = snd_rme_controls_create(mixer);
+ break;
}
return err;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 08aa78007020..849953e5775c 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3346,19 +3346,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
.ifnum = 0,
.type = QUIRK_AUDIO_STANDARD_MIXER,
},
- /* Capture */
- {
- .ifnum = 1,
- .type = QUIRK_IGNORE_INTERFACE,
- },
/* Playback */
{
- .ifnum = 2,
+ .ifnum = 1,
.type = QUIRK_AUDIO_FIXED_ENDPOINT,
.data = &(const struct audioformat) {
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.channels = 2,
- .iface = 2,
+ .iface = 1,
.altsetting = 1,
.altset_idx = 1,
.attributes = UAC_EP_CS_ATTR_FILL_MAX |
diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c
index fa7dca5a68c8..83d76c345940 100644
--- a/sound/x86/intel_hdmi_audio.c
+++ b/sound/x86/intel_hdmi_audio.c
@@ -30,7 +30,6 @@
#include <linux/pm_runtime.h>
#include <linux/dma-mapping.h>
#include <linux/delay.h>
-#include <asm/set_memory.h>
#include <sound/core.h>
#include <sound/asoundef.h>
#include <sound/pcm.h>
@@ -1141,8 +1140,7 @@ static int had_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
struct snd_intelhad *intelhaddata;
- unsigned long addr;
- int pages, buf_size, retval;
+ int buf_size, retval;
intelhaddata = snd_pcm_substream_chip(substream);
buf_size = params_buffer_bytes(hw_params);
@@ -1151,17 +1149,6 @@ static int had_pcm_hw_params(struct snd_pcm_substream *substream,
return retval;
dev_dbg(intelhaddata->dev, "%s:allocated memory = %d\n",
__func__, buf_size);
- /* mark the pages as uncached region */
- addr = (unsigned long) substream->runtime->dma_area;
- pages = (substream->runtime->dma_bytes + PAGE_SIZE - 1) / PAGE_SIZE;
- retval = set_memory_uc(addr, pages);
- if (retval) {
- dev_err(intelhaddata->dev, "set_memory_uc failed.Error:%d\n",
- retval);
- return retval;
- }
- memset(substream->runtime->dma_area, 0, buf_size);
-
return retval;
}
@@ -1171,21 +1158,11 @@ static int had_pcm_hw_params(struct snd_pcm_substream *substream,
static int had_pcm_hw_free(struct snd_pcm_substream *substream)
{
struct snd_intelhad *intelhaddata;
- unsigned long addr;
- u32 pages;
intelhaddata = snd_pcm_substream_chip(substream);
had_do_reset(intelhaddata);
- /* mark back the pages as cached/writeback region before the free */
- if (substream->runtime->dma_area != NULL) {
- addr = (unsigned long) substream->runtime->dma_area;
- pages = (substream->runtime->dma_bytes + PAGE_SIZE - 1) /
- PAGE_SIZE;
- set_memory_wb(addr, pages);
- return snd_pcm_lib_free_pages(substream);
- }
- return 0;
+ return snd_pcm_lib_free_pages(substream);
}
/*
@@ -1860,7 +1837,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev)
* try to allocate 600k buffer as default which is large enough
*/
snd_pcm_lib_preallocate_pages_for_all(pcm,
- SNDRV_DMA_TYPE_DEV, NULL,
+ SNDRV_DMA_TYPE_DEV_UC, NULL,
HAD_DEFAULT_BUFFER, HAD_MAX_BUFFER);
/* create controls */
diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c
index 129180e17db1..2cbd9679aca1 100644
--- a/sound/xen/xen_snd_front_alsa.c
+++ b/sound/xen/xen_snd_front_alsa.c
@@ -637,31 +637,31 @@ static int alsa_pb_fill_silence(struct snd_pcm_substream *substream,
* to know when the buffer can be transferred to the backend.
*/
-static struct snd_pcm_ops snd_drv_alsa_playback_ops = {
- .open = alsa_open,
- .close = alsa_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = alsa_hw_params,
- .hw_free = alsa_hw_free,
- .prepare = alsa_prepare,
- .trigger = alsa_trigger,
- .pointer = alsa_pointer,
- .copy_user = alsa_pb_copy_user,
- .copy_kernel = alsa_pb_copy_kernel,
- .fill_silence = alsa_pb_fill_silence,
+static const struct snd_pcm_ops snd_drv_alsa_playback_ops = {
+ .open = alsa_open,
+ .close = alsa_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = alsa_hw_params,
+ .hw_free = alsa_hw_free,
+ .prepare = alsa_prepare,
+ .trigger = alsa_trigger,
+ .pointer = alsa_pointer,
+ .copy_user = alsa_pb_copy_user,
+ .copy_kernel = alsa_pb_copy_kernel,
+ .fill_silence = alsa_pb_fill_silence,
};
-static struct snd_pcm_ops snd_drv_alsa_capture_ops = {
- .open = alsa_open,
- .close = alsa_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = alsa_hw_params,
- .hw_free = alsa_hw_free,
- .prepare = alsa_prepare,
- .trigger = alsa_trigger,
- .pointer = alsa_pointer,
- .copy_user = alsa_cap_copy_user,
- .copy_kernel = alsa_cap_copy_kernel,
+static const struct snd_pcm_ops snd_drv_alsa_capture_ops = {
+ .open = alsa_open,
+ .close = alsa_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = alsa_hw_params,
+ .hw_free = alsa_hw_free,
+ .prepare = alsa_prepare,
+ .trigger = alsa_trigger,
+ .pointer = alsa_pointer,
+ .copy_user = alsa_cap_copy_user,
+ .copy_kernel = alsa_cap_copy_kernel,
};
static int new_pcm_instance(struct xen_snd_front_card_info *card_info,