March 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: cpcap: replace codec to component
by Kuninori Morimoto 28 Mar '18

28 Mar '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 0 -> .idle_bias_on = 1 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/cpcap.c | 152 +++++++++++++++++++++++------------------------ 1 file changed, 73 insertions(+), 79 deletions(-) diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c index aedb267..d7f05b3 100644 --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -243,7 +243,7 @@ enum cpcap_dai { }; struct cpcap_audio { - struct snd_soc_codec *codec; + struct snd_soc_component *component; struct regmap *regmap; u16 vendor; @@ -256,8 +256,8 @@ struct cpcap_audio { static int cpcap_st_workaround(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); int err = 0; /* Only CPCAP from ST requires workaround */ @@ -357,8 +357,8 @@ static SOC_ENUM_SINGLE_EXT_DECL(cpcap_input_right_mux_enum, static int cpcap_output_mux_get_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int shift = e->shift_l; int reg_voice, reg_hifi, reg_ext, status; @@ -400,8 +400,8 @@ static int cpcap_output_mux_get_enum(struct snd_kcontrol *kcontrol, static int cpcap_output_mux_put_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -445,8 +445,8 @@ static int cpcap_output_mux_put_enum(struct snd_kcontrol *kcontrol, static int cpcap_input_right_mux_get_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); int regval, mask; int err; @@ -484,8 +484,8 @@ static int cpcap_input_right_mux_get_enum(struct snd_kcontrol *kcontrol, static int cpcap_input_right_mux_put_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -529,8 +529,8 @@ static int cpcap_input_right_mux_put_enum(struct snd_kcontrol *kcontrol, static int cpcap_input_left_mux_get_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); int regval, mask; int err; @@ -560,8 +560,8 @@ static int cpcap_input_left_mux_get_enum(struct snd_kcontrol *kcontrol, static int cpcap_input_left_mux_put_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -969,13 +969,13 @@ static int cpcap_set_sysclk(struct cpcap_audio *cpcap, enum cpcap_dai dai, clkidshift = CPCAP_BIT_CLK_IN_SEL; break; default: - dev_err(cpcap->codec->dev, "invalid DAI: %d", dai); + dev_err(cpcap->component->dev, "invalid DAI: %d", dai); return -EINVAL; } /* setup clk id */ if (clk_id < 0 || clk_id > 1) { - dev_err(cpcap->codec->dev, "invalid clk id %d", clk_id); + dev_err(cpcap->component->dev, "invalid clk id %d", clk_id); return -EINVAL; } err = regmap_update_bits(cpcap->regmap, clkidreg, BIT(clkidshift), @@ -1015,7 +1015,7 @@ static int cpcap_set_sysclk(struct cpcap_audio *cpcap, enum cpcap_dai dai, clkfreqval = 0x06 << clkfreqshift; break; default: - dev_err(cpcap->codec->dev, "unsupported freq %u", freq); + dev_err(cpcap->component->dev, "unsupported freq %u", freq); return -EINVAL; } @@ -1035,7 +1035,7 @@ static int cpcap_set_sysclk(struct cpcap_audio *cpcap, enum cpcap_dai dai, static int cpcap_set_samprate(struct cpcap_audio *cpcap, enum cpcap_dai dai, int samplerate) { - struct snd_soc_codec *codec = cpcap->codec; + struct snd_soc_component *component = cpcap->component; u16 sampreg, sampmask, sampshift, sampval, sampreset; int err, sampreadval; @@ -1053,7 +1053,7 @@ static int cpcap_set_samprate(struct cpcap_audio *cpcap, enum cpcap_dai dai, BIT(CPCAP_BIT_CDC_CLOCK_TREE_RESET); break; default: - dev_err(codec->dev, "invalid DAI: %d", dai); + dev_err(component->dev, "invalid DAI: %d", dai); return -EINVAL; } @@ -1087,7 +1087,7 @@ static int cpcap_set_samprate(struct cpcap_audio *cpcap, enum cpcap_dai dai, sampval = 0x0 << sampshift; break; default: - dev_err(codec->dev, "unsupported samplerate %d", samplerate); + dev_err(component->dev, "unsupported samplerate %d", samplerate); return -EINVAL; } err = regmap_update_bits(cpcap->regmap, sampreg, @@ -1103,7 +1103,7 @@ static int cpcap_set_samprate(struct cpcap_audio *cpcap, enum cpcap_dai dai, return err; if (sampreadval & sampreset) { - dev_err(codec->dev, "reset self-clear failed: %04x", + dev_err(component->dev, "reset self-clear failed: %04x", sampreadval); return -EIO; } @@ -1115,20 +1115,20 @@ static int cpcap_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_codec *codec = dai->codec; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); int rate = params_rate(params); - dev_dbg(codec->dev, "HiFi setup HW params: rate=%d", rate); + dev_dbg(component->dev, "HiFi setup HW params: rate=%d", rate); return cpcap_set_samprate(cpcap, CPCAP_DAI_HIFI, rate); } static int cpcap_hifi_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); - struct device *dev = codec->dev; + struct snd_soc_component *component = codec_dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); + struct device *dev = component->dev; dev_dbg(dev, "HiFi setup sysclk: clk_id=%u, freq=%u", clk_id, freq); return cpcap_set_sysclk(cpcap, CPCAP_DAI_HIFI, clk_id, freq); @@ -1137,9 +1137,9 @@ static int cpcap_hifi_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, static int cpcap_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); - struct device *dev = codec->dev; + struct snd_soc_component *component = codec_dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); + struct device *dev = component->dev; static const u16 reg = CPCAP_REG_SDACDI; static const u16 mask = BIT(CPCAP_BIT_SMB_ST_DAC) | @@ -1218,8 +1218,8 @@ static int cpcap_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai, static int cpcap_hifi_set_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); static const u16 reg = CPCAP_REG_RXSDOA; static const u16 mask = BIT(CPCAP_BIT_ST_DAC_SW); u16 val; @@ -1229,7 +1229,7 @@ static int cpcap_hifi_set_mute(struct snd_soc_dai *dai, int mute) else val = BIT(CPCAP_BIT_ST_DAC_SW); - dev_dbg(codec->dev, "HiFi mute: %d", mute); + dev_dbg(component->dev, "HiFi mute: %d", mute); return regmap_update_bits(cpcap->regmap, reg, mask, val); } @@ -1244,9 +1244,9 @@ static int cpcap_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_codec *codec = dai->codec; - struct device *dev = codec->dev; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = dai->component; + struct device *dev = component->dev; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); static const u16 reg_cdi = CPCAP_REG_CDI; int rate = params_rate(params); int channels = params_channels(params); @@ -1283,10 +1283,10 @@ static int cpcap_voice_hw_params(struct snd_pcm_substream *substream, static int cpcap_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = codec_dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); - dev_dbg(codec->dev, "Voice setup sysclk: clk_id=%u, freq=%u", + dev_dbg(component->dev, "Voice setup sysclk: clk_id=%u, freq=%u", clk_id, freq); return cpcap_set_sysclk(cpcap, CPCAP_DAI_VOICE, clk_id, freq); } @@ -1294,8 +1294,8 @@ static int cpcap_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = codec_dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); static const u16 mask = BIT(CPCAP_BIT_SMB_CDC) | BIT(CPCAP_BIT_CLK_INV) | BIT(CPCAP_BIT_FS_INV) | @@ -1304,7 +1304,7 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, u16 val = 0x0000; int err; - dev_dbg(codec->dev, "Voice setup dai format (%08x)", fmt); + dev_dbg(component->dev, "Voice setup dai format (%08x)", fmt); /* * "Voice Playback" and "Voice Capture" should always be @@ -1316,7 +1316,7 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, val &= ~BIT(CPCAP_BIT_SMB_CDC); break; default: - dev_err(codec->dev, "Voice dai fmt failed: CPCAP should be the master"); + dev_err(component->dev, "Voice dai fmt failed: CPCAP should be the master"); val &= ~BIT(CPCAP_BIT_SMB_CDC); break; } @@ -1339,7 +1339,7 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, val &= ~BIT(CPCAP_BIT_FS_INV); break; default: - dev_err(codec->dev, "Voice dai fmt failed: unsupported clock invert mode"); + dev_err(component->dev, "Voice dai fmt failed: unsupported clock invert mode"); break; } @@ -1361,7 +1361,7 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, break; } - dev_dbg(codec->dev, "Voice dai format: val=%04x", val); + dev_dbg(component->dev, "Voice dai format: val=%04x", val); err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI, mask, val); if (err) return err; @@ -1372,8 +1372,8 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); static const u16 reg = CPCAP_REG_RXCOA; static const u16 mask = BIT(CPCAP_BIT_CDC_SW); u16 val; @@ -1383,7 +1383,7 @@ static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute) else val = BIT(CPCAP_BIT_CDC_SW); - dev_dbg(codec->dev, "Voice mute: %d", mute); + dev_dbg(component->dev, "Voice mute: %d", mute); return regmap_update_bits(cpcap->regmap, reg, mask, val); }; @@ -1460,13 +1460,13 @@ static int cpcap_dai_mux(struct cpcap_audio *cpcap, bool swap_dai_configuration) return 0; } -static int cpcap_audio_reset(struct snd_soc_codec *codec, +static int cpcap_audio_reset(struct snd_soc_component *component, bool swap_dai_configuration) { - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); int i, err = 0; - dev_dbg(codec->dev, "init audio codec"); + dev_dbg(component->dev, "init audio codec"); for (i = 0; i < ARRAY_SIZE(cpcap_default_regs); i++) { err = regmap_update_bits(cpcap->regmap, @@ -1500,40 +1500,41 @@ static int cpcap_audio_reset(struct snd_soc_codec *codec, return 0; } -static int cpcap_soc_probe(struct snd_soc_codec *codec) +static int cpcap_soc_probe(struct snd_soc_component *component) { struct cpcap_audio *cpcap; int err; - cpcap = devm_kzalloc(codec->dev, sizeof(*cpcap), GFP_KERNEL); + cpcap = devm_kzalloc(component->dev, sizeof(*cpcap), GFP_KERNEL); if (!cpcap) return -ENOMEM; - snd_soc_codec_set_drvdata(codec, cpcap); - cpcap->codec = codec; + snd_soc_component_set_drvdata(component, cpcap); + cpcap->component = component; - cpcap->regmap = dev_get_regmap(codec->dev->parent, NULL); + cpcap->regmap = dev_get_regmap(component->dev->parent, NULL); if (!cpcap->regmap) return -ENODEV; - snd_soc_codec_init_regmap(codec, cpcap->regmap); + snd_soc_component_init_regmap(component, cpcap->regmap); - err = cpcap_get_vendor(codec->dev, cpcap->regmap, &cpcap->vendor); + err = cpcap_get_vendor(component->dev, cpcap->regmap, &cpcap->vendor); if (err) return err; - return cpcap_audio_reset(codec, false); + return cpcap_audio_reset(component, false); } -static struct snd_soc_codec_driver soc_codec_dev_cpcap = { - .probe = cpcap_soc_probe, - - .component_driver = { - .controls = cpcap_snd_controls, - .num_controls = ARRAY_SIZE(cpcap_snd_controls), - .dapm_widgets = cpcap_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(cpcap_dapm_widgets), - .dapm_routes = intercon, - .num_dapm_routes = ARRAY_SIZE(intercon), - }, +static struct snd_soc_component_driver soc_codec_dev_cpcap = { + .probe = cpcap_soc_probe, + .controls = cpcap_snd_controls, + .num_controls = ARRAY_SIZE(cpcap_snd_controls), + .dapm_widgets = cpcap_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cpcap_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; static int cpcap_codec_probe(struct platform_device *pdev) @@ -1543,19 +1544,12 @@ static int cpcap_codec_probe(struct platform_device *pdev) pdev->dev.of_node = codec_node; - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cpcap, + return devm_snd_soc_register_component(&pdev->dev, &soc_codec_dev_cpcap, cpcap_dai, ARRAY_SIZE(cpcap_dai)); } -static int cpcap_codec_remove(struct platform_device *pdev) -{ - snd_soc_unregister_codec(&pdev->dev); - return 0; -} - static struct platform_driver cpcap_codec_driver = { .probe = cpcap_codec_probe, - .remove = cpcap_codec_remove, .driver = { .name = "cpcap-codec", }, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: mediatek: remove "simple-mfd" in the example
by Ryder Lee 28 Mar '18

28 Mar '18

Remove compatible string "simple-mfd" in the example as we have already added devm_of_platform_populate() in the parent driver. Signed-off-by: Ryder Lee <ryder.lee(a)mediatek.com> --- Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt index 6df87b9..e2f7f49 100644 --- a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt +++ b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt @@ -53,7 +53,7 @@ See ../arm/mediatek/mediatek,audsys.txt for details about the parent node. Example: audsys: audio-subsystem@11220000 { - compatible = "mediatek,mt2701-audsys", "syscon", "simple-mfd"; + compatible = "mediatek,mt2701-audsys", "syscon"; ... afe: audio-controller { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: tscs42xx: Remove owner assignment from i2c_driver
by Fabio Estevam 28 Mar '18

28 Mar '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Structure i2c_driver does not need to set the owner field, as this will be populated by the driver core. Generated by scripts/coccinelle/api/platform_no_drv_owner.cocci. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/tscs42xx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index cfc71b5..bbfc73a 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -1443,7 +1443,6 @@ MODULE_DEVICE_TABLE(of, tscs42xx_of_match); static struct i2c_driver tscs42xx_i2c_driver = { .driver = { .name = "tscs42xx", - .owner = THIS_MODULE, .of_match_table = tscs42xx_of_match, }, .probe = tscs42xx_i2c_probe, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: Add accessor macros for TLV items
by Ranjani Sridharan 28 Mar '18

28 Mar '18

This patch adds macros for accessing the type, length, min, mute and step items in TLV data. These will be used by drivers to extract the TLV data while loading topology. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- include/sound/tlv.h | 5 +++++ include/uapi/sound/tlv.h | 7 +++++++ 2 files changed, 12 insertions(+) diff --git a/include/sound/tlv.h b/include/sound/tlv.h index 3677ebb928d5..8fa379e8a568 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -49,6 +49,11 @@ #define TLV_DB_GAIN_MUTE SNDRV_CTL_TLVD_DB_GAIN_MUTE +#define TLV_ITEM_TYPE SNDRV_CTL_TLVD_ITEM_TYPE +#define TLV_ITEM_LEN SNDRV_CTL_TLVD_ITEM_LEN +#define TLV_DB_MIN SNDRV_CTL_TLVD_DB_SCALE_MIN +#define TLV_DB_MUTE_AND_STEP SNDRV_CTL_TLVD_DB_SCALE_MUTE_AND_STEP + /* * The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR. * This is an old fasion and obsoleted by commit bf1d1c9b6179("ALSA: tlv: add diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h index be5371f09a62..f3fba49ef6e8 100644 --- a/include/uapi/sound/tlv.h +++ b/include/uapi/sound/tlv.h @@ -60,6 +60,13 @@ unsigned int name[] = { \ SNDRV_CTL_TLVD_DB_SCALE_ITEM(min, step, mute) \ } +/* Accessor macros for TLV type and len */ +#define SNDRV_CTL_TLVD_ITEM_TYPE 0 +#define SNDRV_CTL_TLVD_ITEM_LEN 1 + +/* Accessor macros for min, mute and step values from dB scale type TLV */ +#define SNDRV_CTL_TLVD_DB_SCALE_MIN 2 +#define SNDRV_CTL_TLVD_DB_SCALE_MUTE_AND_STEP 3 /* dB scale specified with min/max values instead of step */ #define SNDRV_CTL_TLVD_DB_MINMAX_ITEM(min_dB, max_dB) \ -- 2.14.1

3 3

[alsa-devel] [RFC, alsa-utils] RFC: alsamixer: Improve description of playback switches
by Kirill Marinushkin 27 Mar '18

27 Mar '18

Currently, all playback switches are described as mute. It is shown as: [OO] == sound is on (left and right channels) [MM] == sound is muted (left and right channels) But cards can have different kinds of switch controls, not only mute. At least usb-audio provides boolean controls as playback switches for: * clock source validity (read only switch) * processing features on/off (rw switch) * potentially other similar use-cases It becames confusing, because it is shown as: [OO] == clock source is valid [MM] == clock source is invalid [OO] == processing feature is on [MM] == processing feature is off And from the help, it is not clear how to toggle such switches. In this RFC, I suggest to show switches more generic: [yy] == sound on / clock source is valid / processing feature is on [nn] == sound muted / clock source is invalid / processing feature is off And give a more generic description in the help. Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- alsamixer/mixer_display.c | 8 ++++---- alsamixer/mixer_widget.c | 4 ++-- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/alsamixer/mixer_display.c b/alsamixer/mixer_display.c index b1f79d0..f73b11a 100644 --- a/alsamixer/mixer_display.c +++ b/alsamixer/mixer_display.c @@ -513,13 +513,13 @@ static void display_control(unsigned int control_index) mvwaddch(mixer_widget.window, base_y + 1, frame_left + 1, switches[0] /* TRANSLATORS: playback on; one character */ - ? _("O")[0] | (control->flags & IS_ACTIVE ? attr_ctl_nomute : 0) + ? _("y")[0] | (control->flags & IS_ACTIVE ? attr_ctl_nomute : 0) /* TRANSLATORS: playback muted; one character */ - : _("M")[0] | (control->flags & IS_ACTIVE ? attr_ctl_mute : 0)); + : _("n")[0] | (control->flags & IS_ACTIVE ? attr_ctl_mute : 0)); waddch(mixer_widget.window, switches[1] - ? _("O")[0] | (control->flags & IS_ACTIVE ? attr_ctl_nomute : 0) - : _("M")[0] | (control->flags & IS_ACTIVE ? attr_ctl_mute : 0)); + ? _("y")[0] | (control->flags & IS_ACTIVE ? attr_ctl_nomute : 0) + : _("n")[0] | (control->flags & IS_ACTIVE ? attr_ctl_mute : 0)); } if (control->flags & TYPE_CSWITCH) { diff --git a/alsamixer/mixer_widget.c b/alsamixer/mixer_widget.c index 0c2b9e0..93d1923 100644 --- a/alsamixer/mixer_widget.c +++ b/alsamixer/mixer_widget.c @@ -210,9 +210,9 @@ static void show_help(void) _("Z X C Decrease left/both/right volumes"), _("B Balance left and right volumes"), "", - _("M Toggle mute"), + _("M Toggle mute / playback switch"), /* TRANSLATORS: or , . */ - _("< > Toggle left/right mute"), + _("< > Toggle left/right mute / playback switch"), "", _("Space Toggle capture"), /* TRANSLATORS: or Insert Delete */ -- 2.13.6

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[alsa-devel] [PATCH v2 1/3] ASoC: AMD: Use single dai for da7219 playback and capture
by Akshu Agrawal 27 Mar '18

27 Mar '18

BT I2S is a bi-directional dai, we will use the same cpu dai for playback and capture. V2: reworded the commit message. TEST=aplay -D hw:0,0 -vv <file> arecord -D hw:0,0 -f dat -d 5 -vv <file> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index bbe0f10..d07c2a1 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -124,8 +124,8 @@ static int cz_fe_startup(struct snd_pcm_substream *substream) static struct snd_soc_dai_link cz_dai_7219_98357[] = { { - .name = "amd-da7219-play", - .stream_name = "Playback", + .name = "amd-da7219-play-cap", + .stream_name = "Playback and Capture", .platform_name = "acp_audio_dma.0.auto", .cpu_dai_name = "designware-i2s.3.auto", .codec_dai_name = "da7219-hifi", @@ -134,16 +134,6 @@ static int cz_fe_startup(struct snd_pcm_substream *substream) | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, - }, - { - .name = "amd-da7219-cap", - .stream_name = "Capture", - .platform_name = "acp_audio_dma.0.auto", - .cpu_dai_name = "designware-i2s.4.auto", - .codec_dai_name = "da7219-hifi", - .codec_name = "i2c-DLGS7219:00", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, .ops = &cz_da7219_cap_ops, }, -- 1.9.1

2 4

[alsa-devel] [PATCH v2] ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format()
by Kirill Marinushkin 27 Mar '18

27 Mar '18

The values of bclk and fsync are inverted WRT the codec. But the existing solution already works for Broadwell, see the alsa-lib config: `alsa-lib/src/conf/topology/broadwell/broadwell.conf` This commit provides the backwards-compatible solution to fix this misuse. This commit goes in pair with the corresponding patch for alsa-lib. Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Pan Xiuli <xiuli.pan(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- include/uapi/sound/asoc.h | 16 ++++++++++++++-- sound/soc/soc-topology.c | 12 +++++++----- 2 files changed, 21 insertions(+), 7 deletions(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 69c37ecbff7e..f0e5e21efa54 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -160,6 +160,18 @@ #define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) #define SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP (1 << 3) +/* DAI topology BCLK parameter + * For the backwards capability, by default codec is bclk master + */ +#define SND_SOC_TPLG_BCLK_CM 0 /* codec is bclk master */ +#define SND_SOC_TPLG_BCLK_CS 1 /* codec is bclk slave */ + +/* DAI topology FSYNC parameter + * For the backwards capability, by default codec is fsync master + */ +#define SND_SOC_TPLG_FSYNC_CM 0 /* codec is fsync master */ +#define SND_SOC_TPLG_FSYNC_CS 1 /* codec is fsync slave */ + /* * Block Header. * This header precedes all object and object arrays below. @@ -315,8 +327,8 @@ struct snd_soc_tplg_hw_config { __u8 clock_gated; /* 1 if clock can be gated to save power */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ - __u8 bclk_master; /* 1 for master of BCLK, 0 for slave */ - __u8 fsync_master; /* 1 for master of FSYNC, 0 for slave */ + __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ + __u8 fsync_master; /* SND_SOC_TPLG_FSYNC_ value */ __u8 mclk_direction; /* 0 for input, 1 for output */ __le16 reserved; /* for 32bit alignment */ __le32 mclk_rate; /* MCLK or SYSCLK freqency in Hz */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 01a50413c66f..c5bdc673b195 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1994,13 +1994,15 @@ static void set_link_hw_format(struct snd_soc_dai_link *link, link->dai_fmt |= SND_SOC_DAIFMT_IB_IF; /* clock masters */ - bclk_master = hw_config->bclk_master; - fsync_master = hw_config->fsync_master; - if (!bclk_master && !fsync_master) + bclk_master = (hw_config->bclk_master == + SND_SOC_TPLG_BCLK_CM); + fsync_master = (hw_config->fsync_master == + SND_SOC_TPLG_FSYNC_CM); + if (bclk_master && fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - else if (bclk_master && !fsync_master) - link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; else if (!bclk_master && fsync_master) + link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + else if (bclk_master && !fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; else link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; -- 2.13.6

3 13

[alsa-devel] [PATCH 3/3 v2] ASoC: TSCS42xx: Add CCF support to get sysclk
by Steven Eckhoff 27 Mar '18

27 Mar '18

The TSCS42xx relies on set_sysclk to get a unique clock id and rate, which prevents it from being used with the simple-card. Remove set_sysclk callback Add CCF support to get clock id and rate Add clocks and clock-names to device tree binding Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- .../devicetree/bindings/sound/tscs42xx.txt | 6 ++ sound/soc/codecs/tscs42xx.c | 104 ++++++++++++++------- sound/soc/codecs/tscs42xx.h | 2 +- 3 files changed, 75 insertions(+), 37 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tscs42xx.txt b/Documentation/devicetree/bindings/sound/tscs42xx.txt index 2ac2f0996697..7eea32e9d078 100644 --- a/Documentation/devicetree/bindings/sound/tscs42xx.txt +++ b/Documentation/devicetree/bindings/sound/tscs42xx.txt @@ -8,9 +8,15 @@ Required Properties: - reg : <0x71> for analog mic <0x69> for digital mic + - clock-names: Must one of the following "mclk1", "xtal", "mclk2" + + - clocks: phandle of the clock that provides the codec sysclk + Example: wookie: codec@69 { compatible = "tempo,tscs42A2"; reg = <0x69>; + clock-names = "xtal"; + clocks = <&audio_xtal>; }; diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index d41852a37ac6..69cd4c1ed1d6 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/delay.h> #include <linux/mutex.h> +#include <linux/clk.h> #include <sound/tlv.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -40,6 +41,9 @@ struct tscs42xx { struct mutex pll_lock; struct regmap *regmap; + + struct clk *sysclk; + int sysclk_src_id; }; struct coeff_ram_ctl { @@ -1251,13 +1255,46 @@ static int tscs42xx_set_dai_bclk_ratio(struct snd_soc_dai *codec_dai, return 0; } -static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) +static const struct snd_soc_dai_ops tscs42xx_dai_ops = { + .hw_params = tscs42xx_hw_params, + .mute_stream = tscs42xx_mute_stream, + .set_fmt = tscs42xx_set_dai_fmt, + .set_bclk_ratio = tscs42xx_set_dai_bclk_ratio, +}; + +static int part_is_valid(struct tscs42xx *tscs42xx) { - struct snd_soc_component *component = codec_dai->component; + int val; int ret; + unsigned int reg; + + ret = regmap_read(tscs42xx->regmap, R_DEVIDH, &reg); + if (ret < 0) + return ret; - switch (clk_id) { + val = reg << 8; + ret = regmap_read(tscs42xx->regmap, R_DEVIDL, &reg); + if (ret < 0) + return ret; + + val |= reg; + + switch (val) { + case 0x4A74: + case 0x4A73: + return true; + default: + return false; + }; +} + +static int set_sysclk(struct snd_soc_component *component) +{ + struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); + unsigned long freq; + int ret; + + switch (tscs42xx->sysclk_src_id) { case TSCS42XX_PLL_SRC_XTAL: case TSCS42XX_PLL_SRC_MCLK1: ret = snd_soc_component_write(component, R_PLLREFSEL, @@ -1285,6 +1322,7 @@ static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return -EINVAL; } + freq = clk_get_rate(tscs42xx->sysclk); ret = set_pll_ctl_from_input_freq(component, freq); if (ret < 0) { dev_err(component->dev, @@ -1295,41 +1333,13 @@ static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -static const struct snd_soc_dai_ops tscs42xx_dai_ops = { - .hw_params = tscs42xx_hw_params, - .mute_stream = tscs42xx_mute_stream, - .set_fmt = tscs42xx_set_dai_fmt, - .set_bclk_ratio = tscs42xx_set_dai_bclk_ratio, - .set_sysclk = tscs42xx_set_dai_sysclk, -}; - -static int part_is_valid(struct tscs42xx *tscs42xx) +static int tscs42xx_probe(struct snd_soc_component *component) { - int val; - int ret; - unsigned int reg; - - ret = regmap_read(tscs42xx->regmap, R_DEVIDH, &reg); - if (ret < 0) - return ret; - - val = reg << 8; - ret = regmap_read(tscs42xx->regmap, R_DEVIDL, &reg); - if (ret < 0) - return ret; - - val |= reg; - - switch (val) { - case 0x4A74: - case 0x4A73: - return true; - default: - return false; - }; + return set_sysclk(component); } static const struct snd_soc_component_driver soc_codec_dev_tscs42xx = { + .probe = tscs42xx_probe, .dapm_widgets = tscs42xx_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tscs42xx_dapm_widgets), .dapm_routes = tscs42xx_intercon, @@ -1387,11 +1397,15 @@ static const struct reg_sequence tscs42xx_patch[] = { { R_AIC2, RV_AIC2_BLRCM_DAC_BCLK_LRCLK_SHARED }, }; +static char const * const src_names[TSCS42XX_PLL_SRC_CNT] = { + "xtal", "mclk1", "mclk2"}; + static int tscs42xx_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct tscs42xx *tscs42xx; - int ret = 0; + int src; + int ret; tscs42xx = devm_kzalloc(&i2c->dev, sizeof(*tscs42xx), GFP_KERNEL); if (!tscs42xx) { @@ -1402,6 +1416,24 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, } i2c_set_clientdata(i2c, tscs42xx); + for (src = TSCS42XX_PLL_SRC_XTAL; src < TSCS42XX_PLL_SRC_CNT; src++) { + tscs42xx->sysclk = devm_clk_get(&i2c->dev, src_names[src]); + if (!IS_ERR(tscs42xx->sysclk)) { + break; + } else if (PTR_ERR(tscs42xx->sysclk) != -ENOENT) { + ret = PTR_ERR(tscs42xx->sysclk); + dev_err(&i2c->dev, "Failed to get sysclk (%d)\n", ret); + return ret; + } + } + if (src > TSCS42XX_PLL_SRC_CNT) { + ret = -EINVAL; + dev_err(&i2c->dev, "Failed to get a valid clock name (%d)\n", + ret); + return ret; + } + tscs42xx->sysclk_src_id = src; + tscs42xx->regmap = devm_regmap_init_i2c(i2c, &tscs42xx_regmap); if (IS_ERR(tscs42xx->regmap)) { ret = PTR_ERR(tscs42xx->regmap); diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h index d4a30bcbf64b..814c8f3c4a68 100644 --- a/sound/soc/codecs/tscs42xx.h +++ b/sound/soc/codecs/tscs42xx.h @@ -7,10 +7,10 @@ #define __WOOKIE_H__ enum { - TSCS42XX_PLL_SRC_NONE, TSCS42XX_PLL_SRC_XTAL, TSCS42XX_PLL_SRC_MCLK1, TSCS42XX_PLL_SRC_MCLK2, + TSCS42XX_PLL_SRC_CNT, }; #define R_HPVOLL 0x0 -- 2.15.1

1 0

[alsa-devel] No audio on Lenovo Miix 320
by Freaky 27 Mar '18

27 Mar '18

Hi there, we're having some issues getting the audio working on a Lenovo Miix 320. On some forums they say it should work after blacklisting snd_hdmi_lpe_audio. Have blacklisted this, and there's one audio card less now, but still no go unfortunately. Currently running with 4.15.10 from Fedora 27. It logs a lot of: "Audio Port: ASoC: no backend DAIs enabled for Audio Port" lines. Any ideas on how to troubleshoot this? Saw similar problems in https://bugzilla.kernel.org/show_bug.cgi?id=96691 but this seems to be for another codec/DAI. Also, their system seems to be a Baytrail Atom, this is a Cherrytrail. Not sure if any of the patches are applicable. dmesg is attached. I can patch, build and test kernels. Coding is another story however :). Any help is appreciated. Kind regards,

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: fix memory leak on cval
by Colin King 27 Mar '18

27 Mar '18

From: Colin Ian King <colin.king(a)canonical.com> With the current exit return path of the ctl_info allocation failure cval is not being freed resulting in a memory leak. Fix this by kfree'ing it on the return. Detected by CoverityScan, CID#1466878 ("Resource Leak") Fixes: 21e9b3e931f7 ("ALSA: usb-audio: fix uac control query argument") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/mixer.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 04dab6f65535..301ad61ed426 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1365,8 +1365,10 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, cval->cmask = ctl_mask; ctl_info = get_feature_control_info(control); - if (!ctl_info) + if (!ctl_info) { + kfree(cval); return; + } if (state->mixer->protocol == UAC_VERSION_1) cval->val_type = ctl_info->type; else /* UAC_VERSION_2 */ -- 2.15.1

2 1