July 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: rt5682: cancel jack_detect_work if hs_jack is set to null even soundwire mode
by Oder Chiou 02 Jul '20

02 Jul '20

Base on https://patchwork.kernel.org/patch/11237953/ Soundwire mode also should follow it. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5682.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index e9514c81b9ba..de40b6cd16cf 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -992,16 +992,17 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, rt5682->hs_jack = hs_jack; - if (!rt5682->is_sdw) { - if (!hs_jack) { - regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, - RT5682_JD1_EN_MASK, RT5682_JD1_DIS); - regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, - RT5682_POW_JDH | RT5682_POW_JDL, 0); - cancel_delayed_work_sync(&rt5682->jack_detect_work); - return 0; - } + if (!hs_jack) { + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK, RT5682_JD1_DIS); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_JDH | RT5682_POW_JDL, 0); + cancel_delayed_work_sync(&rt5682->jack_detect_work); + return 0; + } + + if (!rt5682->is_sdw) { switch (rt5682->pdata.jd_src) { case RT5682_JD1: snd_soc_component_update_bits(component, -- 2.27.0

2 1

[PATCH 0/2] ASoC: Intel: atom: fix kernel-doc and W=1 warnings
by Pierre-Louis Bossart 02 Jul '20

02 Jul '20

Yet another series to fix broken kernel-doc and mark unused variables as such. Pierre-Louis Bossart (2): ASoC: Intel: atom: fix kernel-doc ASoC: Intel: atom: fix 'defined but not used' warning sound/soc/intel/atom/sst-atom-controls.c | 65 ++++++++++++++++-------- sound/soc/intel/atom/sst/sst_loader.c | 4 ++ sound/soc/intel/atom/sst/sst_stream.c | 43 +++++++++------- 3 files changed, 74 insertions(+), 38 deletions(-) base-commit: a2b782d59c57cb6f673dbb4804ffc500336d7a54 -- 2.25.1

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[PATCH] ASoC: amd: Enable interrupt in dma_open
by Akshu Agrawal 02 Jul '20

02 Jul '20

Fixes interrupt enable condition check with which now interrupt gets enabled in dma_open. Prior to this patch it was getting enabled in runtime_resume only. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index e6386de20ac7..5bd458e0fe31 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -238,7 +238,7 @@ static int acp3x_dma_open(struct snd_soc_component *component, } if (!adata->play_stream && !adata->capture_stream && - adata->i2ssp_play_stream && !adata->i2ssp_capture_stream) + !adata->i2ssp_play_stream && !adata->i2ssp_capture_stream) rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); i2s_data->acp3x_base = adata->acp3x_base; -- 2.20.1

2 1

[PATCH] ASoC: dt-bindings: keembay-i2s: Fix reg descriptions
by Fabio Estevam 02 Jul '20

02 Jul '20

intel,keembay-i2s has two register regions: - I2S registers - I2S gen configuration Describe these regions accordingly to fix the following warning seen with 'make dt_binding_check': Documentation/devicetree/bindings/sound/intel,keembay-i2s.example.dt.yaml: example-0: i2s@20140000:reg:0: [538181632, 512, 539623588, 4] is too long Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- .../devicetree/bindings/sound/intel,keembay-i2s.yaml | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml index 175e4fb0c315..2e0bbc1c868a 100644 --- a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml @@ -23,7 +23,8 @@ properties: reg: items: - - description: I2S configuration + - description: I2S registers + - description: I2S gen configuration reg-names: items: @@ -60,7 +61,8 @@ examples: i2s3: i2s@20140000 { compatible = "intel,keembay-i2s"; #sound-dai-cells = <0>; - reg = <0x20140000 0x200 0x202a00a4 0x4>; + reg = <0x20140000 0x200>, /* I2S registers */ + <0x202a00a4 0x4>; /* I2S gen configuration */ reg-names = "i2s-regs", "i2s_gen_cfg"; interrupts = <GIC_SPI 120 IRQ_TYPE_LEVEL_HIGH>; clock-names = "osc", "apb_clk"; -- 2.17.1

2 1

[PATCH] ASoC: fsl_sai: Refine regcache usage with pm runtime
by Shengjiu Wang 02 Jul '20

02 Jul '20

When there is dedicated power domain bound with device, after probing the power will be disabled, then registers are not accessible in fsl_sai_dai_probe(), so regcache only need to be enabled in end of probe() and regcache_mark_dirty should be moved to pm runtime resume callback function. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 9d436b0c5718..a22562f2df47 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1016,6 +1016,7 @@ static int fsl_sai_probe(struct platform_device *pdev) platform_set_drvdata(pdev, sai); pm_runtime_enable(&pdev->dev); + regcache_cache_only(sai->regmap, true); ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, &fsl_sai_dai, 1); @@ -1107,7 +1108,6 @@ static int fsl_sai_runtime_suspend(struct device *dev) clk_disable_unprepare(sai->bus_clk); regcache_cache_only(sai->regmap, true); - regcache_mark_dirty(sai->regmap); return 0; } @@ -1137,6 +1137,7 @@ static int fsl_sai_runtime_resume(struct device *dev) } regcache_cache_only(sai->regmap, false); + regcache_mark_dirty(sai->regmap); regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), FSL_SAI_CSR_SR); regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), FSL_SAI_CSR_SR); usleep_range(1000, 2000); -- 2.21.0

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[PATCH] ASoC: rockchip: add format and rate constraints on rk3399
by Yu-Hsuan Hsu 02 Jul '20

02 Jul '20

S8 and S24 formats does not work on this machine driver so force to use S16_LE instead. In addition, add constraint to limit the max value of rate because the rate higher than 96000(172000, 192000) is not stable either. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/rockchip/rk3399_gru_sound.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index f45e5aaa4b302..9539b0d024fed 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -219,19 +219,32 @@ static int rockchip_sound_dmic_hw_params(struct snd_pcm_substream *substream, return 0; } +static int rockchip_sound_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + return snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, + 8000, 96000); +} + static const struct snd_soc_ops rockchip_sound_max98357a_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_max98357a_hw_params, }; static const struct snd_soc_ops rockchip_sound_rt5514_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_rt5514_hw_params, }; static const struct snd_soc_ops rockchip_sound_da7219_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_da7219_hw_params, }; static const struct snd_soc_ops rockchip_sound_dmic_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_dmic_hw_params, }; -- 2.27.0.212.ge8ba1cc988-goog

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[PATCH v4 0/3] ASoC: ti: Add support for audio on J721e EVM
by Peter Ujfalusi 02 Jul '20

02 Jul '20

Hi, Changes since v3: - Fix the single clock source handling and typo Changes since v2: - DT binding: - use proper (?) patch subject for the binding docuemtn patch - drop pll4 and pll15 from DT - driver should check the rate via clk_get_parent. If it is not available (as it is not currently) then use the match_data provided rates. - add simple explanation for the clocking setup - Use descriptive names for clocks: cpb/ivi-mcasp-auxclk and cpb/ivi-codec-scki - dt_binding_check shows no errors/warnings - ASoC machine driver: - Try to read the PLL4/15 rate with clk API (parent of the two clock divider) if it is not available then use the match_data provided numbers. - Support for single PLL setup Changes since v1: - Fixed DT binding documentation errors - Rebased on ASoC head and updated the driver to compile and work This series adds support for the analog audio setup on the j721e EVM. The audio setup of the EVM is: Common Processor Board (CPB): McASP10 <-> pcm3168a Infotainment Expansion Board (IVI): McASP0 <-> 2x pcm3168a Both CPB and IVI wired in parallel serializer setup. The first patch adds the stream_name for McASP driver as it is needed in multicodec (and would be needed in DPCM) setup for proper DAPM handling. The second patch adds two DT schema, one for the cpb and one for the cpb+ivi card. Regards, Peter --- Peter Ujfalusi (3): ASoC: ti: davinci-mcasp: Specify stream_name for playback/capture ASoC: dt-bindings: Add documentation for TI j721e EVM (CPB and IVI) ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI) .../bindings/sound/ti,j721e-cpb-audio.yaml | 95 ++ .../sound/ti,j721e-cpb-ivi-audio.yaml | 150 +++ sound/soc/ti/Kconfig | 8 + sound/soc/ti/Makefile | 2 + sound/soc/ti/davinci-mcasp.c | 3 + sound/soc/ti/j721e-evm.c | 896 ++++++++++++++++++ 6 files changed, 1154 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml create mode 100644 sound/soc/ti/j721e-evm.c -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[PATCH v3] ASoC: fsl_asrc: Add an option to select internal ratio mode
by Shengjiu Wang 02 Jul '20

02 Jul '20

The ASRC not only supports ideal ratio mode, but also supports internal ratio mode. For internal rato mode, the rate of clock source should be divided with no remainder by sample rate, otherwise there is sound distortion. Add function fsl_asrc_select_clk() to find proper clock source for internal ratio mode, if the clock source is available then internal ratio mode will be selected. With change, the ideal ratio mode is not the only option for user. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v3 - convert fsl_asrc_select_clk to void type changes in v2 - update according to Nicolin's comments sound/soc/fsl/fsl_asrc.c | 46 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 44 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 95f6a9617b0b..462ce9f9ab48 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -582,11 +582,51 @@ static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, &fsl_asrc_rate_constraints); } +/* Select proper clock source for internal ratio mode */ +static void fsl_asrc_select_clk(struct fsl_asrc_priv *asrc_priv, + struct fsl_asrc_pair *pair, + int in_rate, + int out_rate) +{ + struct fsl_asrc_pair_priv *pair_priv = pair->private; + struct asrc_config *config = pair_priv->config; + int rate[2], select_clk[2]; /* Array size 2 means IN and OUT */ + int clk_rate, clk_index; + int i = 0, j = 0; + + rate[IN] = in_rate; + rate[OUT] = out_rate; + + /* Select proper clock source for internal ratio mode */ + for (j = 0; j < 2; j++) { + for (i = 0; i < ASRC_CLK_MAP_LEN; i++) { + clk_index = asrc_priv->clk_map[j][i]; + clk_rate = clk_get_rate(asrc_priv->asrck_clk[clk_index]); + /* Only match a perfect clock source with no remainder */ + if (clk_rate != 0 && (clk_rate / rate[j]) <= 1024 && + (clk_rate % rate[j]) == 0) + break; + } + + select_clk[j] = i; + } + + /* Switch to ideal ratio mode if there is no proper clock source */ + if (select_clk[IN] == ASRC_CLK_MAP_LEN || select_clk[OUT] == ASRC_CLK_MAP_LEN) { + select_clk[IN] = INCLK_NONE; + select_clk[OUT] = OUTCLK_ASRCK1_CLK; + } + + config->inclk = select_clk[IN]; + config->outclk = select_clk[OUT]; +} + static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct fsl_asrc *asrc = snd_soc_dai_get_drvdata(dai); + struct fsl_asrc_priv *asrc_priv = asrc->private; struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; struct fsl_asrc_pair_priv *pair_priv = pair->private; @@ -605,8 +645,6 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.pair = pair->index; config.channel_num = channels; - config.inclk = INCLK_NONE; - config.outclk = OUTCLK_ASRCK1_CLK; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config.input_format = params_format(params); @@ -620,6 +658,10 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } + fsl_asrc_select_clk(asrc_priv, pair, + config.input_sample_rate, + config.output_sample_rate); + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); -- 2.21.0

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[PATCH] riscv: asm/gdb_xml.h: use __maybe_used to make W=1 warnings go away
by Pierre-Louis Bossart 01 Jul '20

01 Jul '20

0day/kbuild reports warnings with the ASoC codecs compiled with W=1. In file included from arch/riscv/include/asm/kgdb.h:109, from include/linux/kgdb.h:20, from include/linux/fb.h:5, from include/drm/drm_crtc.h:31, from sound/soc/codecs/hdmi-codec.c:19: arch/riscv/include/asm/gdb_xml.h:23:19: warning: 'riscv_gdb_stub_cpuxml' defined but not used [-Wunused-const-variable=] 23 | static const char riscv_gdb_stub_cpuxml[2048] = | ^~~~~~~~~~~~~~~~~~~~~ arch/riscv/include/asm/gdb_xml.h:16:19: warning: 'riscv_gdb_stub_target_desc' defined but not used [-Wunused-const-variable=] 16 | static const char riscv_gdb_stub_target_desc[256] = | ^~~~~~~~~~~~~~~~~~~~~~~~~~ arch/riscv/include/asm/gdb_xml.h:13:19: warning: 'gdb_xfer_read_cpuxml' defined but not used [-Wunused-const-variable=] 13 | static const char gdb_xfer_read_cpuxml[39] = | ^~~~~~~~~~~~~~~~~~~~ arch/riscv/include/asm/gdb_xml.h:10:19: warning: 'gdb_xfer_read_target' defined but not used [-Wunused-const-variable=] 10 | static const char gdb_xfer_read_target[31] = "qXfer:features:read:target.xml:"; | ^~~~~~~~~~~~~~~~~~~~ arch/riscv/include/asm/gdb_xml.h:7:19: warning: 'riscv_gdb_stub_feature' defined but not used [-Wunused-const-variable=] 7 | static const char riscv_gdb_stub_feature[64] = | ^~~~~~~~~~~~~~~~~~~~~~ Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- I don't know if this is the right way of solving this issue but this error is now consistently thrown in kbuild compile-test reports w/ W=1. arch/riscv/include/asm/gdb_xml.h | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/arch/riscv/include/asm/gdb_xml.h b/arch/riscv/include/asm/gdb_xml.h index 041b45f5b997..c28fc1a1d230 100644 --- a/arch/riscv/include/asm/gdb_xml.h +++ b/arch/riscv/include/asm/gdb_xml.h @@ -4,23 +4,23 @@ #define __ASM_GDB_XML_H_ #define kgdb_arch_gdb_stub_feature riscv_gdb_stub_feature -static const char riscv_gdb_stub_feature[64] = +static __maybe_unused const char riscv_gdb_stub_feature[64] = "PacketSize=800;qXfer:features:read+;"; -static const char gdb_xfer_read_target[31] = "qXfer:features:read:target.xml:"; +static __maybe_unused const char gdb_xfer_read_target[31] = "qXfer:features:read:target.xml:"; #ifdef CONFIG_64BIT -static const char gdb_xfer_read_cpuxml[39] = +static __maybe_unused const char gdb_xfer_read_cpuxml[39] = "qXfer:features:read:riscv-64bit-cpu.xml"; -static const char riscv_gdb_stub_target_desc[256] = +static __maybe_unused const char riscv_gdb_stub_target_desc[256] = "l<?xml version=\"1.0\"?>" "<!DOCTYPE target SYSTEM \"gdb-target.dtd\">" "<target>" "<xi:include href=\"riscv-64bit-cpu.xml\"/>" "</target>"; -static const char riscv_gdb_stub_cpuxml[2048] = +static __maybe_unused const char riscv_gdb_stub_cpuxml[2048] = "l<?xml version=\"1.0\"?>" "<!DOCTYPE feature SYSTEM \"gdb-target.dtd\">" "<feature name=\"org.gnu.gdb.riscv.cpu\">" -- 2.25.1

1 0

[PATCH] ASoC: Intel: bxt-da7219-max98357a: support MAX98390 speaker amp
by Brent Lu 01 Jul '20

01 Jul '20

Support MAX98390 speaker amplifier on cometlake platform. Driver now detects amplifier type in the probe function and installs corresponding controls and DAPM widgets/routes in the late_probe function. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 20 ++++ sound/soc/intel/boards/bxt_da7219_max98357a.c | 129 ++++++++++++++++++++-- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 4 +- 3 files changed, 139 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 3d820e1..b3b863e 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -291,9 +291,17 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC select SND_SOC_DMIC select SND_SOC_HDAC_HDMI +config SND_SOC_INTEL_DA7219_MAX98390_GENERIC + tristate + select SND_SOC_DA7219 + select SND_SOC_MAX98390 + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON tristate select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC + select SND_SOC_INTEL_DA7219_MAX98390_GENERIC if SND_SOC_INTEL_APL @@ -309,6 +317,18 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +config SND_SOC_INTEL_BXT_DA7219_MAX98390_MACH + tristate "Broxton with DA7219 and MAX98390 in I2S Mode" + depends on I2C && ACPI && GPIOLIB + depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI + select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON + help + This adds support for ASoC machine driver for Broxton-P platforms + with DA7219 + MAX98390 I2S audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + config SND_SOC_INTEL_BXT_RT298_MACH tristate "Broxton with RT298 I2S mode" depends on I2C && ACPI && GPIOLIB diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 44016c1..12f07e1 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -25,9 +25,14 @@ #define BXT_DIALOG_CODEC_DAI "da7219-hifi" #define BXT_MAXIM_CODEC_DAI "HiFi" +#define MAX98390_DEV0_NAME "i2c-MX98390:00" +#define MAX98390_DEV1_NAME "i2c-MX98390:01" #define DUAL_CHANNEL 2 #define QUAD_CHANNEL 4 +#define SPKAMP_MAX98357A 1 +#define SPKAMP_MAX98390 2 + static struct snd_soc_jack broxton_headset; static struct snd_soc_jack broxton_hdmi[3]; @@ -40,6 +45,7 @@ struct bxt_hdmi_pcm { struct bxt_card_private { struct list_head hdmi_pcm_list; bool common_hdmi_codec_drv; + int spkamp; }; enum { @@ -85,13 +91,20 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, static const struct snd_kcontrol_new broxton_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static const struct snd_kcontrol_new max98357a_controls[] = { SOC_DAPM_PIN_SWITCH("Spk"), }; +static const struct snd_kcontrol_new max98390_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + static const struct snd_soc_dapm_widget broxton_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), SND_SOC_DAPM_MIC("SoC DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), @@ -100,14 +113,20 @@ static const struct snd_soc_dapm_widget broxton_widgets[] = { platform_clock_control, SND_SOC_DAPM_POST_PMD|SND_SOC_DAPM_PRE_PMU), }; +static const struct snd_soc_dapm_widget max98357a_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_widget max98390_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + static const struct snd_soc_dapm_route audio_map[] = { /* HP jack connectors - unknown if we have jack detection */ {"Headphone Jack", NULL, "HPL"}, {"Headphone Jack", NULL, "HPR"}, - /* speaker */ - {"Spk", NULL, "Speaker"}, - /* other jacks */ {"MIC", NULL, "Headset Mic"}, @@ -134,6 +153,17 @@ static const struct snd_soc_dapm_route audio_map[] = { { "Headset Mic", NULL, "Platform Clock" }, }; +static const struct snd_soc_dapm_route max98357a_routes[] = { + /* speaker */ + {"Spk", NULL, "Speaker"}, +}; + +static const struct snd_soc_dapm_route max98390_routes[] = { + /* Speaker */ + {"Left Spk", NULL, "Left BE_OUT"}, + {"Right Spk", NULL, "Right BE_OUT"}, +}; + static const struct snd_soc_dapm_route broxton_map[] = { {"HiFi Playback", NULL, "ssp5 Tx"}, {"ssp5 Tx", NULL, "codec0_out"}, @@ -404,6 +434,10 @@ SND_SOC_DAILINK_DEF(ssp5_pin, SND_SOC_DAILINK_DEF(ssp5_codec, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", BXT_MAXIM_CODEC_DAI))); +SND_SOC_DAILINK_DEF(max98390_codec, + DAILINK_COMP_ARRAY( + /* Left */ COMP_CODEC(MAX98390_DEV0_NAME, "max98390-aif1"), + /* Right */ COMP_CODEC(MAX98390_DEV1_NAME, "max98390-aif1"))); SND_SOC_DAILINK_DEF(ssp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); @@ -601,15 +635,69 @@ static struct snd_soc_dai_link broxton_dais[] = { }, }; +static struct snd_soc_codec_conf max98390_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(MAX98390_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(MAX98390_DEV1_NAME), + .name_prefix = "Right", + }, +}; + #define NAME_SIZE 32 static int bxt_card_late_probe(struct snd_soc_card *card) { struct bxt_card_private *ctx = snd_soc_card_get_drvdata(card); struct bxt_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; - int err, i = 0; + const struct snd_kcontrol_new *controls; + const struct snd_soc_dapm_widget *widgets; + const struct snd_soc_dapm_route *routes; + int num_controls, num_widgets, num_routes, ret, i = 0; char jack_name[NAME_SIZE]; + switch (ctx->spkamp) { + case SPKAMP_MAX98357A: + controls = max98357a_controls; + num_controls = ARRAY_SIZE(max98357a_controls); + widgets = max98357a_widgets; + num_widgets = ARRAY_SIZE(max98357a_widgets); + routes = max98357a_routes; + num_routes = ARRAY_SIZE(max98357a_routes); + break; + case SPKAMP_MAX98390: + controls = max98390_controls; + num_controls = ARRAY_SIZE(max98390_controls); + widgets = max98390_widgets; + num_widgets = ARRAY_SIZE(max98390_widgets); + routes = max98390_routes; + num_routes = ARRAY_SIZE(max98390_routes); + break; + default: + dev_err(card->dev, "Invalid speaker amplifier %d\n", ctx->spkamp); + break; + } + + ret = snd_soc_dapm_new_controls(&card->dapm, widgets, num_widgets); + if (ret) { + dev_err(card->dev, "Fail to new widgets\n"); + return ret; + } + + ret = snd_soc_add_card_controls(card, controls, num_controls); + if (ret) { + dev_err(card->dev, "Fail to add controls\n"); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, routes, num_routes); + if (ret) { + dev_err(card->dev, "Fail to add routes\n"); + return ret; + } + if (soc_intel_is_glk()) snd_soc_dapm_add_routes(&card->dapm, gemini_map, ARRAY_SIZE(gemini_map)); @@ -631,17 +719,17 @@ static int bxt_card_late_probe(struct snd_soc_card *card) 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, + ret = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &broxton_hdmi[i], NULL, 0); - if (err) - return err; + if (ret) + return ret; - err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + ret = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &broxton_hdmi[i]); - if (err < 0) - return err; + if (ret < 0) + return ret; i++; } @@ -678,6 +766,11 @@ static int broxton_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + if (acpi_dev_present("MX98390", NULL, -1)) + ctx->spkamp = SPKAMP_MAX98390; + else + ctx->spkamp = SPKAMP_MAX98357A; + broxton_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&broxton_audio_card, ctx); if (soc_intel_is_glk()) { @@ -702,7 +795,13 @@ static int broxton_audio_probe(struct platform_device *pdev) } else if (soc_intel_is_cml()) { unsigned int i; - broxton_audio_card.name = "cmlda7219max"; + if (ctx->spkamp == SPKAMP_MAX98390) { + broxton_audio_card.name = "cml_max98390_da7219"; + + broxton_audio_card.codec_conf = max98390_codec_confs; + broxton_audio_card.num_configs = ARRAY_SIZE(max98390_codec_confs); + } else + broxton_audio_card.name = "cmlda7219max"; for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) { /* MAXIM_CODEC is connected to SSP1. */ @@ -710,6 +809,11 @@ static int broxton_audio_probe(struct platform_device *pdev) BXT_MAXIM_CODEC_DAI)) { broxton_dais[i].name = "SSP1-Codec"; broxton_dais[i].cpus->dai_name = "SSP1 Pin"; + + if (ctx->spkamp == SPKAMP_MAX98390) { + broxton_dais[i].codecs = max98390_codec; + broxton_dais[i].num_codecs = ARRAY_SIZE(max98390_codec); + } } /* DIALOG_CODEC is connected to SSP0 */ else if (!strcmp(broxton_dais[i].codecs->dai_name, @@ -759,6 +863,7 @@ MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); MODULE_AUTHOR("Conrad Cooke <conrad.cooke(a)intel.com>"); MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); +MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:bxt_da7219_max98357a"); MODULE_ALIAS("platform:glk_da7219_max98357a"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index cdea0c0..7dc9ba9 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -15,8 +15,8 @@ static struct snd_soc_acpi_codecs rt1011_spk_codecs = { }; static struct snd_soc_acpi_codecs max98357a_spk_codecs = { - .num_codecs = 1, - .codecs = {"MX98357A"} + .num_codecs = 2, + .codecs = {"MX98357A", "MX98390"} }; /* -- 2.7.4

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