October 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: audio-graph: fixup graph_dai_link_of_dpcm() comment
by Kuninori Morimoto 15 Oct '19

15 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> The comment is wrong. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 6007e63..9ad35d9 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -232,7 +232,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { int is_single_links = 0; - /* BE is dummy */ + /* Codec is dummy */ codecs->of_node = NULL; codecs->dai_name = "snd-soc-dummy-dai"; codecs->name = "snd-soc-dummy"; @@ -263,7 +263,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, } else { struct snd_soc_codec_conf *cconf; - /* FE is dummy */ + /* CPU is dummy */ cpus->of_node = NULL; cpus->dai_name = "snd-soc-dummy-dai"; cpus->name = "snd-soc-dummy"; -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: pcm3168a: Fix serial mode dependent format support
by Peter Ujfalusi 15 Oct '19

15 Oct '19

fmt 0 is perfectly valid (PCM3168A_FMT_I2S). Remove the return in case fmt == 0. Fixes: ("ASoC: pcm3168a: Use fixup instead of constraint for channels and formats") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/pcm3168a.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 708dac27feff..df8395b361d4 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -322,8 +322,6 @@ static void pcm3168a_update_fixup_pcm_stream(struct snd_soc_dai *dai) u64 formats = SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE; unsigned int channel_max = dai->id == PCM3168A_DAI_DAC ? 8 : 6; - if (!pcm3168a->io_params[dai->id].fmt) - return; if (pcm3168a->io_params[dai->id].fmt == PCM3168A_FMT_RIGHT_J) { /* S16_LE is only supported in RIGHT_J mode */ -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

1 1

[alsa-devel] [PATCH v3] drm/bridge: dw-hdmi: Restore audio when setting a mode
by Cheng-Yi Chiang 15 Oct '19

15 Oct '19

From: Daniel Kurtz <djkurtz(a)chromium.org> When setting a new display mode, dw_hdmi_setup() calls dw_hdmi_enable_video_path(), which disables all hdmi clocks, including the audio clock. We should only (re-)enable the audio clock if audio was already enabled when setting the new mode. Without this patch, on RK3288, there will be HDMI audio on some monitors if i2s was played to headphone when the monitor was plugged. ACER H277HU and ASUS PB278 are two of the monitors showing this issue. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Change from v2 to v3: - Remove spinlock around setting clock. drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index a15fbf71b9d7..af060162b0af 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -2054,7 +2054,7 @@ static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) /* HDMI Initialization Step E - Configure audio */ hdmi_clk_regenerator_update_pixel_clock(hdmi); - hdmi_enable_audio_clk(hdmi, true); + hdmi_enable_audio_clk(hdmi, hdmi->audio_enable); } /* not for DVI mode */ -- 2.23.0.581.g78d2f28ef7-goog

2 1

[alsa-devel] [RFC PATCH 0/2] Introduce for-dpcm DT property
by Daniel Baluta 15 Oct '19

15 Oct '19

We need to be able to create DPCM links even if we have a single CPU DAI or just a dummy CPU DAI. Daniel Baluta (2): ASoC: simple-card: Introduce force-dpcm DT property ASoC: simple-card: Add documentation for force-dpcm property .../devicetree/bindings/sound/simple-card.txt | 1 + include/sound/simple_card_utils.h | 4 +++ sound/soc/generic/simple-card-utils.c | 17 +++++++++++++ sound/soc/generic/simple-card.c | 25 +++++++++++++++++-- 4 files changed, 45 insertions(+), 2 deletions(-) -- 2.17.1

4 7

Re: [alsa-devel] [PATCH 2/2] ASoC: simple_card_utils.h: Fix potential multiple redefinition error
by Kuninori Morimoto 15 Oct '19

15 Oct '19

Hi > asoc_simple_debug_info and asoc_simple_debug_dai must be static > otherwise we might a compilation error if the compiler decides > not to inline the given function. > > Fixes: 0580dde59438686d ("ASoC: simple-card-utils: add asoc_simple_debug_info()") > Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> > --- Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Thank you for your help !! Best regards --- Kuninori Morimoto

1 0

[alsa-devel] [PATCH] ASoC: Intel: eve: Enable mclk and ssp sclk early
by Brent Lu 14 Oct '19

14 Oct '19

From: Naveen M <naveen.m(a)intel.com> rt5663 and rt5514 needs mclk/sclk early to synchronize its internal clocks. Signed-off-by: Naveen M <naveen.m(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 100 +++++++++++++++++++++ 2 files changed, 101 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5c27f7a..d5f167e 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -315,6 +315,7 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SPI + select SND_SOC_INTEL_SKYLAKE_SSP_CLK select SND_SOC_RT5663 select SND_SOC_RT5514 select SND_SOC_RT5514_SPI diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda87..7b6e70b 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -22,6 +22,9 @@ #include "../../codecs/rt5514.h" #include "../../codecs/rt5663.h" #include "../../codecs/hdac_hdmi.h" +#include <linux/clk.h> +#include <linux/clk-provider.h> +#include <linux/clkdev.h> #define KBL_REALTEK_CODEC_DAI "rt5663-aif" #define KBL_REALTEK_DMIC_CODEC_DAI "rt5514-aif1" @@ -50,6 +53,8 @@ struct kbl_codec_private { struct snd_soc_jack kabylake_headset; struct list_head hdmi_pcm_list; struct snd_soc_jack kabylake_hdmi[2]; + struct clk *mclk; + struct clk *sclk; }; enum { @@ -71,6 +76,67 @@ static const struct snd_kcontrol_new kabylake_controls[] = { SOC_DAPM_PIN_SWITCH("DMIC"), }; +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + /* + * MCLK/SCLK need to be ON early for a successful synchronization of + * codec internal clock. And the clocks are turned off during + * POST_PMD after the stream is stopped. + */ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (__clk_is_enabled(priv->mclk)) + return 0; + + /* Enable MCLK */ + ret = clk_set_rate(priv->mclk, 24000000); + if (ret < 0) { + dev_err(card->dev, "Can't set rate for mclk, err: %d\n", + ret); + return ret; + } + + ret = clk_prepare_enable(priv->mclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + return ret; + } + + /* Enable SCLK */ + ret = clk_set_rate(priv->sclk, 3072000); + if (ret < 0) { + dev_err(card->dev, "Can't set rate for sclk, err: %d\n", + ret); + clk_disable_unprepare(priv->mclk); + return ret; + } + + ret = clk_prepare_enable(priv->sclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); + clk_disable_unprepare(priv->mclk); + } + break; + case SND_SOC_DAPM_POST_PMD: + if (!__clk_is_enabled(priv->mclk)) + return 0; + + clk_disable_unprepare(priv->mclk); + clk_disable_unprepare(priv->sclk); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -79,11 +145,15 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ + { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -92,6 +162,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "Right Spk", NULL, "Right BE_OUT" }, /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, @@ -643,6 +714,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; struct snd_soc_acpi_mach *mach; + int ret = 0; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -658,6 +730,34 @@ static int kabylake_audio_probe(struct platform_device *pdev) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; + ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); + if (IS_ERR(ctx->mclk)) { + ret = PTR_ERR(ctx->mclk); + if (ret == -ENOENT) { + dev_info(&pdev->dev, + "Failed to get ssp1_mclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", + ret); + return ret; + } + + ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); + if (IS_ERR(ctx->sclk)) { + ret = PTR_ERR(ctx->sclk); + if (ret == -ENOENT) { + dev_info(&pdev->dev, + "Failed to get ssp1_sclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", + ret); + return ret; + } + return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.7.4

3 7

[alsa-devel] [PATCH -next] ASoC: SOF: Fix randbuild error
by YueHaibing 14 Oct '19

14 Oct '19

When LEDS_TRIGGER_AUDIO is m and SND_SOC_SOF is y, sound/soc/sof/control.o: In function `snd_sof_switch_put': control.c:(.text+0x587): undefined reference to `ledtrig_audio_set' control.c:(.text+0x593): undefined reference to `ledtrig_audio_set' Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 5d43001ae436 ("ASoC: SOF: acpi led support for switch controls") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sof/control.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 41551e8..2c4abd4 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -36,10 +36,12 @@ static void update_mute_led(struct snd_sof_control *scontrol, scontrol->led_ctl.led_value = temp; +#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) if (!scontrol->led_ctl.direction) ledtrig_audio_set(LED_AUDIO_MUTE, temp ? LED_OFF : LED_ON); else ledtrig_audio_set(LED_AUDIO_MICMUTE, temp ? LED_OFF : LED_ON); +#endif } static inline u32 mixer_to_ipc(unsigned int value, u32 *volume_map, int size) -- 2.7.4

3 2

Re: [alsa-devel] linux-next: Tree for Oct 14 (sound/soc/sof/)
by Randy Dunlap 14 Oct '19

14 Oct '19

On 10/13/19 11:47 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20191011: > on i386: ld: sound/soc/sof/control.o: in function `snd_sof_switch_put': control.c:(.text+0x49a): undefined reference to `ledtrig_audio_set' ld: control.c:(.text+0x4d1): undefined reference to `ledtrig_audio_set' when CONFIG_LEDS_TRIGGER_AUDIO=m CONFIG_SND_SOC_SOF=y This code in <linux/leds.h> does not handle the config combo above: #if IS_ENABLED(CONFIG_LEDS_TRIGGER_AUDIO) enum led_brightness ledtrig_audio_get(enum led_audio type); void ledtrig_audio_set(enum led_audio type, enum led_brightness state); #else static inline enum led_brightness ledtrig_audio_get(enum led_audio type) { return LED_OFF; } static inline void ledtrig_audio_set(enum led_audio type, enum led_brightness state) { } #endif -- ~Randy

1 0

[alsa-devel] TypeError: 'device' is an invalid keyword argument for this function
by GitHub issues - edited 14 Oct '19

14 Oct '19

alsa-project/alsa-lib issue #13 was edited from paddygoat: Hello again! I upgraded Alsa lib to 1.1.7 and tested ok. A small error as below when running in Jupyter Notebook, Python 3.6:: `#inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK) # UltraMic384K 16bit r0, USB Audio inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1)` --------------------------------------------------------------------------- TypeError Traceback (most recent call last) <ipython-input-6-947bfb637aff> in <module> 16 modelName="randomforestPaddyFalse" 17 modelType="randomforest" ---> 18 recordAnalyzeAudio(10, outputWavFile, 2.0, modelName, modelType) <ipython-input-5-8ddddfa5b724> in recordAnalyzeAudio(duration, outputWavFile, midTermBufferSizeSec, modelName, modelType) 87 # UltraMic384K 16bit r0, USB Audio 88 ---> 89 inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') 90 # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1) 91 inp.setchannels(2) TypeError: 'device' is an invalid keyword argument for this function Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/13 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] TypeError: 'device' is an invalid keyword argument for this function
by GitHub issues - opened 14 Oct '19

14 Oct '19

alsa-project/alsa-lib issue #13 was opened from paddygoat: Hello again! I upgraded Alsa lib to 1.1.7 and tested ok. A small error as below when running in Jupyter Notebook, Python 3.6:: ` #inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK) # UltraMic384K 16bit r0, USB Audio inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1)` --------------------------------------------------------------------------- TypeError Traceback (most recent call last) <ipython-input-6-947bfb637aff> in <module> 16 modelName="randomforestPaddyFalse" 17 modelType="randomforest" ---> 18 recordAnalyzeAudio(10, outputWavFile, 2.0, modelName, modelType) <ipython-input-5-8ddddfa5b724> in recordAnalyzeAudio(duration, outputWavFile, midTermBufferSizeSec, modelName, modelType) 87 # UltraMic384K 16bit r0, USB Audio 88 ---> 89 inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') 90 # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1) 91 inp.setchannels(2) TypeError: 'device' is an invalid keyword argument for this function Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/13 Repository URL: https://github.com/alsa-project/alsa-lib

1 0