Alsa-devel March 2018

alsa-devel@alsa-project.org

144 participants
294 discussions

[alsa-devel] [PATCH][RFC] ASoC: wm8350: remove snd_soc_codec
by Kuninori Morimoto 13 Apr '18

13 Apr '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> codec is replace to component. It seems no-one is using it, Let's remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Hi Mark, Liam I think no-one is using snd_soc_codec, but please double-check it. I added [RFC] on this patch include/linux/mfd/wm8350/audio.h | 3 --- 1 file changed, 3 deletions(-) diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h index bd581c6..0bc41c4c 100644 --- a/include/linux/mfd/wm8350/audio.h +++ b/include/linux/mfd/wm8350/audio.h @@ -617,11 +617,8 @@ struct wm8350_audio_platform_data { u32 codec_current_charge:2; /* codec current @ vmid charge */ }; -struct snd_soc_codec; - struct wm8350_codec { struct platform_device *pdev; - struct snd_soc_codec *codec; struct wm8350_audio_platform_data *platform_data; }; -- 1.9.1

3 6

[alsa-devel] [PATCH 1/3] ASoC: TSCS42xx: Shorten lines and other cleanup
by Steven Eckhoff 13 Apr '18

13 Apr '18

Shorten lines greater than 80 chars Add const to struct snd_soc_component_driver Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- sound/soc/codecs/tscs42xx.c | 87 ++++++++++++++++++++++++++++----------------- 1 file changed, 55 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index cfc71b5411ee..37636d6efe60 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -204,7 +204,8 @@ static int power_up_audio_plls(struct snd_soc_component *component) break; default: ret = -EINVAL; - dev_err(component->dev, "Unrecognized PLL output freq (%d)\n", ret); + dev_err(component->dev, + "Unrecognized PLL output freq (%d)\n", ret); return ret; } @@ -261,7 +262,8 @@ static int power_down_audio_plls(struct snd_soc_component *component) static int coeff_ram_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); struct coeff_ram_ctl *ctl = (struct coeff_ram_ctl *)kcontrol->private_value; @@ -280,7 +282,8 @@ static int coeff_ram_get(struct snd_kcontrol *kcontrol, static int coeff_ram_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); struct coeff_ram_ctl *ctl = (struct coeff_ram_ctl *)kcontrol->private_value; @@ -363,7 +366,8 @@ static int dapm_micb_event(struct snd_soc_dapm_widget *w, static int pll_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); + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); int ret; if (SND_SOC_DAPM_EVENT_ON(event)) @@ -377,7 +381,8 @@ static int pll_event(struct snd_soc_dapm_widget *w, static int dac_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); + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); int ret; @@ -819,16 +824,19 @@ static int setup_sample_format(struct snd_soc_component *component, dev_err(component->dev, "Unsupported format width (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_AIC1, RM_AIC1_WL, width); + ret = snd_soc_component_update_bits(component, + R_AIC1, RM_AIC1_WL, width); if (ret < 0) { - dev_err(component->dev, "Failed to set sample width (%d)\n", ret); + dev_err(component->dev, + "Failed to set sample width (%d)\n", ret); return ret; } return 0; } -static int setup_sample_rate(struct snd_soc_component *component, unsigned int rate) +static int setup_sample_rate(struct snd_soc_component *component, + unsigned int rate) { struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); unsigned int br, bm; @@ -881,24 +889,32 @@ static int setup_sample_rate(struct snd_soc_component *component, unsigned int r } /* DAC and ADC share bit and frame clock */ - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBR, br); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBR, br); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBM, bm); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBM, bm); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_DACSR_DBR, br); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_DACSR_DBR, br); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_DACSR_DBM, bm); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_DACSR_DBM, bm); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } @@ -1076,7 +1092,8 @@ static int tscs42xx_hw_params(struct snd_pcm_substream *substream, ret = setup_sample_rate(component, params_rate(params)); if (ret < 0) { - dev_err(component->dev, "Failed to setup sample rate (%d)\n", ret); + dev_err(component->dev, + "Failed to setup sample rate (%d)\n", ret); return ret; } @@ -1087,7 +1104,8 @@ static inline int dac_mute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR1, RM_CNVRTR1_DACMU, + ret = snd_soc_component_update_bits(component, + R_CNVRTR1, RM_CNVRTR1_DACMU, RV_CNVRTR1_DACMU_ENABLE); if (ret < 0) { dev_err(component->dev, "Failed to mute DAC (%d)\n", @@ -1102,7 +1120,8 @@ static inline int dac_unmute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR1, RM_CNVRTR1_DACMU, + ret = snd_soc_component_update_bits(component, + R_CNVRTR1, RM_CNVRTR1_DACMU, RV_CNVRTR1_DACMU_DISABLE); if (ret < 0) { dev_err(component->dev, "Failed to unmute DAC (%d)\n", @@ -1117,8 +1136,8 @@ static inline int adc_mute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR0, RM_CNVRTR0_ADCMU, - RV_CNVRTR0_ADCMU_ENABLE); + ret = snd_soc_component_update_bits(component, + R_CNVRTR0, RM_CNVRTR0_ADCMU, RV_CNVRTR0_ADCMU_ENABLE); if (ret < 0) { dev_err(component->dev, "Failed to mute ADC (%d)\n", ret); @@ -1132,8 +1151,8 @@ static inline int adc_unmute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR0, RM_CNVRTR0_ADCMU, - RV_CNVRTR0_ADCMU_DISABLE); + ret = snd_soc_component_update_bits(component, + R_CNVRTR0, RM_CNVRTR0_ADCMU, RV_CNVRTR0_ADCMU_DISABLE); if (ret < 0) { dev_err(component->dev, "Failed to unmute ADC (%d)\n", ret); @@ -1171,8 +1190,8 @@ static int tscs42xx_set_dai_fmt(struct snd_soc_dai *codec_dai, /* Slave mode not supported since it needs always-on frame clock */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - ret = snd_soc_component_update_bits(component, R_AIC1, RM_AIC1_MS, - RV_AIC1_MS_MASTER); + ret = snd_soc_component_update_bits(component, + R_AIC1, RM_AIC1_MS, RV_AIC1_MS_MASTER); if (ret < 0) { dev_err(component->dev, "Failed to set codec DAI master (%d)\n", ret); @@ -1211,14 +1230,18 @@ static int tscs42xx_set_dai_bclk_ratio(struct snd_soc_dai *codec_dai, return -EINVAL; } - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBCM, value); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBCM, value); if (ret < 0) { - dev_err(component->dev, "Failed to set DAC BCLK ratio (%d)\n", ret); + dev_err(component->dev, + "Failed to set DAC BCLK ratio (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_ADCSR_ABCM, value); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_ADCSR_ABCM, value); if (ret < 0) { - dev_err(component->dev, "Failed to set ADC BCLK ratio (%d)\n", ret); + dev_err(component->dev, + "Failed to set ADC BCLK ratio (%d)\n", ret); return ret; } @@ -1309,7 +1332,7 @@ static int part_is_valid(struct tscs42xx *tscs42xx) }; } -static struct snd_soc_component_driver soc_codec_dev_tscs42xx = { +static const struct snd_soc_component_driver soc_codec_dev_tscs42xx = { .dapm_widgets = tscs42xx_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tscs42xx_dapm_widgets), .dapm_routes = tscs42xx_intercon, @@ -1416,8 +1439,8 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, mutex_init(&tscs42xx->coeff_ram_lock); mutex_init(&tscs42xx->pll_lock); - ret = devm_snd_soc_register_component(tscs42xx->dev, &soc_codec_dev_tscs42xx, - &tscs42xx_dai, 1); + ret = devm_snd_soc_register_component(tscs42xx->dev, + &soc_codec_dev_tscs42xx, &tscs42xx_dai, 1); if (ret) { dev_err(tscs42xx->dev, "Failed to register codec (%d)\n", ret); return ret; -- 2.15.1

2 2

[alsa-devel] [PATCH 2/3] ASoC: TSCS42xx: Cleanup private data members
by Steven Eckhoff 13 Apr '18

13 Apr '18

In the TSCS42xx driver there are some unnecesarry private data members Remove blrcm from private data Remove dev from private data Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- sound/soc/codecs/tscs42xx.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index 37636d6efe60..d41852a37ac6 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -31,7 +31,6 @@ struct tscs42xx { int bclk_ratio; int samplerate; - unsigned int blrcm; struct mutex audio_params_lock; u8 coeff_ram[COEFF_RAM_SIZE]; @@ -41,8 +40,6 @@ struct tscs42xx { struct mutex pll_lock; struct regmap *regmap; - - struct device *dev; }; struct coeff_ram_ctl { @@ -1404,12 +1401,11 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, return ret; } i2c_set_clientdata(i2c, tscs42xx); - tscs42xx->dev = &i2c->dev; tscs42xx->regmap = devm_regmap_init_i2c(i2c, &tscs42xx_regmap); if (IS_ERR(tscs42xx->regmap)) { ret = PTR_ERR(tscs42xx->regmap); - dev_err(tscs42xx->dev, "Failed to allocate regmap (%d)\n", ret); + dev_err(&i2c->dev, "Failed to allocate regmap (%d)\n", ret); return ret; } @@ -1417,21 +1413,21 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, ret = part_is_valid(tscs42xx); if (ret <= 0) { - dev_err(tscs42xx->dev, "No valid part (%d)\n", ret); + dev_err(&i2c->dev, "No valid part (%d)\n", ret); ret = -ENODEV; return ret; } ret = regmap_write(tscs42xx->regmap, R_RESET, RV_RESET_ENABLE); if (ret < 0) { - dev_err(tscs42xx->dev, "Failed to reset device (%d)\n", ret); + dev_err(&i2c->dev, "Failed to reset device (%d)\n", ret); return ret; } ret = regmap_register_patch(tscs42xx->regmap, tscs42xx_patch, ARRAY_SIZE(tscs42xx_patch)); if (ret < 0) { - dev_err(tscs42xx->dev, "Failed to apply patch (%d)\n", ret); + dev_err(&i2c->dev, "Failed to apply patch (%d)\n", ret); return ret; } @@ -1439,10 +1435,10 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, mutex_init(&tscs42xx->coeff_ram_lock); mutex_init(&tscs42xx->pll_lock); - ret = devm_snd_soc_register_component(tscs42xx->dev, + ret = devm_snd_soc_register_component(&i2c->dev, &soc_codec_dev_tscs42xx, &tscs42xx_dai, 1); if (ret) { - dev_err(tscs42xx->dev, "Failed to register codec (%d)\n", ret); + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); return ret; } -- 2.15.1

2 2

[alsa-devel] [PATCH 3/3 v3] ASoC: TSCS42xx: Add CCF support to get sysclk
by Steven Eckhoff 13 Apr '18

13 Apr '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..0aaa6a826096 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

2 4

[alsa-devel] [PATCH] ALSA: hda - Silence PM ops build warning
by Lukas Wunner 13 Apr '18

13 Apr '18

The system sleep PM ops azx_suspend() and azx_resume() were previously called by vga_switcheroo, but commit 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") removed their invocation. Unfortunately the commit neglected to update the #ifdef surrounding the two functions, so if CONFIG_PM_SLEEP is *not* enabled but all three of CONFIG_PM, CONFIG_VGA_SWITCHEROO and CONFIG_SND_HDA_CODEC_HDMI *are* enabled, the compiler now emits the following warning: sound/pci/hda/hda_intel.c:1024:12: warning: 'azx_resume' defined but not used [-Wunused-function] static int azx_resume(struct device *dev) ^~~~~~~~~~ sound/pci/hda/hda_intel.c:989:12: warning: 'azx_suspend' defined but not used [-Wunused-function] static int azx_suspend(struct device *dev) ^~~~~~~~~~~ Silence by updating the #ifdef. Because the #ifdef block now uses the same condition as the one immediately succeeding it, the two blocks can be collapsed together, shaving off another two lines. Fixes: 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") Cc: Takashi Iwai <tiwai(a)suse.com> Reported-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Lukas Wunner <lukas(a)wunner.de> --- sound/pci/hda/hda_intel.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ec4e6b829ee2..d653c56e9141 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -982,7 +982,7 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ -#if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -1063,9 +1063,7 @@ static int azx_resume(struct device *dev) trace_azx_resume(chip); return 0; } -#endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */ -#ifdef CONFIG_PM_SLEEP /* put codec down to D3 at hibernation for Intel SKL+; * otherwise BIOS may still access the codec and screw up the driver */ -- 2.16.2

2 3

[alsa-devel] [PATCH] ASoC: rsnd: mark PM functions __maybe_unused
by Arnd Bergmann 12 Apr '18

12 Apr '18

The suspend/resume callbacks are now optional, leading to a warning when they are unused: sound/soc/sh/rcar/core.c:1548:12: error: 'rsnd_resume' defined but not used [-Werror=unused-function] static int rsnd_resume(struct device *dev) ^~~~~~~~~~~ sound/soc/sh/rcar/core.c:1539:12: error: 'rsnd_suspend' defined but not used [-Werror=unused-function] static int rsnd_suspend(struct device *dev) This marks the as __maybe_unused to avoid the warning. Fixes: f8a9a29c4fe9 ("ASoC: rsnd: set pm_ops in hibernate-compatible way") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sh/rcar/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6a76688a8ba9..94f081b93258 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1536,7 +1536,7 @@ static int rsnd_remove(struct platform_device *pdev) return ret; } -static int rsnd_suspend(struct device *dev) +static int __maybe_unused rsnd_suspend(struct device *dev) { struct rsnd_priv *priv = dev_get_drvdata(dev); @@ -1545,7 +1545,7 @@ static int rsnd_suspend(struct device *dev) return 0; } -static int rsnd_resume(struct device *dev) +static int __maybe_unused rsnd_resume(struct device *dev) { struct rsnd_priv *priv = dev_get_drvdata(dev); -- 2.9.0

2 1

[alsa-devel] [PATCH] ASoC: Intel: atom: fix ACPI/PCI Kconfig
by Pierre-Louis Bossart 12 Apr '18

12 Apr '18

The split between ACPI and PCI platforms generated issues with randconfig: with SND_SST_ATOM_HIFI2_PLATFORM_PCI=y and SND_SST_ATOM_HIFI2_PLATFORM=m, we get this module link failure: ERROR: "sst_context_init" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_context_cleanup" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_alloc_drv_context" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "intel_sst_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_configure_runtime_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! To keep things simple, let's expose two configs for SND_SST_ATOM_HIFI2_PLATFORM_PCI and SND_SST_ATOM_HIFI2_PLATFORM_ACPI, which select a common SND_SST_ATOM_HIFI2_PLATFORM option. To avoid breaking existing solutions with the semantics change, SND_SST_ATOM_HIFI2_PLATFORM_ACPI uses "default ACPI" so that "make oldnoconfig" and "make olddefconfig" still work as expected. Also remove mentions of Medfield while we are at it since it was removed recently. Reported-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 4772c16ede52 ("ASoC: Intel: Kconfig: Simplify-clarify ACPI/PCI dependencies") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/Kconfig | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f2c9e8c5970a..d7ac77b0197b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -72,24 +72,28 @@ config SND_SOC_INTEL_BAYTRAIL for Baytrail Chromebooks but this option is now deprecated and is not recommended, use SND_SST_ATOM_HIFI2_PLATFORM instead. +config SND_SST_ATOM_HIFI2_PLATFORM + tristate + select SND_SOC_COMPRESS + config SND_SST_ATOM_HIFI2_PLATFORM_PCI - tristate "PCI HiFi2 (Medfield, Merrifield) Platforms" + tristate "PCI HiFi2 (Merrifield) Platforms" depends on X86 && PCI select SND_SST_IPC_PCI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM help - If you have a Intel Medfield or Merrifield/Edison platform, then + If you have a Intel Merrifield/Edison platform, then enable this option by saying Y or m. Distros will typically not - enable this option: Medfield devices are not available to - developers and while Merrifield/Edison can run a mainline kernel with - limited functionality it will require a firmware file which - is not in the standard firmware tree + enable this option: while Merrifield/Edison can run a mainline + kernel with limited functionality it will require a firmware file + which is not in the standard firmware tree -config SND_SST_ATOM_HIFI2_PLATFORM +config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" + default ACPI depends on X86 && ACPI select SND_SST_IPC_ACPI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH select IOSF_MBI help -- 2.14.1

4 3

[alsa-devel] [Xen-devel][PATCH v3 0/5] sndif: add explicit back and front synchronization
by Oleksandr Andrushchenko 12 Apr '18

12 Apr '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Hello, all! In order to provide explicit synchronization between backend and frontend the following changes are introduced in the protocol: - bump protocol version to 2 - add new ring buffer for sending asynchronous events from backend to frontend to report number of bytes played by the frontend (XENSND_EVT_CUR_POS) - introduce trigger events for playback control: start/stop/pause/resume - add "req-" prefix to event-channel and ring-ref to unify naming of the Xen event channels for requests and events - add XENSND_OP_HW_PARAM_QUERY request to read/update stream configuration space: request passes desired intervals/formats for the stream parameters and the response returns allowed intervals and formats mask that can be used. Changes since v2: 1. Konrad's r-b tag for version patch 2. MAJOR: changed req/resp/evt packet sizes from 32 to 64 octets 3. Reworked XENSND_OP_HW_PARAM_QUERY so it now sends all parameters at once, allowing to check all the configuration space. 4. Minor documentation cleanup (added missed "reserved" fields) Changes since v1: 1. Changed protocol version definition from string to integer, so it can easily be used in comparisons. Konrad, I have removed your r-b tag for the reason of this change. 2. In order to provide explicit stream parameter negotiation between backend and frontend the following changes are introduced in the protocol: add XENSND_OP_HW_PARAM_QUERY request to read/update configuration space for the parameter given: request passes desired parameter interval (mask) and the response to this request returns min/max interval (mask) for the parameter to be used. Parameters supported by this request/response: - format mask - sample rate interval - number of channels interval - buffer size, interval, frames - period size, interval, frames Oleksandr Andrushchenko (5): sndif: Introduce protocol version sndif: Fix missed "reserved" fields in comments sndif: Make requests and responses 64 octets long sndif: Add explicit back and front synchronization sndif: Add explicit back and front parameter negotiation xen/include/public/io/sndif.h | 322 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 306 insertions(+), 16 deletions(-) -- 2.7.4

4 16

[alsa-devel] [PATCH 1/2] ASoC: AMD: Support headset button on Stoney DA7219
by Akshu Agrawal 12 Apr '18

12 Apr '18

Adds headset button support. TEST=Tested Volume UP/Down functionality Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b205c78..d9491e1 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -33,6 +33,7 @@ #include <linux/gpio.h> #include <linux/module.h> #include <linux/i2c.h> +#include <linux/input.h> #include <linux/acpi.h> #include "../codecs/da7219.h" @@ -51,6 +52,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = rtd->card; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_component *component = codec_dai->component; + struct snd_soc_jack *jack; dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); @@ -80,6 +82,12 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } + jack = &cz_jack; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + da7219_aad_jack_det(component, &cz_jack); return 0; -- 1.9.1

4 10

[alsa-devel] [PATCH v2 1/3] dell-led: Change dell-led.h to dell-common.h
by Kai-Heng Feng 07 Apr '18

07 Apr '18

This header will be used for more than just led. Change it to a more generic name. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v2: Mario suggested to squash the HDA part into the same series. drivers/platform/x86/dell-laptop.c | 2 +- include/linux/{dell-led.h => dell-common.h} | 4 ++-- sound/pci/hda/dell_wmi_helper.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) rename include/linux/{dell-led.h => dell-common.h} (61%) diff --git a/drivers/platform/x86/dell-laptop.c b/drivers/platform/x86/dell-laptop.c index c52c6723374b..8ba820e6c3d0 100644 --- a/drivers/platform/x86/dell-laptop.c +++ b/drivers/platform/x86/dell-laptop.c @@ -29,7 +29,7 @@ #include <linux/mm.h> #include <linux/i8042.h> #include <linux/debugfs.h> -#include <linux/dell-led.h> +#include <linux/dell-common.h> #include <linux/seq_file.h> #include <acpi/video.h> #include "dell-rbtn.h" diff --git a/include/linux/dell-led.h b/include/linux/dell-common.h similarity index 61% rename from include/linux/dell-led.h rename to include/linux/dell-common.h index 92521471517f..37e4b614dd74 100644 --- a/include/linux/dell-led.h +++ b/include/linux/dell-common.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0 */ -#ifndef __DELL_LED_H__ -#define __DELL_LED_H__ +#ifndef __DELL_COMMON_H__ +#define __DELL_COMMON_H__ int dell_micmute_led_set(int on); diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c index 1b48a8c19d28..56050cc3c0ee 100644 --- a/sound/pci/hda/dell_wmi_helper.c +++ b/sound/pci/hda/dell_wmi_helper.c @@ -4,7 +4,7 @@ */ #if IS_ENABLED(CONFIG_DELL_LAPTOP) -#include <linux/dell-led.h> +#include <linux/dell-common.h> enum { MICMUTE_LED_ON, -- 2.15.1

6 24

← Newer
1
2
3
4
5
6
7
...
30
Older →

Jump to page:

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

Alsa-devel March 2018