May 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH][next] ALSA: hda/ca0132: Make a const array static, makes object smaller
by Colin King 27 May '21

27 May '21

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the const array dsp_dma_stream_ids the stack but instead make it static. Makes the object code smaller by 21 bytes. Before: text data bss dec hex filename 189012 70376 192 259580 3f5fc ./sound/pci/hda/patch_ca0132.o After: text data bss dec hex filename 188927 70440 192 259559 3f5e7 ./sound/pci/hda/patch_ca0132.o (gcc version 10.3.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 49b4fdd2feab..b66e7bdbf483 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -7598,7 +7598,7 @@ static void ca0132_alt_free_active_dma_channels(struct hda_codec *codec) */ static void ca0132_alt_start_dsp_audio_streams(struct hda_codec *codec) { - const unsigned int dsp_dma_stream_ids[] = { 0x0c, 0x03, 0x04 }; + static const unsigned int dsp_dma_stream_ids[] = { 0x0c, 0x03, 0x04 }; struct ca0132_spec *spec = codec->spec; unsigned int i, tmp; -- 2.31.1

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[PATCH 1/2] ALSA: drivers: opl3: fix useless self-comparison
by Pierre-Louis Bossart 27 May '21

27 May '21

Sparse throws the following warning: sound/drivers/opl3/opl3_midi.c:183:60: error: self-comparison always evaluates to false This is likely a 16+ year old confusion between vp2 and vp. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/drivers/opl3/opl3_midi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index eb23c55323ae..e1b69c65c3c8 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -180,8 +180,7 @@ static int opl3_get_voice(struct snd_opl3 *opl3, int instr_4op, if (vp2->state == SNDRV_OPL3_ST_ON_2OP) { /* kill two voices, EXPENSIVE */ bp++; - voice_time = (voice_time > vp->time) ? - voice_time : vp->time; + voice_time = max(voice_time, vp2->time); } } else { /* allocate 2op voice */ -- 2.25.1

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[PATCH] sound/oss/dmasound: Remove superfluous "break"
by zuoqilin1＠163.com 27 May '21

27 May '21

From: zuoqilin <zuoqilin(a)yulong.com> Remove superfluous "break", as there is a "return" before them. Signed-off-by: zuoqilin <zuoqilin(a)yulong.com> --- sound/oss/dmasound/dmasound_core.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 22cef0c..0c95828 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -1229,31 +1229,22 @@ static char *get_afmt_string(int afmt) switch(afmt) { case AFMT_MU_LAW: return "mu-law"; - break; case AFMT_A_LAW: return "A-law"; - break; case AFMT_U8: return "unsigned 8 bit"; - break; case AFMT_S8: return "signed 8 bit"; - break; case AFMT_S16_BE: return "signed 16 bit BE"; - break; case AFMT_U16_BE: return "unsigned 16 bit BE"; - break; case AFMT_S16_LE: return "signed 16 bit LE"; - break; case AFMT_U16_LE: return "unsigned 16 bit LE"; - break; case 0: return "format not set" ; - break ; default: break ; } -- 1.9.1

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Soft jack injection for USB audio?
by Kai-Heng Feng 27 May '21

27 May '21

I have the need to use soft jack injection to debug userpsace, but currently USB audio doesn't have that ability. The problem I am facing is that the USB audio doesn't use snd_jack_* to control the jack, so we can add the support for USB audio separately. It's not hard to add the support if we open code soft injection in USB audio's build_connector_control(), but would it be possible to use snd_jack_* in USB audio? Kai-Heng

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[PATCH v2 0/7] ASoC: adds new .get_fmt support
by Kuninori Morimoto 27 May '21

27 May '21

Hi Mark These are v2 of .get_fmt support. This is a little bit challenging patch-set. The idea/code is almost same as v1. We need to set dai_link->dai_fmt to select CPU/Codec settings, and it is selected by Sound Card Driver, today. Because of it, Sound Card user need to know both CPU / Codec available dai_fmt, and needs to select it. For example simple-card / audio-graph case, it is selected by "format" and "bitclock/frame-master/inversion" on DT. But, it can be automatically selected if both CPU and Codec drivers indicate it to ALSA SoC Framework, somehow. By this patch, dai_fmt can be automatically selected from each driver if both CPU / Codec driver had .get_dai_fmt callback. Automatically selectable *field* is depends on each drivers. For example, some driver want to select format "automatically", but want to select other fields "manually", because of complex limitation. Or other example, in case of both CPU and Codec are possible to be clock provider, but the quality was different. In these case, user need/want to *manually* select each fields from Sound Card driver. It uses Sound Card specified fields preferentially, and try to select non-specific fields from CPU and Codec driver settings if driver had callbacks. In other words, we can select all dai_fmt via Sound Card driver same as before. Select dai_fmt 100% automatically is very difficult and will be very complex, but select automatically some fields only is very easy, I guess. This patch-set is based on such assumption. v1 -> v2 - Add more detail explanation on git-log, code, comment. - Possible to be Clock/Frame provider is depends on driver's situation. Link: https://lore.kernel.org/r/871rb3hypy.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (7): ASoC: soc-core: move snd_soc_runtime_set_dai_fmt() to upside ASoC: soc-core: add snd_soc_runtime_get_dai_fmt() ASoC: ak4613: add .get_fmt support ASoC: pcm3168a: add .get_fmt support ASoC: rsnd: add .get_fmt support ASoC: fsi: add .get_fmt support ASoC: hdmi-codec: add .get_fmt support include/sound/soc-dai.h | 45 +++++++ sound/soc/codecs/ak4613.c | 14 ++ sound/soc/codecs/hdmi-codec.c | 22 ++++ sound/soc/codecs/pcm3168a.c | 19 +++ sound/soc/sh/fsi.c | 19 +++ sound/soc/sh/rcar/adg.c | 8 ++ sound/soc/sh/rcar/core.c | 24 +++- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/soc-core.c | 234 +++++++++++++++++++++++++--------- sound/soc/soc-dai.c | 18 +++ sound/soc/soc-utils.c | 32 +++++ 11 files changed, 372 insertions(+), 64 deletions(-) -- 2.25.1

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Working to make ALSA work on a gemini lake tablet
by Sergio Callegari 26 May '21

26 May '21

Hi, I am trying to understand the issues that ALSA has on a recent gemini lake tablet (the CHUWI HI 10 X) for which otherwise linux support is in rather good shape. In case I am successful, I'd be glad to try to set up a guide for users encountering the same or a similar setup. On this machine, even on very recent kernels, it is impossible to have the soundcard recognized (only HDMI audio outputs are shown for aplay -l). The tablet advertises Intel SST and to the best of my understanding that implies that in addition to the basic Intel HDA support there should be some codec chip that needs support too (this used to be the bytcrrt5651 in a previous version of the tablet not based on gemini lake). I also see that in many similar pieces of hardware, regular (non HDMI) audio support can only be activated by setting some GPIO outputs (e.g., https://github.com/danielotero/linux-on-hi10/issues/8). My biggest element of concern at this point is that I am not able to identify the CODEC part, nor to understand if some specific UCM file or firmware file is needed. I wonder if anybody could point me to some hints at how to properly identify the hardware and the issue. I have windows pre-installed on the system but even from there I have so far been unable to identify the audio chipset properly. Thanks for the attention and even more thanks for any help, Best, Sergio

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[PATCH -next] ASoC: cs42l56: use DEVICE_ATTR_WO macro
by YueHaibing 26 May '21

26 May '21

Use DEVICE_ATTR_WO() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/cs42l56.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 7cdffdf6b8cf..3cf8a0b4478c 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1021,9 +1021,8 @@ static int cs42l56_beep_event(struct input_dev *dev, unsigned int type, return 0; } -static ssize_t cs42l56_beep_set(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t beep_store(struct device *dev, struct device_attribute *attr, + const char *buf, size_t count) { struct cs42l56_private *cs42l56 = dev_get_drvdata(dev); long int time; @@ -1038,7 +1037,7 @@ static ssize_t cs42l56_beep_set(struct device *dev, return count; } -static DEVICE_ATTR(beep, 0200, NULL, cs42l56_beep_set); +static DEVICE_ATTR_WO(beep); static void cs42l56_init_beep(struct snd_soc_component *component) { -- 2.17.1

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[PATCH -next] ASoC: cs42l52: use DEVICE_ATTR_WO macro
by YueHaibing 26 May '21

26 May '21

Use DEVICE_ATTR_WO() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/cs42l52.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 88547e2cd53d..80161151b3f2 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -957,9 +957,8 @@ static int cs42l52_beep_event(struct input_dev *dev, unsigned int type, return 0; } -static ssize_t cs42l52_beep_set(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t beep_store(struct device *dev, struct device_attribute *attr, + const char *buf, size_t count) { struct cs42l52_private *cs42l52 = dev_get_drvdata(dev); long int time; @@ -974,7 +973,7 @@ static ssize_t cs42l52_beep_set(struct device *dev, return count; } -static DEVICE_ATTR(beep, 0200, NULL, cs42l52_beep_set); +static DEVICE_ATTR_WO(beep); static void cs42l52_init_beep(struct snd_soc_component *component) { -- 2.17.1

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[PATCH -next] ASoC: cs43130: Use DEVICE_ATTR_RO macro
by YueHaibing 26 May '21

26 May '21

Use DEVICE_ATTR_RO() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/cs43130.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index de7fa3e4bfbb..44b20c1ef851 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1672,14 +1672,14 @@ static int cs43130_show_dc(struct device *dev, char *buf, u8 ch) cs43130->hpload_dc[ch]); } -static ssize_t cs43130_show_dc_l(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t hpload_dc_l_show(struct device *dev, + struct device_attribute *attr, char *buf) { return cs43130_show_dc(dev, buf, HP_LEFT); } -static ssize_t cs43130_show_dc_r(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t hpload_dc_r_show(struct device *dev, + struct device_attribute *attr, char *buf) { return cs43130_show_dc(dev, buf, HP_RIGHT); } @@ -1719,22 +1719,22 @@ static int cs43130_show_ac(struct device *dev, char *buf, u8 ch) } } -static ssize_t cs43130_show_ac_l(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t hpload_ac_l_show(struct device *dev, + struct device_attribute *attr, char *buf) { return cs43130_show_ac(dev, buf, HP_LEFT); } -static ssize_t cs43130_show_ac_r(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t hpload_ac_r_show(struct device *dev, + struct device_attribute *attr, char *buf) { return cs43130_show_ac(dev, buf, HP_RIGHT); } -static DEVICE_ATTR(hpload_dc_l, 0444, cs43130_show_dc_l, NULL); -static DEVICE_ATTR(hpload_dc_r, 0444, cs43130_show_dc_r, NULL); -static DEVICE_ATTR(hpload_ac_l, 0444, cs43130_show_ac_l, NULL); -static DEVICE_ATTR(hpload_ac_r, 0444, cs43130_show_ac_r, NULL); +static DEVICE_ATTR_RO(hpload_dc_l); +static DEVICE_ATTR_RO(hpload_dc_r); +static DEVICE_ATTR_RO(hpload_ac_l); +static DEVICE_ATTR_RO(hpload_ac_r); static struct attribute *hpload_attrs[] = { &dev_attr_hpload_dc_l.attr, -- 2.17.1

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[PATCH] ASoC: cs35l3x: Use neutral language in amp drivers
by David Rhodes 26 May '21

26 May '21

Revise variable names and comments in cs35l35 and cs35l36 amp drivers. Signed-off-by: David Rhodes <drhodes(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l35.c | 16 ++++++++-------- sound/soc/codecs/cs35l35.h | 2 +- sound/soc/codecs/cs35l36.c | 16 ++++++++-------- 3 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index f20ed838b958..f4e2dc0ed5cc 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -367,16 +367,16 @@ static int cs35l35_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct snd_soc_component *component = codec_dai->component; struct cs35l35_private *cs35l35 = snd_soc_component_get_drvdata(component); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, CS35L35_MS_MASK, 1 << CS35L35_MS_SHIFT); - cs35l35->slave_mode = false; + cs35l35->clock_consumer = false; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, CS35L35_MS_MASK, 0 << CS35L35_MS_SHIFT); - cs35l35->slave_mode = true; + cs35l35->clock_consumer = true; break; default: return -EINVAL; @@ -555,8 +555,8 @@ static int cs35l35_hw_params(struct snd_pcm_substream *substream, } sp_sclks = ((cs35l35->sclk / srate) / 4) - 1; - /* Only certain ratios are supported in I2S Slave Mode */ - if (cs35l35->slave_mode) { + /* Only certain ratios supported when device is a clock consumer */ + if (cs35l35->clock_consumer) { switch (sp_sclks) { case CS35L35_SP_SCLKS_32FS: case CS35L35_SP_SCLKS_48FS: @@ -567,7 +567,7 @@ static int cs35l35_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } } else { - /* Only certain ratios supported in I2S MASTER Mode */ + /* Only certain ratios supported when device is a clock provider */ switch (sp_sclks) { case CS35L35_SP_SCLKS_32FS: case CS35L35_SP_SCLKS_64FS: diff --git a/sound/soc/codecs/cs35l35.h b/sound/soc/codecs/cs35l35.h index ffb154cd962c..2e69fef318e3 100644 --- a/sound/soc/codecs/cs35l35.h +++ b/sound/soc/codecs/cs35l35.h @@ -282,7 +282,7 @@ struct cs35l35_private { int sclk; bool pdm_mode; bool i2s_mode; - bool slave_mode; + bool clock_consumer; /* GPIO for /RST */ struct gpio_desc *reset_gpio; struct completion pdn_done; diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index a038bcec2d17..895ceed12cd9 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -756,14 +756,14 @@ static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, { struct cs35l36_private *cs35l36 = snd_soc_component_get_drvdata(component_dai->component); - unsigned int asp_fmt, lrclk_fmt, sclk_fmt, slave_mode, clk_frc; + unsigned int asp_fmt, lrclk_fmt, sclk_fmt, clock_provider, clk_frc; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - slave_mode = 1; + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + clock_provider = 1; break; - case SND_SOC_DAIFMT_CBS_CFS: - slave_mode = 0; + case SND_SOC_DAIFMT_CBC_CFC: + clock_provider = 0; break; default: return -EINVAL; @@ -771,10 +771,10 @@ static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, CS35L36_SCLK_MSTR_MASK, - slave_mode << CS35L36_SCLK_MSTR_SHIFT); + clock_provider << CS35L36_SCLK_MSTR_SHIFT); regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, CS35L36_LRCLK_MSTR_MASK, - slave_mode << CS35L36_LRCLK_MSTR_SHIFT); + clock_provider << CS35L36_LRCLK_MSTR_SHIFT); switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { case SND_SOC_DAIFMT_CONT: -- 2.25.1

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