November 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 0/3] ASoC: mediatek:mt8186: fix both the speaker and hdmi
by Jiaxin Yu 06 Dec '24

06 Dec '24

The speaker and hdmi of mt8186 platform are shared the same port of I2S, when connect the external display, use build-in speakers to play audio, they both playback at the same time. So we want to manage the playback device through DAPM events. Change since v1: Support audio function for it6505 bridge to hdmi-codec Jiaxin Yu (3): ASoC: hdmi-codec: Add event handler for hdmi TX ASoC: mediatek: mt8186: correct the HDMI widgets drm/bridge: it6505: Add audio support drivers/gpu/drm/bridge/ite-it6505.c | 81 +++++++++++++++++-- include/sound/hdmi-codec.h | 6 ++ sound/soc/codecs/hdmi-codec.c | 37 ++++++++- .../mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 5 files changed, 116 insertions(+), 12 deletions(-) -- 2.18.0

3 16

[PATCH] ASoC: tas2781: Fix redundant logical jump
by Tang Bin 02 Dec '24

02 Dec '24

In these functions, some logical jump of "goto" and variable are redundant, thus remove them. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/codecs/tas2781-i2c.c | 68 +++++++++++++--------------------- 1 file changed, 25 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 68887799e..27f2f84a4 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -489,14 +489,11 @@ static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); const int sum = ARRAY_SIZE(tas2563_cali_start_reg); - int rc = 1; int i, j; guard(mutex)(&tas_priv->codec_lock); - if (tas_priv->chip_id != TAS2563) { - rc = -1; - goto out; - } + if (tas_priv->chip_id != TAS2563) + return -1; for (i = 0; i < tas_priv->ndev; i++) { struct tasdevice *tasdev = tas_priv->tasdevice; @@ -523,8 +520,8 @@ static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, q[j].val, 4); } } -out: - return rc; + + return 1; } static void tas2563_calib_stop_put(struct tasdevice_priv *tas_priv) @@ -576,7 +573,7 @@ static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, struct cali_reg *p = &cali_data->cali_reg_array; unsigned char *src = ucontrol->value.bytes.data; unsigned char *dst = cali_data->data; - int rc = 1, i = 0; + int i = 0; int j; guard(mutex)(&priv->codec_lock); @@ -605,7 +602,7 @@ static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, i += 3; memcpy(dst, &src[i], cali_data->total_sz); - return rc; + return 1; } static int tas2781_latch_reg_get(struct snd_kcontrol *kcontrol, @@ -1115,25 +1112,21 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) char *conf_name, *prog_name; int nr_controls = 4; int mix_index = 0; - int ret; /* Alloc kcontrol via devm_kzalloc, which don't manually * free the kcontrol */ dsp_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, sizeof(dsp_ctrls[0]), GFP_KERNEL); - if (!dsp_ctrls) { - ret = -ENOMEM; - goto out; - } + if (!dsp_ctrls) + return -ENOMEM; /* Create mixer items for selecting the active Program and Config */ prog_name = devm_kstrdup(tas_priv->dev, "Speaker Program Id", GFP_KERNEL); - if (!prog_name) { - ret = -ENOMEM; - goto out; - } + if (!prog_name) + return -ENOMEM; + dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -1143,10 +1136,9 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) conf_name = devm_kstrdup(tas_priv->dev, "Speaker Config Id", GFP_KERNEL); - if (!conf_name) { - ret = -ENOMEM; - goto out; - } + if (!conf_name) + return -ENOMEM; + dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; @@ -1156,10 +1148,9 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) active_dev_num = devm_kstrdup(tas_priv->dev, "Activate Tasdevice Num", GFP_KERNEL); - if (!active_dev_num) { - ret = -ENOMEM; - goto out; - } + if (!active_dev_num) + return -ENOMEM; + dsp_ctrls[mix_index].name = active_dev_num; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_active_num; @@ -1168,21 +1159,17 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) mix_index++; chip_id = devm_kstrdup(tas_priv->dev, "Tasdevice Chip Id", GFP_KERNEL); - if (!chip_id) { - ret = -ENOMEM; - goto out; - } + if (!chip_id) + return -ENOMEM; + dsp_ctrls[mix_index].name = chip_id; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_chip_id; dsp_ctrls[mix_index].get = tasdevice_get_chip_id; mix_index++; - ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, + return snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); - -out: - return ret; } static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) @@ -1469,7 +1456,6 @@ static int tasdevice_hw_params(struct snd_pcm_substream *substream, unsigned int slot_width; unsigned int fsrate; int bclk_rate; - int rc = 0; fsrate = params_rate(params); switch (fsrate) { @@ -1479,8 +1465,7 @@ static int tasdevice_hw_params(struct snd_pcm_substream *substream, default: dev_err(tas_priv->dev, "%s: incorrect sample rate = %u\n", __func__, fsrate); - rc = -EINVAL; - goto out; + return -EINVAL; } slot_width = params_width(params); @@ -1493,20 +1478,17 @@ static int tasdevice_hw_params(struct snd_pcm_substream *substream, default: dev_err(tas_priv->dev, "%s: incorrect slot width = %u\n", __func__, slot_width); - rc = -EINVAL; - goto out; + return -EINVAL; } bclk_rate = snd_soc_params_to_bclk(params); if (bclk_rate < 0) { dev_err(tas_priv->dev, "%s: incorrect bclk rate = %d\n", __func__, bclk_rate); - rc = bclk_rate; - goto out; + return bclk_rate; } -out: - return rc; + return 0; } static int tasdevice_set_dai_sysclk(struct snd_soc_dai *codec_dai, -- 2.33.0

3 2

[PATCH v4] ASoC: Intel: avs: da7219: Remove suspend_pre() and resume_post()
by Marek Maslanka 02 Dec '24

02 Dec '24

The presence of a plugged jack is not detected after resuming the device if the jack was plugged before the device was suspended. This problem is caused by calling the sound/soc/codecs/da7219-aad.c:da7219_aad_jack_det() function on resume, which forces the jack insertion state to be unplugged. Signed-off-by: Marek Maslanka <mmaslanka(a)google.com> --- Changes in v4: - Remove redundant spaces from the commit message - Link to v3: https://lore.kernel.org/all/20241128155926.1774625-1-mmaslanka@google.com/ Changes in v3: - Fix line breaks in commit message - Link to v2: https://lore.kernel.org/all/20241128151239.1666582-1-mmaslanka@google.com/ Changes in v2: - Rephrase commit title and message as suggested in v1 - Link to v1: https://lore.kernel.org/all/20241128122732.1205732-1-mmaslanka@google.com/ --- --- sound/soc/intel/avs/boards/da7219.c | 17 ----------------- 1 file changed, 17 deletions(-) diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index 80c0a1a956542..daf53ca490a14 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -211,21 +211,6 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in return 0; } -static int avs_card_suspend_pre(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - - return snd_soc_component_set_jack(codec_dai->component, NULL, NULL); -} - -static int avs_card_resume_post(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); - - return snd_soc_component_set_jack(codec_dai->component, jack, NULL); -} - static int avs_da7219_probe(struct platform_device *pdev) { struct snd_soc_dai_link *dai_link; @@ -257,8 +242,6 @@ static int avs_da7219_probe(struct platform_device *pdev) card->name = "avs_da7219"; card->dev = dev; card->owner = THIS_MODULE; - card->suspend_pre = avs_card_suspend_pre; - card->resume_post = avs_card_resume_post; card->dai_link = dai_link; card->num_links = 1; card->controls = card_controls; -- 2.47.0.338.g60cca15819-goog

3 2

[PATCH v3] ASoC: Intel: avs: da7219: Remove suspend_pre() and resume_post()
by Marek Maslanka 02 Dec '24

02 Dec '24

The presence of a plugged jack is not detected after resuming the device if the jack was plugged before the device was suspended. This problem is caused by calling the sound/soc/codecs/da7219-aad.c:da7219_aad_jack_det() function on resume, which forces the jack insertion state to be unplugged. Signed-off-by: Marek Maslanka <mmaslanka(a)google.com> --- Changes in v3: - Fix line breaks in commit message - Link to v2: https://lore.kernel.org/all/20241128151239.1666582-1-mmaslanka@google.com/ Changes in v2: - Rephrase commit title and message as suggested in v1 - Link to v1: https://lore.kernel.org/all/20241128122732.1205732-1-mmaslanka@google.com/ --- --- sound/soc/intel/avs/boards/da7219.c | 17 ----------------- 1 file changed, 17 deletions(-) diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index 80c0a1a956542..daf53ca490a14 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -211,21 +211,6 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in return 0; } -static int avs_card_suspend_pre(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - - return snd_soc_component_set_jack(codec_dai->component, NULL, NULL); -} - -static int avs_card_resume_post(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); - - return snd_soc_component_set_jack(codec_dai->component, jack, NULL); -} - static int avs_da7219_probe(struct platform_device *pdev) { struct snd_soc_dai_link *dai_link; @@ -257,8 +242,6 @@ static int avs_da7219_probe(struct platform_device *pdev) card->name = "avs_da7219"; card->dev = dev; card->owner = THIS_MODULE; - card->suspend_pre = avs_card_suspend_pre; - card->resume_post = avs_card_resume_post; card->dai_link = dai_link; card->num_links = 1; card->controls = card_controls; -- 2.47.0.338.g60cca15819-goog

2 1

[PATCH] ASoC: Intel: avs: Remove component->set_jack() on suspend/resume
by Marek Maslanka 02 Dec '24

02 Dec '24

Removed calls to component->set_jack() during suspend/resume to fix an issue where an inserted jack is not detected after resuming from suspend if the jack was inserted before suspend. This problem is caused by calls to the sound/soc/codecs/da7219-aad.c:da7219_aad_jack_det() function during resume, which forces the jack insertion state to be unplugged. Signed-off-by: Marek Maslanka <mmaslanka(a)google.com> --- sound/soc/intel/avs/boards/da7219.c | 17 ----------------- 1 file changed, 17 deletions(-) diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index 80c0a1a956542..daf53ca490a14 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -211,21 +211,6 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in return 0; } -static int avs_card_suspend_pre(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - - return snd_soc_component_set_jack(codec_dai->component, NULL, NULL); -} - -static int avs_card_resume_post(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); - - return snd_soc_component_set_jack(codec_dai->component, jack, NULL); -} - static int avs_da7219_probe(struct platform_device *pdev) { struct snd_soc_dai_link *dai_link; @@ -257,8 +242,6 @@ static int avs_da7219_probe(struct platform_device *pdev) card->name = "avs_da7219"; card->dev = dev; card->owner = THIS_MODULE; - card->suspend_pre = avs_card_suspend_pre; - card->resume_post = avs_card_resume_post; card->dai_link = dai_link; card->num_links = 1; card->controls = card_controls; -- 2.47.0.338.g60cca15819-goog

3 4

[PATCH AUTOSEL 5.10 4/6] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate()
by Sasha Levin 02 Dec '24

02 Dec '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 63c1c87993e0e5bb11bced3d8224446a2bc62338 ] This patch checks if div is less than or equal to zero (div <= 0). If div is zero or negative, the function returns -EINVAL, ensuring the division operation (*prate / div) is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Link: https://patch.msgid.link/20241106014654.206860-1-luoyifan@cmss.chinamobile.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3aa1cf2624020..3a7f0102b4c5c 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -380,8 +380,8 @@ static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, int div; div = stm32_sai_get_clk_div(sai, *prate, rate); - if (div < 0) - return div; + if (div <= 0) + return -EINVAL; mclk->freq = *prate / div; -- 2.43.0

2 1

[PATCH] ALSA: hda/realtek: Enable mute and micmute LED on HP ProBook 430 G8
by Nazar Bilinskyi 01 Dec '24

01 Dec '24

HP ProBook 430 G8 has a mute and micmute LEDs that can be made to work using quirk ALC236_FIXUP_HP_GPIO_LED. Enable already existing quirk. Signed-off-by: Nazar Bilinskyi <nbilinskyi(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2bf5c512e..877c5d20a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10340,6 +10340,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87b7, "HP Laptop 14-fq0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87d3, "HP Laptop 15-gw0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), + SND_PCI_QUIRK(0x103c, 0x87df, "HP ProBook 430 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f1, "HP ProBook 630 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), -- 2.47.1

1 0

Linux Mint Cinnamon 22 Soundblaster AE-9 no sound
by GitHub issues - opened 30 Nov '24

30 Nov '24

alsa-project/alsa-lib issue #424 was opened from Starfoxfs: Hi, i have fresh installed Linux Mint Cinnamon 22 Kernel 6.8.0.49-generic I can´t get the Soundblaster AE-9 to work. Creative Labs has no Drivers for Linux and points to the alsa community. Is there any change to get support for the Soundblaster AE-9 in the future ? I think the big Problem is the external DAC for Headphones from the Soundblaster AE-9. In Windows this "clicks" on startup and activates the DAC, but in Linux this is all dead. Issue URL : https://github.com/alsa-project/alsa-lib/issues/424 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

arecordmidi.1: Some remarks and editorial changes for this man page
by Bjarni Ingi Gislason 30 Nov '24

30 Nov '24

Package: alsa-utils Version: 1.2.12-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -ww -b -z < "man page" [Use "groff -e ' $' <file>" to find trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? troff: backtrace: file '<stdin>':69 troff:<stdin>:69: warning: trailing space in the line troff: backtrace: file '<stdin>':77 troff:<stdin>:77: warning: trailing space in the line Output from "test-nroff -mandoc -t -K utf8 -rF0 -rHY=0 -ww -b -z ": troff: backtrace: file '<stdin>':69 troff:<stdin>:69: warning: trailing space in the line troff: backtrace: file '<stdin>':77 troff:<stdin>:77: warning: trailing space in the line * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.11.10-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.12-1+b1 ii libatopology2t64 1.2.12-1+b1 ii libc6 2.40-3 ii libfftw3-single3 3.3.10-1+b3 ii libncursesw6 6.5-2+b1 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5-2+b1 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

[tiwai-sound:for-next 6/10] sound/core/ump.c:1259:69: warning: '%s' directive output may be truncated writing up to 11 bytes into a region of size between 4 and 22
by kernel test robot 29 Nov '24

29 Nov '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 9b5f8ee43e48c25fbe1a10163ec04343d750acd0 commit: e29e504e7890b9ee438ca6370d0180d607c473f9 [6/10] ALSA: ump: Indicate the inactive group in legacy substream names config: alpha-allyesconfig (https://download.01.org/0day-ci/archive/20241130/202411300103.FrGuTAYp-lkp@…) compiler: alpha-linux-gcc (GCC) 14.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20241130/202411300103.FrGuTAYp-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202411300103.FrGuTAYp-lkp@intel.com/ All warnings (new ones prefixed by >>): sound/core/ump.c: In function 'fill_substream_names': >> sound/core/ump.c:1259:69: warning: '%s' directive output may be truncated writing up to 11 bytes into a region of size between 4 and 22 [-Wformat-truncation=] 1259 | snprintf(s->name, sizeof(s->name), "Group %d (%.16s)%s", | ^~ sound/core/ump.c:1259:17: note: 'snprintf' output between 11 and 40 bytes into a destination of size 32 1259 | snprintf(s->name, sizeof(s->name), "Group %d (%.16s)%s", | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 1260 | idx + 1, name, | ~~~~~~~~~~~~~~ 1261 | ump->groups[idx].active ? "" : " [Inactive]"); | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vim +1259 sound/core/ump.c 1246 1247 static void fill_substream_names(struct snd_ump_endpoint *ump, 1248 struct snd_rawmidi *rmidi, int dir) 1249 { 1250 struct snd_rawmidi_substream *s; 1251 const char *name; 1252 int idx; 1253 1254 list_for_each_entry(s, &rmidi->streams[dir].substreams, list) { 1255 idx = ump->legacy_mapping[s->number]; 1256 name = ump->groups[idx].name; 1257 if (!*name) 1258 name = ump->info.name; > 1259 snprintf(s->name, sizeof(s->name), "Group %d (%.16s)%s", 1260 idx + 1, name, 1261 ump->groups[idx].active ? "" : " [Inactive]"); 1262 } 1263 } 1264 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0