July 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: codecs: va-macro: use fsgen as clock
by Srinivas Kandagatla 27 Jul '22

27 Jul '22

VA Macro fsgen clock is supplied to other LPASS Macros using proper clock apis, however the internal user uses the registers directly without clk apis. This approch has race condition where in external users of the clock might cut the clock while VA macro is actively using this. Moving the internal usage to clk apis would provide a proper refcounting and avoid such race conditions. This issue was noticed while headset was pulled out while recording is in progress and shifting record patch to DMIC. Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/lpass-va-macro.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index d18b56e60433..1ea10dc70748 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -199,6 +199,7 @@ struct va_macro { struct clk *mclk; struct clk *macro; struct clk *dcodec; + struct clk *fsgen; struct clk_hw hw; struct lpass_macro *pds; @@ -467,9 +468,9 @@ static int va_macro_mclk_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - return va_macro_mclk_enable(va, true); + return clk_prepare_enable(va->fsgen); case SND_SOC_DAPM_POST_PMD: - return va_macro_mclk_enable(va, false); + clk_disable_unprepare(va->fsgen); } return 0; @@ -1473,6 +1474,12 @@ static int va_macro_probe(struct platform_device *pdev) if (ret) goto err_clkout; + va->fsgen = clk_hw_get_clk(&va->hw, "fsgen"); + if (IS_ERR(va->fsgen)) { + ret = PTR_ERR(va->fsgen); + goto err_clkout; + } + ret = devm_snd_soc_register_component(dev, &va_macro_component_drv, va_macro_dais, ARRAY_SIZE(va_macro_dais)); -- 2.25.1

3 2

snd_soc_skl - No sound on PopOS
by GitHub issues - opened 27 Jul '22

27 Jul '22

alsa-project/alsa-lib issue #255 was opened from runcros: Hello, I have no sound on skylake nau8825 max98357a here is my alsa-info : http://alsa-project.org/db/?f=c1b90f94c1bb072f507636f3979c0bc8a62ae470 Can you help me please If you need others infos, I am here =) thank you ! Issue URL : https://github.com/alsa-project/alsa-lib/issues/255 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: mediatek: mt8186: remove unnecessary judgments
by Jiaxin Yu 27 Jul '22

27 Jul '22

The afe_priv->dai_priv[] is allocated when platform driver probe(), if it failed, the ASoC platform driver probe() will return fail first. Therefore, this is excessive judgment, and the condition will never be established. Bug report: https://www.spinics.net/lists/alsa-devel/msg145609.html This is a semi-automatic email about new static checker warnings. The patch ae92dcbee8b6: "ASoC: mediatek: mt8186: support tdm in platform driver" from May 23, 2022, leads to the following Smatch complaint: sound/soc/mediatek/mt8186/mt8186-dai-tdm.c:424 mtk_dai_tdm_hw_params() warn: variable dereferenced before check 'tdm_priv' (see line 406) sound/soc/mediatek/mt8186/mt8186-dai-tdm.c 405 struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[tdm_id]; 406 unsigned int tdm_mode = tdm_priv->tdm_mode; ^^^^^^^^^^^^^^^^^^^ Lot's of dereferences 407 unsigned int data_mode = tdm_priv->data_mode; 408 unsigned int rate = params_rate(params); 409 unsigned int channels = params_channels(params); 410 snd_pcm_format_t format = params_format(params); 411 unsigned int bit_width = 412 snd_pcm_format_physical_width(format); 413 unsigned int tdm_channels = (data_mode == TDM_DATA_ONE_PIN) ? 414 get_tdm_ch_per_sdata(tdm_mode, channels) : 2; 415 unsigned int lrck_width = 416 get_tdm_lrck_width(format, tdm_mode); 417 unsigned int tdm_con = 0; 418 bool slave_mode = tdm_priv->slave_mode; 419 bool lrck_inv = tdm_priv->lck_invert; 420 bool bck_inv = tdm_priv->bck_invert; 421 unsigned int tran_rate; 422 unsigned int tran_relatch_rate; 423 424 if (!tdm_priv) { ^^^^^^^^^ Checked too late 425 dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); 426 return -EINVAL; Fixes: ae92dcbee8b6 ("ASoC: mediatek: mt8186: support tdm in platform driver") Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 5 -- sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 63 --------------------- sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 5 -- sound/soc/mediatek/mt8186/mt8186-dai-tdm.c | 55 +----------------- 4 files changed, 1 insertion(+), 127 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index 6be6d4f3b585..f6f23c7c3ff5 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -655,11 +655,6 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, dev_dbg(afe->dev, "%s(), id %d, stream %d, rate %d\n", __func__, id, substream->stream, rate); - if (!adda_priv) { - dev_err(afe->dev, "%s(), adda_priv == NULL", __func__); - return -EINVAL; - } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { unsigned int dl_src2_con0; unsigned int dl_src2_con1; diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c index 5c1290b950e8..ec79e2f2a54d 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c @@ -119,12 +119,6 @@ static int mt8186_i2s_hd_get(struct snd_kcontrol *kcontrol, struct mtk_afe_i2s_priv *i2s_priv; i2s_priv = get_i2s_priv_by_name(afe, kcontrol->id.name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - ucontrol->value.integer.value[0] = i2s_priv->low_jitter_en; return 0; @@ -148,12 +142,6 @@ static int mt8186_i2s_hd_set(struct snd_kcontrol *kcontrol, __func__, kcontrol->id.name, hd_en); i2s_priv = get_i2s_priv_by_name(afe, kcontrol->id.name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - if (i2s_priv->low_jitter_en == hd_en) return 0; @@ -377,11 +365,6 @@ static int mtk_i2s_en_event(struct snd_soc_dapm_widget *w, i2s_priv = get_i2s_priv_by_name(afe, w->name); - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x\n", __func__, w->name, event); @@ -442,11 +425,6 @@ static int mtk_mclk_en_event(struct snd_soc_dapm_widget *w, i2s_priv = get_i2s_priv_by_name(afe, w->name); - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - switch (event) { case SND_SOC_DAPM_PRE_PMU: mt8186_mck_enable(afe, i2s_priv->mclk_id, i2s_priv->mclk_rate); @@ -566,12 +544,6 @@ static int mtk_afe_i2s_share_connect(struct snd_soc_dapm_widget *source, struct mtk_afe_i2s_priv *i2s_priv; i2s_priv = get_i2s_priv_by_name(afe, sink->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - if (i2s_priv->share_i2s_id < 0) return 0; @@ -587,12 +559,6 @@ static int mtk_afe_i2s_hd_connect(struct snd_soc_dapm_widget *source, struct mtk_afe_i2s_priv *i2s_priv; i2s_priv = get_i2s_priv_by_name(afe, sink->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - if (get_i2s_id_by_name(afe, sink->name) == get_i2s_id_by_name(afe, source->name)) return i2s_priv->low_jitter_en; @@ -618,15 +584,8 @@ static int mtk_afe_i2s_apll_connect(struct snd_soc_dapm_widget *source, int i2s_need_apll; i2s_priv = get_i2s_priv_by_name(afe, w->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - /* which apll */ cur_apll = mt8186_get_apll_by_name(afe, source->name); - /* choose APLL from i2s rate */ i2s_need_apll = mt8186_get_apll_by_rate(afe, i2s_priv->rate); @@ -642,12 +601,6 @@ static int mtk_afe_i2s_mclk_connect(struct snd_soc_dapm_widget *source, struct mtk_afe_i2s_priv *i2s_priv; i2s_priv = get_i2s_priv_by_name(afe, sink->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - if (get_i2s_id_by_name(afe, sink->name) == get_i2s_id_by_name(afe, source->name)) return (i2s_priv->mclk_rate > 0) ? 1 : 0; @@ -672,12 +625,6 @@ static int mtk_afe_mclk_apll_connect(struct snd_soc_dapm_widget *source, int cur_apll; i2s_priv = get_i2s_priv_by_name(afe, w->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - /* which apll */ cur_apll = mt8186_get_apll_by_name(afe, source->name); @@ -980,11 +927,6 @@ static int mtk_dai_i2s_config(struct mtk_base_afe *afe, dev_dbg(afe->dev, "%s(), id %d, rate %d, format %d\n", __func__, i2s_id, rate, format); - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - i2s_priv->rate = rate; switch (i2s_id) { @@ -1053,11 +995,6 @@ static int mtk_dai_i2s_set_sysclk(struct snd_soc_dai *dai, int apll; int apll_rate; - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - if (dir != SND_SOC_CLOCK_OUT) { dev_err(afe->dev, "%s(), dir != SND_SOC_CLOCK_OUT", __func__); return -EINVAL; diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c b/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c index 0b0032ecfe6d..41221a66111c 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c @@ -287,11 +287,6 @@ static int mtk_dai_pcm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct mt8186_afe_private *afe_priv = afe->platform_priv; struct mtk_afe_pcm_priv *pcm_priv = afe_priv->dai_priv[dai->id]; - if (!pcm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - /* DAI mode*/ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c b/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c index c6ead7c252f0..4148dceb3a4c 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c @@ -106,11 +106,6 @@ static int mtk_tdm_en_event(struct snd_soc_dapm_widget *w, int dai_id = get_tdm_id_by_name(w->name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x\n", __func__, w->name, event); @@ -138,11 +133,6 @@ static int mtk_tdm_mck_en_event(struct snd_soc_dapm_widget *w, int dai_id = get_tdm_id_by_name(w->name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x, dai_id %d\n", __func__, w->name, event, dai_id); @@ -215,11 +205,6 @@ static int mtk_afe_tdm_mclk_connect(struct snd_soc_dapm_widget *source, int dai_id = get_tdm_id_by_name(w->name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return 0; - } - return (tdm_priv->mclk_rate > 0) ? 1 : 0; } @@ -250,11 +235,6 @@ static int mtk_afe_tdm_hd_connect(struct snd_soc_dapm_widget *source, int dai_id = get_tdm_id_by_name(w->name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return 0; - } - return tdm_priv->low_jitter_en; } @@ -270,11 +250,6 @@ static int mtk_afe_tdm_apll_connect(struct snd_soc_dapm_widget *source, int cur_apll; int tdm_need_apll; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return 0; - } - /* which apll */ cur_apll = mt8186_get_apll_by_name(afe, source->name); @@ -303,11 +278,6 @@ static int mt8186_tdm_hd_get(struct snd_kcontrol *kcontrol, int dai_id = get_tdm_id_by_name(kcontrol->id.name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - ucontrol->value.integer.value[0] = tdm_priv->low_jitter_en; return 0; @@ -332,11 +302,6 @@ static int mt8186_tdm_hd_set(struct snd_kcontrol *kcontrol, dev_dbg(afe->dev, "%s(), kcontrol name %s, hd_en %d\n", __func__, kcontrol->id.name, hd_en); - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - if (tdm_priv->low_jitter_en == hd_en) return 0; @@ -421,22 +386,14 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream, unsigned int tran_rate; unsigned int tran_relatch_rate; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - tdm_priv->rate = rate; - tran_rate = mt8186_rate_transform(afe->dev, rate, dai->id); tran_relatch_rate = mt8186_tdm_relatch_rate_transform(afe->dev, rate); /* calculate mclk_rate, if not set explicitly */ if (!tdm_priv->mclk_rate) { tdm_priv->mclk_rate = rate * tdm_priv->mclk_multiple; - mtk_dai_tdm_cal_mclk(afe, - tdm_priv, - tdm_priv->mclk_rate); + mtk_dai_tdm_cal_mclk(afe, tdm_priv, tdm_priv->mclk_rate); } /* ETDM_IN1_CON0 */ @@ -508,11 +465,6 @@ static int mtk_dai_tdm_set_sysclk(struct snd_soc_dai *dai, struct mt8186_afe_private *afe_priv = afe->platform_priv; struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai->id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - if (dir != SND_SOC_CLOCK_IN) { dev_err(afe->dev, "%s(), dir != SND_SOC_CLOCK_OUT", __func__); return -EINVAL; @@ -529,11 +481,6 @@ static int mtk_dai_tdm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct mt8186_afe_private *afe_priv = afe->platform_priv; struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai->id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - /* DAI mode*/ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: -- 2.25.1

2 1

[PATCH] ASoC: amd: acp: uninitialized variable in i2s_irq_handler()
by Dan Carpenter 27 Jul '22

27 Jul '22

The "ext_intr_stat1" is used without being initialized. Fixes: 93f53881473c ("ASoC: amd: acp: Modify local variables name to generic") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/amd/acp/acp-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 10730d33c3b0..f561d39b33e2 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -100,7 +100,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) return IRQ_NONE; if (adata->rsrc->no_of_ctrls == 2) - ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); + ext_intr_stat1 = readl(ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); -- 2.35.1

2 1

[PATCH 0/2] Add support for CLSA0101
by Lucas Tanure 27 Jul '22

27 Jul '22

Add Support for the CLSA0101 laptop, an Intel version of CLSA0100. This patch has been tested using the CLSA0100, ensuring it doesn't break the sound for it. We appreciate it if someone with CLSA0101 could verify that this the patch works for them. Lucas Tanure (2): ALSA: hda: cs35l41: Use the CS35L41 HDA internal define ALSA: hda: cs35l41: Support CLSA0101 sound/pci/hda/cs35l41_hda.c | 67 +++++++++++++++++++++-------------- sound/pci/hda/patch_realtek.c | 12 +++++++ 2 files changed, 53 insertions(+), 26 deletions(-) -- 2.37.1

3 6

[PATCH 0/4] ALSA: Defer async signal handling
by Takashi Iwai 27 Jul '22

27 Jul '22

Hi, this series is another attempt to address the potential deadlocks via kill_fasync() calls that have been reported by syzbot for long time. Instead of the previous series to drop the async handler[*], this tries to defer the kill_fasync() call. A few new common helpers are introduced at first, then the actual usages are replaced in the following patches. The patches passed the quick tests with alsa-lib test cases. Takashi [*] https://lore.kernel.org/r/20220717070549.5993-1-tiwai@suse.de === Takashi Iwai (4): ALSA: core: Add async signal helpers ALSA: timer: Use deferred fasync helper ALSA: pcm: Use deferred fasync helper ALSA: control: Use deferred fasync helper include/sound/control.h | 2 +- include/sound/core.h | 8 ++++ include/sound/pcm.h | 2 +- sound/core/control.c | 7 ++-- sound/core/misc.c | 93 +++++++++++++++++++++++++++++++++++++++++ sound/core/pcm.c | 1 + sound/core/pcm_lib.c | 2 +- sound/core/pcm_native.c | 2 +- sound/core/timer.c | 11 ++--- 9 files changed, 116 insertions(+), 12 deletions(-) -- 2.35.3

2 6

HDA ATI HDMI not supported
by GitHub issues - opened 27 Jul '22

27 Jul '22

alsa-project/alsa-ucm-conf issue #194 was opened from yangxiaohua2009: I am using the latest ucm2 file and alsalib alsainfo [http://alsa-project.org/db/?f=92c58f3a6b72fee05b678386858dbc816d25dbf3](htt… When nothing is changed, the alsaucm reload command returns error saying: `ALSA lib parser.c:244:(error_node) UCM is not supported for this HDA model (HDA ATI HDMI at 0xfeb60000 irq 28)` and the HDMI is not working. After changing the file /ucm2/HDA/HDA.conf, from `Define.AcpCardId "$${find-card:field=name,return=id,regex='^(acp|acp6x)$'}"` to `Define.AcpCardId "$${find-card:field=name,return=id,regex='^(acp|acp6x|acpes8336)$'}"` alsucm reload returns no error, but alsaucm listcards returns: `ALSA lib main.c:1541:(snd_use_case_mgr_open) error: failed to import hw:1 use case configuration -2` ` 0: hw:0` ` HDA ATI HDMI at 0xfeb60000 irq 28` despite the fact that I already added the acpes8336.conf in conf.d and acpes8336 folder. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/194 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: mediatek: mt8186: set the correct string to strncmp()
by Jiaxin Yu 26 Jul '22

26 Jul '22

Fix Smatch static checker warning. strncmp() here only needs to compare the first seven bytes, so in order to make the code more clear, only the first seven bytes of the string used as the comparison are reserved. Bug report: https://www.spinics.net/lists/alsa-devel/msg145608.html sound/soc/mediatek/mt8186/mt8186-dai-adda.c:78 get_adda_priv_by_name() warn: strncmp() with weird length: 17 vs 7 sound/soc/mediatek/mt8186/mt8186-dai-adda.c 72 static struct mtk_afe_adda_priv *get_adda_priv_by_name(struct mtk_base_afe *afe, 73 const char *name) 74 { 75 struct mt8186_afe_private *afe_priv = afe->platform_priv; 76 int dai_id; 77 --> 78 if (strncmp(name, "aud_dac_hires_clk", 7) == 0 || 79 strncmp(name, "aud_adc_hires_clk", 7) == 0) Fixes: b65c466220b3 ("ASoC: mediatek: mt8186: support adda in platform driver") Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index db71b032770d..81976f7c444e 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -75,8 +75,7 @@ static struct mtk_afe_adda_priv *get_adda_priv_by_name(struct mtk_base_afe *afe, struct mt8186_afe_private *afe_priv = afe->platform_priv; int dai_id; - if (strncmp(name, "aud_dac_hires_clk", 7) == 0 || - strncmp(name, "aud_adc_hires_clk", 7) == 0) + if (strncmp(name, "aud_dac", 7) == 0 || strncmp(name, "aud_adc", 7) == 0) dai_id = MT8186_DAI_ADDA; else return NULL; -- 2.25.1

2 1

[PATCH v1] ASoC: Intel: cirrus-common: Use UID to map correct amp to prefix
by Stefan Binding 26 Jul '22

26 Jul '22

Since the order of the amps in the ACPI determines the device name, and the ACPI order may change depending on hardware configuration, use UID to dynamically compute the dai links, allowing dynamic assignment of the name_prefix. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/intel/boards/sof_cirrus_common.c | 73 +++++++++------------- 1 file changed, 31 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index f4192df962d6..a96bc70a34d7 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -10,6 +10,9 @@ #include "../../codecs/cs35l41.h" #include "sof_cirrus_common.h" +#define CS35L41_HID "CSC3541" +#define CS35L41_MAX_AMPS 4 + /* * Cirrus Logic CS35L41/CS35L53 */ @@ -35,50 +38,12 @@ static const struct snd_soc_dapm_route cs35l41_dapm_routes[] = { {"TR Spk", NULL, "TR SPK"}, }; -static struct snd_soc_dai_link_component cs35l41_components[] = { - { - .name = CS35L41_DEV0_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV1_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV2_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV3_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, -}; +static struct snd_soc_dai_link_component cs35l41_components[CS35L41_MAX_AMPS]; /* * Mapping between ACPI instance id and speaker position. - * - * Four speakers: - * 0: Tweeter left, 1: Woofer left - * 2: Tweeter right, 3: Woofer right */ -static struct snd_soc_codec_conf cs35l41_codec_conf[] = { - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV0_NAME), - .name_prefix = "TL", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV1_NAME), - .name_prefix = "WL", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV2_NAME), - .name_prefix = "TR", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV3_NAME), - .name_prefix = "WR", - }, -}; +static struct snd_soc_codec_conf cs35l41_codec_conf[CS35L41_MAX_AMPS]; static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) { @@ -117,10 +82,10 @@ static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) static const struct { unsigned int rx[2]; } cs35l41_channel_map[] = { - {.rx = {0, 1}}, /* TL */ {.rx = {0, 1}}, /* WL */ - {.rx = {1, 0}}, /* TR */ {.rx = {1, 0}}, /* WR */ + {.rx = {0, 1}}, /* TL */ + {.rx = {1, 0}}, /* TR */ }; static int cs35l41_hw_params(struct snd_pcm_substream *substream, @@ -175,8 +140,32 @@ static const struct snd_soc_ops cs35l41_ops = { .hw_params = cs35l41_hw_params, }; +static const char * const cs35l41_name_prefixes[] = { "WL", "WR", "TL", "TR" }; + +static const char * const cs35l41_uid_strings[] = { "0", "1", "2", "3" }; + +static void cs35l41_compute_codec_conf(void) +{ + int uid; + struct acpi_device *adev; + struct device *physdev; + + for (uid = 0; uid < CS35L41_MAX_AMPS; uid++) { + adev = acpi_dev_get_first_match_dev(CS35L41_HID, cs35l41_uid_strings[uid], -1); + if (!adev) + return; + physdev = get_device(acpi_get_first_physical_node(adev)); + cs35l41_components[uid].name = dev_name(physdev); + cs35l41_components[uid].dai_name = CS35L41_CODEC_DAI; + cs35l41_codec_conf[uid].dlc.name = dev_name(physdev); + cs35l41_codec_conf[uid].name_prefix = cs35l41_name_prefixes[uid]; + acpi_dev_put(adev); + } +} + void cs35l41_set_dai_link(struct snd_soc_dai_link *link) { + cs35l41_compute_codec_conf(); link->codecs = cs35l41_components; link->num_codecs = ARRAY_SIZE(cs35l41_components); link->init = cs35l41_init; -- 2.34.1

3 2

[PATCH v7] ASoc: dt-bindings: tas2780: remove tas2780 and will merge it to tas27xx
by Raphael-Xu 26 Jul '22

26 Jul '22

remove tas2780 Signed-off-by: Raphael-Xu <13691752556(a)139.com> --- .../devicetree/bindings/sound/tas2780.yaml | 75 ------------------- 1 file changed, 75 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2780.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2780.yaml b/Documentation/devicetree/bindings/sound/tas2780.yaml deleted file mode 100644 index 21389a16fdb9..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2780.yaml +++ /dev/null @@ -1,75 +0,0 @@ -# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) -# Copyright (C) 2020-2022 Texas Instruments Incorporated -%YAML 1.2 ---- -$id: "http://devicetree.org/schemas/sound/tas2780.yaml#" -$schema: "http://devicetree.org/meta-schemas/core.yaml#" - -title: Texas Instruments TAS2780 Smart PA - -maintainers: - - Raphael Xu <raphael-xu(a)ti.com> - -description: | - The TAS2780 is a mono, digital input Class-D audio amplifier optimized for - efficiently driving high peak power into small loudspeakers. - Integrated speaker voltage and current sense provides for - real time monitoring of loudspeaker behavior. - -properties: - compatible: - enum: - - ti,tas2780 - - reg: - description: - I2C address of the device can be between 0x38 to 0x3f. - minimum: 0x38 - maximum: 0x3f - - reset-gpios: - description: GPIO used to reset the device. - maxItems: 1 - - interrupts: - maxItems: 1 - - ti,imon-slot-no: - description: TDM TX current sense time slot. - $ref: /schemas/types.yaml#/definitions/uint32 - minimum: 0 - maximum: 3 - - ti,vmon-slot-no: - description: TDM TX voltage sense time slot. - $ref: /schemas/types.yaml#/definitions/uint32 - minimum: 0 - maximum: 3 - - '#sound-dai-cells': - const: 1 - -required: - - compatible - - reg - -additionalProperties: false - -examples: - - | - #include <dt-bindings/gpio/gpio.h> - i2c0 { - #address-cells = <1>; - #size-cells = <0>; - codec: codec@38 { - compatible = "ti,tas2780"; - reg = <0x38>; - #sound-dai-cells = <1>; - interrupts = <14>; - reset-gpios = <&gpio1 15 0>; - ti,imon-slot-no = <0>; - ti,vmon-slot-no = <2>; - }; - }; - -... -- 2.35.1

3 4