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[GIT PULL] ASoC fixes for v6.13-rc6
by Mark Brown 10 Jan '25

10 Jan '25

The following changes since commit 32c9c06adb5b157ef259233775a063a43746d699: ASoC: mediatek: disable buffer pre-allocation (2024-12-19 11:15:09 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-rc6 for you to fetch changes up to 139fa599cea0fd9d38e00246ea9f79af6c59acbd: ASoC: rsnd: check rsnd_adg_clk_enable() return value (2025-01-09 13:17:02 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A collection of device specific fixes that came in over the holidays, plus a MAINTAINERS update and some documentation to help users debug problems with some of the Cirrus CODECs found in modern laptops. ---------------------------------------------------------------- Charles Keepax (3): ASoC: wm8994: Add depends on MFD core ASoC: samsung: Add missing selects for MFD_WM8994 ASoC: samsung: Add missing depends on I2C Kuninori Morimoto (2): MAINTAINERS: add missing maintainers for Simple Audio Card ASoC: rsnd: check rsnd_adg_clk_enable() return value Maciej Strozek (1): ASoC: cs42l43: Add codec force suspend/resume ops Marian Postevca (1): ASoC: codecs: es8316: Fix HW rate calculation for 48Mhz MCLK Richard Fitzgerald (2): ALSA: doc: cs35l56: Add information about Cirrus Logic CS35L54/56/57 ALSA: doc: Add codecs/index.rst to top-level index Shenghao Ding (1): ASoC: tas2781: Fix occasional calibration failture Zhang Yi (1): ASoC: codecs: ES8326: Adjust ANA_MICBIAS to reduce pop noise Documentation/sound/codecs/cs35l56.rst | 292 +++++++++++++++++++++++++++++++++ Documentation/sound/codecs/index.rst | 9 + Documentation/sound/index.rst | 1 + MAINTAINERS | 2 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/cs42l43.c | 1 + sound/soc/codecs/es8316.c | 10 +- sound/soc/codecs/es8326.c | 23 ++- sound/soc/codecs/tas2781-i2c.c | 2 +- sound/soc/renesas/rcar/adg.c | 28 +++- sound/soc/renesas/rcar/core.c | 4 +- sound/soc/renesas/rcar/rsnd.h | 2 +- sound/soc/samsung/Kconfig | 6 +- 13 files changed, 364 insertions(+), 17 deletions(-) create mode 100644 Documentation/sound/codecs/cs35l56.rst create mode 100644 Documentation/sound/codecs/index.rst

2 1

alsactl info fails with exit code 25
by GitHub issues - opened 09 Jan '25

09 Jan '25

alsa-project/alsa-utils issue #287 was opened from z-s-e: Here is the output of running alsactl info on my machine: ``` $ alsactl info 0 # # Sound card # - card: 0 id: Generic name: HD-Audio Generic longname: HD-Audio Generic at 0xb88c8000 irq 87 driver_name: HDA-Intel mixer_name: ATI R6xx HDMI components: HDA:1002aa01,00aa0100,00100800 controls_count: 15 pcm: - stream: PLAYBACK devices: - device: 3 id: HDMI 0 name: G246HYL subdevices: - subdevice: 0 name: subdevice #0 - device: 7 id: HDMI 1 name: HDMI 1 subdevices: - subdevice: 0 name: subdevice #0 alsactl: rawmidi_device_list:105: snd_ctl_rawmidi_next_device $ echo $? 25 ``` Apparently it fails when trying to enumerate raw midi devices. I don't think this is supposed to happen, right? alsa-info for reference: [alsa-info.txt](https://github.com/user-attachments/files/18362369/alsa-info… Issue URL : https://github.com/alsa-project/alsa-utils/issues/287 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v1] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 09 Jan '25

09 Jan '25

Due to internal re-org, Kevin has no longer maintained audio driver. With the job responsibility change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 30cbc3d44cd5..4c806ad2fb16 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23172,25 +23172,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: linux-sound(a)vger.kernel.org -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: linux-sound(a)vger.kernel.org +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -23200,9 +23217,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

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Topology: NHLT: Intel: Fix mono DMIC configure for MTL platform
by GitHub pull_request - opened 08 Jan '25

08 Jan '25

alsa-project/alsa-utils pull request #286 was opened from singalsu: This change fixes the blob generator for mono microphone configuration. As difference to previous platforms the FIFO packer mono/stereo mode set up in OUTCONTROLx IPM_SOURCE_MODE bit-field. Request URL : https://github.com/alsa-project/alsa-utils/pull/286 Patch URL : https://github.com/alsa-project/alsa-utils/pull/286.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aseqdump version 1.2.11, There are small bugs
by GitHub issues - opened 07 Jan '25

07 Jan '25

alsa-project/alsa-utils issue #285 was opened from masahirokakishita: Hello, I found small bugs in aseqdump.c static void dump_ump_midi2_event(const unsigned int *ump) ... case SND_UMP_MSG_PITCHBEND: printf("Channel pressure %2d, value %s", channel, midi2_pitchbend(m->channel_pressure.data)); break; It is not "Channel pressure", but "Pitch Bend". case SND_UMP_MSG_NOTE_ON: printf("Note on %2d, note %d, velocity %s, attr type = %d, data = 0x%x", channel, m->note_off.note, midi2_velocity(m->note_off.velocity), m->note_off.attr_type, m->note_off.attr_data); break; "note->on" is correct. This bug does not affect the display. Issue URL : https://github.com/alsa-project/alsa-utils/issues/285 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: rt5682: Support the ALC5682I-VE codec
by derek.fang＠realtek.com 07 Jan '25

07 Jan '25

From: Derek Fang <derek.fang(a)realtek.com> The ALC5682I-VD and ALC5682I-VE use the same I2C codec driver with different calibration settings. This patch aims to handle their differences. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index ff9e14fad0cd..a8820435d1e0 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -186,6 +186,12 @@ static int rt5682_i2c_probe(struct i2c_client *i2c) return -ENODEV; } + regmap_read(rt5682->regmap, RT5682_INT_DEVICE_ID, &val); + if (val == 0x6956) { + dev_dbg(&i2c->dev, "ALC5682I-VE device\n"); + rt5682->ve_ic = true; + } + mutex_init(&rt5682->calibrate_mutex); rt5682_calibrate(rt5682); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index aa163ec40862..b4d72fc4a44d 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -395,6 +395,7 @@ bool rt5682_volatile_register(struct device *dev, unsigned int reg) case RT5682_4BTN_IL_CMD_1: case RT5682_AJD1_CTRL: case RT5682_HP_CALIB_CTRL_1: + case RT5682_INT_DEVICE_ID: case RT5682_DEVICE_ID: case RT5682_I2C_MODE: case RT5682_HP_CALIB_CTRL_10: @@ -419,6 +420,7 @@ bool rt5682_readable_register(struct device *dev, unsigned int reg) { switch (reg) { case RT5682_RESET: + case RT5682_INT_DEVICE_ID: case RT5682_VERSION_ID: case RT5682_VENDOR_ID: case RT5682_DEVICE_ID: @@ -3139,7 +3141,10 @@ void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100); regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); + if (rt5682->ve_ic) + regmap_write(rt5682->regmap, RT5682_CHOP_ADC, 0x7005); + else + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); @@ -3168,7 +3173,10 @@ void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); + if (rt5682->ve_ic) + regmap_write(rt5682->regmap, RT5682_CHOP_ADC, 0x2005); + else + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0c0c); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index b2d9e87af259..de43a5d99403 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -22,6 +22,7 @@ /* Info */ #define RT5682_RESET 0x0000 +#define RT5682_INT_DEVICE_ID 0x00f9 #define RT5682_VERSION_ID 0x00fd #define RT5682_VENDOR_ID 0x00fe #define RT5682_DEVICE_ID 0x00ff @@ -1446,6 +1447,7 @@ struct rt5682_priv { bool hw_init; bool first_hw_init; bool is_sdw; + bool ve_ic; #ifdef CONFIG_COMMON_CLK struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; -- 2.34.1

2 1

[PATCH v1] ASoC: tas2781: Fix occasional calibration failture
by Shenghao Ding 07 Jan '25

07 Jan '25

The root cause is that TAS2781 internal register is not unlocked before writing data into TAS2781_PRM_TEST_57_REG. Fixes: 49e2e353fb0d ("ASoC: tas2781: Add Calibration Kcontrols for Chromebook") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index fb8cd2284fe8..728bf78ae71f 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -78,7 +78,7 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { X2781_CL_STT_VAL(TAS2781_PRM_INT_MASK_REG, 0xfe, false), X2781_CL_STT_VAL(TAS2781_PRM_CLK_CFG_REG, 0xdd, false), X2781_CL_STT_VAL(TAS2781_PRM_RSVD_REG, 0x20, false), - X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, false), + X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, true), X2781_CL_STT_VAL(TAS2781_PRM_TEST_62_REG, 0x45, true), X2781_CL_STT_VAL(TAS2781_PRM_PVDD_UVLO_REG, 0x03, false), X2781_CL_STT_VAL(TAS2781_PRM_CHNL_0_REG, 0xa8, false), -- 2.34.1

2 1

[PATCH RESEND] ASoC: codecs: es8316: Fix HW rate calculation for 48Mhz MCLK
by Marian Postevca 07 Jan '25

07 Jan '25

For 48Mhz MCLK systems the calculation of the HW rate is broken, and will not produce even one sane rate. Since es83xx supports the option to halve MCLK, calculate also rates with MCLK/2. Signed-off-by: Marian Postevca <posteuca(a)mutex.one> --- sound/soc/codecs/es8316.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 61729e5b50a8..f508df01145b 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -39,7 +39,9 @@ struct es8316_priv { struct snd_soc_jack *jack; int irq; unsigned int sysclk; - unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios)]; + /* ES83xx supports halving the MCLK so it supports twice as many rates + */ + unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios) * 2]; struct snd_pcm_hw_constraint_list sysclk_constraints; bool jd_inverted; }; @@ -386,6 +388,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (freq % ratio == 0) es8316->allowed_rates[count++] = freq / ratio; + + /* We also check if the halved MCLK produces a valid rate + * since the codec supports halving the MCLK. + */ + if ((freq / ratio) % 2 == 0) + es8316->allowed_rates[count++] = freq / ratio / 2; } if (count) { -- 2.45.2

2 1

[PATCH] ASoC: codecs: ES8326: Adjust ANA_MICBIAS to reduce pop noise
by Zhang Yi 07 Jan '25

07 Jan '25

We modify the value of ES8326_ANA_MICBIAS to reduce the pop noise Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index a5603b617688..b06eead7e0f6 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -616,7 +616,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) 0x0F, 0x0F); if (es8326->version > ES8326_VERSION_B) { regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); } } } else { @@ -631,6 +631,8 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) regmap_write(es8326->regmap, ES8326_HPR_OFFSET_INI, offset_r); es8326->calibrated = true; } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0x00); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x00); if (direction == SNDRV_PCM_STREAM_PLAYBACK) { regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x01); usleep_range(1000, 5000); @@ -645,7 +647,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) } else { msleep(300); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x50); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x70); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00); } regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, @@ -676,6 +678,10 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x00); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x20); regmap_update_bits(es8326->regmap, ES8326_RESET, 0x02, 0x00); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); + } break; case SND_SOC_BIAS_PREPARE: break; @@ -683,6 +689,12 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x00); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0xc0); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x80); break; case SND_SOC_BIAS_OFF: clk_disable_unprepare(es8326->mclk); @@ -773,7 +785,10 @@ static void es8326_jack_button_handler(struct work_struct *work) case 0x6f: case 0x4b: /* button volume up */ - cur_button = SND_JACK_BTN_1; + if ((iface == 0x6f) && (es8326->version > ES8326_VERSION_B)) + cur_button = SND_JACK_BTN_0; + else + cur_button = SND_JACK_BTN_1; break; case 0x27: /* button volume down */ @@ -1082,7 +1097,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x10); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); } -- 2.17.1

2 1

headers: make more headers self-contained
by GitHub pull_request - opened 06 Jan '25

06 Jan '25

alsa-project/alsa-lib pull request #435 was opened from timblechmann: Some IDEs parse headers in isolation to populate their code model. This exposes some missing headers, mainly for alsa internal struct definitions, ssize_t (requiring sys/types.h), size_t (requiring string.h), va_args (requiring stdargs.h) or FILE (requiring stdio.h) This patch makes the main headers parsable by qtcreator Request URL : https://github.com/alsa-project/alsa-lib/pull/435 Patch URL : https://github.com/alsa-project/alsa-lib/pull/435.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

bug in the asla driver asla-sink-viritio-snd
by GitHub issues - opened 05 Jan '25

05 Jan '25

alsa-project/alsa-lib issue #434 was opened from LoreFiore: <img width="1440" alt="Screenshot 2025-01-05 alle 23 31 08" src="https://github.com/user-attachments/assets/34b279a0-b256-4e7b-82ec-a9df46f9…" /> Issue URL : https://github.com/alsa-project/alsa-lib/issues/434 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ld.lld default version symbol snd_dlopen@@ALSA_1.1.6 must be defined
by GitHub issues - edited 05 Jan '25

05 Jan '25

alsa-project/alsa-lib issue #433 was edited from trcrsired: ``` clang: warning: argument unused during compilation: '--unwindlib=libunwind' [-Wunused-command-line-argument] CCLD libasound.la ld.lld: error: seq/.libs/libseq.a(seq.o at 50494) <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:1:1: default version symbol snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_format_first, snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:2:1: default version symbol snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_channels_near, snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:3:1: default version symbol snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_rate_near, snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:4:1: default version symbol snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_time_near, snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:5:1: default version symbol snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_size_near, snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:6:1: default version symbol snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_periods_near, snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:7:1: default version symbol snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_time_near, snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:8:1: default version symbol snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_size_near, snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:9:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: .libs/conf.o <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_hw.o at 720480) <inline asm>:1:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:1:1: default version symbol snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_access, snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:2:1: default version symbol snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_format, snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:3:1: default version symbol snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_subformat, snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:4:1: default version symbol snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_channels, snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:5:1: default version symbol snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_rate, snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:6:1: default version symbol snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_time, snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:7:1: default version symbol snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_size, snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:8:1: default version symbol snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_buffer_size, snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 ^ ld.lld: error: too many errors emitted, stopping now (use --error-limit=0 to see all errors) clang: error: linker command failed with exit code 1 (use -v to see invocation) make[3]: *** [Makefile:506: libasound.la] Error 1 make[3]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[2]: *** [Makefile:567: all-recursive] Error 1 make[2]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[1]: *** [Makefile:436: all] Error 2 make[1]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make: *** [Makefile:414: all-recursive] Error 1 alsa-lib build failure ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/433 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

default version symbol snd_dlopen@@ALSA_1.1.6 must be defined
by GitHub issues - opened 05 Jan '25

05 Jan '25

alsa-project/alsa-lib issue #433 was opened from trcrsired: ``` clang: warning: argument unused during compilation: '--unwindlib=libunwind' [-Wunused-command-line-argument] CCLD libasound.la ld.lld: error: seq/.libs/libseq.a(seq.o at 50494) <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:1:1: default version symbol snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_format_first, snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:2:1: default version symbol snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_channels_near, snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:3:1: default version symbol snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_rate_near, snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:4:1: default version symbol snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_time_near, snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:5:1: default version symbol snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_size_near, snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:6:1: default version symbol snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_periods_near, snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:7:1: default version symbol snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_time_near, snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:8:1: default version symbol snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_size_near, snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:9:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: .libs/conf.o <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_hw.o at 720480) <inline asm>:1:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:1:1: default version symbol snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_access, snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:2:1: default version symbol snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_format, snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:3:1: default version symbol snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_subformat, snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:4:1: default version symbol snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_channels, snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:5:1: default version symbol snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_rate, snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:6:1: default version symbol snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_time, snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:7:1: default version symbol snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_size, snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:8:1: default version symbol snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_buffer_size, snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 ^ ld.lld: error: too many errors emitted, stopping now (use --error-limit=0 to see all errors) clang: error: linker command failed with exit code 1 (use -v to see invocation) make[3]: *** [Makefile:506: libasound.la] Error 1 make[3]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[2]: *** [Makefile:567: all-recursive] Error 1 make[2]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[1]: *** [Makefile:436: all] Error 2 make[1]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make: *** [Makefile:414: all-recursive] Error 1 alsa-lib build failure ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/433 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ucm2: USB-Audio: add Presonus Revelator IO 44 (USB194f:0424)
by GitHub pull_request - edited 04 Jan '25

04 Jan '25

alsa-project/alsa-ucm-conf pull request #495 was edited from markulovi: [alsa-info.txt](https://github.com/user-attachments/files/18306150/alsa-info… This adds support for [Presonus Revelator io44](https://eu.presonus.com/products/revelator-io44). This device has 3 virtual stereo channels out and 4 virtual stereo channels in as follows: ![out](https://github.com/user-attachments/assets/f4cacf14-e75e-4d33-ad64-cca4ca7ee18e) ![in](https://github.com/user-attachments/assets/60515091-51a5-4299-97c7-a12d3c091b8a) Microphone inputs have been separated into two mono channels for convenience. As far as I can tell, the [Universal Control](https://www.presonus.com/pages/support-documents-downloads?product… software is required to configure levels and effects of these channels, but the settings can be saved to the device and operation without the software is possible outside Windows. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: USB-Audio: add Presonus Revelator IO 44 (USB194f:0424)
by GitHub pull_request - opened 04 Jan '25

04 Jan '25

alsa-project/alsa-ucm-conf pull request #495 was opened from markulovi: [alsa-info.txt](https://github.com/user-attachments/files/18306150/alsa-info… This adds support for [Presonus Revelator io44](https://eu.presonus.com/products/revelator-io44). This device has 3 virtual stereo channels out and 4 virtual stereo channels in as follows: ![out](https://github.com/user-attachments/assets/f4cacf14-e75e-4d33-ad64-cca4ca7ee18e) ![in](https://github.com/user-attachments/assets/60515091-51a5-4299-97c7-a12d3c091b8a) Microphone inputs have been separated to two mono channels for convenience. As far as I can tell, the [Universal Control](https://www.presonus.com/pages/support-documents-downloads?product… software is required to configure levels and effects of these channels, but the settings can be saved to the device and operation without the software is possible outside Windows. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/5] soundwire: amd: code improvements and new platform
by Vijendar Mukunda 02 Jan '25

02 Jan '25

This patch series consists of code improvements and new platform support for ACP7.0 & ACP7.1 platforms. Vijendar Mukunda (5): soundwire: amd: change the soundwire wake enable/disable sequence soundwire: amd: add debug log for soundwire wake event soundwire: amd: add support for ACP7.0 & ACP7.1 platforms soundwire: amd: add soundwire host wake interrupt enable/disable sequence soundwire: amd: clear soundwire pme wake status drivers/soundwire/amd_manager.c | 102 ++++++++++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 27 ++++++++ include/linux/soundwire/sdw_amd.h | 2 + 3 files changed, 131 insertions(+) -- 2.34.1

2 7

Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - edited 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was edited from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when (at most week after Baldur's Gate 3 release ;). What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. After connecting external monitor alsacap reports max 48000 for both cards. FFmpeg reports sampling rate problem with both audio cards: ``` [alsa @ 0x586316542b00] sample rate 96000 not available, nearest is 48000 ``` I had this monitor connected before and alsacap reported "flipped" results - not like now that both cards are limited to 48000. What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - edited 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was edited from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when (at most week after Baldur's Gate 3 release ;). What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. After connecting external monitor alsacap reports max 48000 for both cards. FFmpeg reports sampling rate problem with both audio cards: ``` [alsa @ 0x586316542b00] sample rate 96000 not available, nearest is 48000 ``` What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - edited 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was edited from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when. What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. After connecting external monitor alsacap reports max 48000 for both cards. FFmpeg reports sampling rate problem with both audio cards: ``` [alsa @ 0x586316542b00] sample rate 96000 not available, nearest is 48000 ``` What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - opened 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was opened from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when. What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Add Roland QUAD-CAPTURE USB 2.0 support
by GitHub pull_request - opened 31 Dec '24

31 Dec '24

alsa-project/alsa-ucm-conf pull request #494 was opened from bjornmp: Add Roland QUAD-CAPTURE USB 2.0 support Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/494 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/494.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

alsa-lib-1.2.13 is breaking dssi-1.1.1 because of missing header
by GitHub issues - opened 30 Dec '24

30 Dec '24

alsa-project/alsa-lib issue #431 was opened from kobboi: What was working with 1.2.12 no longer isn't: ``` /usr/include/alsa/ump_msg.h:1119:65: error: unknown type name ‘size_t’ 1119 | int snd_ump_msg_sysex_expand(const uint32_t *ump, uint8_t *buf, size_t maxlen, | ^~~~~~ /usr/include/alsa/ump_msg.h:13:1: note: ‘size_t’ is defined in header ‘<stddef.h>’; did you forget to ‘#include <stddef.h>’? 12 | #include <stdint.h> +++ |+#include <stddef.h> 13 | /usr/include/alsa/ump_msg.h:1120:30: error: unknown type name ‘size_t’ 1120 | size_t *filled); | ^~~~~~ /usr/include/alsa/ump_msg.h:1120:30: note: ‘size_t’ is defined in header ‘<stddef.h>’; did you forget to ‘#include <stddef.h>’? ``` Can this be addressed? Happy holidays! Issue URL : https://github.com/alsa-project/alsa-lib/issues/431 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Ask snd_seq_set_client_midi_version
by Correo Alternativo 30 Dec '24

30 Dec '24

Hi, I'm trying out MIDI 2.0 and found a problem, when using snd_seq_set_client_midi_version With SND_SEQ_CLIENT_LEGACY_MIDI I get the following sequence of events when connecting a midi device (using the method snd_seq_event_input or snd_seq_ump_event_input) (timidity command:timidity --interface=A --buffer-fragments=2,8 --output-mode=s1l --sampling-freq=44100): SND_SEQ_EVENT_CLIENT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START But with SND_SEQ_CLIENT_UMP_MIDI_1_0 and SND_SEQ_CLIENT_UMP_MIDI_2_0 I don't get the SND_SEQ_EVENT_PORT_START events, I only get the following: SND_SEQ_EVENT_CLIENT_START SND_SEQ_EVENT_SYSTEM SND_SEQ_EVENT_SYSTEM SND_SEQ_EVENT_SYSTEM I don't know if I'm asking this in the right place, what mistake am I making? Just by modifying that it stops working.

2 1

[PATCH v1] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 30 Dec '24

30 Dec '24

Due to internal re-org, Kevin has no longer maintained audio driver. With the job responsibility change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 910305c11e8a..db50e677c9ca 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23170,25 +23170,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: linux-sound(a)vger.kernel.org -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: linux-sound(a)vger.kernel.org +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -23198,9 +23215,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

UCM2 macro breaks on updates for Motu 828
by GitHub issues - opened 30 Dec '24

30 Dec '24

alsa-project/alsa-ucm-conf issue #493 was opened from beatboxa: It appears that every time there is an update, my UCM profile breaks. So that when I boot my machine, I have no sound (unless I manually map things in a pro audio application). This appears to be due to an issue in the SplitPCM macro--specifically, some parsing. Here is the thread describing the creation of the UCM profile I am using, including a patch that I had to apply in order to get things working: #416 And specifically, here is the patch I apply in order to fix it: https://github.com/alsa-project/alsa-ucm-conf/commit/b68aa52acdd2763fedad5e… It involves adding a single hyphen character to the file: ucm2/common/pcm/split.conf Because it is a single character, I have been manually changing the file. It's hard for me to follow that thread--but is this fixed and I am on older versions, or is this still an open item? Or is there a way to automate this so that my audio doesn't break on every update? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/493 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RESEND PATCH v2] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 30 Dec '24

30 Dec '24

Hi Mark, resending this as it failed to apply in my last submission. Added Neil Armstrong to Cc: as hopefully he will be able to better review this. Thanks, Geraldo Nascimento --- Starting with version 2.10a of Synopsys DesignWare HDMI controller the insert_pcuv bit was introduced. On RK3399pro SoM (Radxa Rock Pi N10), for example, if we neglect to set this bit and proceed to enable hdmi_sound and i2s2 on the device tree there will be extreme clipping of sound output, to the point that music sounds like white noise. Problem could also manifest as just mild cracking depending of HDMI audio implementation of sink. Setting insert_pcuv bit (bit 2 of aud_conf2 Audio Sample register) fixes this. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- v1->v2: SoC->SoM on description, better commenting, minor style changes, conditional application of fix for L-PCM only --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c @@ -42,6 +42,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, struct dw_hdmi *hdmi = audio->hdmi; u8 conf0 = 0; u8 conf1 = 0; + u8 conf2 = 0; u8 inputclkfs = 0; /* it cares I2S only */ @@ -101,6 +102,28 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, return -EINVAL; } + /* + * dw-hdmi introduced insert_pcuv bit in + * version 2.10a. + * + * This single bit (bit 2 of HDMI_AUD_CONF2) + * when set to 1 will enable the insertion of the PCUV + * (Parity, Channel Status, User bit and Validity) + * bits on the incoming audio stream. + * + * Support is limited to Linear PCM audio. If + * neglected, the lack of valid PCUV bits + * on L-PCM streams will cause anything from + * mild cracking to full blown extreme + * clipping depending on the HDMI audio + * implementation of the sink. + * + */ + + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21 && + !(hparms->iec.status[0] & IEC958_AES0_NONAUDIO)) + conf2 = HDMI_AUD_CONF2_INSERT_PCUV; + dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); @@ -109,6 +120,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, inputclkfs, HDMI_AUD_INPUTCLKFS); hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + hdmi_write(audio, conf2, HDMI_AUD_CONF2); return 0; } --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h @@ -931,6 +931,11 @@ enum { HDMI_AUD_CONF1_WIDTH_16 = 0x10, HDMI_AUD_CONF1_WIDTH_24 = 0x18, +/* AUD_CONF2 field values */ + HDMI_AUD_CONF2_HBR = 0x01, + HDMI_AUD_CONF2_NLPCM = 0x02, + HDMI_AUD_CONF2_INSERT_PCUV = 0x04, + /* AUD_CTS3 field values */ HDMI_AUD_CTS3_N_SHIFT_OFFSET = 5, HDMI_AUD_CTS3_N_SHIFT_MASK = 0xe0,

3 6

[PATCH v4 0/9] ASoC: samsung: midas-audio: Add GPIO-based headset jack detection
by Artur Weber 29 Dec '24

29 Dec '24

Many of Samsung's Exynos 4 devices share the same midas-audio driver to handle the codec setup. While most of these devices, including the Midas itself, use the jack detection provided by the WM8994 driver, other devices such as the Samsung Galaxy Tab 3 8.0 (lt01) use two GPIOs and an ADC channel to determine jack insertion, the jack's type, and button presses (for headsets with volume up/down/play buttons). In the downstream kernel, this behavior is implemented in the sec-jack driver[1], and the per-device settings are configured in *-jack.c files in the mach folder (see e.g. the Tab 3's implementation[2]). This patchset implements this mechanism in the midas_wm1811.c driver, and adds new DTS options to allow for its configuration. It also enables jack detection for the Samsung Galaxy Tab 3 8.0. A very similar mechanism was already present in the aries_wm8994.c driver[3]; this implementation heavily borrows from it, though there are a few extra cleanups as well. Signed-off-by: Artur Weber <aweber.kernel(a)gmail.com> [1] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [2] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [3] https://github.com/torvalds/linux/blob/master/sound/soc/samsung/aries_wm899… --- Changes in v4: - Replaced manual regulator toggles with SND_SOC_DAPM_REGULATOR_SUPPLY, both for existing mic bias regulators and for the new headset mic bias regulator - Link to v3: https://lore.kernel.org/r/20240519-midas-wm1811-gpio-jack-v3-0-0c1736144c0e… Changes in v3: - Re-added pipe (|) character to samsung,headset-button-threshold-microvolt description to prevent it from being parsed as a mapping (fixes syntax error) - Split out "ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection" into two patches - Link to v2: https://lore.kernel.org/r/20240508-midas-wm1811-gpio-jack-v2-0-b4d36cd02c6e… Changes in v2: - Added vendor prefix to threshold properties - Added separate headset mic bias regulator - Changed some cases of dev_err + return with return dev_err_probe - Added an extra patch to replace some previous dev_err + return cases with dev_err_probe - Moved tab3 DTS wm1811 codec config changes to separate commit --- Artur Weber (9): ASoC: dt-bindings: samsung,midas-audio: Add headset mic bias supply ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Use SND_SOC_DAPM_REGULATOR_SUPPLY for bias regulators ASoC: samsung: midas_wm1811: Add headset mic bias supply support ASoC: samsung: midas_wm1811: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Use dev_err_probe where appropriate ARM: dts: samsung: exynos4212-tab3: Fix headset mic, add jack detection ARM: dts: samsung: exynos4212-tab3: Add MCLK2 clock to WM1811 codec config ARM: dts: samsung: exynos4212-tab3: Drop interrupt from WM1811 codec .../bindings/sound/samsung,midas-audio.yaml | 33 ++ arch/arm/boot/dts/samsung/exynos4212-tab3.dtsi | 31 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/midas_wm1811.c | 348 +++++++++++++++++---- 4 files changed, 343 insertions(+), 71 deletions(-) --- base-commit: a38297e3fb012ddfa7ce0321a7e5a8daeb1872b6 change-id: 20240502-midas-wm1811-gpio-jack-b10226b17ecc Best regards, -- Artur Weber <aweber.kernel(a)gmail.com>

3 12

USB-Audio: ALC4080 - add ASUS ROG Crosshair X870E Hero (USB 0b05:1b7c)
by GitHub pull_request - opened 26 Dec '24

26 Dec '24

alsa-project/alsa-ucm-conf pull request #492 was opened from jetm: https://rog.asus.com/motherboards/rog-crosshair/rog-crosshair-x870e-hero/ Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/492 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/492.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: Bug: slab-out-of-bounds in snd_seq_oss_synth_sysex
by Kun Hu 26 Dec '24

26 Dec '24

>> BUG: KASAN: slab-out-of-bounds in snd_seq_oss_synth_sysex+0x5d1/0x6c0 sound/core/seq/oss/seq_oss_synth.c:516 > > We further analyzed the issue at line 516 in ./sound/core/seq/oss/seq_oss_synth.c. > The slab-out-of-bounds crash occurs in line 509, when sysex->len = 128. Specifically, the write operation to dest[0] accesses memory beyond the bounds of sysex->buf (128 byte). > To resolve this issue, we suggest adding 6 lines of code to validate the legality of the address write to sysex->buf before entering the loop: > > if (sysex->len >= MAX_SYSEX_BUFLEN) { > sysex->len = 0; > sysex->skip = 1; > return -EINVAL; /* Exit early if sysex->len is out of bounds */ > } If you need any more information, please don't hesitate to let me know. ————— Best regards, Kun Hu > 2024年12月25日 13:37，Kun Hu <huk23(a)m.fudan.edu.cn> 写道： > > Hello, > >> BUG: KASAN: slab-out-of-bounds in snd_seq_oss_synth_sysex+0x5d1/0x6c0 sound/core/seq/oss/seq_oss_synth.c:516 > > We further analyzed the issue at line 516 in ./sound/core/seq/oss/seq_oss_synth.c. > The slab-out-of-bounds crash occurs in line 509, when sysex->len = 128. Specifically, the write operation to dest[0] accesses memory beyond the bounds of sysex->buf (128 byte). > To resolve this issue, we suggest adding 6 lines of code to validate the legality of the address write to sysex->buf before entering the loop: > > if (sysex->len >= MAX_SYSEX_BUFLEN) { > sysex->len = 0; > sysex->skip = 1; > return -EINVAL; /* Exit early if sysex->len is out of bounds */ > } > > If you fix this issue, please add the following tag to the commit: > Reported-by: Kun Hu <huk23(a)m.fudan.edu.cn> > > ————— > Thanks, > Kun Hu > >> 2024年12月24日 19:16，Kun Hu <huk23(a)m.fudan.edu.cn> 写道： >> >> Hello, >> >> When using fuzzer tool to fuzz the latest Linux kernel, the following crash >> was triggered. >> >> HEAD commit: 78d4f34e2115b517bcbfe7ec0d018bbbb6f9b0b8 >> git tree: upstream >> Console output:https://drive.google.com/file/d/17oCyKDW_kNhSW5Bbvm23vnpD1eo0MHFi/vi… >> Kernel config: https://drive.google.com/file/d/1RhT5dFTs6Vx1U71PbpenN7TPtnPoa3NI/view?usp=… >> C reproducer: https://drive.google.com/file/d/177HJht6a7-6F3YLudKb_d4kiPGd1VA_i/view?usp=… >> Syzlang reproducer: https://drive.google.com/file/d/1AuP5UGGc47rEXXPuvjmCKgJ3d0U1P84j/view?usp=… >> >> >> If you fix this issue, please add the following tag to the commit: >> Reported-by: Kun Hu <huk23(a)m.fudan.edu.cn> >> >> ================================================================== >> BUG: KASAN: slab-out-of-bounds in snd_seq_oss_synth_sysex+0x5d1/0x6c0 sound/core/seq/oss/seq_oss_synth.c:516 >> Write of size 1 at addr ff1100000588e288 by task syz-executor411/824 >> >> CPU: 2 UID: 0 PID: 824 Comm: syz-executor411 Not tainted 6.13.0-rc3 #5 >> Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.13.0-1ubuntu1.1 04/01/2014 >> Call Trace: >> <TASK> >> __dump_stack lib/dump_stack.c:94 [inline] >> dump_stack_lvl+0x116/0x1b0 lib/dump_stack.c:120 >> print_address_description mm/kasan/report.c:378 [inline] >> print_report+0xcf/0x5f0 mm/kasan/report.c:489 >> kasan_report+0x93/0xc0 mm/kasan/report.c:602 >> snd_seq_oss_synth_sysex+0x5d1/0x6c0 sound/core/seq/oss/seq_oss_synth.c:516 >> snd_seq_oss_process_event+0x46a/0x2620 sound/core/seq/oss/seq_oss_event.c:61 >> insert_queue sound/core/seq/oss/seq_oss_rw.c:167 [inline] >> snd_seq_oss_write+0x261/0x7f0 sound/core/seq/oss/seq_oss_rw.c:135 >> odev_write+0x53/0xa0 sound/core/seq/oss/seq_oss.c:168 >> vfs_write fs/read_write.c:677 [inline] >> vfs_write+0x2e3/0x10f0 fs/read_write.c:659 >> ksys_write+0x122/0x240 fs/read_write.c:731 >> do_syscall_x64 arch/x86/entry/common.c:52 [inline] >> do_syscall_64+0xc3/0x1d0 arch/x86/entry/common.c:83 >> entry_SYSCALL_64_after_hwframe+0x77/0x7f >> RIP: 0033:0x7f69b379994d >> Code: c3 e8 97 2a 00 00 0f 1f 80 00 00 00 00 f3 0f 1e fa 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 c7 c1 b8 ff ff ff f7 d8 64 89 01 48 >> RSP: 002b:00007f69b3718d78 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 >> RAX: ffffffffffffffda RBX: 00007f69b382d2d0 RCX: 00007f69b379994d >> RDX: 0000000000000008 RSI: 0000000020000140 RDI: 0000000000000003 >> RBP: 00007f69b382d2d8 R08: 0000000000000000 R09: 0000000000000000 >> R10: 0000000000000000 R11: 0000000000000246 R12: 00007f69b382d2dc >> R13: 0000000020000140 R14: 00007f69b37fa008 R15: 000000000000000d >> </TASK> >> >> Allocated by task 823: >> kasan_save_stack+0x24/0x50 mm/kasan/common.c:47 >> kasan_save_track+0x14/0x30 mm/kasan/common.c:68 >> poison_kmalloc_redzone mm/kasan/common.c:377 [inline] >> __kasan_kmalloc+0xaa/0xb0 mm/kasan/common.c:394 >> kmalloc_noprof include/linux/slab.h:901 [inline] >> kzalloc_noprof include/linux/slab.h:1037 [inline] >> snd_seq_oss_synth_sysex+0x1d9/0x6c0 sound/core/seq/oss/seq_oss_synth.c:502 >> snd_seq_oss_process_event+0x46a/0x2620 sound/core/seq/oss/seq_oss_event.c:61 >> insert_queue sound/core/seq/oss/seq_oss_rw.c:167 [inline] >> snd_seq_oss_write+0x261/0x7f0 sound/core/seq/oss/seq_oss_rw.c:135 >> odev_write+0x53/0xa0 sound/core/seq/oss/seq_oss.c:168 >> vfs_write fs/read_write.c:677 [inline] >> vfs_write+0x2e3/0x10f0 fs/read_write.c:659 >> ksys_write+0x122/0x240 fs/read_write.c:731 >> do_syscall_x64 arch/x86/entry/common.c:52 [inline] >> do_syscall_64+0xc3/0x1d0 arch/x86/entry/common.c:83 >> entry_SYSCALL_64_after_hwframe+0x77/0x7f >> >> The buggy address belongs to the object at ff1100000588e200 >> which belongs to the cache kmalloc-192 of size 192 >> The buggy address is located 0 bytes to the right of >> allocated 136-byte region [ff1100000588e200, ff1100000588e288) >> >> The buggy address belongs to the physical page: >> page: refcount:1 mapcount:0 mapping:0000000000000000 index:0x0 pfn:0x588e >> anon flags: 0x100000000000000(node=0|zone=1) >> page_type: f5(slab) >> raw: 0100000000000000 ff1100000103c3c0 ffd4000000162340 dead000000000003 >> raw: 0000000000000000 0000000080100010 00000001f5000000 0000000000000000 >> page dumped because: kasan: bad access detected >> >> Memory state around the buggy address: >> ff1100000588e180: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc >> ff1100000588e200: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 >>> ff1100000588e280: 00 fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc >> ^ >> ff1100000588e300: fa fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb >> ff1100000588e380: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc >> ================================================================== >> >> --------------- >> thanks, >> Kun Hu > >

1 0

[PATCH 00/21] ASoC: amd: acp70: add soundwire and acp pdm support
by Vijendar Mukunda 25 Dec '24

25 Dec '24

This patch series includes below changes - Add Audio IO support for ACP7.0 and ACP7.1 platforms for SoundWire IO and ACP PDM controller combination. - Add common driver modules for ACP7.0 and ACP7.1 platforms (acp pci driver, Soundwire dma driver, pdm platform driver). - Add SoundWire generic machine driver changes for legacy stack (No DSP enabled) for ACP7.0 & ACP7.1 platforms. - Add SoundWire machines for ACP7.0 & ACP7.1 platforms. Vijendar Mukunda (21): ASoC: amd: add register header file for ACP7.0 platform ASoC: amd: acp70: add acp pci driver for ACP7.0 and ACP7.1 platforms ASoC: amd: acp70: add acp init and de-init functions ASoC: amd: acp70: add logic for scanning acp child devices ASoC: amd: acp70: create platform devices for acp child nodes ASoC: amd: acp70: enable driver build for ACP7.0 platform ASoC: amd: acp70: add acp pdm platform driver ASoC: amd: acp70: add acp pdm driver dma ops and dai ops ASoC: amd: acp70: add acp soundwire dma driver ASoC: amd: update ACP7.0 KConfig option description ASoC: amd: acp70: add soundwire dma driver dma ops ASoC: amd: acp70: add acp ip interrupt handler ASoC: amd: acp70: add acp pdm driver pm ops ASoC: amd: acp70: add pm ops support for soundwire dma driver ASoC: amd: acp70: add acp driver pm ops support ASoC: amd: acp70: enable wake capability for acp pci driver ASoC: amd: acp70: add soundwire host wake interrupt handling ASoC: amd: acp70: create a device node for soundwire machine driver ASoC: amd: acp: add machine driver changes for ACP7.0 and ACP7.1 platforms ASoC: amd: acp: add RT711, RT714 & RT1316 support for ACP7.0 platform ASoC: amd: acp: amd-acp70-acpi-match: Add rt722 support include/sound/acp70_chip_offset_byte.h | 484 ++++++++++++++ sound/soc/amd/Kconfig | 12 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp/Makefile | 2 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 16 + sound/soc/amd/acp/acp-sdw-mach-common.c | 34 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 132 ++++ sound/soc/amd/acp/soc_amd_sdw_common.h | 11 + sound/soc/amd/acp70/Makefile | 9 + sound/soc/amd/acp70/acp70-pdm-dma.c | 463 +++++++++++++ sound/soc/amd/acp70/acp70-sdw-dma.c | 586 ++++++++++++++++ sound/soc/amd/acp70/acp70.h | 283 ++++++++ sound/soc/amd/acp70/pci-acp70.c | 807 +++++++++++++++++++++++ sound/soc/amd/mach-config.h | 1 + 14 files changed, 2840 insertions(+), 1 deletion(-) create mode 100644 include/sound/acp70_chip_offset_byte.h create mode 100644 sound/soc/amd/acp/amd-acp70-acpi-match.c create mode 100644 sound/soc/amd/acp70/Makefile create mode 100644 sound/soc/amd/acp70/acp70-pdm-dma.c create mode 100644 sound/soc/amd/acp70/acp70-sdw-dma.c create mode 100644 sound/soc/amd/acp70/acp70.h create mode 100644 sound/soc/amd/acp70/pci-acp70.c -- 2.34.1

3 23

Acer 315-24P Microphone indicator is not working
by GitHub issues - edited 23 Dec '24

23 Dec '24

alsa-project/alsa-utils issue #284 was edited from River000000: ![изображение](https://github.com/user-attachments/assets/dc8b0b9a-f877-4edc-a9eb-d1c4d1d2fcb7) ![изображение](https://github.com/user-attachments/assets/5f8f1c69-294b-4833-8bdf-9c2f8033c379) alsa-info: [alsa.txt](https://github.com/user-attachments/files/18227950/alsa.txt) Issue URL : https://github.com/alsa-project/alsa-utils/issues/284 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Acer 315-24P Microphone indicator is not working
by GitHub issues - edited 23 Dec '24

23 Dec '24

alsa-project/alsa-utils issue #284 was edited from River000000: ![изображение](https://github.com/user-attachments/assets/dc8b0b9a-f877-4edc-a9eb-d1c4d1d2fcb7) ![изображение](https://github.com/user-attachments/assets/5f8f1c69-294b-4833-8bdf-9c2f8033c379) [alsa.txt](https://github.com/user-attachments/files/18227950/alsa.txt) Issue URL : https://github.com/alsa-project/alsa-utils/issues/284 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Acer 315-24P Microphone indicator is not working
by GitHub issues - opened 23 Dec '24

23 Dec '24

alsa-project/alsa-utils issue #284 was opened from River000000: [alsa.txt](https://github.com/user-attachments/files/18227950/alsa.txt) ![изображение](https://github.com/user-attachments/assets/dc8b0b9a-f877-4edc-a9eb-d1c4d1d2fcb7) ![изображение](https://github.com/user-attachments/assets/5f8f1c69-294b-4833-8bdf-9c2f8033c379) Issue URL : https://github.com/alsa-project/alsa-utils/issues/284 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH RESEND] soundwire: amd: clear wake enable register for power off mode
by Vijendar Mukunda 23 Dec '24

23 Dec '24

As per design for power off mode, clear the wake enable register during resume sequence. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 5a4bfaef65fb..96a3aa6da711 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1190,6 +1190,7 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { return amd_sdw_clock_stop_exit(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; -- 2.34.1

3 3

[PATCH] soundwire: amd: clear wake enable register for power off mode
by Vijendar Mukunda 23 Dec '24

23 Dec '24

As per design for power off mode, clear the wake enable register during resume sequence. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 5a4bfaef65fb..96a3aa6da711 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1190,6 +1190,7 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { return amd_sdw_clock_stop_exit(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; -- 2.34.1

3 2

[PATCH V2] soundwire: amd: clear wake enable register for power off mode
by Vijendar Mukunda 23 Dec '24

23 Dec '24

As per design for AMD SoundWire manager power off mode, clear the wake enable register during resume sequence. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- Changes Since v1: - Replaced 0x00 with 0 in register write - Updte Commit message drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 5a4bfaef65fb..ca271a728a65 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1190,6 +1190,7 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { return amd_sdw_clock_stop_exit(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + writel(0, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; -- 2.34.1

1 0

[PATCH RESEND] soundwire: amd: clear wake enable register for
by Vijendar Mukunda 23 Dec '24

23 Dec '24

As per design for power off mode, clear the wake enable register during resume sequence. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 5a4bfaef65fb..96a3aa6da711 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1190,6 +1190,7 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { return amd_sdw_clock_stop_exit(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; -- 2.34.1

3 3

make check fails to compile playmidi1.c: "error: expected identifier or '(' before numeric constant"
by GitHub issues - opened 22 Dec '24

22 Dec '24

alsa-project/alsa-lib issue #430 was opened from 746B61: Getting the latest version from branch master (352cbc5e) and running `./gitcompile` followed by `make check` fails with: ``` Making check in test make[1]: Entering directory '/tmp/alsa-lib-master/test' Making check in . make[2]: Entering directory '/tmp/alsa-lib-master/test' make control pcm pcm_min latency seq seq-ump-example playmidi1 timer rawmidi midiloop umpinfo oldapi queue_timer namehint client_event_filter chmap audio_time user-ctl-element-set pcm-multi-thread make[3]: Entering directory '/tmp/alsa-lib-master/test' CC control.o CCLD control CC pcm.o CCLD pcm CC pcm_min.o CCLD pcm_min CC latency.o CCLD latency CC seq.o In file included from seq.c:12: seq-sender.c: In function 'event_sender': seq-sender.c:150:62: warning: variable 'pcm_flag' set but not used [-Wunused-but-set-variable] 150 | int client, port, queue, max, err, v1, v2, time = 0, pcm_flag = 0; | ^~~~~~~~ CCLD seq CC seq-ump-example.o CCLD seq-ump-example CC playmidi1.o In file included from playmidi1.c:48: midifile.h:55:33: error: expected identifier or '(' before numeric constant 55 | #define note_on 0x90 | ^~~~ ../include/alsa/ump_msg.h:170:41: note: in expansion of macro 'note_on' 170 | snd_ump_msg_midi1_note_t note_on; /**< MIDI1 note-on message */ | ^~~~~~~ midifile.h:54:33: error: expected identifier or '(' before numeric constant 54 | #define note_off 0x80 | ^~~~ ../include/alsa/ump_msg.h:171:41: note: in expansion of macro 'note_off' 171 | snd_ump_msg_midi1_note_t note_off; /**< MIDI1 note-off message */ | ^~~~~~~~ midifile.h:57:33: error: expected identifier or '(' before numeric constant 57 | #define control_change 0xb0 | ^~~~ ../include/alsa/ump_msg.h:173:41: note: in expansion of macro 'control_change' 173 | snd_ump_msg_midi1_cc_t control_change; /**< MIDI1 control-change message */ | ^~~~~~~~~~~~~~ midifile.h:55:33: error: expected identifier or '(' before numeric constant 55 | #define note_on 0x90 | ^~~~ ../include/alsa/ump_msg.h:432:41: note: in expansion of macro 'note_on' 432 | snd_ump_msg_midi2_note_t note_on; /**< MIDI2 note-on message */ | ^~~~~~~ midifile.h:54:33: error: expected identifier or '(' before numeric constant 54 | #define note_off 0x80 | ^~~~ ../include/alsa/ump_msg.h:433:41: note: in expansion of macro 'note_off' 433 | snd_ump_msg_midi2_note_t note_off; /**< MIDI2 note-off message */ | ^~~~~~~~ midifile.h:57:33: error: expected identifier or '(' before numeric constant 57 | #define control_change 0xb0 | ^~~~ ../include/alsa/ump_msg.h:438:41: note: in expansion of macro 'control_change' 438 | snd_ump_msg_midi2_cc_t control_change; /**< MIDI2 control-change message */ | ^~~~~~~~~~~~~~ In file included from playmidi1.c:49: midifile.c:57:33: error: expected identifier or '(' before numeric constant 57 | #define set_tempo 0x51 | ^~~~ ../include/alsa/ump_msg.h:711:41: note: in expansion of macro 'set_tempo' 711 | snd_ump_msg_set_tempo_t set_tempo; /**< Set Tempo */ | ^~~~~~~~~ make[3]: *** [Makefile:622: playmidi1.o] Error 1 make[3]: Leaving directory '/tmp/alsa-lib-master/test' make[2]: *** [Makefile:819: check-am] Error 2 make[2]: Leaving directory '/tmp/alsa-lib-master/test' make[1]: *** [Makefile:669: check-recursive] Error 1 make[1]: Leaving directory '/tmp/alsa-lib-master/test' make: *** [Makefile:414: check-recursive] Error 1 ``` The issue was introduced with commit 07cee0ba. playmidi1.c first [includes](https://github.com/alsa-project/alsa-lib/blob/352cbc5eb94a271a9c3… midifile.h from the test directory, which [has](https://github.com/alsa-project/alsa-lib/blob/352cbc5eb94a271a9c3c0ff5… `#define note_on 0x90`. After commit 07cee0ba, ump_msg.h gets included later, which [has](https://github.com/alsa-project/alsa-lib/blob/352cbc5eb94a271a9c3c0ff5… ` snd_ump_msg_midi1_note_t note_on; /**< MIDI1 note-on message */`. The preprocessor replaces `note_on` with `0x90` and the compilation fails with the error above. Issue URL : https://github.com/alsa-project/alsa-lib/issues/430 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

sun4i-codec: add routing for headphones and internal speaker
by GitHub pull_request - opened 21 Dec '24

21 Dec '24

alsa-project/alsa-ucm-conf pull request #491 was opened from tokyovigilante: The sun4i-codec kernel driver covers a large number of Allwinner SoCs. The H616 codec has a single line-out route, which is used in concert with a Toshiba mux chip to send audio to either an internal speaker or headphone jack on a number of Anbernic handheld gaming devices. Add a UCM configuration to allow enabling/disabling the speaker amp depending on whether headphones are in use or not. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/491 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/491.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

API memory allocator
by GitHub issues - opened 20 Dec '24

20 Dec '24

alsa-project/alsa-lib issue #429 was opened from sylware: If I recall properly some data returned from the alsa lib API must be free using the C runtime free(). Would it be appropriate to move away from the C allocator to an API install-able memory allocator (like vulkan3D)? Backward compatibility would be achieved since the "default memory allocator" would be the C one, if the C runtime is around. Unless that part of the API is already deprecated (and I should not have used it). Issue URL : https://github.com/alsa-project/alsa-lib/issues/429 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 00/35] Reorganize kerneldoc parameter names
by Julia Lawall 20 Dec '24

20 Dec '24

Reorganize kerneldoc parameter names to match the parameter order in the function header. The misordered cases were identified using the following Coccinelle semantic patch: // <smpl> @initialize:ocaml@ @@ let parse_doc l = let pieces = List.map String.trim (String.split_on_char '*' l) in let l = String.concat " " pieces in match String.split_on_char ':' l with x::xs -> x | _ -> "" let params ps = List.rev (List.fold_left (fun prev (pm,_) -> let ty = String.trim(Pretty_print_c.string_of_fullType pm.Ast_c.p_type) in if ty = "void" && pm.Ast_c.p_namei = None then prev else let name = match pm.Ast_c.p_namei with Some name -> name | None -> failwith "function parameter has no name" in (String.trim (Pretty_print_c.string_of_name name),ty)::prev) [] ps) @r@ comments c; identifier fn; position p; parameter list ps; type T; @@ T@c fn@p(ps) { ... } @script:ocaml@ p << r.p; c << r.c; (_,ps) << r.ps; @@ let isdoc c ps = List.length ps > 1 && (let c = String.trim c in String.length c > 3 && String.sub c 0 3 = "/**" && String.get c 3 != '*') in let subset l1 l2 = List.for_all (fun x -> List.mem x l2) l1 in let (cb,cm,ca) = List.hd c in match List.rev cb with c::_ when isdoc c ps -> let pieces = String.split_on_char '@' c in (match pieces with _::tl -> let d_names = List.map parse_doc tl in (* check parameters *) let p_names = List.map fst (params ps) in if d_names <> [] && not(d_names = p_names) then begin if List.sort compare d_names = List.sort compare p_names then Coccilib.print_main "out of order" p else if subset d_names p_names then Coccilib.print_main "doc is missing a parameter" p else if subset d_names p_names then Coccilib.print_main "doc has an extra parameter" p end | _ -> ()) | _ -> () // </smpl> --- arch/arm/mach-omap2/prm2xxx_3xxx.c | 1 - arch/powerpc/platforms/ps3/interrupt.c | 2 +- arch/powerpc/platforms/ps3/repository.c | 2 +- drivers/base/firmware_loader/main.c | 2 +- drivers/comedi/drivers/comedi_8254.c | 2 +- drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c | 2 +- drivers/gpu/drm/amd/display/dc/core/dc.c | 2 +- drivers/gpu/drm/drm_gem_framebuffer_helper.c | 3 +-- drivers/gpu/drm/drm_gpuvm.c | 2 +- drivers/gpu/drm/radeon/radeon_ib.c | 2 +- drivers/iommu/iommu.c | 2 +- drivers/leds/leds-gpio-register.c | 2 +- drivers/mfd/atmel-smc.c | 4 ++-- drivers/misc/mei/bus.c | 2 +- drivers/mtd/ubi/eba.c | 2 +- drivers/net/ethernet/broadcom/bnxt/bnxt_hwrm.c | 2 +- drivers/net/ethernet/intel/e1000/e1000_hw.c | 2 +- drivers/net/ethernet/intel/i40e/i40e_common.c | 7 +++---- drivers/net/ethernet/intel/ice/ice_common.c | 2 +- drivers/net/ethernet/intel/ixgbe/ixgbe_common.c | 2 +- drivers/nvdimm/dimm_devs.c | 2 +- drivers/pci/hotplug/pci_hotplug_core.c | 2 +- drivers/pinctrl/pinmux.c | 2 +- drivers/slimbus/messaging.c | 2 +- drivers/soc/qcom/qmi_interface.c | 2 +- drivers/soundwire/stream.c | 2 +- drivers/usb/gadget/config.c | 4 ++-- fs/char_dev.c | 2 +- fs/dcache.c | 4 ++-- fs/seq_file.c | 2 +- kernel/audit.c | 2 +- kernel/resource.c | 2 +- kernel/sysctl.c | 1 - kernel/trace/ring_buffer.c | 2 +- lib/lru_cache.c | 2 +- lib/maple_tree.c | 2 +- mm/mmu_notifier.c | 2 +- net/dccp/feat.c | 2 +- net/mac80211/mesh_hwmp.c | 6 +++--- net/mac80211/mesh_pathtbl.c | 10 +++++----- net/socket.c | 2 +- net/sunrpc/xprt.c | 2 +- net/tipc/link.c | 14 +++++++------- net/tipc/msg.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- 45 files changed, 60 insertions(+), 64 deletions(-)

4 5

[alsa-ucm-conf] sof-soundwire: Use correct speaker data port and PCM for cs35l56 bridge
by Maciej Strozek 20 Dec '24

20 Dec '24

Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 176f880..2e8c1c6 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -23,8 +23,8 @@ SectionDevice."Speaker" { EnableSequence [ cset "name='cs42l43 Speaker L Input 1' 'ASPRX1'" cset "name='cs42l43 Speaker R Input 1' 'ASPRX2'" - cset "name='cs42l43 ASPTX1 Input' 'DP5RX1'" - cset "name='cs42l43 ASPTX2 Input' 'DP5RX2'" + cset "name='cs42l43 ASPTX1 Input' 'DP6RX1'" + cset "name='cs42l43 ASPTX2 Input' 'DP6RX2'" cset "name='AMPL ASP1 TX1 Source' 'DSP1TX1'" cset "name='AMPL ASP1 TX2 Source' 'None'" @@ -54,7 +54,7 @@ SectionDevice."Speaker" { Value { PlaybackPriority 100 - PlaybackPCM "hw:${CardId},0" + PlaybackPCM "hw:${CardId},2" PlaybackMixer "default:${CardId}" PlaybackMixerElem "Speaker" } -- 2.39.5

2 1

hda: how to implement component master_unbind?
by Kai Vehmanen 20 Dec '24

20 Dec '24

Hi Takashi et al, I've been having multiple discussions with our i915 team w.r.t. audio driver behaviour when the aggregate component is unbound, triggered by i915 unbind. This came up again in the recent regression with devres allocations and I now dug into the topic again. In short, I'm puzzled how to really implement this. ALSA (or ASoC) don't really have support for individual components of a card disappearing in a hotplug fashion. At card level, we do have such support (USB, firewire and recent work for PCI hotplug). But not individual components of a card getting unplugged. In current code we have this: static void hdac_component_master_unbind(struct device *dev) { » struct drm_audio_component *acomp = hdac_get_acomp(dev); » if (acomp->audio_ops && acomp->audio_ops->master_unbind) » » acomp->audio_ops->master_unbind(dev, acomp); » module_put(acomp->ops->owner); » component_unbind_all(dev, acomp); » WARN_ON(acomp->ops || acomp->dev); } ... when e.g. i915 driver is unbound (but could be any driver using the component framework, not jus Intel hw), i915 calls component_del() and HDA gets call to the above. After this, acomp ops are null are audio no longer has ability to talk to i915 driver (which makes sense given it's unbound). If audio was runtime suspended, this kind of works (but relies on some good luck). Upon HDA controller resume, we note acomp ops are NULL and HDMI/DP operations (like snd_hdac_display_power()) silently become no-ops. PCM streaming will go to /dev/null, but this is ok'ish since no monitor can be connected anyways. If audio was active, we start to get warnings or worse. Audio expects hw display codec to be powered and programmed, but suddenly it mey lose state. If the audio controller is actually part of the display hardware (e.g. discrete GPUs), then controller registers can stop functioning (they should be still available, but given the main diplay driver is unbound, odds of suprising behaviour are high). So this is less than ideal. I've been looking at options: 1) prevent/block the unbind if audio device is busy The component framework does not allow individual components to return -EBUSY, so there's no way to let others know that unbind is not possible now. This would force anyone testing DRM driver unbind to first unbind the audio driver and stop any active audio use-cases if needed. 2) unbind the ALSA card I've experimented doing a device_unregister() from the above callback, but this has not really worked (onion peeling exercise of new probelms). The code is shared by multiple controllers, so getting a handle to an snd_card object is not straighforward (drvdata is differnet for SOF and snd-hda-intel for instance). But with some new work, maybe we could call snd_card_disconnect() to detach the card and inform user-space. 3) continue as-is and try to fix bugs If audio is active, maybe we could force a acomp->put_power() to ensure a clean unregister of the display part. But this leaves a big chunk of issues especially with HDA controllers that require the display to be powered on, to function. .. Any ideas, and/or has there been prior work? It seems Takashi it's been mostly you who has been active working on the acomp integration recently. I also included Chris, Daniel and Jani who've had patches to hdac_component.c. Br, Kai

3 5

[GIT PULL] ASoC fixes for v6.13-rc3
by Mark Brown 20 Dec '24

20 Dec '24

The following changes since commit 78d4f34e2115b517bcbfe7ec0d018bbbb6f9b0b8: Linux 6.13-rc3 (2024-12-15 15:58:23 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-rc3 for you to fetch changes up to 32c9c06adb5b157ef259233775a063a43746d699: ASoC: mediatek: disable buffer pre-allocation (2024-12-19 11:15:09 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A mix of quirks and small fixes, nothing too major anywhere. ---------------------------------------------------------------- Chen-Yu Tsai (2): ASoC: dt-bindings: realtek,rt5645: Fix CPVDD voltage comment ASoC: mediatek: disable buffer pre-allocation Mark Brown (1): ASoC: Intel: sof_sdw: Update DMI matches for Lenovo Peter Ujfalusi (1): ASoC: SOF: Intel: hda-dai: Do not release the link DMA on STOP Richard Fitzgerald (2): ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 21Q6 and 21Q7 ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 21QA and 21QB Shuming Fan (1): ASoC: rt722: add delay time to wait for the calibration procedure Venkata Prasad Potturu (1): ASoC: amd: ps: Fix for enabling DMIC on acp63 platform via _DSD entry .../devicetree/bindings/sound/realtek,rt5645.yaml | 2 +- sound/soc/amd/ps/pci-ps.c | 17 ++++++++++++++- sound/soc/codecs/rt722-sdca.c | 7 +++++- sound/soc/intel/boards/sof_sdw.c | 23 +++++++++++++++++--- .../soc/mediatek/common/mtk-afe-platform-driver.c | 4 ++-- sound/soc/sof/intel/hda-dai.c | 25 ++++++++++++++++------ sound/soc/sof/intel/hda.h | 2 -- 7 files changed, 64 insertions(+), 16 deletions(-)

2 1

[PATCH] ASoC: codecs: ES8326: Reduce pop noise
by Zhang Yi 20 Dec '24

20 Dec '24

We modify the value of ES8326_ANA_MICBIAS to reduce the pop noise Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index a5603b617688..27736c6933c5 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -616,7 +616,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) 0x0F, 0x0F); if (es8326->version > ES8326_VERSION_B) { regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); } } } else { @@ -631,6 +631,8 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) regmap_write(es8326->regmap, ES8326_HPR_OFFSET_INI, offset_r); es8326->calibrated = true; } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0x00); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x00); if (direction == SNDRV_PCM_STREAM_PLAYBACK) { regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x01); usleep_range(1000, 5000); @@ -645,7 +647,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) } else { msleep(300); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x50); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x70); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00); } regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, @@ -676,6 +678,10 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x00); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x20); regmap_update_bits(es8326->regmap, ES8326_RESET, 0x02, 0x00); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); + } break; case SND_SOC_BIAS_PREPARE: break; @@ -683,6 +689,12 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x00); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0xc0); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x80); break; case SND_SOC_BIAS_OFF: clk_disable_unprepare(es8326->mclk); @@ -771,6 +783,10 @@ static void es8326_jack_button_handler(struct work_struct *work) cur_button = SND_JACK_BTN_0; break; case 0x6f: + if (es8326->version > ES8326_VERSION_B) { + cur_button = SND_JACK_BTN_0; + break; + } case 0x4b: /* button volume up */ cur_button = SND_JACK_BTN_1; @@ -1082,7 +1098,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x10); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); } -- 2.17.1

1 0

USB-Audio: Add support for TASCAM Model 12
by GitHub pull_request - opened 20 Dec '24

20 Dec '24

alsa-project/alsa-ucm-conf pull request #490 was opened from Unveiled53: Adds support for the TASCAM Model 12 USB multi-track mixer. Couple of things to note: - This profile assumes the "USB AUDIO" setting on the mixer itself is set to "MULTI INPUT". - Under the "MULTI INPUT" mode, this mixer has 12 logical input channels (Last two are a stereo mix) and 10 logical output channels. - Most of the accessible I/O is combo XLR/TRS jacks, so I opted to have the mono capture devices (With the exception of physical inputs 8 and 10 which are TRS only) named as "Mic" with the rest as "Line". Otherwise, the profile is tested and working with the latest USB-Audio profiles. [alsa-info.txt](https://github.com/user-attachments/files/18207924/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/490 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/490.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RFC PATCH] soc: audio-graph-card2: use correct endpoint when getting link parameters
by Ivaylo Dimitrov 20 Dec '24

20 Dec '24

We may have multiple links between ports, with each link having different parameters. Currently, no matter the topology, it is always port endpoint 0 that is used when setting parameters. On a complex sound system, like the one found on Motorola droid4, hifi and voice DAIs require differents formats (i2s vs dsp_a) and curently it is impossible to use DT to set that. Implementing the change leads to partially dropping of at least 0dedbde5062d (ASoC: cpcap: Implement set_tdm_slot for voice call support), as core does most of what is needed to configure voice DAI. We (on Maemo Leste ) use the patch (along with few others) to have voice calls working properly on d4 through UCM. The patch is for linux 6.6, I want to know whether the approach would be accepted before sending a proper patch for current master. the original commit message follows: When link parameters are parsed, it is always endpoint@0 that is used and parameters set to other endpoints are ignored. Fix that by using endpoint that is set in DT when parsing link parameters. Signed-off-by: Ivaylo Dimitrov <ivo.g.dimitrov.75(a)gmail.com> --- sound/soc/generic/audio-graph-card2.c | 59 +++++++++++++-------------- 1 file changed, 28 insertions(+), 31 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index b1c675c6b6db..163a20c8ffee 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -508,17 +508,16 @@ static int __graph_parse_node(struct asoc_simple_priv *priv, static int graph_parse_node(struct asoc_simple_priv *priv, enum graph_type gtype, - struct device_node *port, + struct device_node *ep, struct link_info *li, int is_cpu) { - struct device_node *ep; int ret = 0; + struct device_node *port = of_get_parent(ep); + bool is_multi = graph_lnk_is_multi(port); - if (graph_lnk_is_multi(port)) { + if (is_multi) { int idx; - of_node_get(port); - for (idx = 0;; idx++) { ep = graph_get_next_multi_ep(&port); if (!ep) @@ -532,9 +531,8 @@ static int graph_parse_node(struct asoc_simple_priv *priv, } } else { /* Single CPU / Codec */ - ep = port_to_endpoint(port); + of_node_put(port); ret = __graph_parse_node(priv, gtype, ep, li, is_cpu, 0); - of_node_put(ep); } return ret; @@ -591,22 +589,20 @@ static void graph_parse_daifmt(struct device_node *node, } static void graph_link_init(struct asoc_simple_priv *priv, - struct device_node *port, + struct device_node *ep, struct link_info *li, int is_cpu_node) { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); - struct device_node *ep; + struct device_node *port = of_get_parent(ep); + bool is_multi = graph_lnk_is_multi(port); struct device_node *ports; unsigned int daifmt = 0, daiclk = 0; unsigned int bit_frame = 0; - if (graph_lnk_is_multi(port)) { - of_node_get(port); + if (is_multi) { ep = graph_get_next_multi_ep(&port); port = of_get_parent(ep); - } else { - ep = port_to_endpoint(port); } ports = of_get_parent(port); @@ -642,6 +638,9 @@ static void graph_link_init(struct asoc_simple_priv *priv, dai_link->ops = &graph_ops; if (priv->ops) dai_link->ops = priv->ops; + + of_node_put(port); + of_node_put(ports); } int audio_graph2_link_normal(struct asoc_simple_priv *priv, @@ -650,7 +649,7 @@ int audio_graph2_link_normal(struct asoc_simple_priv *priv, { struct device_node *cpu_port = lnk; struct device_node *cpu_ep = port_to_endpoint(cpu_port); - struct device_node *codec_port = of_graph_get_remote_port(cpu_ep); + struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep); int ret; /* @@ -658,20 +657,20 @@ int audio_graph2_link_normal(struct asoc_simple_priv *priv, * see * __graph_parse_node() :: DAI Naming */ - ret = graph_parse_node(priv, GRAPH_NORMAL, codec_port, li, 0); + ret = graph_parse_node(priv, GRAPH_NORMAL, codec_ep, li, 0); if (ret < 0) goto err; /* * call CPU, and set DAI Name */ - ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_port, li, 1); + ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_ep, li, 1); if (ret < 0) goto err; - graph_link_init(priv, cpu_port, li, 1); + graph_link_init(priv, cpu_ep, li, 1); err: - of_node_put(codec_port); + of_node_put(codec_ep); of_node_put(cpu_ep); return ret; @@ -684,7 +683,6 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, { struct device_node *ep = port_to_endpoint(lnk); struct device_node *rep = of_graph_get_remote_endpoint(ep); - struct device_node *rport = of_graph_get_remote_port(ep); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); int is_cpu = asoc_graph_is_ports0(lnk); @@ -718,7 +716,7 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; - ret = graph_parse_node(priv, GRAPH_DPCM, rport, li, 1); + ret = graph_parse_node(priv, GRAPH_DPCM, rep, li, 1); if (ret) goto err; } else { @@ -751,7 +749,7 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; - ret = graph_parse_node(priv, GRAPH_DPCM, rport, li, 0); + ret = graph_parse_node(priv, GRAPH_DPCM, rep, li, 0); if (ret < 0) goto err; } @@ -761,11 +759,10 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, snd_soc_dai_link_set_capabilities(dai_link); - graph_link_init(priv, rport, li, is_cpu); + graph_link_init(priv, rep, li, is_cpu); err: of_node_put(ep); of_node_put(rep); - of_node_put(rport); return ret; } @@ -777,7 +774,7 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct device_node *port0, *port1, *ports; - struct device_node *codec0_port, *codec1_port; + struct device_node *codec0_ep, *codec1_ep; struct device_node *ep0, *ep1; u32 val = 0; int ret = -EINVAL; @@ -834,31 +831,31 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, ep0 = port_to_endpoint(port0); ep1 = port_to_endpoint(port1); - codec0_port = of_graph_get_remote_port(ep0); - codec1_port = of_graph_get_remote_port(ep1); + codec0_ep = of_graph_get_remote_endpoint(ep0); + codec1_ep = of_graph_get_remote_endpoint(ep1); /* * call Codec first. * see * __graph_parse_node() :: DAI Naming */ - ret = graph_parse_node(priv, GRAPH_C2C, codec1_port, li, 0); + ret = graph_parse_node(priv, GRAPH_C2C, codec1_ep, li, 0); if (ret < 0) goto err2; /* * call CPU, and set DAI Name */ - ret = graph_parse_node(priv, GRAPH_C2C, codec0_port, li, 1); + ret = graph_parse_node(priv, GRAPH_C2C, codec0_ep, li, 1); if (ret < 0) goto err2; - graph_link_init(priv, codec0_port, li, 1); + graph_link_init(priv, codec0_ep, li, 1); err2: of_node_put(ep0); of_node_put(ep1); - of_node_put(codec0_port); - of_node_put(codec1_port); + of_node_put(codec0_ep); + of_node_put(codec1_ep); err1: of_node_put(ports); of_node_put(port0); -- 2.30.2

1 0

CS42L43 headset mic enable/disable not applied when jack insert state changes
by GitHub issues - opened 19 Dec '24

19 Dec '24

alsa-project/alsa-ucm-conf issue #489 was opened from rfvirgil: Originally the configuration for sof-soundwire/cs42l43*.conf had the headset mic and internal mics on the same ALSA capture device 4. The Headset Mic enable and disable sequences ran as expected when the headset jack was inserted and removed. But SOF expects that device 4 is the internal mics, and headset mic should be on device 1. So we made this change to move the headset mic to capture device 1: https://github.com/alsa-project/alsa-ucm-conf/commit/30989bd0c2aa3f9f4b6f5e… But now the Headset Mic enable and disable sequences do not run when the headset jack state changes. Seen on multiple systems with Pipewire + alsa-ucm + CS42L43 on sof-soundwire. Example from a system that booted with jack unplugged. The jack was then plugged in. The CS42L43 has reported that the headset mic is present: ``` Simple mixer control 'Headset Mic',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] ``` But the enable sequence has not been applied (DP2TX1 and DP2TX2 should have been set to Decimator 1): ``` Simple mixer control 'cs42l43 DP2TX1 Input',0 Capabilities: enum Items: 'None' 'Tone Generator 1' 'Tone Generator 2' 'Decimator 1' 'Decimator 2' 'Decimator 3' 'Decimator 4' 'ASPRX1' 'ASPRX2' 'ASPRX3' 'ASPRX4' 'ASPRX5' 'ASPRX6' 'DP5RX1' 'DP5RX2' 'DP6RX1' 'DP6RX2' 'DP7RX1' 'DP7RX2' 'ASRC INT1' 'ASRC INT2' 'ASRC INT3' 'ASRC INT4' 'ASRC DEC1' 'ASRC DEC2' 'ASRC DEC3' 'ASRC DEC4' 'ISRC1 INT1' 'ISRC1 INT2' 'ISRC1 DEC1' 'ISRC1 DEC2' 'ISRC2 INT1' 'ISRC2 INT2' 'ISRC2 DEC1' 'ISRC2 DEC2' 'EQ1' 'EQ2' Item0: 'None' Simple mixer control 'cs42l43 DP2TX2 Input',0 Capabilities: enum Items: 'None' 'Tone Generator 1' 'Tone Generator 2' 'Decimator 1' 'Decimator 2' 'Decimator 3' 'Decimator 4' 'ASPRX1' 'ASPRX2' 'ASPRX3' 'ASPRX4' 'ASPRX5' 'ASPRX6' 'DP5RX1' 'DP5RX2' 'DP6RX1' 'DP6RX2' 'DP7RX1' 'DP7RX2' 'ASRC INT1' 'ASRC INT2' 'ASRC INT3' 'ASRC INT4' 'ASRC DEC1' 'ASRC DEC2' 'ASRC DEC3' 'ASRC DEC4' 'ISRC1 INT1' 'ISRC1 INT2' 'ISRC1 DEC1' 'ISRC1 DEC2' 'ISRC2 INT1' 'ISRC2 INT2' 'ISRC2 DEC1' 'ISRC2 DEC2' 'EQ1' 'EQ2' Item0: 'None' ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/489 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

snd_pcm_writei() takes to much time to write data
by GitHub issues - edited 19 Dec '24

19 Dec '24

alsa-project/alsa-lib issue #428 was edited from nnamera: I am using the ALSA open-source library to read and write audio data for my project. To write audio data, I am calling the snd_pcm_writei function. Initially, it works as expected, but after writing 2-3 frames, the function starts taking significantly more time (around 1 second) to complete. Normally, snd_pcm_writei should execute within 1-3 ms, but this delay is affecting my application's overall performance, causing audio chopping issues. Here are the details of my setup: Frame size: ~8000 bytes (writing 8000 bytes per call) Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Soft volume PCM Control: amba-playback min_dB: -20 max_dB: 20 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Hardware PCM card 0 'es8389' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAV[ 22.206911] ambarella-i2c e400a000.i2c: No ACK from address 0x6c, 0:0! ED format [ 22.214533] os04c10: probe of 2-0001 failed with error -5 : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 appl_ptr : 0 hw_ptr : 0 I would appreciate any guidance or suggestions on why this delay is happening and how I can resolve it. Let me know if you need any additional details or logs. Issue URL : https://github.com/alsa-project/alsa-lib/issues/428 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_pcm_readi() takes to much time to write data
by GitHub issues - opened 19 Dec '24

19 Dec '24

alsa-project/alsa-lib issue #428 was opened from nnamera: I am using the ALSA open-source library to read and write audio data for my project. To write audio data, I am calling the snd_pcm_writei function. Initially, it works as expected, but after writing 2-3 frames, the function starts taking significantly more time (around 1 second) to complete. Normally, snd_pcm_writei should execute within 1-3 ms, but this delay is affecting my application's overall performance, causing audio chopping issues. Here are the details of my setup: Frame size: ~8000 bytes (writing 8000 bytes per call) Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Soft volume PCM Control: amba-playback min_dB: -20 max_dB: 20 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Hardware PCM card 0 'es8389' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAV[ 22.206911] ambarella-i2c e400a000.i2c: No ACK from address 0x6c, 0:0! ED format [ 22.214533] os04c10: probe of 2-0001 failed with error -5 : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 appl_ptr : 0 hw_ptr : 0 I would appreciate any guidance or suggestions on why this delay is happening and how I can resolve it. Let me know if you need any additional details or logs. Issue URL : https://github.com/alsa-project/alsa-lib/issues/428 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 1/1] ASoC: codecs: nau8824: fix max volume for Speaker output
by Maxim Kochetkov 19 Dec '24

19 Dec '24

There is no audio output if Speaker volume is set above 25. According to datasheet Rev 2.5 maximum allowed value for the Speaker output is 0b11001 (25) 0x6D CLASSD_GAIN_1/ 0x6E CLASSD_GAIN_2: Left/Right Channel Class-D Driver Gain For DAC Left/Right Input (Step size is 1dB.) 00000 = Mute (DEFAULT) 00001 = 0dB 00002 = 1dB ... 11000 = 23dB 11001 = 24dB So adjust this value in accordance with the datasheet. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/codecs/nau8824.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 12540397fd4d..5aaf8c496300 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -368,13 +368,13 @@ static const struct snd_kcontrol_new nau8824_snd_controls[] = { SOC_ENUM("DAC Oversampling Rate", nau8824_dac_oversampl_enum), SOC_SINGLE_TLV("Speaker Right DACR Volume", - NAU8824_REG_CLASSD_GAIN_1, 8, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_1, 8, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Left DACL Volume", - NAU8824_REG_CLASSD_GAIN_2, 0, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_2, 0, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Left DACR Volume", - NAU8824_REG_CLASSD_GAIN_1, 0, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_1, 0, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Right DACL Volume", - NAU8824_REG_CLASSD_GAIN_2, 8, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_2, 8, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Headphone Right DACR Volume", NAU8824_REG_ATT_PORT0, 8, 0x1f, 0, hp_vol_tlv), -- 2.45.2

1 0

[PATCH v2] ASoC: amd: yc: Fix the wrong return value
by Venkata Prasad Potturu 17 Dec '24

17 Dec '24

With the current implementation, when ACP driver fails to read ACPI _WOV entry then the DMI overrides code won't invoke, may cause regressions for some BIOS versions. Add a condition check to jump to check the DMI entries incase of ACP driver fail to read ACPI _WOV method. Fixes: 4095cf872084 (ASoC: amd: yc: Fix for enabling DMIC on acp6x via _DSD entry) Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- Changes since v1: - removed redundant "&& wov_en" check sound/soc/amd/yc/acp6x-mach.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index e38c5885dadf..ecf57a6cb7c3 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -578,14 +578,19 @@ static int acp6x_probe(struct platform_device *pdev) handle = ACPI_HANDLE(pdev->dev.parent); ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); - if (!ACPI_FAILURE(ret)) + if (!ACPI_FAILURE(ret)) { wov_en = dmic_status; + if (!wov_en) + return -ENODEV; + } else { + /* Incase of ACPI method read failure then jump to check_dmi_entry */ + goto check_dmi_entry; + } - if (is_dmic_enable && wov_en) + if (is_dmic_enable) platform_set_drvdata(pdev, &acp6x_card); - else - return 0; +check_dmi_entry: /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); if (dmi_id) -- 2.25.1

5 5

UCM2: Intel: sof-hda-dsp: Enable Dmic0 pre-processing and SDW speaker post-processing
by GitHub pull_request - edited 16 Dec '24

16 Dec '24

alsa-project/alsa-ucm-conf pull request #481 was edited from singalsu: This patch adds enable of the microphone capture pre-processing features as BootSequence inital values if the controls for them are available. If the user later switches these off, they will remain off. The DRC is dynamic range control that boosts the capture level, and the TDFB (time-domain fixed beamformer) improves capture signa-to-noise ratio for set beam angle direction. The used 0 degrees angle is defined as angle towards notebook computer user. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/481 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/481.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: amd: ps: Fix for enabling DMIC on acp63 platform via _DSD entry
by Venkata Prasad Potturu 16 Dec '24

16 Dec '24

Add condition check to register ACP PDM sound card by reading _WOV acpi entry. Fixes: 0386d765f27a ("ASoC: amd: ps: refactor acp device configuration read logic") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index a7583844f5b4..983d7fa13938 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -374,11 +374,18 @@ static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *a { struct acpi_device *pdm_dev; const union acpi_object *obj; + acpi_handle handle; + acpi_integer dmic_status; u32 config; bool is_dmic_dev = false; bool is_sdw_dev = false; + bool wov_en, dmic_en; int ret; + /* IF WOV entry not found, enable dmic based on acp-audio-device-type entry*/ + wov_en = true; + dmic_en = false; + config = readl(acp_data->acp63_base + ACP_PIN_CONFIG); switch (config) { case ACP_CONFIG_4: @@ -411,10 +418,18 @@ static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *a if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type", ACPI_TYPE_INTEGER, &obj) && obj->integer.value == ACP_DMIC_DEV) - is_dmic_dev = true; + dmic_en = true; } + + handle = ACPI_HANDLE(&pci->dev); + ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); + if (!ACPI_FAILURE(ret)) + wov_en = dmic_status; } + if (dmic_en && wov_en) + is_dmic_dev = true; + if (acp_data->is_sdw_config) { ret = acp_scan_sdw_devices(&pci->dev, ACP63_SDW_ADDR); if (!ret && acp_data->info.link_mask) -- 2.25.1

3 2

[PATCH v7 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 13 Dec '24

13 Dec '24

This function is base on the accelerator implementation for compress API: 04177158cf98 ("ALSA: compress_offload: introduce accel operation mode") Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v7: - add 'select SND_COMPRESS_OFFLOAD' for fixing compile issue with imx_v6_v7_defconfig changes in v6: - rebase to latest tree, change to use 'MODULE_IMPORT_NS("DMA_BUF")' - Add Acked-by: Vinod Koul for patch 1/6 changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 2 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1261 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

3 9

[PATCH v6 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 13 Dec '24

13 Dec '24

This function is base on the accelerator implementation for compress API: 04177158cf98 ("ALSA: compress_offload: introduce accel operation mode") Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v6: - rebase to latest tree, change to use 'MODULE_IMPORT_NS("DMA_BUF")' - Add Acked-by: Vinod Koul for patch 1/6 changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

2 7

Re: [syzbot] [tipc?] kernel BUG in __pskb_pull_tail
by syzbot 13 Dec '24

13 Dec '24

syzbot has bisected this issue to: commit de4f5fed3f231a8ff4790bf52975f847b95b85ea Author: Jens Axboe <axboe(a)kernel.dk> Date: Wed Mar 29 14:52:15 2023 +0000 iov_iter: add iter_iovec() helper bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=17424730580000 start commit: 96b6fcc0ee41 Merge branch 'net-dsa-cleanup-eee-part-1' git tree: net-next final oops: https://syzkaller.appspot.com/x/report.txt?x=14c24730580000 console output: https://syzkaller.appspot.com/x/log.txt?x=10c24730580000 kernel config: https://syzkaller.appspot.com/x/.config?x=1362a5aee630ff34 dashboard link: https://syzkaller.appspot.com/bug?extid=4f66250f6663c0c1d67e syz repro: https://syzkaller.appspot.com/x/repro.syz?x=166944f8580000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=1287ecdf980000 Reported-by: syzbot+4f66250f6663c0c1d67e(a)syzkaller.appspotmail.com Fixes: de4f5fed3f23 ("iov_iter: add iter_iovec() helper") For information about bisection process see: https://goo.gl/tpsmEJ#bisection

2 1

[GIT PULL] ASoC fixes for v6.12-rc2
by Mark Brown 12 Dec '24

12 Dec '24

The following changes since commit fac04efc5c793dccbd07e2d59af9f90b7fc0dca4: Linux 6.13-rc2 (2024-12-08 14:03:39 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.12-rc2 for you to fetch changes up to 255cc582e6e16191a20d54bcdbca6c91d3e90c5e: ASoC: Intel: sof_sdw: Add space for a terminator into DAIs array (2024-12-12 11:08:49 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A small pile of driver specific fixes, all quite small and not particularly major. ---------------------------------------------------------------- Charles Keepax (1): ASoC: Intel: sof_sdw: Add space for a terminator into DAIs array Shenghao Ding (1): ASoC: tas2781: Fix calibration issue in stress test Shengjiu Wang (2): ASoC: fsl_xcvr: change IFACE_PCM to IFACE_MIXER ASoC: fsl_spdif: change IFACE_PCM to IFACE_MIXER Stephen Gordon (1): ASoC: audio-graph-card: Call of_node_put() on correct node Venkata Prasad Potturu (1): ASoC: amd: yc: Fix the wrong return value sound/soc/amd/yc/acp6x-mach.c | 13 +++++++++---- sound/soc/codecs/tas2781-i2c.c | 2 +- sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 2 +- sound/soc/generic/audio-graph-card2.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 8 ++++++-- 6 files changed, 19 insertions(+), 10 deletions(-)

2 1

Re: [PATCH v6 4/6] ASoC: fsl_asrc_m2m: Add memory to memory function
by Shengjiu Wang 12 Dec '24

12 Dec '24

On Wed, Dec 11, 2024 at 11:28 PM Mark Brown <broonie(a)kernel.org> wrote: > > On Wed, Dec 11, 2024 at 11:08:47AM +0800, Shengjiu Wang wrote: > > Implement the ASRC memory to memory function using > > the compress framework, user can use this function with > > compress ioctl interface. > > > > This feature can be shared by ASRC and EASRC drivers > > This breaks an imx_v6_v7_defconfig build: > > arm-linux-gnueabihf-ld: sound/soc/fsl/fsl_asrc_m2m.o: in function `fsl_asrc_m2m_ > init': > fsl_asrc_m2m.c:(.text+0x7ac): undefined reference to `snd_compress_new' > > Looks like we're missing some select statements. Thanks for noticing this issue. Sorry about that, I will fix it. Best regards Shengjiu Wang

1 0

git clone failure
by GitHub issues - opened 12 Dec '24

12 Dec '24

alsa-project/tinycompress issue #27 was opened from antonio-fr: Working on embedded project, there's a recipe for a system that uses tinycompress, and points to the alsa-project git server. Unfortunately, tinycompress can't be downloaded, leading the building system unable to make the system image. ``` $ git clone git://git.alsa-project.org/tinycompress.git Cloning into 'tinycompress'... fatal: Could not read from remote repository. Please make sure you have the correct access rights and the repository exists. ``` It worked yesterday, And I can see the project [there in the web interface](https://git.alsa-project.org/?p=tinycompress.git;a=tree). But right now, no luck. Issue URL : https://github.com/alsa-project/tinycompress/issues/27 Repository URL: https://github.com/alsa-project/tinycompress

1 0

Re: [PATCH v2 30/37] ALSA: seq: Automatic conversion of UMP events
by Mike Lothian 11 Dec '24

11 Dec '24

Hi I'm not sure when this first appeared but I see the following when compiling/linking with Clang 19.1.5 vmlinux.o: warning: objtool: cc_ev_to_ump_midi2+0x5a5: return with modified stack frame This is the only commit I can see that has cc_ev_to_ump_midi2 in it Cheers Mike

2 1

[PATCH v1] ASoC: tas2781: Fix calibration issue in stress test
by Shenghao Ding 11 Dec '24

11 Dec '24

One specific test condition: the default registers of p[j].reg ~ p[j+3].reg are 0, TASDEVICE_REG(0x00, 0x14, 0x38)(PLT_FLAG_REG), TASDEVICE_REG(0x00, 0x14, 0x40)(SINEGAIN_REG), and TASDEVICE_REG(0x00, 0x14, 0x44)(SINEGAIN2_REG). After first calibration, they are freshed to TASDEVICE_REG(0x00, 0x1a, 0x20), TASDEVICE_REG(0x00, 0x16, 0x58)(PLT_FLAG_REG), TASDEVICE_REG(0x00, 0x14, 0x44)(SINEGAIN_REG), and TASDEVICE_REG(0x00, 0x16, 0x64)(SINEGAIN2_REG) via "Calibration Start" kcontrol. In second calibration, the p[j].reg ~ p[j+3].reg have already become tas2781_cali_start_reg. However, p[j+2].reg, TASDEVICE_REG(0x00, 0x14, 0x44)(SINEGAIN_REG), will be freshed to TASDEVICE_REG(0x00, 0x16, 0x64), which is the third register in the input params of the kcontrol. This is why only first calibration can work, the second-time, third-time or more-time calibration always failed without reboot. Of course, if no p[j].reg is in the list of tas2781_cali_start_reg, this stress test can work well. Fixes: 49e2e353fb0d ("ASoC: tas2781: Add Calibration Kcontrols for Chromebook") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index be2ca5eb6c93..fb8cd2284fe8 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -370,7 +370,7 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, tasdevice_dev_read(tas_priv, i, p[j].reg, (int *)&p[j].val[0]); } else { - switch (p[j].reg) { + switch (tas2781_cali_start_reg[j].reg) { case 0: { if (!reg[0]) continue; -- 2.34.1

1 0

rt722: add headset mic led control
by GitHub pull_request - edited 11 Dec '24

11 Dec '24

alsa-project/alsa-ucm-conf pull request #487 was edited from shumingfan: rt722: add headset mic led control Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/487 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/487.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v4 00/24] Add audio support for the Renesas RZ/G3S SoC
by Claudiu 10 Dec '24

10 Dec '24

From: Claudiu Beznea <claudiu.beznea.uj(a)bp.renesas.com> Hi, Series enables the audio support for the Renesas RZ/G3S SoC along with runtime PM and suspend to RAM. Patches: - 01-03/24 - update versaclock3 clock generator driver to support the 5L35023 hardware variant; versaclock3 provides clocks for the audio devices (SSIF, DA7212 codec) - 04-19/24 - add SSIF support for the RZ/G3S SoC; fixes and cleanups were also included - 20-24/24 - add device tree support Merge strategy, if any: - clock patches (01-03/24) can go the clock tree - audio patches (04-19/24) can go through the audio tree - device tree patches (20-24/24) can go through the Renesas tree Thank you, Claudiu Beznea Changes in v4: - added patch 18/24 "ASoC: dt-bindings: renesas,rz-ssi: Remove DMA description" - collected tags - addressed review comments - dropped already integrated patches - the changelog for each patch is detailed within the individual patches Changes in v3: - use renesas instead of sh in audio patches title - use proper fixes tag for patch "ASoC: renesas: rz-ssi: Terminate all the DMA transactions" - collected tags Changes in v2: - fixed typos pointed out in the review process - dropped da7213 patches already applied - dropped patch "ASoC: sh: rz-ssi: Use a proper bitmask for clear bits" as requested in the review process Claudiu Beznea (24): clk: versaclock3: Prepare for the addition of 5L35023 device dt-bindings: clock: versaclock3: Document 5L35023 Versa3 clock generator clk: versaclock3: Add support for the 5L35023 variant ASoC: renesas: rz-ssi: Terminate all the DMA transactions ASoC: renesas: rz-ssi: Use only the proper amount of dividers ASoC: renesas: rz-ssi: Fix typo on SSI_RATES macro comment ASoC: renesas: rz-ssi: Remove pdev member of struct rz_ssi_priv ASoC: renesas: rz-ssi: Remove the rz_ssi_get_dai() function ASoC: renesas: rz-ssi: Remove the first argument of rz_ssi_stream_is_play() ASoC: renesas: rz-ssi: Use readl_poll_timeout_atomic() ASoC: renesas: rz-ssi: Use temporary variable for struct device ASoC: renesas: rz-ssi: Use goto label names that specify their actions ASoC: renesas: rz-ssi: Rely on the ASoC subsystem to runtime resume/suspend the SSI ASoC: renesas: rz-ssi: Enable runtime PM autosuspend support ASoC: renesas: rz-ssi: Add runtime PM support ASoC: renesas: rz-ssi: Issue software reset in hw_params API ASoC: renesas: rz-ssi: Add suspend to RAM support ASoC: dt-bindings: renesas,rz-ssi: Remove DMA description ASoC: dt-bindings: renesas,rz-ssi: Document the Renesas RZ/G3S SoC arm64: dts: renesas: r9a08g045: Add SSI nodes arm64: dts: renesas: rzg3s-smarc-som: Add versa3 clock generator node arm64: dts: renesas: Add da7212 audio codec node arm64: dts: renesas: rzg3s-smarc: Enable SSI3 arm64: dts: renesas: rzg3s-smarc: Add sound card .../bindings/clock/renesas,5p35023.yaml | 1 + .../bindings/sound/renesas,rz-ssi.yaml | 19 +- arch/arm64/boot/dts/renesas/r9a08g045.dtsi | 94 ++++++++ .../boot/dts/renesas/rzg3s-smarc-som.dtsi | 47 +++- arch/arm64/boot/dts/renesas/rzg3s-smarc.dtsi | 63 +++++ drivers/clk/clk-versaclock3.c | 67 ++++-- sound/soc/renesas/rz-ssi.c | 226 +++++++++++------- 7 files changed, 388 insertions(+), 129 deletions(-) -- 2.39.2

2 27

[PATCH] ASoC: amd: yc: Fix the wrong return value
by Venkata Prasad Potturu 10 Dec '24

10 Dec '24

With the current implementation, when ACP driver fails to read ACPI _WOV entry then the DMI overrides code won't invoke, may cause regressions for some BIOS versions. Add a condition check to jump to check the DMI entries incase of ACP driver fail to read ACPI _WOV method. Fixes: 4095cf872084 (ASoC: amd: yc: Fix for enabling DMIC on acp6x via _DSD entry) Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/yc/acp6x-mach.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index e38c5885dadf..13af1d485eb0 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -578,14 +578,19 @@ static int acp6x_probe(struct platform_device *pdev) handle = ACPI_HANDLE(pdev->dev.parent); ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); - if (!ACPI_FAILURE(ret)) + if (!ACPI_FAILURE(ret)) { wov_en = dmic_status; + if (!wov_en) + return -ENODEV; + } else { + /* Incase of ACPI method read failure then jump to check_dmi_entry */ + goto check_dmi_entry; + } if (is_dmic_enable && wov_en) platform_set_drvdata(pdev, &acp6x_card); - else - return 0; +check_dmi_entry: /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); if (dmi_id) -- 2.25.1

4 5

Support for ASUS ROG STRIX X870E-E GAMING WIFI
by GitHub pull_request - opened 10 Dec '24

10 Dec '24

alsa-project/alsa-ucm-conf pull request #486 was opened from aidvu: Add support for [ASUS ROG STRIX X870E-E GAMING WIFI](https://rog.asus.com/motherboards/rog-strix/rog-strix-x870e-e-gaming-… ). ```Bus 003 Device 004: ID 0b05:1b9b ASUSTek Computer, Inc. USB Audio``` Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/486 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/486.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: sof-soundwire: Correct FixedBootSequence for dmic info
by GitHub pull_request - opened 10 Dec '24

10 Dec '24

alsa-project/alsa-ucm-conf pull request #485 was opened from ujfalusi: Replace the curly brackets with square brackets. The curly ones break the FixedBootSequence parsing by resetting the sequence. If the /var/lib/alsa/card0.conf.d is not present then UCM will fail because all sequences before the dmic is lost, including the card-init: # alsactl init alsa-lib main.c:619:(execute_cfgsave) unable to open file '/var/lib/alsa/card0.conf.d/42-sof-hdmi.conf': No such file or directory alsa-lib main.c:2456:(set_fixedboot_user) Unable to execute force boot sequence The full sequence supposed to be: card-init ctl-remap cs42l43 dmic hdmi Fixes: 9fa70add7638 ("sof-hda-dsp,sof-soundwire: add CaptureMicInfoFile fields for dmics") Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/485 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/485.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] dt-bindings: sound:atmel-at91sam9g20ek: convert bindings to json-schema
by Balakrishnan Sambath 10 Dec '24

10 Dec '24

Convert atmel-at91sam9g20ek-wm8731-audio DT binding to yaml based json-schema.Change file name to match json-scheme naming. Signed-off-by: Balakrishnan Sambath <balakrishnan.s(a)microchip.com> --- .../bindings/sound/atmel,at91sam9g20ek-wm8731.yaml | 60 ++++++++++++++++++++++ .../sound/atmel-at91sam9g20ek-wm8731-audio.txt | 26 ---------- 2 files changed, 60 insertions(+), 26 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml b/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml new file mode 100644 index 000000000000..f6330707fe1b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,at91sam9g20ek-wm8731.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel at91sam9g20ek wm8731 audio complex + +maintainers: + - Balakrishnan Sambath <balakrishnan.s(a)microchip.com> + +properties: + compatible: + const: atmel,at91sam9g20ek-wm8731-audio + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + atmel,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: A list of the connections between audio components. + minItems: 2 + items: + enum: + # Board Connectors + - "Ext Spk" + - "Int MIC" + + # CODEC Pins + - LHPOUT + - MICIN + atmel,ssc-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the SSC controller + atmel,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of WM8731 audio codec + +required: + - compatible + - atmel,model + - atmel,audio-routing + - atmel,ssc-controller + - atmel,audio-codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "atmel,at91sam9g20ek-wm8731-audio"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pck0_as_mck>; + atmel,model = "wm8731 @ AT91SAMG20EK"; + atmel,audio-routing = + "Ext Spk", "LHPOUT", + "Int MIC", "MICIN"; + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8731>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt b/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt deleted file mode 100644 index 9c5a9947b64d..000000000000 --- a/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt +++ /dev/null @@ -1,26 +0,0 @@ -* Atmel at91sam9g20ek wm8731 audio complex - -Required properties: - - compatible: "atmel,at91sam9g20ek-wm8731-audio" - - atmel,model: The user-visible name of this sound complex. - - atmel,audio-routing: A list of the connections between audio components. - - atmel,ssc-controller: The phandle of the SSC controller - - atmel,audio-codec: The phandle of the WM8731 audio codec -Optional properties: - - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt - -Example: -sound { - compatible = "atmel,at91sam9g20ek-wm8731-audio"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_pck0_as_mck>; - - atmel,model = "wm8731 @ AT91SAMG20EK"; - - atmel,audio-routing = - "Ext Spk", "LHPOUT", - "Int MIC", "MICIN"; - - atmel,ssc-controller = <&ssc0>; - atmel,audio-codec = <&wm8731>; -}; --- base-commit: 0dd3ee31125508cd67f7e7172247f05b7fd1753a change-id: 20240214-at91sam9g20ek-wm8731-yaml-404ddddbb91a Best regards, -- Balakrishnan Sambath <balakrishnan.s(a)microchip.com>

5 5

[PATCH 00/17] ASoC: remove snd_soc_dai_set_tristate()
by Kuninori Morimoto 10 Dec '24

10 Dec '24

Hi Mark snd_soc_dai_set_tristate() has added in v2.6.27 (2008/Oct) by this commit 8c6529dbf881303920a415c2d14a500218661949 ("ALSA: asoc: core - add Digital Audio Interface (DAI) control functions.") But it has never been used during this 16 years. Let's remove it. Kuninori Morimoto (17): ASoC: pxa: remove snd_soc_dai_set_tristate() ASoC: codecs: cs35l32: remove snd_soc_dai_set_tristate() ASoC: codecs: cs35l33: remove snd_soc_dai_set_tristate() ASoC: codecs: cs35l34: remove snd_soc_dai_set_tristate() ASoC: codecs: cs42l73: remove snd_soc_dai_set_tristate() ASoC: codecs: cs53l30: remove snd_soc_dai_set_tristate() ASoC: codecs: wm8961: remove snd_soc_dai_set_tristate() ASoC: codecs: wm8994: remove snd_soc_dai_set_tristate() ASoC: codecs: wm8995: remove snd_soc_dai_set_tristate() ASoC: codecs: wm9713: remove snd_soc_dai_set_tristate() ASoC: codecs: adau1372: remove snd_soc_dai_set_tristate() ASoC: codecs: adau1977: remove snd_soc_dai_set_tristate() ASoC: codecs: adau7118: remove snd_soc_dai_set_tristate() ASoC: codecs: madera: remove snd_soc_dai_set_tristate() ASoC: codecs: arizona: remove snd_soc_dai_set_tristate() ASoC: codecs: twl4030: remove snd_soc_dai_set_tristate() ASoC: remove snd_soc_dai_set_tristate() include/sound/soc-dai.h | 3 --- sound/soc/codecs/adau1372.c | 14 -------------- sound/soc/codecs/adau1977.c | 15 --------------- sound/soc/codecs/adau7118.c | 19 ------------------- sound/soc/codecs/arizona.c | 17 ----------------- sound/soc/codecs/cs35l32.c | 9 --------- sound/soc/codecs/cs35l33.c | 21 --------------------- sound/soc/codecs/cs35l34.c | 15 --------------- sound/soc/codecs/cs42l73.c | 10 ---------- sound/soc/codecs/cs53l30.c | 10 ---------- sound/soc/codecs/madera.c | 22 ---------------------- sound/soc/codecs/twl4030.c | 28 ---------------------------- sound/soc/codecs/wm8961.c | 14 -------------- sound/soc/codecs/wm8994.c | 28 ---------------------------- sound/soc/codecs/wm8995.c | 33 --------------------------------- sound/soc/codecs/wm9713.c | 17 ----------------- sound/soc/pxa/pxa-ssp.c | 21 --------------------- sound/soc/soc-dai.c | 19 ------------------- 18 files changed, 315 deletions(-) -- 2.43.0

3 20

Re: [vdr] Identifying MPEG-4 HE-AAC (LATM, LAOS) audio formats
by schorpp 09 Dec '24

09 Dec '24

Am 06.12.24 um 21:32 schrieb Marko Mäkelä: > A couple of weeks ago, Finland took the penultimate step to get rid of > DVB-T. Since years, we had DVB-T and DVB-T2 running in parallel. The > DVB-T transponders should be shut down in 2025. same here but h.265 video and aac audio. > > The step was to introduce a second free-to-air DVB-T2 bouquet to carry > some channels that were previously only available with DVB-T and MPEG-2. > The VDR channel search failed to find this transponder; I ran a patched > version of https://github.com/mighty-p/t2scan/pull/15/ to generate a new > channels.conf. (Is this a bug of VDR, or a feature? I remember that when > I originally set up VDR, I had to get a channels.conf from tscan or > t2scan.) have You tried the latest git version of scan in dvb-utils @linuxtx.org repo? > > Unlike our first free-to-air DVB-T2 bouquet, which uses MPEG-4 H.264 > video with AC3 audio, the second one uses H.264 video and HE-AAC audio, > also known as LATM or LOAS. This had to be explicitly enabled for my > output device: https://github.com/reufer/rpihddevice/issues/20 > > The HE-AAC support in rpihddevice is experimental, as it has been in ffmpeg until today and rpihddevive is based on it? > maybe because the > sampling rate is hard-coded as 48 kHz and the number of channels as 2. > Unlike AC3, it does not look like these parameters would be available in > any audio frame header. As far I understood from a NorDig.org > specification, the sampling rate would always be 48 kHz, but the number > of channels could be identified by something in the container, > specifically, Table 12.28 and Table 12.29 at page 164 of > https://nordig.org/wp-content/uploads/2016/03/NorDig-Unified- > Requirements-ver.-3.2.1.pdf Have You tried the latest git ffprobe version, does it detect the aac flavor of Your dvb-t2 DVB channels correctly? Have You tried latest git mediainfo? > > What would have to be changed in VDR or rpihddevice or both to correctly > switch between 2-channel and 5.1-channel HE-AAC audio? This is mostly an > academic question and exercise, because I only have stereo audio > hardware and don't know if any programming is actually available with > multichannel audio. Not at all academic. Since AVRs do not implement Fraunhofer codecs we need a HDNI PCM Multichannel driver for the raspberry pi to playback 5.1 channel aac without transcoding it to ac3 which can be done with mplayer on the fly. I've made a HDMI PCM Multichannel output to AVR driver patch for the Allwinner SoC alsa audio driver, look for it on the alsa-dev list archives. > > Best regards, > > Marko y tom

2 1

[PATCH treewide 00/11] ASoC: Clean up {hp,mic}-det-gpio handling
by Geert Uytterhoeven 09 Dec '24

09 Dec '24

Hi all, The "gpio" suffix for GPIO consumers was deprecated a while ago, in favor of the "gpios" suffix. However, there are still several users of the "hp-det-gpio" and "mic-det-gpio" properties, in DT bindings, drivers, and DT source files. Drivers that use gpiolib handle both the deprecated and new properties transparently, but that is not the case when using of_property_*() calls directly. - The first patch fixes a bug, as the Freescale Generic ASoC Sound driver handles only the deprecated properties, while there is already a user of the new variants, - The second patch updates the Audio Graph and Simple Audio Card DT bindings, - Patches 3-9 converts various DTS files to use the new properties, - The last 2 patches convert Freescale sound device nodes to use the new properties. All patches can be applied independently, except for the last two, which depend on the first patch to avoid regressions. This series was compile-tested only. Thanks for your comments! Geert Uytterhoeven (11): ASoC: fsl-asoc-card: Add missing handling of {hp,mic}-dt-gpios ASoC: dt-bindings: Deprecate {hp,mic}-det-gpio ARM: dts: marvell: mmp2-olpc-xo-1-75: Switch to {hp,mic}-det-gpios arm64: dts: freescale: imx: Switch to simple-audio-card,hp-det-gpios arm64: dts: renesas: beacon-renesom: Switch to mic-det-gpios arm64: dts: rockchip: Switch to simple-audio-card,hp-det-gpios arm64: dts: rockchip: Switch to hp-det-gpios arm64: dts: uniphier: Switch to hp-det-gpios mips: dts: ingenic: Switch to simple-audio-card,hp-det-gpios ARM: dts: nxp: imx: Switch to {hp,mic}-det-gpios arm64: dts: freescale: imx: Switch to hp-det-gpios .../devicetree/bindings/sound/audio-graph.yaml | 6 ++++++ .../devicetree/bindings/sound/simple-card.yaml | 12 ++++++++++++ arch/arm/boot/dts/marvell/mmp2-olpc-xo-1-75.dts | 4 ++-- arch/arm/boot/dts/nxp/imx/imx6qdl-sabresd.dtsi | 4 ++-- arch/arm/boot/dts/nxp/imx/imx6sl-evk.dts | 2 +- arch/arm/boot/dts/nxp/imx/imx6sll-evk.dts | 2 +- arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi | 2 +- arch/arm/boot/dts/nxp/imx/imx6ul-14x14-evk.dtsi | 2 +- arch/arm/boot/dts/nxp/imx/imx7d-sdb.dts | 2 +- .../boot/dts/freescale/imx8mq-librem5-devkit.dts | 2 +- arch/arm64/boot/dts/freescale/imx8mq-librem5.dtsi | 2 +- arch/arm64/boot/dts/freescale/imx8qm-mek.dts | 2 +- arch/arm64/boot/dts/freescale/imx8qxp-mek.dts | 2 +- arch/arm64/boot/dts/freescale/imx95-19x19-evk.dts | 2 +- .../boot/dts/renesas/beacon-renesom-baseboard.dtsi | 2 +- .../boot/dts/rockchip/rk3326-anbernic-rg351m.dtsi | 2 +- .../arm64/boot/dts/rockchip/rk3326-gameforce-chi.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3326-odroid-go.dtsi | 2 +- arch/arm64/boot/dts/rockchip/rk3368-lba3368.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-eaidk-610.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-pinebook-pro.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-roc-pc-plus.dts | 2 +- .../boot/dts/rockchip/rk3399-rock-pi-4a-plus.dts | 2 +- .../boot/dts/rockchip/rk3399-rock-pi-4b-plus.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-rock-pi-4c.dts | 2 +- .../boot/dts/rockchip/rk3399-sapphire-excavator.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg-arc.dtsi | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353p.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353ps.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353v.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353vs.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg503.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3566-pinetab2.dtsi | 2 +- arch/arm64/boot/dts/rockchip/rk3566-powkiddy-x55.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3568-odroid-m1.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-armsom-sige7.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-evb1-v10.dts | 2 +- .../dts/rockchip/rk3588-friendlyelec-cm3588-nas.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-nanopc-t6.dtsi | 2 +- arch/arm64/boot/dts/rockchip/rk3588-ok3588-c.dts | 2 +- .../boot/dts/rockchip/rk3588-orangepi-5-plus.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-quartzpro64.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-rock-5-itx.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-rock-5b.dts | 2 +- .../boot/dts/rockchip/rk3588s-gameforce-ace.dts | 2 +- .../boot/dts/socionext/uniphier-ld11-global.dts | 2 +- .../boot/dts/socionext/uniphier-ld20-global.dts | 2 +- arch/mips/boot/dts/ingenic/gcw0.dts | 2 +- arch/mips/boot/dts/ingenic/rs90.dts | 2 +- sound/soc/fsl/fsl-asoc-card.c | 8 +++++--- 50 files changed, 72 insertions(+), 52 deletions(-) -- 2.34.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

8 30

Audio Crackling with some Steam games
by orbea 06 Dec '24

06 Dec '24

Hi, OS: Gentoo Kernel: 6.12.1 alsa-lib: 1.2.12 Product Name: X670E PG Lightning GPU: [AMD/ATI] Navi 10 [Radeon RX 5600 OEM/5600 XT / 5700/5700 XT] (rev c1) When using HDMI audio I am having an issue where some Steam games as run using Proton will have audio crackling or static after a short period of time after starting the game. The games that I know do this are 'Tales of Symphonia', 'Ys VIII: Lacrimosa of DANA' and 'Metaphor: Refantazio'. While many other games do not have this problem. I am not sure if this is a problem specific to the Steam ecosystem, the games, alsa, the kernel or something else? At least in the case of 'Ys VIII' I can only reproduce the problem when running the game in Steam where the same game files when run using the system wine or self compiled proton there are no audio issues. In the Proton logs it shows many underruns as soon as the issue starts: ALSA lib pcm.c:8545:(snd_pcm_recover) underrun occurred The problem also appears when using a simple /etc/asound.conf that contains only: defaults.pcm.card 0 defaults.pcm.device 9 defaults.ctl.card 0 I have also tried asking about this issue on the Proton Github issue tracker and the FAudio issue tracker. https://github.com/FNA-XNA/FAudio/issues/367 https://github.com/ValveSoftware/Proton/issues/3820#issuecomment-2477846139 https://github.com/ValveSoftware/Proton/issues/822#issuecomment-2326999328 In the FAudio issue it was suggested it was related to protonmediaconverter where the system wine and proton use ffmpeg, but I am not sure how to confirm this. I attached the alsa-info.txt. Is there anything I can do to further diagnose, resolve or workaround this? Any further info I neglected to provide that can help this? Thanks!

1 0

[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 v5 RESEND 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 06 Dec '24

06 Dec '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20241002093904.180979… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

2 7

[GIT PULL] ASoC fixes for v6.13-rc1
by Mark Brown 05 Dec '24

05 Dec '24

The following changes since commit 40384c840ea1944d7c5a392e8975ed088ecf0b37: Linux 6.13-rc1 (2024-12-01 14:28:56 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-rc1 for you to fetch changes up to ec16a3cdf37e507013062f9c4a2067eacdd12b62: ASoC: mediatek: mt8188-mt6359: Remove hardcoded dmic codec (2024-12-05 13:39:58 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A few small fixes for v6.13, all system specific - the biggest thing is the fix for jack handling over suspend on some Intel laptops. ---------------------------------------------------------------- Dan Carpenter (1): ASoC: SOF: ipc3-topology: fix resource leaks in sof_ipc3_widget_setup_comp_dai() Marek Maslanka (1): ASoC: Intel: avs: da7219: Remove suspend_pre() and resume_post() Nícolas F. R. A. Prado (1): ASoC: mediatek: mt8188-mt6359: Remove hardcoded dmic codec sound/soc/intel/avs/boards/da7219.c | 17 ----------------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 4 +--- sound/soc/sof/ipc3-topology.c | 7 ++++--- 3 files changed, 5 insertions(+), 23 deletions(-)

2 1

With SOF enabled in sof-hda-generic.tplg device (no DMIC) the control 'Capture Source' is not controlled by UCM
by GitHub issues - edited 05 Dec '24

05 Dec '24

alsa-project/alsa-ucm-conf issue #484 was edited from singalsu: The configuration is not usual. If there is no DMIC in the system usually the legacy mode is selected (as with `options snd_intel_dspcfg dsp_driver=1`). In such PCs SOF can be enabled with `options snd_intel_dspcfg dsp_driver=3`. When a headset is plugged/unplugged the playback correctly switches between speaker and headphone/headset earbuds. But the microphone capture remains muted. I noticed with legacy mode where internal microphones work that control name='Capture Source' changes to 1 when headset is removed: ``` singalsu@asteri:~$ amixer -c0 cget name='Capture Source' numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Internal Mic' : values=1 singalsu@asteri:~$ amixer -c0 cget name='Capture Source' numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Internal Mic' : values=0 ``` This is not happening when SOF is enabled. But I could make internal microphones capture work with `amixer -c0 cset name='Capture Source' 1`. Is there a way to fix this in ucm2? I'm not absolutely sure about this issue in Fedora 41 but this happens with my ucm2 development code with plain git master branch (commit 0782bf435b8332833e7b3fc91501d78473dd6edf). Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/484 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

With SOF enabled in sof-hda-generic.tplg device (no DMIC) the control 'Capture Source' is not controlled by UCM
by GitHub issues - opened 05 Dec '24

05 Dec '24

alsa-project/alsa-ucm-conf issue #484 was opened from singalsu: The configuration is not usual. If there is no DMIC in the system usually the legacy mode is selected (as with `options snd_intel_dspcfg dsp_driver=1`). In such PCs SOF can be enabled with `options snd_intel_dspcfg dsp_driver=3`. When a headset is plugged/unplugged the playback correctly switches between speaker and headphone/headset earbuds. But the microphone capture remains muted. I noticed with legacy mode where internal microphones work that control name='Capture Source' changes to 1 when headset is removed: ``` singalsu@asteri:~$ amixer -c0 cget name='Capture Source' numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Internal Mic' : values=1 singalsu@asteri:~$ amixer -c0 cget name='Capture Source' numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Internal Mic' : values=0 ``` This is not happening when SOF is enabled. But I could make internal microphones capture work with `amixer -c0 cset name='Capture Source' 1`. Is there a way to fix this in ucm2? I'm not absolutely sure about this issue in Fedora 41 but this happens with my modified ucm2 development code with git master branch. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/484 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

scsi: aic7xxx: Fix build 'aicasm' warning
by wangdicheng 05 Dec '24

05 Dec '24

1 0

aplay doesn't play audio sometimes "unable to open slave"
by GitHub issues - opened 05 Dec '24

05 Dec '24

alsa-project/alsa-lib issue #426 was opened from haoyi-codes: I have initially thought it was a bug that was related to chromium as stated in #425. However, now I realize it has to be something else that is wrong. `aplay -D default /dev/zero` ``` ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave aplay: main:850: audio open error: No such file or directory ``` `aplay -D plughw:0 /dev/zero` ``` aplay: main:850: audio open error: No such file or directory ``` `aplay -l` ``` **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 3: Generic Digital [Generic Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: Generic Digital [Generic Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Audio [SA9123 USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `cat /proc/asound/cards` ``` 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0x81080000 irq 17 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0x81220000 irq 165 2 [Audio ]: USB-Audio - SA9123 USB Audio SYC SA9123 USB Audio at usb-0000:00:14.0-3.2, full speed 3 [Edition ]: USB-Audio - Logi 4K Stream Edition Logi 4K Stream Edition at usb-0000:00:14.0-3.3, super speed ``` `ls -la /dev/snd/*` ``` crw-rw----+ 1 root audio 116, 0 Dec 4 23:18 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 32 Dec 4 23:18 /dev/snd/controlC1 crw-rw----+ 1 root audio 116, 64 Dec 4 23:18 /dev/snd/controlC2 crw-rw----+ 1 root audio 116, 96 Dec 4 23:18 /dev/snd/controlC3 crw-rw----+ 1 root audio 116, 19 Dec 4 23:18 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 51 Dec 4 23:18 /dev/snd/pcmC1D3p crw-rw----+ 1 root audio 116, 80 Dec 4 23:18 /dev/snd/pcmC2D0p crw-rw----+ 1 root audio 116, 120 Dec 4 23:18 /dev/snd/pcmC3D0c crw-rw----+ 1 root audio 116, 1 Dec 4 23:18 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Dec 4 23:18 /dev/snd/timer /dev/snd/by-id: total 0 drwxr-xr-x 2 root root 80 Dec 4 23:18 . drwxr-xr-x 4 root root 280 Dec 4 23:17 .. lrwxrwxrwx 1 root root 12 Dec 4 23:18 usb-046d_Logi_4K_Stream_Edition_51C07EAF-03 -> ../controlC3 lrwxrwxrwx 1 root root 12 Dec 4 23:18 usb-SYC_SA9123_USB_Audio-01 -> ../controlC2 /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 160 Dec 4 23:18 . drwxr-xr-x 4 root root 280 Dec 4 23:17 .. lrwxrwxrwx 1 root root 12 Dec 4 23:18 pci-0000:00:14.0-usb-0:3.2:1.1 -> ../controlC2 lrwxrwxrwx 1 root root 12 Dec 4 23:18 pci-0000:00:14.0-usb-0:3.3:1.3 -> ../controlC3 lrwxrwxrwx 1 root root 12 Dec 4 23:18 pci-0000:00:14.0-usbv2-0:3.2:1.1 -> ../controlC2 lrwxrwxrwx 1 root root 12 Dec 4 23:18 pci-0000:00:14.0-usbv3-0:3.3:1.3 -> ../controlC3 lrwxrwxrwx 1 root root 12 Dec 4 23:18 pci-0000:01:00.1 -> ../controlC0 lrwxrwxrwx 1 root root 12 Dec 4 23:18 pci-0000:05:00.1 -> ../controlC1 ``` For reference I am still able to play audio on my system as pulseaudio is running but for applications that use alsa directly it does not work. ``` Server String: /run/user/1011/pulse/native Library Protocol Version: 35 Server Protocol Version: 35 Is Local: yes Client Index: 59 Tile Size: 65472 User Name: user Host Name: system Server Name: PulseAudio (on PipeWire 1.2.7) Server Version: 15.0.0 Default Sample Specification: float32le 2ch 48000Hz Default Channel Map: front-left,front-right Default Sink: alsa_output.usb-SYC_SA9123_USB_Audio-01.iec958-stereo Default Source: alsa_output.usb-SYC_SA9123_USB_Audio-01.iec958-stereo.monitor Cookie: XXXX:XXXX ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/426 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

pulseaudio bails out on 1.2.13 (SIGSEGV), with MOTU Ultralite mk5
by GitHub issues - opened 04 Dec '24

04 Dec '24

alsa-project/alsa-ucm-conf issue #483 was opened from karolyi: Hey, I have this sound card and with the latest update (1.2.13), pulseaudio started to SIGSEGV and I have no sound. Upon going to the pulseaudio IRC channel and talking to the devs, we've managed to point out that it's the UCM configuration that makes pulseaudio crash. Reverting to 1.2.12 made the crash cease. It seems that the newly added configuration for the Ultralite makes pulseaudio crash. According to the below gdb traces, it seems the configuration drives pulseaudio into an infinite loop. Beginning of the trace: ``` Debuginfod has been enabled. To make this setting permanent, add 'set debuginfod enabled on' to .gdbinit. [Thread debugging using libthread_db enabled] Using host libthread_db library "/usr/lib/libthread_db.so.1". Program received signal SIGSEGV, Segmentation fault. 0x00007ffff7ae98df in _int_malloc (av=av@entry=0x7ffff7c2dac0 <main_arena>, bytes=bytes@entry=2080) at malloc.c:4059 4059 size_t tc_idx = csize2tidx (nb); Thread 1 (Thread 0x7ffff73298c0 (LWP 40542) "pulseaudio"): #0 0x00007ffff7ae98df in _int_malloc (av=av@entry=0x7ffff7c2dac0 <main_arena>, bytes=bytes@entry=2080) at malloc.c:4059 #1 0x00007ffff7aec069 in __libc_calloc (n=<optimized out>, elem_size=<optimized out>) at malloc.c:3754 #2 0x00007ffff7e5e366 in pa_xmalloc0 (size=2080) at ../src/pulse/xmalloc.c:74 #3 0x00007ffff7e7d3d4 in pa_idxset_new (hash_func=0x7ffff7e7d371 <pa_idxset_trivial_hash_func>, compare_func=0x7ffff7e7d37f <pa_idxset_trivial_compare_func>) at ../src/pulsecore/idxset.c:86 #4 0x00007ffff7e7f5a9 in pa_idxset_copy (s=0x5555556f5890, copy_func=0x0) at ../src/pulsecore/idxset.c:623 #5 0x00007ffff6b93ff0 in iterate_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1399 #6 0x00007ffff6b94108 in iterate_maximal_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1412 #7 0x00007ffff6b9418a in iterate_maximal_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1422 #8 0x00007ffff6b9418a in iterate_maximal_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1422 ``` Inbetween there's just a lot of recursive calls. End of the trace: ``` #130916 0x00007ffff6b9418a in iterate_maximal_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1422 #130917 0x00007ffff6b9418a in iterate_maximal_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1422 #130918 0x00007ffff6b9418a in iterate_maximal_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1422 #130919 0x00007ffff6b9418a in iterate_maximal_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1422 #130920 0x00007ffff6b96635 in ucm_create_verb_profiles (ucm=0x55555567bc60, ps=0x5555556ca560, verb=0x5555556c8970, verb_name=0x5555556c6d60 "HiFi", verb_desc=0x5555556a3120 "Default") at ../src/modules/alsa/alsa-ucm.c:2042 #130921 0x00007ffff6b973bf in pa_alsa_ucm_add_profile_set (ucm=0x55555567bc60, default_channel_map=0x555555570de0) at ../src/modules/alsa/alsa-ucm.c:2305 #130922 0x00007ffff6c6bb43 in module_alsa_card_LTX_pa__init (m=0x55555566d9d0) at ../src/modules/alsa/module-alsa-card.c:966 #130923 0x00007ffff7f26919 in pa_module_load (module=0x7fffffffccf0, c=0x555555570d20, name=0x7ffff6c7747d "module-alsa-card", argument=0x555555587f10 "device_id=\"2\" name=\"usb-MOTU_UltraLite-mk5_UL5LFF34E6-00\" card_name=\"alsa_card.usb-MOTU_UltraLite-mk5_UL5LFF34E6-00\" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=y"...) at ../src/pulsecore/module.c:208 #130924 0x00007ffff6c74260 in verify_access (u=0x55555566ae40, d=0x55555566e430) at ../src/modules/module-udev-detect.c:342 #130925 0x00007ffff6c7488f in card_changed (u=0x55555566ae40, dev=0x55555568ae50) at ../src/modules/module-udev-detect.c:437 #130926 0x00007ffff6c74c6f in process_device (u=0x55555566ae40, dev=0x55555568ae50) at ../src/modules/module-udev-detect.c:479 #130927 0x00007ffff6c74cf8 in process_path (u=0x55555566ae40, path=0x5555556879b0 "/sys/devices/pci0000:00/0000:00:01.2/0000:02:00.0/0000:03:08.0/0000:07:00.1/usb1/1-1/1-1.1/1-1.1:1.0/sound/card2") at ../src/modules/module-udev-detect.c:496 #130928 0x00007ffff6c75ed0 in module_udev_detect_LTX_pa__init (m=0x555555666220) at ../src/modules/module-udev-detect.c:812 #130929 0x00007ffff7f26919 in pa_module_load (module=0x7fffffffcf28, c=0x555555570d20, name=0x555555668c20 "module-udev-detect", argument=0x0) at ../src/pulsecore/module.c:208 #130930 0x00007ffff7f073ba in pa_cli_command_load (c=0x555555570d20, t=0x555555668c40, buf=0x5555555717b0, fail=0x55555556f6b5) at ../src/pulsecore/cli-command.c:440 #130931 0x00007ffff7f0fac4 in pa_cli_command_execute_line_stateful (c=0x555555570d20, s=0x7fffffffd100 "load-module module-udev-detect", buf=0x5555555717b0, fail=0x55555556f6b5, ifstate=0x7fffffffd0f8) at ../src/pulsecore/cli-command.c:2207 #130932 0x00007ffff7f0fda2 in pa_cli_command_execute_file_stream (c=0x555555570d20, f=0x55555556fce0, buf=0x5555555717b0, fail=0x55555556f6b5) at ../src/pulsecore/cli-command.c:2247 #130933 0x00007ffff7f10031 in pa_cli_command_execute_file (c=0x555555570d20, fn=0x555555570c70 "/home/karolyi/Work/opensource/pulseaudio/build/src/daemon/default.pa", buf=0x5555555717b0, fail=0x55555556f6b5) at ../src/pulsecore/cli-command.c:2278 #130934 0x00007ffff7f0f7d8 in pa_cli_command_execute_line_stateful (c=0x555555570d20, s=0x555555570560 ".include /home/karolyi/Work/opensource/pulseaudio/build/src/daemon/default.pa", buf=0x5555555717b0, fail=0x55555556f6b5, ifstate=0x7fffffffdb1c) at ../src/pulsecore/cli-command.c:2170 #130935 0x00007ffff7f10239 in pa_cli_command_execute (c=0x555555570d20, s=0x555555577bc0 ".include /home/karolyi/Work/opensource/pulseaudio/build/src/daemon/default.pa\n", buf=0x5555555717b0, fail=0x55555556f6b5) at ../src/pulsecore/cli-command.c:2304 #130936 0x0000555555564162 in main (argc=7, argv=0x7fffffffde98) at ../src/daemon/main.c:1267 Continuing. Program terminated with signal SIGSEGV, Segmentation fault. The program no longer exists. ``` Requested debug output from the pulseaudio project: https://termbin.com/n04r Please investigate. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/483 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Should SOF ucm2/blobs/sof user_configs and product_configs be without .conf suffix?
by GitHub issues - edited 04 Dec '24

04 Dec '24

alsa-project/alsa-ucm-conf issue #482 was edited from singalsu: It seems updates have dropped .conf from looked up SOF blobs customization files. Is the change intentional? Then the example in ucm2/blobs/sof/product_configs/AAEON/UPX-TGL01.conf should be renamed to be without .conf to work and to be applied correctly. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/482 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Should SOF ucm2/blobs/sof user_configs and product_configs be without .conf suffix?
by GitHub issues - opened 04 Dec '24

04 Dec '24

alsa-project/alsa-ucm-conf issue #482 was opened from singalsu: It seems updates have dropped .conf from looked up SOF blobs customization files. Is the change intentional? Then the example in ucm2/blobs/sof/product_configs/AAEON/UPX-TGL01.conf should be renamed to be without .conf to work and be applied correctly. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/482 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

axfer-transfer.1: Some remarks and editorial changes for this man page
by Bjarni Ingi Gislason 04 Dec '24

04 Dec '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>':1006 troff:<stdin>:1006: 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

[PATCH 1/2] ASoC: amd: ps: update mach params subsystem_rev variable
by Vijendar Mukunda 03 Dec '24

03 Dec '24

Update mach_params subsystem_rev with acp_rev variable for ACP6.3 platform. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index aef73ec6f7ef..4ed50cebadb3 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -312,6 +312,7 @@ static struct snd_soc_acpi_mach *acp63_sdw_machine_select(struct device *dev) if (mach && mach->link_mask) { mach->mach_params.links = mach->links; mach->mach_params.link_mask = mach->link_mask; + mach->mach_params.subsystem_rev = acp_data->acp_rev; return mach; } } -- 2.34.1

2 2

UCM2: Intel: sof-hda-dsp: Enable Dmic0 DRC and TDFB pre-processing
by GitHub pull_request - opened 03 Dec '24

03 Dec '24

alsa-project/alsa-ucm-conf pull request #481 was opened from singalsu: This patch adds enable of the microphone capture pre-processing features as BootSequence inital values if the controls for them are available. If the user later switches these off, they will remain off. The DRC is dynamic range control that boosts the capture level, and the TDFB (time-domain fixed beamformer) improves capture signa-to-noise ratio for set beam angle direction. The used 0 degrees angle is defined as angle towards notebook computer user. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/481 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/481.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: native DSD on PCI linux driver
by Takashi Iwai 03 Dec '24

03 Dec '24

On Tue, 26 Nov 2024 10:41:21 +0100, Filip Van Lunteren wrote: > > > Hello, > > I am currently developing PCI audio output hardware and driver and want to > integrate native DSD playback. I am struggling to find documentation on this. > Is there DSD native support in ALSA? And how do I incorporate it? PCM is > already set up and works. AFAIK, in the driver side, there is not much special; it assumes that DSD samples are passed over the standard PCM stream, and the driver uses special formats SNDRV_PCM_FORMAT_DSD_U8, *_U16 and *_U32 (with _LE and _BE suffix) for 8, 16 and 32bit, instead of the standard SNDRV_PCM_FORMAT_U8, *_S16, etc. For those formats, the silence data is assumed to be 0x69, 0x6969, etc. Takashi

1 1

[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

alsa-lib-1.2.13 prevents chromium-131.0.6778.85 from playing audio on gentoo.
by GitHub issues - edited 02 Dec '24

02 Dec '24

alsa-project/alsa-lib issue #425 was edited from haoyi-codes: I have recently compiled chromium against alsa-lib-1.2.13 for two different systems. However, I am unable to play audio. The terminal output shows: System 1: ``` Failed to create secure directory (/run/user/1011//pulse): Permission denied ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194540.301739:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194540.302803:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory libpng warning: iCCP: known incorrect sRGB profile Warning: disabling flag --expose_wasm due to conflicting flags Warning: disabling flag --expose_wasm due to conflicting flags ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194614.580896:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194614.582080:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory libpng warning: iCCP: known incorrect sRGB profile ``` System 2: ``` socket(): Operation not permitted ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230412.353283:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230412.353697:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration libpng warning: iCCP: known incorrect sRGB profile libpng warning: iCCP: known incorrect sRGB profile ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230432.095492:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230432.095901:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory libpng warning: iCCP: known incorrect sRGB profile ``` After downgrading alsa-lib to 1.2.12 and recompiling chromium I was able to play audio just fine. Further information can also be found at: https://bugs.gentoo.org/945749 Issue URL : https://github.com/alsa-project/alsa-lib/issues/425 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsa-lib-1.2.13 prevents chromium-131.0.6778.85 from playing audio on gentoo.
by GitHub issues - opened 02 Dec '24

02 Dec '24

alsa-project/alsa-lib issue #425 was opened from haoyi-codes: I have recently compiled chromium against alsa-lib-1.2.13 for two different systems, however I am unable to play audio. The terminal output shows: System 1: ``` Failed to create secure directory (/run/user/1011//pulse): Permission denied ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194540.301739:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194540.302803:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory libpng warning: iCCP: known incorrect sRGB profile Warning: disabling flag --expose_wasm due to conflicting flags Warning: disabling flag --expose_wasm due to conflicting flags ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194614.580896:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194614.582080:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory libpng warning: iCCP: known incorrect sRGB profile ``` System 2: ``` socket(): Operation not permitted ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230412.353283:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230412.353697:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration libpng warning: iCCP: known incorrect sRGB profile libpng warning: iCCP: known incorrect sRGB profile ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230432.095492:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230432.095901:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory libpng warning: iCCP: known incorrect sRGB profile ``` After downgrading alsa-lib to 1.2.12 and recompiling chromium I was able to play audio just fine. Further information can also be found at: https://bugs.gentoo.org/945749 Issue URL : https://github.com/alsa-project/alsa-lib/issues/425 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[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

No microphone USB audio Behringer UMC204HD ALSA update
by GitHub issues - edited 02 Dec '24

02 Dec '24

alsa-project/alsa-ucm-conf issue #479 was edited from emiltoacs: No input on my Behringer UMC204HD because of an update in the following packages on Arch Linux alsa-card-profiles,alsa-lib,alsa-ucm-conf,alsa-utils,lib32-alsa-lib alsa-card-profiles 1:1.2.6-1 -> 1:1.2.7-1 alsa-lib 1.2.12-1 -> 1.2.13-1 alsa-ucm-conf 1.2.12-1 -> 1.2.13-2 alsa-utils 1.2.12-1 -> 1.2.13-2 lib32-alsa-lib 1.2.12-1 -> 1.2.13-1 The error messages related by pipewire : ``` pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (0 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:6/6 s:0 a:0 f:0 waiting:0 process:0 status:triggered pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (4 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:1/1 s:2663559587040 a:2663559606728 f:2663559630434 waiting:19688 process:23706 status:triggered pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (0 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:1/1 s:5553029600968 a:5553029610813 f:5553029623757 waiting:9845 process:12944 status:triggered ``` If I revert package the package to these version it works again : alsa-card-profiles-1:1.2.6-1-x86_64.pkg.tar.zst alsa-lib-1.2.12-1-x86_64.pkg.tar.zst lib32-alsa-lib-1.2.12-1-x86_64.pkg.tar.zst alsa-utils-1.2.12-1-x86_64.pkg.tar.zst alsa-ucm-conf-1.2.12-1-any.pkg.tar.zst In the meantime I add : --ignore alsa-card-profiles,alsa-lib,alsa-ucm-conf,alsa-utils,lib32-alsa-lib to my pacman update. **EDIT** only the alsa-ucm-conf package is faulty somewhere between these two version alsa-ucm-conf 1.2.12-1 -> 1.2.13-2 Workaround: Downgrade to alsa-ucm-conf-1.2.12-1 With this package built by the maintainer @christian-heusel for testing https://pkgbuild.com/\~gromit/alsa-ucm-conf-v1.2.13.r6.g62acef2-1-any.pkg.t… the microphone seens to work but in the logs I have the same error message as above : ``` pipewire[2522]: pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-56) graph xrun not-triggered (0 suppressed) pipewire[2522]: pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-56) xrun state:0x7e7d78f97008 pending:6/6 s:69949980861 a:69950023367 f:69950232612 waiting:42506 process:209245 status:triggered ``` although I manage to receive the signal of my microphone. See https://bbs.archlinux.org/viewtopic.php?pid=2211786 for the details of our discussion with @christian-heusel Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/479 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Unable to request ucm2 to load BootSequence controls defaults
by GitHub issues - opened 02 Dec '24

02 Dec '24

alsa-project/alsa-ucm-conf issue #480 was opened from singalsu: I'm stuck with my ucm2 patch to set by default control switches for SOF DRC and beamformer "on" when the controls appear for first time. I don't seem to be able to reset the controls for the SOF sound card to defaults. There's a similar feature in https://github.com/alsa-project/alsa-ucm-conf/blob/62acef2efffdcb75a35f90d2… for "Dmic0 Capture Volume". But I haven't been able to see the volume to go to 70% in my tries to reset the configuration. In Fedora 41 I've tried to remove /var/lib/alsa/asound.state and rename /usr/sbin/alsactl, reboot, and rename it back. But no success. The volume control remains in 100%. Similarly my tries to get DRC and TDFB enabled fail. What is the right trick needed to trigger the BootSequence? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/480 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[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

No microphone USB audio Behringer UMC204HD ALSA update
by GitHub issues - edited 02 Dec '24

02 Dec '24

alsa-project/alsa-ucm-conf issue #479 was edited from emiltoacs: No input on my Behringer UMC204HD because of an update in the following packages on Arch Linux alsa-card-profiles,alsa-lib,alsa-ucm-conf,alsa-utils,lib32-alsa-lib alsa-card-profiles 1:1.2.6-1 -> 1:1.2.7-1 alsa-lib 1.2.12-1 -> 1.2.13-1 alsa-ucm-conf 1.2.12-1 -> 1.2.13-2 alsa-utils 1.2.12-1 -> 1.2.13-2 lib32-alsa-lib 1.2.12-1 -> 1.2.13-1 The error messages related by pipewire : pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (0 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:6/6 s:0 a:0 f:0 waiting:0 process:0 status:triggered pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (4 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:1/1 s:2663559587040 a:2663559606728 f:2663559630434 waiting:19688 process:23706 status:triggered pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (0 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:1/1 s:5553029600968 a:5553029610813 f:5553029623757 waiting:9845 process:12944 status:triggered If I revert package the package to these version it works again : alsa-card-profiles-1:1.2.6-1-x86_64.pkg.tar.zst alsa-lib-1.2.12-1-x86_64.pkg.tar.zst lib32-alsa-lib-1.2.12-1-x86_64.pkg.tar.zst alsa-utils-1.2.12-1-x86_64.pkg.tar.zst alsa-ucm-conf-1.2.12-1-any.pkg.tar.zst In the meantime I add : --ignore alsa-card-profiles,alsa-lib,alsa-ucm-conf,alsa-utils,lib32-alsa-lib to my pacman update. **EDIT** only the alsa-ucm-conf package is faulty somewhere between these two version alsa-ucm-conf 1.2.12-1 -> 1.2.13-2 Workaround: Downgrade to alsa-ucm-conf-1.2.12-1 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/479 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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No microphone USB audio Behringer UMC204HD ALSA update
by GitHub issues - opened 02 Dec '24

02 Dec '24

alsa-project/alsa-ucm-conf issue #479 was opened from emiltoacs: No input on my Behringer UMC204HD because of an update in the following packages on Arch Linux alsa-card-profiles,alsa-lib,alsa-ucm-conf,alsa-utils,lib32-alsa-lib alsa-card-profiles 1:1.2.6-1 -> 1:1.2.7-1 alsa-lib 1.2.12-1 -> 1.2.13-1 alsa-ucm-conf 1.2.12-1 -> 1.2.13-2 alsa-utils 1.2.12-1 -> 1.2.13-2 lib32-alsa-lib 1.2.12-1 -> 1.2.13-1 The error messages related by pipewire : pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (0 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:6/6 s:0 a:0 f:0 waiting:0 process:0 status:triggered pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (4 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:1/1 s:2663559587040 a:2663559606728 f:2663559630434 waiting:19688 process:23706 status:triggered pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (0 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:1/1 s:5553029600968 a:5553029610813 f:5553029623757 waiting:9845 process:12944 status:triggered If I revert package the package to these version it works again : alsa-card-profiles-1:1.2.6-1-x86_64.pkg.tar.zst alsa-lib-1.2.12-1-x86_64.pkg.tar.zst lib32-alsa-lib-1.2.12-1-x86_64.pkg.tar.zst alsa-utils-1.2.12-1-x86_64.pkg.tar.zst alsa-ucm-conf-1.2.12-1-any.pkg.tar.zst In the meantime I add : --ignore alsa-card-profiles,alsa-lib,alsa-ucm-conf,alsa-utils,lib32-alsa-lib to my pacman update. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/479 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[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

snd_seq_open and pulse audio??
by Mark Lentczner 01 Dec '24

01 Dec '24

My ALSA Seq port manager runs as a service. The service is installed under its own service user, which has the home directory set to some place non-existent and non-writable. When my service starts I see this surprising log message: *amidiminder[136345]: Failed to create secure directory (/var/lib/misc/.config/pulse): No such file or directory* A little investigation reveals that this error is generated by this call, opening the ALSA Seq: serr = snd_seq_open(&seq, "default", SND_SEQ_OPEN_DUPLEX, 0); It's a little surprising that opening ALSA Seq would somehow invoke something in pulse audio, and further that pulse audio would then be trying to create its user directory (.configure/pulse is where pulse audio sticks things under each user's home directory). This seems to be a reprise of this Debian bug from 2013 involving alsa-utils and pulse audio: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=712980 Which was close via this patch to alsa-utils: alsactl: Make homedir configurable <https://mailman.alsa-project.org/hyperkitty/list/alsa-devel@alsa-project.or…> The fix there seems to be adding some mechanism to change the $HOME environment variable to point at /var/run/alsa. I agree with Takashi's reply to that fix: Overriding $HOME sounds scary to me, too! Is there any insight into this... and did that fix stick or was some other method found to deal with this. In my case, I don't understand why just opening the ALSA Seq is causing any pulse audio code to run... but perhaps things are more entwined at a lower level of ALSA lib than I understand. - Mark

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

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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

amd-soundwire: add support for AMD generic legacy machine driver
by GitHub pull_request - opened 29 Nov '24

29 Nov '24

alsa-project/alsa-ucm-conf pull request #478 was opened from perexg: The card components string is similar to Intel's SOF driver like, so we can reuse sof-soundwire configuration and it is logical to reduce future maintenance, because the chips connected to soundwire bus are similar for multiple PCI bridges. Example of the components string: Components : ' cfg-amp:1 hs:rt722 spk:rt722 mic:rt722' Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/478 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/478.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC fixes for v6.13-merge-window
by Mark Brown 28 Nov '24

28 Nov '24

The following changes since commit 82ff5abc2edcfba0c0f1a1be807795e2876f46e9: ASoC: hdmi-codec: reorder channel allocation list (2024-11-15 13:43:00 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-merge-window for you to fetch changes up to b682aa788e5f9f1ddacdfbb453e49fd3f4e83721: ASoC: amd: yc: Add a quirk for microfone on Lenovo ThinkPad P14s Gen 5 21MES00B00 (2024-11-27 13:45:45 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A pile of driver specific quirks and fixes that came in since the merge window. One of the AMD fixes is a bit broken for some systems, I'm expecting an incremental change to fix that but it seems better overall to merge the rest of the fixes. There's also one small documentation update that seemed sensible to apply now, pointing to the dapm-graph tool. ---------------------------------------------------------------- Alex Far (1): ASoC: amd: yc: fix internal mic on Redmi G 2022 Bard Liao (1): ASoC: SOF: ipc3-topology: Convert the topology pin index to ALH dai index Charles Han (1): ASoC: imx-audmix: Add NULL check in imx_audmix_probe Ilya Zverev (1): ASoC: amd: yc: Add a quirk for microfone on Lenovo ThinkPad P14s Gen 5 21MES00B00 Mario Limonciello (1): ASoC: amd: Fix build dependencies for `SND_SOC_AMD_PS` Nícolas F. R. A. Prado (1): ASoC: mediatek: Check num_codecs is not zero to avoid panic during probe Uwe Kleine-König (1): ASoC: amd: yc: Add quirk for microphone on Lenovo Thinkpad T14s Gen 6 21M1CTO1WW Venkata Prasad Potturu (1): ASoC: amd: yc: Fix for enabling DMIC on acp6x via _DSD entry anish kumar (1): ASoC: doc: dapm: Add location information for dapm-graph tool liujing (1): ASoC: apple: Fix the wrong format specifier Documentation/sound/soc/dapm.rst | 3 ++ sound/soc/amd/Kconfig | 1 + sound/soc/amd/yc/acp6x-mach.c | 39 +++++++++++++++++++++- sound/soc/apple/mca.c | 2 +- sound/soc/fsl/imx-audmix.c | 3 ++ sound/soc/mediatek/mt8188/mt8188-mt6359.c | 9 +++-- .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 4 +-- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 9 +++-- sound/soc/sof/ipc3-topology.c | 26 +++++++++++++-- 9 files changed, 86 insertions(+), 10 deletions(-)

2 1

aplay.1: Some remarks and editorial changes for this man page
by Bjarni Ingi Gislason 27 Nov '24

27 Nov '24

Package: alsa-utils Version: 1.2.12-1 (Debian testing) 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>':3 troff:<stdin>:3: warning: trailing space in the line troff: backtrace: file '<stdin>':87 troff:<stdin>:87: warning: trailing space in the line troff: backtrace: file '<stdin>':91 troff:<stdin>:91: warning: trailing space in the line troff: backtrace: file '<stdin>':95 troff:<stdin>:95: warning: trailing space in the line troff: backtrace: file '<stdin>':99 troff:<stdin>:99: warning: trailing space in the line troff: backtrace: file '<stdin>':103 troff:<stdin>:103: warning: trailing space in the line troff: backtrace: file '<stdin>':107 troff:<stdin>:107: warning: trailing space in the line troff: backtrace: file '<stdin>':111 troff:<stdin>:111: warning: trailing space in the line troff: backtrace: file '<stdin>':114 troff:<stdin>:114: warning: trailing space in the line troff: backtrace: file '<stdin>':115 troff:<stdin>:115: warning: trailing space in the line troff: backtrace: file '<stdin>':118 troff:<stdin>:118: warning: trailing space in the line troff: backtrace: file '<stdin>':121 troff:<stdin>:121: warning: trailing space in the line troff: backtrace: file '<stdin>':131 troff:<stdin>:131: warning: trailing space in the line troff: backtrace: file '<stdin>':215 troff:<stdin>:215: warning: trailing space in the line troff: backtrace: file '<stdin>':225 troff:<stdin>:225: warning: trailing space in the line troff: backtrace: file '<stdin>':258 troff:<stdin>:258: 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 6.11.9-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

[PATCH] ASoC: amd: yc: Fix for enabling DMIC on acp6x via _DSD entry
by Venkata Prasad Potturu 27 Nov '24

27 Nov '24

Add condition check to register ACP PDM sound card by reading _WOV acpi entry. Fixes: 5426f506b584 ("ASoC: amd: Add support for enabling DMIC on acp6x via _DSD") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/yc/acp6x-mach.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 438865d5e376..c301dac8b979 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -523,8 +523,14 @@ static int acp6x_probe(struct platform_device *pdev) struct acp6x_pdm *machine = NULL; struct snd_soc_card *card; struct acpi_device *adev; + acpi_handle handle; + acpi_integer dmic_status; int ret; + bool is_dmic_enable, wov_en; + /* IF WOV entry not found, enable dmic based on AcpDmicConnected entry*/ + is_dmic_enable = false; + wov_en = true; /* check the parent device's firmware node has _DSD or not */ adev = ACPI_COMPANION(pdev->dev.parent); if (adev) { @@ -532,9 +538,19 @@ static int acp6x_probe(struct platform_device *pdev) if (!acpi_dev_get_property(adev, "AcpDmicConnected", ACPI_TYPE_INTEGER, &obj) && obj->integer.value == 1) - platform_set_drvdata(pdev, &acp6x_card); + is_dmic_enable = true; } + handle = ACPI_HANDLE(pdev->dev.parent); + ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); + if (!ACPI_FAILURE(ret)) + wov_en = dmic_status; + + if (is_dmic_enable && wov_en) + platform_set_drvdata(pdev, &acp6x_card); + else + return 0; + /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); if (dmi_id) -- 2.39.2

4 5

[BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR, partly fixed but HDMI AC3 passthrough
by schorpp 27 Nov '24

27 Nov '24

>>> >>> [BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR >>> >>> ALSA versions 1.0.28 >>> >>> Kernel Driver versions 3.12.40 > Partly FIXED in Longterm Stable Kernel 4.19.324 and using HDMI DEV 1 instead of 0. > > 2 CH PCM HDMI to AVR works now but AC3/DTS passthrough still not or is not recognized by AVR or some wrong AES setting but iec958 ALC886 optical toslink AC3/DTS output to AVR works. > > New BUG: No more 5.1 Analog Output on the CM106 USB Box if AC3 stream is selected. No. Sorry, VDR xineliboutput config error, needs speaker setup 5.1 instead of pass through. y tom > > Your ALSA information is located at http://alsa-project.org/db/?f=5ef8b56450583ba8afbcc705afc02228da4a3072 > > y > tom > >>> >>> Hardware: >>> >>> DFI Mainboard MB330-CRM >>> >>> https://www.dfi.com/de/product/index/273#specification >>> >>> Reproducible Always >>> >>> Steps to reproduce and full report: >>> >>> https://www-vdr--portal-de.translate.goog/forum/index.php?thread/136519-gel… >> Output of alsa-info.sh is here >> >> http://alsa-project.org/db/?f=fc4734deddd9a92331692bf94b15cf84b7722cd4 >> >>> >>> Software crosscheck with CM106 USB Soundbox iec958: works OK. >>> >>> Where to start in the drivers source code to fix it? >>> >>> y >>> tom >>> >>>

1 0

alsa-ucm-conf update from 1.2.12 to 1.2.13 causes HDMI/DsiplayPort outputs to disappear
by GitHub issues - opened 27 Nov '24

27 Nov '24

alsa-project/alsa-ucm-conf issue #477 was opened from mwsltn: Basic Info: Linux Distro: Arch Kernel Version: 6.12.1 I have very reliably been using a USB-C connected external monitor that has internal speakers represented by the output device: HDMI / DisplayPort 1 After running a package update this AM all the HDMI/DisplayPort outputs disappeared. First I downgraded the kernel which didn't help, then checked the pacman logs to see what else had changed and the list included the following: ``` [2024-11-26T05:07:01-0600] [ALPM] upgraded alsa-ucm-conf (1.2.12-1 -> 1.2.13-1) [2024-11-26T05:07:01-0600] [ALPM] upgraded alsa-lib (1.2.12-1 -> 1.2.13-1) [2024-11-26T05:07:01-0600] [ALPM] upgraded alsa-utils (1.2.12-1 -> 1.2.13-1) ``` So I downgrade: ``` [2024-11-26T06:51:22-0600] [PACMAN] Running 'pacman -U file:///var/cache/pacman/pkg/alsa-ucm-conf-1.2.12-1-any.pkg.tar.zst file:///var/cache/pacman/pkg/alsa-lib-1.2.12-1-x86_64.pkg.tar.zst file:///var/cache/pacman/pkg/alsa-utils-1.2.12-1-x86_64.pkg.tar.zst' [2024-11-26T06:51:24-0600] [ALPM] downgraded alsa-ucm-conf (1.2.13-1 -> 1.2.12-1) [2024-11-26T06:51:24-0600] [ALPM] downgraded alsa-lib (1.2.13-1 -> 1.2.12-1) [2024-11-26T06:51:24-0600] [ALPM] downgraded alsa-utils (1.2.13-1 -> 1.2.12-1) ``` Problem fixed. Please let me know if I can provide any other info. Originally I had opened an issue on the `alsa-lib` project and others are reporting the same: https://github.com/alsa-project/alsa-lib/issues/423 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/477 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

alsactl.1: Some remarks and editorial changes for this man page
by Bjarni Ingi Gislason 27 Nov '24

27 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>':93 troff:<stdin>:93: warning: trailing space in the line troff: backtrace: file '<stdin>':224 troff:<stdin>:224: 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 6.11.9-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

alsabat.1: Some remarks and editorial changes for this man page
by Bjarni Ingi Gislason 27 Nov '24

27 Nov '24

Package: alsa-utils Version: 1.2.12-1 (Debian testing) 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>':175 troff:<stdin>:175: 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 6.11.9-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

RE: Set pcbeep to default value
by Kailang 26 Nov '24

26 Nov '24

Sorry! Attach again. Add ALSA: hda/realtek: to Title. > -----Original Message----- > From: Kailang > Sent: Tuesday, November 26, 2024 10:36 AM > To: 'Takashi Iwai' <tiwai(a)suse.de> > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org>; > 'linux-sound(a)vger.kernel.org' <linux-sound(a)vger.kernel.org> > Subject: RE: Set pcbeep to default value > > OK. > > > -----Original Message----- > > From: Takashi Iwai <tiwai(a)suse.de> > > Sent: Monday, November 25, 2024 10:45 PM > > To: Kailang <kailang(a)realtek.com> > > Cc: Takashi Iwai <tiwai(a)suse.de> > > Subject: Re: Set pcbeep to default value > > > > > > External mail. > > > > > > > > On Mon, 25 Nov 2024 03:00:26 +0100, > > Kailang wrote: > > > > > > > > > > > > > -----Original Message----- > > > > From: Takashi Iwai <tiwai(a)suse.de> > > > > Sent: Friday, November 22, 2024 5:34 PM > > > > To: Kailang <kailang(a)realtek.com> > > > > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > > > > Subject: Re: Set pcbeep to default value > > > > > > > > > > > > External mail. > > > > > > > > > > > > > > > > On Fri, 22 Nov 2024 08:29:56 +0100, Kailang wrote: > > > > > > > > > > Hi Takashi, > > > > > > > > > > Attach patch will set pcbeep to default value for ALC274. > > > > > > > > Could you resubmit with your Signed-off-by tag? > > > > > > Sorry! Attach again. > > > > You seem to have dropped Cc to alsa-devel ML. > > Also, nowadays we use linux-sound(a)vger.kernel.org for the main ML for > > kernel patches. Care to resubmit to there? > > > > > > thanks, > > > > Takashi

2 1

alsa-lib update from 1.2.12 to 1.2.13 causes HDMI/DsiplayPort outputs to disappear
by GitHub issues - opened 26 Nov '24

26 Nov '24

alsa-project/alsa-lib issue #423 was opened from mwsltn: Basic Info: Linux Distro: Arch Kernel Version: 6.12.1 I have very reliably been using a USB-C connected external monitor that has internal speakers represented by the output device: `HDMI / DisplayPort 1` After running a package update this AM all the HDMI/DisplayPort outputs disappeared. First I downgraded the kernel which didn't help, then checked the pacman logs to see what else had changed and the list included the following: ``` [2024-11-26T05:07:01-0600] [ALPM] upgraded alsa-ucm-conf (1.2.12-1 -> 1.2.13-1) [2024-11-26T05:07:01-0600] [ALPM] upgraded alsa-lib (1.2.12-1 -> 1.2.13-1) [2024-11-26T05:07:01-0600] [ALPM] upgraded alsa-utils (1.2.12-1 -> 1.2.13-1) ``` So I downgrade: ``` [2024-11-26T06:51:22-0600] [PACMAN] Running 'pacman -U file:///var/cache/pacman/pkg/alsa-ucm-conf-1.2.12-1-any.pkg.tar.zst file:///var/cache/pacman/pkg/alsa-lib-1.2.12-1-x86_64.pkg.tar.zst file:///var/cache/pacman/pkg/alsa-utils-1.2.12-1-x86_64.pkg.tar.zst' [2024-11-26T06:51:24-0600] [ALPM] downgraded alsa-ucm-conf (1.2.13-1 -> 1.2.12-1) [2024-11-26T06:51:24-0600] [ALPM] downgraded alsa-lib (1.2.13-1 -> 1.2.12-1) [2024-11-26T06:51:24-0600] [ALPM] downgraded alsa-utils (1.2.13-1 -> 1.2.12-1) ``` Problem fixed. Please let me know if I can provide any other info. Issue URL : https://github.com/alsa-project/alsa-lib/issues/423 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

kernel snd seq bugs: SND_SEQ_EVENT_CLIENT_CHANGE & SND_SEQ_EVENT_PORT_CHANGE
by Mark Lentczner 26 Nov '24

26 Nov '24

I'm the author of amidiminder - a utility that keeps track of desired connections between ALSA Seq (MIDI) ports. I've found a bug in the kernel part of the ALSA Sequencer, and am not sure where or how to report it. Summary: The events: * SND_SEQ_EVENT_CLIENT_CHANGE is never sent * SND_SEQ_EVENT_PORT_CHANGE is not sent on common port changes (name and port info) I have a lot of detail on these bugs, what other software causes them to be a problem (PureData), and where the missing code in the kernel sound/core/seq files is. Before I dump all the details in this forum.... where is the right place for me to report and/or discuss this? - Mark

2 3

Build failures with GCC 15 (-std=gnu23)
by GitHub issues - edited 26 Nov '24

26 Nov '24

alsa-project/alsa-tools issue #25 was edited from kostadinsh: Some of the programs in alsa-tools fail to build with upcoming GCC 15 which uses `-std=gnu23` by default. This can be reproduced on older GCC and Clang versions by adding `-std=gnu23` to CFLAGS. Downstream Gentoo bug report: https://bugs.gentoo.org/943946 hdajackretask: ``` apply-changes.c:358:10: error: conflicting types for ‘reset_changes_boot’; have ‘gboolean(GError **)’ {aka ‘int(struct _GError **)’} 358 | gboolean reset_changes_boot(GError** err) | ^~~~~~~~~~~~~~~~~~ In file included from apply-changes.c:12: apply-changes.h:12:10: note: previous declaration of ‘reset_changes_boot’ with type ‘gboolean(void)’ {aka ‘int(void)’} 12 | gboolean reset_changes_boot(); | ^~~~~~~~~~~~~~~~~~ main-gtk.c: In function ‘reset_boot_clicked’: main-gtk.c:375:5: error: too many arguments to function ‘reset_changes_boot’ 375 | reset_changes_boot(&err); | ^~~~~~~~~~~~~~~~~~ In file included from main-gtk.c:11: apply-changes.h:12:10: note: declared here 12 | gboolean reset_changes_boot(); | ^~~~~~~~~~~~~~~~~~ ``` envy24control: ``` new_process.c: In function ‘new_process’: new_process.c:67:18: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘__sighandler_t’ {aka ‘void (*)(int)’} [-Wincompatible-pointer-types] 67 | int_stat = signal(SIGINT, SIG_IGN); | ^ new_process.c:68:19: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘__sighandler_t’ {aka ‘void (*)(int)’} [-Wincompatible-pointer-types] 68 | quit_stat = signal(SIGQUIT, SIG_IGN); | ^ new_process.c:69:19: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘__sighandler_t’ {aka ‘void (*)(int)’} [-Wincompatible-pointer-types] 69 | usr2_stat = signal(SIGUSR2, SIG_IGN); | ^ new_process.c:81:24: error: passing argument 2 of ‘signal’ from incompatible pointer type [-Wincompatible-pointer-types] 81 | signal(SIGINT, int_stat); | ^~~~~~~~ | | | void (*)(void) In file included from envy24control.h:4: /usr/include/signal.h:88:57: note: expected ‘__sighandler_t’ {aka ‘void (*)(int)’} but argument is of type ‘void (*)(void)’ 88 | extern __sighandler_t signal (int __sig, __sighandler_t __handler) | ~~~~~~~~~~~~~~~^~~~~~~~~ new_process.c:82:25: error: passing argument 2 of ‘signal’ from incompatible pointer type [-Wincompatible-pointer-types] 82 | signal(SIGQUIT, quit_stat); | ^~~~~~~~~ | | | void (*)(void) /usr/include/signal.h:88:57: note: expected ‘__sighandler_t’ {aka ‘void (*)(int)’} but argument is of type ‘void (*)(void)’ 88 | extern __sighandler_t signal (int __sig, __sighandler_t __handler) | ~~~~~~~~~~~~~~~^~~~~~~~~ new_process.c:83:25: error: passing argument 2 of ‘signal’ from incompatible pointer type [-Wincompatible-pointer-types] 83 | signal(SIGUSR2, usr2_stat); | ^~~~~~~~~ | | | void (*)(void) /usr/include/signal.h:88:57: note: expected ‘__sighandler_t’ {aka ‘void (*)(int)’} but argument is of type ‘void (*)(void)’ 88 | extern __sighandler_t signal (int __sig, __sighandler_t __handler) | ~~~~~~~~~~~~~~~^~~~~~~~~ ``` Issue URL : https://github.com/alsa-project/alsa-tools/issues/25 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

Build failures with GCC 15 (-std=gnu23)
by GitHub issues - opened 26 Nov '24

26 Nov '24

alsa-project/alsa-tools issue #25 was opened from kostadinsh: Some of the programs in alsa-tools fail to build with upcoming GCC 15 which uses `-std=gnu23` by default. This can be reproduced on older GCC and Clang versions by adding `-std=gnu23` to CFLAGS. Downstream Gentoo bug report: https://bugs.gentoo.org/943946 hdajackretask: ``` main-gtk.c: In function ‘reset_boot_clicked’: main-gtk.c:375:5: error: too many arguments to function ‘reset_changes_boot’ 375 | reset_changes_boot(&err); | ^~~~~~~~~~~~~~~~~~ In file included from main-gtk.c:11: apply-changes.h:12:10: note: declared here 12 | gboolean reset_changes_boot(); | ^~~~~~~~~~~~~~~~~~ ``` envy24control: ``` new_process.c: In function ‘new_process’: new_process.c:67:18: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘__sighandler_t’ {aka ‘void (*)(int)’} [-Wincompatible-pointer-types] 67 | int_stat = signal(SIGINT, SIG_IGN); | ^ new_process.c:68:19: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘__sighandler_t’ {aka ‘void (*)(int)’} [-Wincompatible-pointer-types] 68 | quit_stat = signal(SIGQUIT, SIG_IGN); | ^ new_process.c:69:19: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘__sighandler_t’ {aka ‘void (*)(int)’} [-Wincompatible-pointer-types] 69 | usr2_stat = signal(SIGUSR2, SIG_IGN); | ^ new_process.c:81:24: error: passing argument 2 of ‘signal’ from incompatible pointer type [-Wincompatible-pointer-types] 81 | signal(SIGINT, int_stat); | ^~~~~~~~ | | | void (*)(void) In file included from envy24control.h:4: /usr/include/signal.h:88:57: note: expected ‘__sighandler_t’ {aka ‘void (*)(int)’} but argument is of type ‘void (*)(void)’ 88 | extern __sighandler_t signal (int __sig, __sighandler_t __handler) | ~~~~~~~~~~~~~~~^~~~~~~~~ new_process.c:82:25: error: passing argument 2 of ‘signal’ from incompatible pointer type [-Wincompatible-pointer-types] 82 | signal(SIGQUIT, quit_stat); | ^~~~~~~~~ | | | void (*)(void) /usr/include/signal.h:88:57: note: expected ‘__sighandler_t’ {aka ‘void (*)(int)’} but argument is of type ‘void (*)(void)’ 88 | extern __sighandler_t signal (int __sig, __sighandler_t __handler) | ~~~~~~~~~~~~~~~^~~~~~~~~ new_process.c:83:25: error: passing argument 2 of ‘signal’ from incompatible pointer type [-Wincompatible-pointer-types] 83 | signal(SIGUSR2, usr2_stat); | ^~~~~~~~~ | | | void (*)(void) /usr/include/signal.h:88:57: note: expected ‘__sighandler_t’ {aka ‘void (*)(int)’} but argument is of type ‘void (*)(void)’ 88 | extern __sighandler_t signal (int __sig, __sighandler_t __handler) | ~~~~~~~~~~~~~~~^~~~~~~~~ ``` Issue URL : https://github.com/alsa-project/alsa-tools/issues/25 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

No built-in microphone in Thinkpad X1 Carbon (Gen 12) after update to v1.2.13
by GitHub issues - opened 26 Nov '24

26 Nov '24

alsa-project/alsa-ucm-conf issue #476 was opened from iyanmv: I think the title is self-explanatory. After updating from v1.2.12 to v1.2.13 in Arch Linux, the built-in microphone on my laptop is not working. Downgrading to previous versions fixes the issue. In addition, I also noticed that when attached to an external monitor using a docking station, the speakers of the monitor are not listed as before, and only the laptop's built-in speaker appears. I'm running linux 6.12.1 and KDE Plasma 6.2.3. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/476 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

how do I create configure a ucm2 conf file for my card just like an acp file
by GitHub issues - opened 26 Nov '24

26 Nov '24

alsa-project/alsa-ucm-conf issue #475 was opened from beh-10257: 1_ how do I know if my card is using acp or ucm profile 2_ can ucm config file be availble in user directories 3_ how do i test a ucm profile 4_ if my card is using ucm is there a way to cooverride that file 5_ for more context https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4421 6_ as a tldr I want to take advanatage of indep_hp=yes (IDK why this is not enabled by default its a great feature compared to other platforms after all) either way I want to make hw:0,0 as speaker and hw:0,2 as headphones can a ucm profile replace the indep_hp hint and udev rule and just porceed to seperate them directly also indep_hp seperates speaker and headphones output is there a way to seperate internal mic and headphone speaker also is there a way to put priority for headphones when the jack port is actually put or even disable it entirely and just default back to speaker automatically thats all thanks thats all and thanks further editing will be filled once I have more questions and thanks Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/475 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC225 update depop procedure
by Kailang 26 Nov '24

26 Nov '24

Hi Takashi, ALC225 update depop procedure. Many Thanks. BR, Kailang

2 1

Enable speaker pins for Medion E15443 platform
by Kailang 26 Nov '24

26 Nov '24

Hi Takashi, Attach was fixed speaker no sound issue. BR, Kailang

2 1

Device selection regression in 1.2.13 for LNL
by GitHub issues - edited 26 Nov '24

26 Nov '24

alsa-project/alsa-ucm-conf issue #474 was edited from CarelessCaution: On Intel Lunar Lake, there seems to be a regression in alsa-ucm-conf that affects device selection in KDE Plasma and probably in other software. In 1.2.12, there are the options Play HiFi Quality Music and Pro Audio for both input and output. However, when upgrading to 1.2.13, there is only the option for Pro Audio. This problem is immediately remedied by downgrading just the alsa-ucm-conf package on Arch Linux. According to `aplay -l`, the audio card is sofhdadsp. Before (1.2.12): ![before](https://github.com/user-attachments/assets/c7fa81e0-97f1-45ae-a14e-8549f85cdd27) After (1.2.13): ![after](https://github.com/user-attachments/assets/5867886b-6686-4265-ad68-50c7f53e1a47) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/474 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Device selection regression in 1.2.13 for LNL
by GitHub issues - edited 26 Nov '24

26 Nov '24

alsa-project/alsa-ucm-conf issue #474 was edited from CarelessCaution: On Intel Lunar Lake, there seems to be a regression in alsa-ucm-conf that affects device selection in KDE Plasma and probably in other software. In 1.2.12, there are the options Play HiFi Quality Music and Pro Audio for both input and output. However, when upgrading to 1.2.13, there is only the option for Pro Audio. This problem is immediately remedied by downgrading just the alsa-ucm-conf package on Arch Linux. According to `aplay -l`, the audio card is sofhdadsp . Before (1.2.12): ![before](https://github.com/user-attachments/assets/c7fa81e0-97f1-45ae-a14e-8549f85cdd27) After (1.2.13): ![after](https://github.com/user-attachments/assets/5867886b-6686-4265-ad68-50c7f53e1a47) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/474 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Device selection regression in 1.2.13 for LNL
by GitHub issues - opened 26 Nov '24

26 Nov '24

alsa-project/alsa-ucm-conf issue #474 was opened from CarelessCaution: On Intel Lunar Lake, there seems to be a regression in alsa-ucm-conf that affects device selection in KDE Plasma and probably in other software. In 1.2.12, there are the options Play HiFi Quality Music and Pro Audio for both input and output. However, when upgrading to 1.2.13, there is only the option for Pro Audio. This problem is immediately remedied by downgrading just the alsa-ucm-conf package on Arch Linux. Before (1.2.12): ![before](https://github.com/user-attachments/assets/c7fa81e0-97f1-45ae-a14e-8549f85cdd27) After (1.2.13): ![after](https://github.com/user-attachments/assets/5867886b-6686-4265-ad68-50c7f53e1a47) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/474 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

RE: Set pcbeep to default value
by Kailang 26 Nov '24

26 Nov '24

OK. > -----Original Message----- > From: Takashi Iwai <tiwai(a)suse.de> > Sent: Monday, November 25, 2024 10:45 PM > To: Kailang <kailang(a)realtek.com> > Cc: Takashi Iwai <tiwai(a)suse.de> > Subject: Re: Set pcbeep to default value > > > External mail. > > > > On Mon, 25 Nov 2024 03:00:26 +0100, > Kailang wrote: > > > > > > > > > -----Original Message----- > > > From: Takashi Iwai <tiwai(a)suse.de> > > > Sent: Friday, November 22, 2024 5:34 PM > > > To: Kailang <kailang(a)realtek.com> > > > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > > > Subject: Re: Set pcbeep to default value > > > > > > > > > External mail. > > > > > > > > > > > > On Fri, 22 Nov 2024 08:29:56 +0100, > > > Kailang wrote: > > > > > > > > Hi Takashi, > > > > > > > > Attach patch will set pcbeep to default value for ALC274. > > > > > > Could you resubmit with your Signed-off-by tag? > > > > Sorry! Attach again. > > You seem to have dropped Cc to alsa-devel ML. > Also, nowadays we use linux-sound(a)vger.kernel.org for the main ML for kernel > patches. Care to resubmit to there? > > > thanks, > > Takashi

1 0

GoXLR not working in 1.2.13
by GitHub issues - opened 25 Nov '24

25 Nov '24

alsa-project/alsa-ucm-conf issue #473 was opened from AruAki: GoXLR not splitting into multiple channels. `alsaucm -c hw:1 dump text` returns ``` ALSA lib parser.c:270:(parse_syntax_field) Incompatible syntax 7 in USB-Audio.conf ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:1 use case configuration -22 alsaucm: error failed to open sound card hw:1: Invalid argument ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/473 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RFC PATCH 0/2] Add support for stacked and parallel memories
by Amit Kumar Mahapatra 25 Nov '24

25 Nov '24

Hello Everyone, Following an email discussion with Miquel regarding the binding changes and overall architecture for implementing support for stacked and parallel memories, I’m sharing this RFC to initiate a discussion on the proposed updates to current bindings and to finalize the implementation architecture. Before diving into the main topic, here is some background on stacked and parallel memories. The AMD QSPI controller supports two advanced connection modes(Stacked and Parallel) which allow the controller to treat two different flashes as one storage. Stacked: Flashes share the same SPI bus, but different CS line, controller driver asserts the CS of the flash to which it needs to communicate. Stacked mode is a software abstraction rather than a controller feature or capability. At any given time, the controller communicates with one of the two connected flash devices, as determined by the requested address and data length. If an operation starts on one flash and ends on the other, the core needs to split it into two separate operations and adjust the data length accordingly. Parallel(Multi-CS): Both the flashes have their separate SPI bus, CS of both the flashes will be asserted/de-asserted at the same time. In this mode data will be split across both the flashes by enabling the STRIPE setting in the controller. Parallel mode is a controller feature where if the STRIPE bit is set then the controller internally handles the data split during data write to the flashes and while reading data from the flash the controller internally merges data from both the flashes before writing to the controller FIFO. If STRIPE is not enabled, then same data will be sent to both the devices. In parallel mode both the flashes should be identical. For more information on the modes please feel free to go through the controller flash interface below [1]. Mirochip QSPI controller[2] also supports "Dual Parallel 8-bit IO mode", but they call it "Twin Quad Mode". Initially in [3] [4] [5] Miquel had tried to extend MTD-CONCAT driver to support Stacked mode, but the bindings were not accepted. So, the MTD-CONCAT approach was dropped and the DT bindings that got accepted [6] [7] [8] that describes the two flash devices as being one. SPI core changes to support the above bindings were added [9]. While adding the support in SPI-NOR Tudor provided additional feedback, leading to a discussion on updating the current stacked and parallel DT bindings. Proposed Solution: The solution has two parts: 1. Update MTD-CONCAT Update MTD-CONCAT to create virtual concatinated mtd devices as defined in the device tree. 2. Add a New Layer Add a new layer between the SPI-NOR and MTD layers to support stacked and parallel configurations. This new layer will be part of spi-nor, located in mtd/spi-nor/, can be included/excluded via Kconfig, will be maintained by AMD and will: - During probing, store information from all connected flashes in stacked or parallel mode and present them as a single device to the MTD layer. - Register callbacks and manage MTD device registration within the new layer instead of spi-nor/core.c. - Make minimal changes in spi-nor/core.c, as stacked and parallel handling will be managed by the new layer on top of SPI-NOR. - Handle odd byte count requests from the MTD layer during flash operations in parallel mode. [1] https://docs.amd.com/r/en-US/am011-versal-acap-trm/QSPI-Flash-Device-Interf… [2] https://ww1.microchip.com/downloads/aemDocuments/documents/MPU32/ProductDoc… [3] https://lore.kernel.org/all/20191113171505.26128-4-miquel.raynal@bootlin.co… [4] https://lore.kernel.org/all/20191127105522.31445-5-miquel.raynal@bootlin.co… [5]https://lore.kernel.org/all/20211112152411.818321-1-miquel.raynal@bootlin… [6] https://github.com/torvalds/linux/commit/f89504300e94524d5d5846ff8b728592ac… [7] https://github.com/torvalds/linux/commit/eba5368503b4291db7819512600fa014ea… [8] https://github.com/torvalds/linux/commit/e2edd1b64f1c79e8abda365149ed62a2a9… [9]https://github.com/torvalds/linux/commit/4d8ff6b0991d5e86b17b235fc46ec62e… Thanks, Amit Amit Kumar Mahapatra (2): dt-bindings: mtd: Add bindings for describing concatinated MTD devices dt-bindings: spi: Update stacked and parallel bindings .../mtd/partitions/fixed-partitions.yaml | 18 +++++++++++++++ .../bindings/mtd/partitions/partitions.yaml | 6 +++++ .../bindings/spi/spi-controller.yaml | 23 +++++++++++++++++-- .../bindings/spi/spi-peripheral-props.yaml | 9 +++----- 4 files changed, 48 insertions(+), 8 deletions(-) -- 2.34.1

6 15

ABI break in 1.2.12 -> 1.2.13
by GitHub issues - edited 25 Nov '24

25 Nov '24

alsa-project/alsa-lib issue #422 was edited from bbhtt: This seems a bit unusual for a patch version bump. Any idea what this is about? Thanks! ``` ┌───────────────────────────┐ │ ABI Break: libasound.so.2 │ └───────────────────────────┘ Functions changes summary: 1 Removed, 3 Changed (250 filtered out), 35 Added functions Variables changes summary: 0 Removed, 0 Changed, 0 Added variable 1 Removed function: [D] 'function void snd_ump_block_info_set_block_id(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_block_id@@ALSA_1.2.10} 35 Added functions: [A] 'function const unsigned char* snd_pcm_hw_params_get_sync(const snd_pcm_hw_params_t*)' {snd_pcm_hw_params_get_sync@@ALSA_1.2.13} [A] 'function int snd_seq_create_ump_block(snd_seq_t*, int, const snd_ump_block_info_t*)' {snd_seq_create_ump_block@@ALSA_1.2.13} [A] 'function int snd_seq_create_ump_endpoint(snd_seq_t*, const snd_ump_endpoint_info_t*, unsigned int)' {snd_seq_create_ump_endpoint@@ALSA_1.2.13} [A] 'function int snd_seq_has_queue_tempo_base(snd_seq_t*)' {snd_seq_has_queue_tempo_base@@ALSA_0.9} [A] 'function int snd_seq_port_info_get_ump_is_midi1(const snd_seq_port_info_t*)' {snd_seq_port_info_get_ump_is_midi1@@ALSA_1.2.13} [A] 'function void snd_seq_port_info_set_ump_is_midi1(snd_seq_port_info_t*, int)' {snd_seq_port_info_set_ump_is_midi1@@ALSA_1.2.13} [A] 'function unsigned int snd_seq_queue_tempo_get_tempo_base(const snd_seq_queue_tempo_t*)' {snd_seq_queue_tempo_get_tempo_base@@ALSA_1.2.13} [A] 'function void snd_seq_queue_tempo_set_tempo_base(snd_seq_queue_tempo_t*, unsigned int)' {snd_seq_queue_tempo_set_tempo_base@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_clear(snd_ump_block_info_t*)' {snd_ump_block_info_clear@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_active(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_active@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_card(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_card@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_device(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_device@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_direction(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_direction@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_first_group(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_first_group@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_flags(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_flags@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_midi_ci_version(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_midi_ci_version@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_name(snd_ump_block_info_t*, const char*)' {snd_ump_block_info_set_name@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_num_groups(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_num_groups@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_sysex8_streams(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_sysex8_streams@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_ui_hint(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_ui_hint@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_clear(snd_ump_endpoint_info_t*)' {snd_ump_endpoint_info_clear@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_card(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_card@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_device(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_device@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_family_id(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_family_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_flags(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_flags@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_manufacturer_id(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_manufacturer_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_model_id(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_model_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_name(snd_ump_endpoint_info_t*, const char*)' {snd_ump_endpoint_info_set_name@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_num_blocks(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_num_blocks@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_product_id(snd_ump_endpoint_info_t*, const char*)' {snd_ump_endpoint_info_set_product_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_protocol(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_protocol@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_protocol_caps(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_protocol_caps@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_sw_revision(snd_ump_endpoint_info_t*, const unsigned char*)' {snd_ump_endpoint_info_set_sw_revision@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_version(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_version@@ALSA_1.2.13} [A] 'function int snd_ump_packet_length(unsigned int)' {snd_ump_packet_length@@ALSA_1.2.13} 3 functions with some indirect sub-type change: [C] 'function int __old_snd_pcm_hw_params_get_access(const snd_pcm_hw_params_t*)' at pcm.c:8062:1 has some indirect sub-type changes: Please note that the exported symbol of this function is snd_pcm_hw_params_get_access(a)ALSA_0.9 parameter 1 of type 'const snd_pcm_hw_params_t*' has sub-type changes: in pointed to type 'const snd_pcm_hw_params_t': in unqualified underlying type 'typedef snd_pcm_hw_params_t' at pcm.h:64:1: underlying type 'struct snd_pcm_hw_params' at asound.h:414:1 changed: type size hasn't changed 1 data member insertion: 'unsigned char sync[16]', at offset 4352 (in bits) at asound.h:421:1 1 data member change: type of 'unsigned char reserved[64]' changed: type name changed from 'unsigned char[64]' to 'unsigned char[48]' array type size changed from 512 to 384 array type subrange 1 changed length from 64 to 48 array subrange changed: upper bound of '<anonymous range>[48]' change from '63' to '47' and offset changed from 4352 to 4480 (in bits) (by +128 bits) [C] 'function int snd_ctl_pcm_info(snd_ctl_t*, snd_pcm_info_t*)' at control.c:1219:1 has some indirect sub-type changes: parameter 2 of type 'snd_pcm_info_t*' has sub-type changes: in pointed to type 'typedef snd_pcm_info_t' at pcm.h:48:1: underlying type 'struct snd_pcm_info' at asound.h:345:1 changed: type size hasn't changed 1 data member change: type of 'snd_pcm_sync_id sync' changed: entity changed from 'union snd_pcm_sync_id' to 'unsigned char[16]' and name of 'snd_pcm_info::sync' changed to 'snd_pcm_info::pad1' at asound.h:349:1 [C] 'function int snd_seq_get_queue_tempo(snd_seq_t*, int, snd_seq_queue_tempo_t*)' at seq.c:3880:1 has some indirect sub-type changes: parameter 3 of type 'snd_seq_queue_tempo_t*' has sub-type changes: in pointed to type 'typedef snd_seq_queue_tempo_t' at seq.h:441:1: underlying type 'struct snd_seq_queue_tempo' at asequencer.h:540:1 changed: type size hasn't changed 1 data member insertion: 'unsigned short int tempo_base', at offset 160 (in bits) at asequencer.h:532:1 1 data member change: type of 'char reserved[24]' changed: type name changed from 'char[24]' to 'char[22]' array type size changed from 192 to 176 array type subrange 1 changed length from 24 to 22 array subrange changed: upper bound of '<anonymous range>[22]' change from '23' to '21' and offset changed from 160 to 176 (in bits) (by +16 bits) ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/422 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ABI break in 1.2.12 -> 1.2.13
by GitHub issues - opened 25 Nov '24

25 Nov '24

alsa-project/alsa-lib issue #422 was opened from bbhtt: This seems a bit unusual for a patch version bump. ``` ┌───────────────────────────┐ │ ABI Break: libasound.so.2 │ └───────────────────────────┘ Functions changes summary: 1 Removed, 3 Changed (250 filtered out), 35 Added functions Variables changes summary: 0 Removed, 0 Changed, 0 Added variable 1 Removed function: [D] 'function void snd_ump_block_info_set_block_id(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_block_id@@ALSA_1.2.10} 35 Added functions: [A] 'function const unsigned char* snd_pcm_hw_params_get_sync(const snd_pcm_hw_params_t*)' {snd_pcm_hw_params_get_sync@@ALSA_1.2.13} [A] 'function int snd_seq_create_ump_block(snd_seq_t*, int, const snd_ump_block_info_t*)' {snd_seq_create_ump_block@@ALSA_1.2.13} [A] 'function int snd_seq_create_ump_endpoint(snd_seq_t*, const snd_ump_endpoint_info_t*, unsigned int)' {snd_seq_create_ump_endpoint@@ALSA_1.2.13} [A] 'function int snd_seq_has_queue_tempo_base(snd_seq_t*)' {snd_seq_has_queue_tempo_base@@ALSA_0.9} [A] 'function int snd_seq_port_info_get_ump_is_midi1(const snd_seq_port_info_t*)' {snd_seq_port_info_get_ump_is_midi1@@ALSA_1.2.13} [A] 'function void snd_seq_port_info_set_ump_is_midi1(snd_seq_port_info_t*, int)' {snd_seq_port_info_set_ump_is_midi1@@ALSA_1.2.13} [A] 'function unsigned int snd_seq_queue_tempo_get_tempo_base(const snd_seq_queue_tempo_t*)' {snd_seq_queue_tempo_get_tempo_base@@ALSA_1.2.13} [A] 'function void snd_seq_queue_tempo_set_tempo_base(snd_seq_queue_tempo_t*, unsigned int)' {snd_seq_queue_tempo_set_tempo_base@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_clear(snd_ump_block_info_t*)' {snd_ump_block_info_clear@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_active(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_active@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_card(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_card@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_device(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_device@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_direction(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_direction@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_first_group(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_first_group@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_flags(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_flags@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_midi_ci_version(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_midi_ci_version@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_name(snd_ump_block_info_t*, const char*)' {snd_ump_block_info_set_name@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_num_groups(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_num_groups@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_sysex8_streams(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_sysex8_streams@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_ui_hint(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_ui_hint@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_clear(snd_ump_endpoint_info_t*)' {snd_ump_endpoint_info_clear@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_card(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_card@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_device(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_device@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_family_id(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_family_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_flags(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_flags@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_manufacturer_id(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_manufacturer_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_model_id(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_model_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_name(snd_ump_endpoint_info_t*, const char*)' {snd_ump_endpoint_info_set_name@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_num_blocks(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_num_blocks@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_product_id(snd_ump_endpoint_info_t*, const char*)' {snd_ump_endpoint_info_set_product_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_protocol(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_protocol@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_protocol_caps(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_protocol_caps@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_sw_revision(snd_ump_endpoint_info_t*, const unsigned char*)' {snd_ump_endpoint_info_set_sw_revision@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_version(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_version@@ALSA_1.2.13} [A] 'function int snd_ump_packet_length(unsigned int)' {snd_ump_packet_length@@ALSA_1.2.13} 3 functions with some indirect sub-type change: [C] 'function int __old_snd_pcm_hw_params_get_access(const snd_pcm_hw_params_t*)' at pcm.c:8062:1 has some indirect sub-type changes: Please note that the exported symbol of this function is snd_pcm_hw_params_get_access(a)ALSA_0.9 parameter 1 of type 'const snd_pcm_hw_params_t*' has sub-type changes: in pointed to type 'const snd_pcm_hw_params_t': in unqualified underlying type 'typedef snd_pcm_hw_params_t' at pcm.h:64:1: underlying type 'struct snd_pcm_hw_params' at asound.h:414:1 changed: type size hasn't changed 1 data member insertion: 'unsigned char sync[16]', at offset 4352 (in bits) at asound.h:421:1 1 data member change: type of 'unsigned char reserved[64]' changed: type name changed from 'unsigned char[64]' to 'unsigned char[48]' array type size changed from 512 to 384 array type subrange 1 changed length from 64 to 48 array subrange changed: upper bound of '<anonymous range>[48]' change from '63' to '47' and offset changed from 4352 to 4480 (in bits) (by +128 bits) [C] 'function int snd_ctl_pcm_info(snd_ctl_t*, snd_pcm_info_t*)' at control.c:1219:1 has some indirect sub-type changes: parameter 2 of type 'snd_pcm_info_t*' has sub-type changes: in pointed to type 'typedef snd_pcm_info_t' at pcm.h:48:1: underlying type 'struct snd_pcm_info' at asound.h:345:1 changed: type size hasn't changed 1 data member change: type of 'snd_pcm_sync_id sync' changed: entity changed from 'union snd_pcm_sync_id' to 'unsigned char[16]' and name of 'snd_pcm_info::sync' changed to 'snd_pcm_info::pad1' at asound.h:349:1 [C] 'function int snd_seq_get_queue_tempo(snd_seq_t*, int, snd_seq_queue_tempo_t*)' at seq.c:3880:1 has some indirect sub-type changes: parameter 3 of type 'snd_seq_queue_tempo_t*' has sub-type changes: in pointed to type 'typedef snd_seq_queue_tempo_t' at seq.h:441:1: underlying type 'struct snd_seq_queue_tempo' at asequencer.h:540:1 changed: type size hasn't changed 1 data member insertion: 'unsigned short int tempo_base', at offset 160 (in bits) at asequencer.h:532:1 1 data member change: type of 'char reserved[24]' changed: type name changed from 'char[24]' to 'char[22]' array type size changed from 192 to 176 array type subrange 1 changed length from 24 to 22 array subrange changed: upper bound of '<anonymous range>[22]' change from '23' to '21' and offset changed from 160 to 176 (in bits) (by +16 bits) ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/422 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

1.2.13 will lead to mute of speakers and headphone jack on MTL laptop Lenovo Youga Pro 7 14IMH9
by GitHub issues - edited 25 Nov '24

25 Nov '24

alsa-project/alsa-ucm-conf issue #472 was edited from phush0: Changing version from 1.2.12 to 1.2.13 will lead to not working sound during plug/unplug of the head phone jack or mute during reboot on Arch install. Other ppl with similar laptops have reported same problem. On my version codec is ALC287 with bass speaker output. I can unmute speaker or headphones trough alsamixer, but plugin headphone and the unplugin will lead to both mute. Sometimes reboot will lead to same problem and I have to operate channels by hand again. Similar problems: [https://discuss.cachyos.org/t/sound-problems-kde/4350](https://discuss.cach… edit: reading more seems like problem is with the new HDMI macro, because other report problems with HDMI output. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/472 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

1.2.13 will lead to mute of speakers and headphone jack on MTL laptop Lenovo Youga Pro 7 14IMH9
by GitHub issues - edited 25 Nov '24

25 Nov '24

alsa-project/alsa-ucm-conf issue #472 was edited from phush0: Changing version from 1.2.12 to 1.2.13 will lead to not working sound during plug/unplug of the head phone jack or mute during reboot on Arch install. Other ppl with similar laptops have reported same problem. On my version codec is ALC287 with bass speaker output. I can unmute speaker or headphones trough alsamixer, but plugin headphone and the unplugin will lead to both mute. Sometimes reboot will lead to same problem and I have to operate channels by hand again. Similar problems: [https://discuss.cachyos.org/t/sound-problems-kde/4350](https://discuss.cach… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/472 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

1.2.13 will lead to mute of speakers and head phone jack on MTL laptop Lenovo Youga Pro 7 14IMH9
by GitHub issues - opened 25 Nov '24

25 Nov '24

alsa-project/alsa-ucm-conf issue #472 was opened from phush0: Changing version from 1.2.12 to 1.2.13 will lead to not working sound during plug/unplug of the head phone jack or mute during reboot on Arch install. Other ppl with similar laptops have reported same problem. On my version codec is ALC287 with bass speaker output. I can unmute speaker or headphones trough alsamixer, but plugin headphone and the unplugin will lead to both mute. Sometimes reboot will lead to same problem and I have to operate channels by hand again. Similar problems: [https://discuss.cachyos.org/t/sound-problems-kde/4350](https://discuss.cach… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/472 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR, partly fixed but fix broke CM106
by schorpp 25 Nov '24

25 Nov '24

>> >> [BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR >> >> ALSA versions 1.0.28 >> >> Kernel Driver versions 3.12.40 Partly FIXED in Longterm Stable Kernel 4.19.324 and using HDMI DEV 1 instead of 0. 2 CH PCM HDMI to AVR works now but AC3/DTS passthrough still not or is not recognized by AVR or some wrong AES setting but iec958 optical out AC3/DTS output to AVR works. New BUG: No more 5.1 Analog Output on the CM106 USB Box if AC3 stream is selected. Your ALSA information is located at http://alsa-project.org/db/?f=5ef8b56450583ba8afbcc705afc02228da4a3072 y tom >> >> Hardware: >> >> DFI Mainboard MB330-CRM >> >> https://www.dfi.com/de/product/index/273#specification >> >> Reproducible Always >> >> Steps to reproduce and full report: >> >> https://www-vdr--portal-de.translate.goog/forum/index.php?thread/136519-gel… > Output of alsa-info.sh is here > > http://alsa-project.org/db/?f=fc4734deddd9a92331692bf94b15cf84b7722cd4 > >> >> Software crosscheck with CM106 USB Soundbox iec958: works OK. >> >> Where to start in the drivers source code to fix it? >> >> y >> tom >> >>

1 0

Motu D828 (2024) not working again
by GitHub issues - opened 25 Nov '24

25 Nov '24

alsa-project/alsa-ucm-conf issue #471 was opened from beatboxa: I had previously gotten my MOTU D828 working with UCM. I recently used Ubuntu 24.04's software updater, and now my UCM appears to have stopped working. The UCM files are still present in the directories where they previously worked. The output of spa-acp-tool -c 0 -vvvv info now begins with the following: ``` UCM is not supported for this USB device (MOTU 828 at usb-0000:59:00.0-6, high speed @ USB07fd:000e) error: failed to import hw:0 use case configuration -6 UCM is not supported for this USB device (MOTU 828 at usb-0000:59:00.0-6, high speed @ USB07fd:000e) error: failed to import 828 use case configuration -6 UCM not available for card 828 Check for file: /home/beatboxa/.config/alsa-card-profile/mixer/profile-sets/default.conf Check for file: /home/beatboxa/.config/alsa-card-profile/profile-sets/default.conf Check for file: /etc/alsa-card-profile/mixer/profile-sets/default.conf Check for file: /etc/alsa-card-profile/profile-sets/default.conf Loading profile set: /usr/share/alsa-card-profile/mixer/profile-sets/default.conf Parsing configuration file '/usr/share/alsa-card-profile/mixer/profile-sets/default.conf' Parsing configuration file '/usr/share/alsa-card-profile/mixer/profile-sets/9999-custom.conf' Looking at profile input:analog-stereo Checking for recording on Analog Stereo (analog-stereo) Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ALSA device open 'front:0' capture: 0x5804b63f3d90 snd_pcm_hw_params_set_format(Signed 16 bit Little Endian) failed: Invalid argument snd_pcm_hw_params_set_format(Signed 16 bit Big Endian) failed: Invalid argument snd_pcm_hw_params_set_format(Float 32 bit Little Endian) failed: Invalid argument snd_pcm_hw_params_set_format(Float 32 bit Big Endian) failed: Invalid argument snd_pcm_hw_params_set_format(Signed 32 bit Little Endian) failed: Invalid argument snd_pcm_hw_params_set_format(Signed 32 bit Big Endian) failed: Invalid argument snd_pcm_hw_params_set_format(Signed 24 bit Little Endian) failed: Invalid argument snd_pcm_hw_params_set_format(Signed 24 bit Big Endian) failed: Invalid argument snd_pcm_hw_params_set_channels(2) failed: Invalid argument ALSA device close 0x5804b63f3d90 ``` And goes on from there. Previously, my configuration worked, and is tracked in this thread: #https://github.com/alsa-project/alsa-ucm-conf/pull/416 Is there some little thing I need to change to get UCM working again? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/471 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No sound and alsactl init errors after updating to 1.2.13
by GitHub issues - edited 24 Nov '24

24 Nov '24

alsa-project/alsa-utils issue #283 was edited from wastlnd: I think I have kind of a regression after updating to the latest version. Running alsactl restore as a runit service, I have this error at startup: ``` Found hardware: "acp" "" "" "" "" Hardware is initialized using a generic method ``` And I have no sound at all. I have tried to apply the latest commit fixing the udev rule, but it didn't work. Downgrading to 1.2.12 fixes the issue. I am running an Arch based distro (kernel 6.6) on AMD Ryzen 7 5825U CPU with Radeon graphics. My soundcard is Realtek and I have an AMD microphone array. I don't run Pulseaudio or Pipewire, just plain ALSA. Issue URL : https://github.com/alsa-project/alsa-utils/issues/283 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

No sound and alsactl init errors after updating to 1.2.13
by GitHub issues - edited 24 Nov '24

24 Nov '24

alsa-project/alsa-utils issue #283 was edited from wastlnd: I think I have kind of a regression after updating to the latest version. I have this error at startup: ``` Found hardware: "acp" "" "" "" "" Hardware is initialized using a generic method ``` And I have no sound at all. I have tried to apply the latest commit fixing the udev rule, but it didn't work. Downgrading to 1.2.12 fixes the issue. I am running an Arch based distro (kernel 6.6) on AMD Ryzen 7 5825U CPU with Radeon graphics. My soundcard is Realtek and I have an AMD microphone array. I don't run Pulseaudio or Pipewire, just plain ALSA. Issue URL : https://github.com/alsa-project/alsa-utils/issues/283 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

amd-soundwire: add support for AMD generic soundwire legacy machine driver
by GitHub pull_request - edited 24 Nov '24

24 Nov '24

alsa-project/alsa-ucm-conf pull request #470 was edited from vijendarmukunda: Add support for AMD generic legacy(No DSP) machine driver for ACP6.3 platform using RT722 codec. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/470 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/470.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

amd-soundwire: add support for AMD generic legacy machine driver
by GitHub pull_request - opened 24 Nov '24

24 Nov '24

alsa-project/alsa-ucm-conf pull request #470 was opened from vijendarmukunda: Add support for AMD generic legacy(No DSP) machine driver for ACP6.3 platform using RT722 codec. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/470 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/470.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No sound and alsactl init errors after updating to 1.2.13
by GitHub issues - edited 24 Nov '24

24 Nov '24

alsa-project/alsa-utils issue #283 was edited from wastlnd: I think I have kind of a regression after updating to the latest version. I have this error at startup: ``` Found hardware: "acp" "" "" "" "" Hardware is initialized using a generic method ``` And I have no sound at all. I have tried to apply the latest commit fixing the udev rule, but it didn't work. Downgrading to 1.2.12 fixes the issue. I am running an Arch based distro on AMD Ryzen 7 5825U CPU with Radeon graphics. My soundcard is Realtek and I have an AMD microphone array. Issue URL : https://github.com/alsa-project/alsa-utils/issues/283 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

No sound and alsactl init errors after updating to 1.12.13
by GitHub issues - opened 24 Nov '24

24 Nov '24

alsa-project/alsa-utils issue #283 was opened from wastlnd: I think I have kind of a regression after updating to the latest version. I have this error at startup: ``` Found hardware: "acp" "" "" "" "" Hardware is initialized using a generic method ``` And I have no sound at all. I have tried to apply the latest commit fixing the udev rule, but it didn't work. Downgrading to 1.2.12 fixes the issue. I am running an Arch based distro on AMD Ryzen 7 5825U CPU with Radeon graphics. My soundcard is Realtek and I have an AMD microphone array. Issue URL : https://github.com/alsa-project/alsa-utils/issues/283 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

UCM2 Profile for Roland BridgeCastX
by GitHub pull_request - edited 24 Nov '24

24 Nov '24

alsa-project/alsa-ucm-conf pull request #469 was edited from Konna1337: Added Configuration for the X version of Bridge Cast. Config is based on the non X version but it's wired differently and has two more Hardware Channels. It's for the 2.0 Firmware of the Device. The "Game" channel has a virtual 5.1 mode which doesn't work properly and isn't configured. - Added condition to USB-Audio.conf - Added BridgeCastXV2.conf - Added BridgeCastXV2-Hifi.conf with configuration for channels Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/469 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/469.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

UCM Configuration for Roland BridgeCastX
by GitHub pull_request - edited 24 Nov '24

24 Nov '24

alsa-project/alsa-ucm-conf pull request #469 was edited from Konna1337: Added Configuration for the X version of Bridge Cast. Config is based on the non X version but it's wired differently and has two more Hardware Channels. It's for the 2.0 Firmware of the Device. The "Game" channel has a virtual 5.1 mode which doesn't work properly and isn't configured. - Added condition to USB-Audio.conf - Added BridgeCastXV2.conf - Added BridgeCastXV2-Hifi.conf with configuration for channels Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/469 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/469.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Configuration for Roland BridgeCastX
by GitHub pull_request - opened 24 Nov '24

24 Nov '24

alsa-project/alsa-ucm-conf pull request #469 was opened from Konna1337: Added Configuration for the X version of Bridge Cast. Config is based on the non X version but it's wired differently and has two more Hardware Channels. It's for the 2.0 Firmware of the Device. The "Game" channel has a virtual 5.1 mode which doesn't work properly and isn't configured. - Added condition to USB-Audio.conf - Added BridgeCastXV2.conf - Added BridgeCastXV2-Hifi.conf with configuration for channels Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/469 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/469.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

TLVs with (size % 4) != 0 make amixer and alsactl crash
by GitHub issues - opened 23 Nov '24

23 Nov '24

alsa-project/alsa-utils issue #282 was opened from geoffreybennett: While trying to figure out how TLVs work, I found that if the length field is not a multiple of 4 then amixer and alsactl keep reading past the end of the buffer until they segfault, because size is unsigned. One example here: ``` diff --git a/amixer/amixer.c b/amixer/amixer.c index 8b8000b..03cc9cb 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -587,7 +587,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ #endif default: printf("unk-%u-", type); - while (size > 0) { + while (size >= sizeof(unsigned int)) { printf("0x%08x,", tlv[idx++]); size -= sizeof(unsigned int); } ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/282 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR
by schorpp 23 Nov '24

23 Nov '24

> > [BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR > > ALSA versions 1.0.25...current > > Kernel Driver versions 3.12.40...6.x > > Hardware: > > DFI Mainboard MB330-CRM > > https://www.dfi.com/de/product/index/273#specification > > Reproducible Always > > Steps to reproduce and full report: > > https://www-vdr--portal-de.translate.goog/forum/index.php?thread/136519-gel… Output of alsa-info.sh is here http://alsa-project.org/db/?f=fc4734deddd9a92331692bf94b15cf84b7722cd4 > > Software crosscheck with CM106 USB Soundbox iec958: works OK. > > Where to start in the drivers source code to fix it? > > y > tom > >

1 0

x870-A no S/PDIF output audio (ALC4080)
by GitHub issues - opened 22 Nov '24

22 Nov '24

alsa-project/alsa-ucm-conf issue #468 was opened from SentryII: Hi all, I've been searching for hours for a fix for this and im posting this as a last resort I've checked under If.realtek-alc4080 in the USB-Audio.conf and the X870-A is not listed IDVendor is 0b05 IDProduct is 1b9b 0b05:1b9b Line-Out works fine, issue is with Soundcard 2 USB-Audio Which is using optical / S/PDIF I've attached the alsa-info output below Thank you for your time [alsa-info.txt](https://github.com/user-attachments/files/17876970/alsa-info… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/468 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: [PATCH] ASoC: qcom: apq8016_sbc: Fix second argument of apq8016_dai_init()
by Stephan Gerhold 22 Nov '24

22 Nov '24

Hi, On Mon, Nov 18, 2024 at 12:20:58AM +0100, Bert Karwatzki wrote: > Since commit a78a42fb48b8 the second argument of apq8016_dai_init() has > to be an lpass id returned by qdsp6_dai_get_lpass_id(). > > Fixes: a78a42fb48b8 ("ASoC: qcom: apq8016_sbc: Allow routing audio through QDSP6") > > Signed-off-by: Bert Karwatzki <spasswolf(a)web.de> > --- > sound/soc/qcom/apq8016_sbc.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c > index 3023cf180a75..ddfcd4616895 100644 > --- a/sound/soc/qcom/apq8016_sbc.c > +++ b/sound/soc/qcom/apq8016_sbc.c > @@ -150,7 +150,7 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) > { > struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); > > - return apq8016_dai_init(rtd, cpu_dai->id); > + return apq8016_dai_init(rtd, qdsp6_dai_get_lpass_id(cpu_dai)); > } > Hm no, the code is correct the way it is. There are two cases handled by this driver: - ADSP bypass (lpass-* drivers, qcom,apq8016-sbc-sndcard): This one uses the LPASS IDs as DAI IDs, so cpu_dai->id already contains the correct value. - ADSP (qdsp6 drivers, qcom,msm8916-qdsp6-sndcard): This one uses the QDSP6 IDs (e.g. PRIMARY_MI2S_RX/TX instead of MI2S_PRIMARY), so qdsp6_dai_get_lpass_id() is used to translate that to the LPASS IDs. The code you are changing here handles the first case (ADSP bypass), the DAI IDs do not need any translation there. Thanks, Stephan

2 1

Set pcbeep to default value
by Kailang 22 Nov '24

22 Nov '24

Hi Takashi, Attach patch will set pcbeep to default value for ALC274. BR, Kailang

2 1

Re: [PATCH] ASoC: qcom: apq8016_sbc: Fix second argument of apq8016_dai_init()
by Mark Brown 21 Nov '24

21 Nov '24

On Mon, Nov 18, 2024 at 12:20:58AM +0100, Bert Karwatzki wrote: > Since commit a78a42fb48b8 the second argument of apq8016_dai_init() has > to be an lpass id returned by qdsp6_dai_get_lpass_id(). This breaks the build: /build/stage/linux/sound/soc/qcom/apq8016_sbc.c: In function ‘apq8016_sbc_dai_in it’: /build/stage/linux/sound/soc/qcom/apq8016_sbc.c:153:38: error: implicit declarat ion of function ‘qdsp6_dai_get_lpass_id’ [-Werror=implicit-function-declaration] 153 | return apq8016_dai_init(rtd, qdsp6_dai_get_lpass_id(cpu_dai)); | ^~~~~~~~~~~~~~~~~~~~~~ /build/stage/linux/sound/soc/qcom/apq8016_sbc.c: At top level: /build/stage/linux/sound/soc/qcom/apq8016_sbc.c:165:12: error: static declaratio n of ‘qdsp6_dai_get_lpass_id’ follows non-static declaration 165 | static int qdsp6_dai_get_lpass_id(struct snd_soc_dai *cpu_dai) | ^~~~~~~~~~~~~~~~~~~~~~ /build/stage/linux/sound/soc/qcom/apq8016_sbc.c:153:38: note: previous implicit declaration of ‘qdsp6_dai_get_lpass_id’ with type ‘int()’ 153 | return apq8016_dai_init(rtd, qdsp6_dai_get_lpass_id(cpu_dai)); | ^~~~~~~~~~~~~~~~~~~~~~ cc1: all warnings being treated as errors

1 0

Tiny sound on ASUS Zenbook UM5606 out of the box
by GitHub issues - opened 20 Nov '24

20 Nov '24

alsa-project/alsa-ucm-conf issue #467 was opened from IShashkin: ASUS Zenbook UM5606 doesn't use subwoofers out of the box. So sound is tiny on the laptop. There is a workaround described here https://wiki.archlinux.org/title/ASUS_Zenbook_UM5606#Audio but it's broken in the last alsa release. alsa-info.sh --no-upload [alsa-info.txt.zip](https://github.com/user-attachments/files/17834654/alsa-… Let me know if you need more info. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/467 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

sof-soundwire: cs42l43: Correct CapturePCM and routing
by GitHub pull_request - opened 20 Nov '24

20 Nov '24

alsa-project/alsa-ucm-conf pull request #466 was opened from charleskeepax: For headset microphone capture the correct PCM device to use is "hw:${CardId},1", "hw:${CardId},4" is for built in microphone capture. Adjust the routing as well since :0,1 is connected to data port 2 on the codec. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/466 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/466.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC225 ALC256 update depop procedure
by Kailang 20 Nov '24

20 Nov '24

Hi Takashi, ALC225 ALC256 update depop procedure. Many Thanks. BR, Kailang

2 1

alsactl: 90-alsa-restore.rules - fix AMD acp-pdm-mach link name acp-dmic-codec
by GitHub pull_request - edited 20 Nov '24

20 Nov '24

alsa-project/alsa-utils pull request #281 was edited from pma1: The link name of AMD ACP digital microphones driver acp-pdm-mach is acp-dmic-codec. fix it to make UCM initialize it again. Fixes: 8116639 ("alsactl: 90-alsa-restore.rules - add support for AMD ACP digital microphone") ``` # arecord -l **** List of CAPTURE Hardware Devices **** card 1: Generic_1 [HD-Audio Generic], device 0: ALC257 Analog [ALC257 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: acppdmmach [acp-pdm-mach], device 0: (null) dmic-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 # ls -la /sys/class/sound/controlC2/device/ total 0 drwxr-xr-x 5 root root 0 Nov 20 15:50 . drwxr-xr-x 3 root root 0 Nov 20 15:50 .. drwxr-xr-x 3 root root 0 Nov 20 15:47 controlC2 lrwxrwxrwx 1 root root 0 Nov 20 15:47 device -> ../../../acp-pdm-mach -rw-r--r-- 1 root root 4096 Nov 20 15:50 id -r--r--r-- 1 root root 4096 Nov 20 15:47 number drwxr-xr-x 3 root root 0 Nov 20 15:47 pcmC2D0c drwxr-xr-x 2 root root 0 Nov 20 15:50 power lrwxrwxrwx 1 root root 0 Nov 20 15:47 subsystem -> ../../../../../../../../class/sound -rw-r--r-- 1 root root 4096 Nov 20 15:47 uevent # ls -la /sys/class/sound/controlC2/device/device/ total 0 drwxr-xr-x 5 root root 0 Nov 20 15:50 . drwxr-xr-x 4 root root 0 Nov 20 15:50 .. drwxr-xr-x 3 root root 0 Nov 20 15:50 acp-dmic-codec lrwxrwxrwx 1 root root 0 Nov 20 15:47 driver -> ../../../../../../bus/platform/drivers/acp_mach -rw-r--r-- 1 root root 4096 Nov 20 15:50 driver_override -r--r--r-- 1 root root 4096 Nov 20 15:50 modalias drwxr-xr-x 2 root root 0 Nov 20 15:50 power drwxr-xr-x 3 root root 0 Nov 20 15:50 sound lrwxrwxrwx 1 root root 0 Nov 20 15:47 subsystem -> ../../../../../../bus/platform -rw-r--r-- 1 root root 4096 Nov 20 15:50 uevent ``` After the change of link name acp-dmic-codec, the micmute led can work on ThinkPad T14s Gen 6. Request URL : https://github.com/alsa-project/alsa-utils/pull/281 Patch URL : https://github.com/alsa-project/alsa-utils/pull/281.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

alsactl: 90-alsa-restore.rules - fix AMD acp-pdm-mach link name acp-dmic-codec
by GitHub pull_request - opened 20 Nov '24

20 Nov '24

alsa-project/alsa-utils pull request #281 was opened from pma1: The link name of AMD ACP digital microphones driver acp-pdm-mach is acp-dmic-codec. fix it to make UCM initialize it again. Fixes: 8116639 ("alsactl: 90-alsa-restore.rules - add support for AMD ACP digital microphone") # arecord -l **** List of CAPTURE Hardware Devices **** card 1: Generic_1 [HD-Audio Generic], device 0: ALC257 Analog [ALC257 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: acppdmmach [acp-pdm-mach], device 0: (null) dmic-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 # ls -la /sys/class/sound/controlC2/device/ total 0 drwxr-xr-x 5 root root 0 Nov 20 15:50 . drwxr-xr-x 3 root root 0 Nov 20 15:50 .. drwxr-xr-x 3 root root 0 Nov 20 15:47 controlC2 lrwxrwxrwx 1 root root 0 Nov 20 15:47 device -> ../../../acp-pdm-mach -rw-r--r-- 1 root root 4096 Nov 20 15:50 id -r--r--r-- 1 root root 4096 Nov 20 15:47 number drwxr-xr-x 3 root root 0 Nov 20 15:47 pcmC2D0c drwxr-xr-x 2 root root 0 Nov 20 15:50 power lrwxrwxrwx 1 root root 0 Nov 20 15:47 subsystem -> ../../../../../../../../class/sound -rw-r--r-- 1 root root 4096 Nov 20 15:47 uevent # ls -la /sys/class/sound/controlC2/device/device/ total 0 drwxr-xr-x 5 root root 0 Nov 20 15:50 . drwxr-xr-x 4 root root 0 Nov 20 15:50 .. drwxr-xr-x 3 root root 0 Nov 20 15:50 acp-dmic-codec lrwxrwxrwx 1 root root 0 Nov 20 15:47 driver -> ../../../../../../bus/platform/drivers/acp_mach -rw-r--r-- 1 root root 4096 Nov 20 15:50 driver_override -r--r--r-- 1 root root 4096 Nov 20 15:50 modalias drwxr-xr-x 2 root root 0 Nov 20 15:50 power drwxr-xr-x 3 root root 0 Nov 20 15:50 sound lrwxrwxrwx 1 root root 0 Nov 20 15:47 subsystem -> ../../../../../../bus/platform -rw-r--r-- 1 root root 4096 Nov 20 15:50 uevent After the change of link name acp-dmic-codec, the micmute led can work on ThinkPad T14s Gen 6. Request URL : https://github.com/alsa-project/alsa-utils/pull/281 Patch URL : https://github.com/alsa-project/alsa-utils/pull/281.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] Error patching file pcm_native.c
by Delusion's Master 20 Nov '24

20 Nov '24

Hi, first of all I apologize in case you receive multiple copies of this message, but I forgot to specify a subject in the original mail. I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

3 10

sof-soundwire: cs42l43-spk: Correct PlaybackPCM and routing
by GitHub pull_request - opened 19 Nov '24

19 Nov '24

alsa-project/alsa-ucm-conf pull request #465 was opened from ujfalusi: For speaker the correct PCM device to use is "hw:${CardId},2", the "hw:${CardId},0" is for headset playback. Adjust the routing as well since with :0,2 the DP6RX1/2 needs to be selected for the speaker. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/465 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/465.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC updates for v6.13
by Mark Brown 18 Nov '24

18 Nov '24

The following changes since commit 2ae6da569e34e1d26c5275442d17ffd75fd343b3: ASoC: max9768: Fix event generation for playback mute (2024-11-12 13:16:37 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.13 for you to fetch changes up to 82ff5abc2edcfba0c0f1a1be807795e2876f46e9: ASoC: hdmi-codec: reorder channel allocation list (2024-11-15 13:43:00 +0000) ---------------------------------------------------------------- ASoC: Updates for v6.13 This release was mainly about new drivers, there's a very large batch of new drivers and devices including quite a few from newer vendors which is great to see. Other than the new drivers and the usual routine fixes and enhancements the bulk of the work has been Morimoto-san's continuing work on simplifiying APIs, plus a few other bits: - More API simplifications from Morimoto-san. - Renaming of the sh directory to Renesas to reflect the focus on other architectures. - Factoring out of some of the common code for Realtek devices. - Support for Allwinner H616, AMD ACP 6.3 systems, AWInic AW88081, Cirrus Logic CS32L84, Everest ES8328, Iron Devices SMA1307, Longsoon I2S, NeoFidelity NTP8918 and NTP8835, Philips UDA1342, Qualcomm SM8750, RealTek RT721, and ST Microelectronics STM32MP25. ---------------------------------------------------------------- Advait Dhamorikar (1): ASoC: tlv320adc3xxx: Fix unsigned int compared against 0 Alexey Klimov (1): ASoC: qcom: sm8250: add handling of secondary MI2S clock Amadeusz Sławiński (2): ASoC: Intel: Remove unused code ASoC: Intel: avs: Fix return status of avs_pcm_hw_constraints_init() Andy Shevchenko (1): ASoC: codecs: wcd937x: Remove unused of_gpio.h Anne Onciulescu (1): ASoC: SOF: sof-of-dev: add parameter to override tplg/fw_filename Arnd Bergmann (3): ASoC: loongson: make loongson-i2s.o a separate module ASoC: cs42l84: remove incorrect of_match_ptr() ASoC: sma1307: fix uninitialized variable refence Balamurugan C (1): ASoC: Intel: sof_rt5682: Add HDMI-In capture with rt5682 support for MTL. Bard Liao (6): soundwire: sdw_intel: include linux/acpi.h ASoC: rt-sdw-common: fix rt_sdca_index_update_bits function parameter description ASoC: Intel: Kconfig: make SND_SOC_ACPI_INTEL_MATCH depend on ACPI ASoC: Intel: Kconfig: select SND_SOC_SDCA by SND_SOC_ACPI_INTEL_SDCA_QUIRKS ASoC: sdca: test adev before calling acpi_dev_for_each_child ASoC: Intel: soc-acpi-intel-lnl-match: add rt712_vb + rt1320 support Biju Das (1): ASoC: sh: rz-ssi: Use SSIFCR_FIFO_RST macro Binbin Zhou (7): ASoC: dt-bindings: Add Everest ES8323 Codec ASoC: codecs: Add support for ES8323 ASoC: dt-bindings: Add NXP uda1342 Codec ASoC: codecs: Add uda1342 codec driver ASoC: dt-bindings: Add Loongson I2S controller ASoC: loongson: Add I2S controller driver as platform device ASoC: loongson: Fix build warning when !CONFIG_PCI Chancel Liu (2): ASoC: imx-card: Set mclk for codec ASoC: imx-card: Add CS42888 support Charles Keepax (4): ASoC: sdw_utils: Add support for exclusion DAI quirks ASoC: sdw_utils: Add a quirk to allow the cs42l43 mic DAI to be ignored ASoC: Intel: sof_sdw: Add quirk for cs42l43 system using host DMICs ASoC: Intel: sof_sdw: Add quirks for some new Lenovo laptops Claudiu Beznea (3): ASoC: da7213: Populate max_register to regmap_config ASoC: da7213: Return directly the value of regcache_sync() ASoC: da7213: Add suspend to RAM support Codrin Ciubotariu (4): ASoC: atmel: atmel_ssc_dai: Add stream names ASoC: atmel: atmel_ssc_dai: Drop S24_LE support due to single channel limitation ASoC: atmel: mchp-spdiftx: Remove interface name from stream_name ASoC: atmel: mchp-spdifrx: Remove interface name from stream_name Colin Ian King (3): ASoC: codecs: aw88395: Fix spelling mistake "unsupport" -> "unsupported" ASoC: codecs: aw88399: Fix spelling mistake "unsupport" -> "unsupported" ASoC: SOF: ipc4-topology: remove redundant assignment to variable ret Cristian Ciocaltea (1): ASoC: dt-bindings: everest,es8328: Document audio graph port Daniel Baluta (1): ASoC: SOF: ipc3: Use standard dev_dbg API Deep Harsora (1): ASoC: Intel: sof_sdw: Add missing quirks from some new Dell Dharageswari R (2): ASoC: Intel: sof_rt5682: Add support for ptl_max98360a_rt5682 ASoC: intel: sof_sdw: add RT722 SDCA card for PTL platform Dr. David Alan Gilbert (1): ASoC: soc-devres: Remove unused devm_snd_soc_register_dai Everest K.C (2): ASoC: rt721-sdca: Clean logically deadcode in rt721-sdca.c ASoC: rt722-sdca: Remove logically deadcode in rt722-sdca.c Fabio Estevam (1): ASoC: dt-bindings: fsl_spdif: Document imx6sl/sx compatible fallback Fei Shao (3): ASoC: dt-bindings: mediatek,mt8188-mt6359: Add mediatek,adsp property ASoC: dt-bindings: everest,es8326: Document interrupt property ASoC: dt-bindings: maxim,max98390: Reference common DAI properties Frank Li (2): ASoC: dt-bindings: fsl-esai: Add power-domains for fsl,imx8qm-esai ASoC: dt-bindings: fsl-esai: allow fsl,imx8qm-esai fallback to fsl,imx6ull-esai Geert Uytterhoeven (2): ASoC: fsl-asoc-card: Add missing handling of {hp,mic}-dt-gpios ASoC: dt-bindings: Deprecate {hp,mic}-det-gpio Hao Bui (2): ASoC: da7213: Avoid setting PLL when closing audio stream ASoC: da7213: Extend support for the MCK in range [2, 50] MHz Harshit Mogalapalli (1): ASoC: codecs: fix error code in ntp8835_i2c_probe() Heiko Stuebner (1): ASoC: dt-bindings: rockchip,rk3036-codec: convert to yaml Hendrik v. Raven (2): ASoc: simple-mux: add idle-state support ASoC: dt-bindings: simple-mux: add idle-state property Herve Codina (1): ASoC: fsl: fsl_qmc_audio: Remove the logging when parsing channels Hongbo Li (5): ASoC: improve macro definition on TWL4030_OUTPUT_PGA ASoC: remove unused substream in macro soc_component_mark_pop ASoC: remove unused substream in macro soc_dai_mark_pop ASoC: remove unused substream in macro soc_link_mark_pop ASoC: stm: fix macro definition on STM_SAI_HAS_EXT_SYNC Igor Prusov (6): dt-bindings: vendor-prefixes: Add NeoFidelity, Inc ASoC: codecs: Add NeoFidelity Firmware helpers ASoC: dt-bindings: Add NeoFidelity NTP8918 ASoC: codecs: Add NeoFidelity NTP8918 codec ASoC: dt-bindings: Add NeoFidelity NTP8835 ASoC: codecs: Add NeoFidelity NTP8835 codec Ingyu Jang (2): ASoC: uniphier: Handle regmap_write errors in aio_src_set_param() ASoC: uniphier: Handle regmap_write errors in aio_iecout_set_enable() Jack Yu (6): ASoC: rt-sdw-common: Common functions for Realtek soundwire driver ASoC: rt721-sdca: Add RT721 SDCA driver ASoC: rt721-sdca: Fix issue of warning message ASoC: Intel: Add rt721-sdca support for PTL platform ASoC: rt-sdw-common: Enhance switch case to prevent uninitialized variable ASoC: rt721-sdca: change interrupt mask from XU to GE James Calligeros (1): ASoC: cs42l84: leverage ring sense IRQs to correctly detect headsets Javier Carrasco (1): ASoC: constify snd_soc_component_driver struct Jerome Brunet (1): ASoC: meson: axg-iface: set continuous rates Jiapeng Chong (1): ASoC: cs42l84: Remove unused including <linux/version.h> John Watts (1): ASoC: test-component: Support continuous rates for test component Jonas Karlman (1): ASoC: hdmi-codec: reorder channel allocation list Kiseok Jo (2): ASoC: dt-bindings: irondevice,sma1307: Add initial DT ASoC: sma1307: Add driver for Iron Device SMA1307 Krzysztof Kozlowski (5): ASoC: qcom: sm8250: correct typo in shutdown function name ASoC: dt-bindings: qcom: Add SM8750 LPASS macro codecs ASoC: qcom: x1e80100: Support boards with two speakers ASoC: dt-bindings: qcom,sm8250: Add SM8750 sound card ASoC: qcom: sc8280xp Add SM8750 sound card Kuninori Morimoto (24): ASoC: makes snd_soc_set_runtime_hwparams() inline ASoC: amd: remove dpcm_xxx flags ASoC: fsl: remove dpcm_xxx flags ASoC: sof: remove dpcm_xxx flags ASoC: intel: remove dpcm_xxx flags ASoC: samsung: remove dpcm_xxx flags ASoC: mediatek: remove dpcm_xxx flags ASoC: soc-core: remove dpcm_xxx flags ASoC: soc-compress: remove dpcm_xxx flags ASoC: soc-topology: remove dpcm_xxx flags ASoC: intel: boards: remove dpcm_xxx flags ASoC: soc-pcm: remove dpcm_xxx flags ASoC: doc: remove dpcm_xxx flags ASoC: soc-pcm: merge DPCM and non-DPCM validation check ASoC: rename rtd->num to rtd->id ASoC: fsl: switch to use rtd->id from rtd->num ASoC: meson: switch to use rtd->id from rtd->num ASoC: sh: switch to use rtd->id from rtd->num ASoC: generic: switch to use rtd->id from rtd->num ASoC: remove rtd->num ASoC: soc-core: do rtd->id trick at snd_soc_add_pcm_runtime() ASoC: cleanup function parameter for rtd and its id ASoC: add symmetric_ prefix for dai->rate/channels/sample_bits ASoC: simple-card-utils: care simple_util_dai for dummy DAI Lad Prabhakar (6): ASoC: rsnd: Refactor port handling with helper for endpoint node selection ASoC: Rename "sh" to "renesas" ASoC: renesas, rsnd: Update file path ASoC: audio-graph-card2: Update comment with renamed file path MAINTAINERS: Add entry for Renesas R-Car and FSI ASoC drivers MAINTAINERS: Add entry for Renesas RZ ASoC driver Liu Jing (1): ASoC: rx651: Use card->dev in replace of the &pdev->dev argument in the dev_err function Mac Chiang (3): ASoC: sdw_utils: Add quirk to exclude amplifier function ASoC: intel: sof_sdw: add quirk for Dell SKU ASoC: Intel: soc-acpi-intel-arl-match: Add rt722 and rt1320 support Macpaul Lin (1): ASoC: dt-bindings: mt6359: Update generic node name and dmic-mode Marcus Cooper (2): ASoC: sun4i-codec: Add support for different DAC FIFOC addresses to quirks ASoC: sun4i-codec: Add playback only flag to quirks Marek Vasut (3): ASoC: max98088: Remove duplicate DACs ASoC: max98088: Add left/right DAC volume control ASoC: max98088: Add headphone mixer switch Mario Limonciello (1): ASoC: amd: acp: Add stream name to ACP PDM DMIC devices Mark Brown (41): ASoC: bcm2835-i2s: Use maple tree register cache ASoC: fsl: Use maple tree register cache ASoC: fsl_micfil: fix and improvement ASoC: amd: acp: refactor acp version ASoC: improve some macro definitions and usages Updates for Atmel SSC DAI ASoC: Updates for mchp-spdif(tx/rx) {ASoC,media}: constify snd_soc_component_driver ASoC: Add NTP8918 and NTP8835 codecs support ASoC: Clean up {hp,mic}-det-gpio handling ASoC: Intel: sof_rt5682: add supports for new ASoC: Merge up v6.12 Fixes and improvements related to amd soundwire ASoC: Intel: add rt722/rt721 support for PTL platform ASoC: SOF: ipc4-topology: Rework the module audio ASoC: Some issues about loongson i2s ASoC/SoundWire: Simply interrupt enabling for Intel ASoC: imx-card: add cs42888 codec support ASoC: Merge up fixes ASoC: add support for some new Lenovo laptops with ASoC/soundwire: add initial support for SDCA ASoC: fsl_xcvr: enable some interrupts ASoC: SOF: core: Module parameter updates ASoC: remove dpcm_xxx flags ASoC: add Allwinner H616 audio codec support ASoC: add CS42L84 codec driver ASoC: Intel: add rt722 and rt1320 support ASoC: codecs: adau1373: drop platform data ASoC: Rename "sh" to "renesas" ASoC: cleasnup rtd and its ID Soundwire dai link init logic refactor ASoC: codecs: Add aw88081 amplifier driver ASoC: Intel: Kconfig: fix undefined symbol: ASoC: SOF: ipc3-loader: Handle PROBE_INFO ext_manifest Add a driver for the Iron Device SMA1307 Amp ASoC: Intel: Fix Kconfig for ACPI=n ASoC: stm32: sai: add stm32mp25 support ASoC: stm32: i2s: add stm32mp25 support ASoC: Merge up fixes Add generic AMD Soundwire machine driver for Legacy(No ASoc: simple-mux: Allow to specify an idle-state Markus Elfring (1): ASoC: tas5805m: Improve a size determination in tas5805m_i2c_probe() Martin Povišer (2): ASoC: cs42l84: Add new codec driver ASoC: dt-bindings: Add CS42L84 codec Masahiro Yamada (1): ASoC: codecs: wcd9335: remove unnecessary MODULE_ALIAS() Murad Masimov (1): ASoC: amd: acp: drop bogus NULL check from i2s_irq_handler Naveen Manohar (2): ASoC: intel/sdw_utils: refactor RT multifunction sdca speaker codecs ASoC: Intel: soc-acpi-intel-ptl-match: Add rt721 support Neil Armstrong (1): ASoC: dt-bindings: realtek,rt5640: Convert to dtschema Nuno Sa (5): ASoC: codecs: adau1372: add match table ASoC: codecs: adau1373: add some kconfig text ASoC: dt-bindings: document the adau1373 Codec ASoC: codecs: adau1373: drop patform_data ASoC: codecs: adau1373: add powerdown gpio Olivier Moysan (6): ASoC: dt-bindings: add stm32mp25 support for sai ASoC: stm32: sai: add stm32mp25 support ASoC: dt-bindings: add stm32mp25 support for i2s ASoC: stm32: i2s: add stm32mp25 support ASoC: dt-bindings: stm32: add missing port property ASoC: stm32: dfsdm: change rate upper limits Peter Ujfalusi (16): ASoC: SOF: ipc4-topology: Drop the 'index' from 'Pin index' of format print ASoC: SOF: ipc4-topology: Use local variables in sof_ipc4_init_input_audio_fmt() ASoC: SOF: ipc4-topology: Remove redundant check in sof_ipc4_init_input_audio_fmt() ASoC: SOF: ipc4-topology: Simplify match format print in sof_ipc4_init_input_audio_fmt() ASoC: SOF: ipc4-topology: Use local variables in sof_ipc4_init_output_audio_fmt() ASoC: SOF: ipc4-topology: Simplify code to deal with process modules without output ASoC: SOF: ipc4-topology: Concentrate prints inside of sof_ipc4_init_output_audio_fmt() ASoC: SOF: ipc4-topology: Add helper function to print the module's in/out audio format ASoC: SOF: core: Add debug module parameters to set IPC and boot timeout ASoC: SOF: core: Add fw, tplg and ipc type override parameters ASoC: SOF: ext_manifest: Add missing ext_manifest type for PROBE_INFO ASoC: SOF: ipc3-loader: 'Handle' PROBE_INFO ext_manifest type when parsing ASoC: SOF: Intel: hda-stream: Always use at least two BDLE for transfers ASoC: SOF: Intel: hda: Add support for persistent Code Loader DMA buffers ASoC: Intel: Kconfig: Only select SND_SOC_SDCA if ACPI is enabled ASoC: Intel: Kconfig: Revert make SND_SOC_ACPI_INTEL_MATCH depend on ACPI Pierre-Louis Bossart (12): ASoC: SOF: Intel: hda-mlink: expose unlocked interrupt enable routine ASoC/SoundWire: Intel: lnl: enable interrupts after first power-up/before last power-down ASoC/soundwire: remove sdw_slave_extended_id ASoC: SDCA: add initial module soundwire: slave: lookup SDCA version and functions ASoC: SDCA: add quirk function for RT712_VB match ASoC: rt712-sdca: detect the SMART_MIC function during the probe stage ASoC: soc-acpi: introduce new 'machine check' callback ASoC: sdw_utils: add SmartMic DAI for RT712 VB ASoC: sdw_utils: add SmartMic DAI for RT713 VB ASoC: Intel: soc-acpi: add is_device_rt712_vb() helper ASoC: SOF: Intel: hda: use machine_check() for SoundWire Qiu-ji Chen (1): ASoC: codecs: Fix atomicity violation in snd_soc_component_get_drvdata() Ranjani Sridharan (1): ASoC: SOF: Intel: hda: handle only paused streams in hda_dai_suspend() Ritu Chaudhary (1): ASoC: tegra: Add support for S24_LE audio format Ryan Roberts (1): ASoC: soc-utils: Remove PAGE_SIZE compile-time constant assumption Ryan Walklin (2): ASoC: sun4i-codec: support allwinner H616 codec ASoC: dt-bindings: allwinner: add H616 sun4i audio codec binding Shengjiu Wang (6): ASoC: fsl_micfil: fix regmap_write_bits usage ASoC: fsl_micfil: Add mclk enable flag ASoC: fsl_micfil: Enable micfil error interrupt ASoC: fsl_xcvr: enable interrupt of cmdc status update ASoC: fsl_xcvr: reset RX dpath after wrong preamble ASoC: fsl_mqs: Support accessing registers by scmi interface Shuming Fan (4): ASoC: rt1320: reads patch code from firmware file ASoC: rt1320: fix the range of patch code address ASoC: rt1320: add mic function ASoC: rt722: change the interrupt mask for jack type detection Stanislav Jakubek (2): ASoC: dt-bindings: sprd,pcm-platform: convert to YAML ASoC: dt-bindings: sprd,sc9860-mcdt: convert to YAML Suraj Sonawane (2): ASoC: bcm63xx-pcm-whistler: fix uninit-value in i2s_dma_isr ASoc: SOF: ipc4-pcm: fix uninit-value in sof_ipc4_pcm_dai_link_fixup_rate Tang Bin (7): ASoC: tas2781: Fix redundant parameter assignment ASoC: codecs: Fix error check in es8323_i2c_probe ASoC: mediatek: mt8188: Remove unnecessary variable assignments ASoC: mediatek: mt8188: remove unnecessary variable assignment ASoC: ux500: Remove redundant casts ASoC: mediatek: mt8183: Remove unnecessary variable assignments ASoC: sma1307: Fix invalid logical judgement Venkata Prasad Potturu (1): ASoC: amd: acp: Fix for ACP SOF dmic tplg component load failure Vijendar Mukunda (20): ASoC: amd: acp: simplify platform conditional checks code ASoC: amd: acp: use acp_rev for platform specific conditional checks ASoC: amd: acp: use acp pci revision id for platform differntiation ASoC: amd: acp: store acp pci rev id in platform driver private structure ASoC: amd: acp: pass acp pci revision id as platform data ASoC: amd: acp: update mach_params subsystem_rev field ASoC: amd: acp: remove unused variable from acp_card_drvdata structure ASoC: amd: acp: replace adata->platform conditional check ASoC: amd: acp: remove unused variable from acp platform driver ASoC: amd: acp: fix for inconsistent indenting ASoC: amd: acp: fix for cpu dai index logic ASoC: amd: acp: refactor sof_card_dai_links_create() function ASoC: sdw_utils/intel/amd: refactor dai link init logic ASoC: sdw_utils: Update stream_name in dai_links structure ASoC: amd: acp: add rt722 based soundwire machines ASoC: amd: acp: add RT711, RT714 & RT1316 support for acp 6.3 platform ASoC: amd: ps: add soundwire machines for acp6.3 platform ASoC: amd: acp: move get_acp63_cpu_pin_id() to common file ASoC: amd: acp: add soundwire machine driver for legacy stack ASoC: amd: ps: fix the pcm device numbering for acp 6.3 platform Vitaliy Shevtsov (1): ASoC: nau8821: check regmap_raw_read/regmap_raw_write for failure Weidong Wang (2): ASoC: dt-bindings: Add schema for "awinic,aw88081" ASoC: codecs: Add aw88081 amplifier driver Yang Li (1): ASoC: codecs: Remove unneeded semicolon Zhang Yi (2): ASoC: codecs: ES8326: Modify the configuration of and micbias ASoC: codecs: ES8326: Reduce pop noise Zhu Jun (1): ASoC: aw88399: Correct error handling in aw_dev_get_dsp_status function anish kumar (2): ASoC: doc: update clock api details ASoC: machine: update documentation .mailmap | 17 +- CREDITS | 58 +- Documentation/admin-guide/LSM/ipe.rst | 7 +- Documentation/admin-guide/kernel-parameters.txt | 2 +- Documentation/admin-guide/mm/transhuge.rst | 2 +- Documentation/admin-guide/pm/cpufreq.rst | 20 +- Documentation/core-api/protection-keys.rst | 38 +- .../bindings/display/elgin,jg10309-01.yaml | 54 + .../bindings/display/mediatek/mediatek,dpi.yaml | 24 +- .../bindings/display/mediatek/mediatek,split.yaml | 19 + .../devicetree/bindings/firmware/arm,scmi.yaml | 2 +- .../devicetree/bindings/iio/adc/adi,ad7380.yaml | 21 + .../devicetree/bindings/iio/dac/adi,ad5686.yaml | 53 +- .../devicetree/bindings/iio/dac/adi,ad5696.yaml | 3 +- .../interrupt-controller/fsl,ls-extirq.yaml | 26 +- .../devicetree/bindings/misc/fsl,qoriq-mc.yaml | 2 +- .../devicetree/bindings/net/brcm,unimac-mdio.yaml | 1 + .../devicetree/bindings/net/xlnx,axi-ethernet.yaml | 2 +- .../bindings/phy/qcom,sc8280xp-qmp-pcie-phy.yaml | 5 +- .../devicetree/bindings/sound/adi,adau1373.yaml | 111 + .../bindings/sound/allwinner,sun4i-a10-codec.yaml | 53 +- .../devicetree/bindings/sound/audio-graph.yaml | 6 + .../devicetree/bindings/sound/awinic,aw88395.yaml | 4 +- .../devicetree/bindings/sound/cirrus,cs42l84.yaml | 56 + .../devicetree/bindings/sound/everest,es8316.yaml | 4 +- .../devicetree/bindings/sound/everest,es8326.yaml | 4 + .../devicetree/bindings/sound/everest,es8328.yaml | 4 + .../devicetree/bindings/sound/fsl,esai.yaml | 28 +- .../devicetree/bindings/sound/fsl,spdif.yaml | 27 +- .../devicetree/bindings/sound/inno-rk3036.txt | 20 - .../bindings/sound/irondevice,sma1307.yaml | 53 + .../bindings/sound/loongson,ls2k1000-i2s.yaml | 68 + .../devicetree/bindings/sound/maxim,max98390.yaml | 8 +- .../bindings/sound/mediatek,mt8188-mt6359.yaml | 7 + .../devicetree/bindings/sound/mt6359.yaml | 10 +- .../bindings/sound/neofidelity,ntp8835.yaml | 73 + .../bindings/sound/neofidelity,ntp8918.yaml | 70 + .../devicetree/bindings/sound/nxp,uda1342.yaml | 42 + .../bindings/sound/qcom,lpass-rx-macro.yaml | 1 + .../bindings/sound/qcom,lpass-tx-macro.yaml | 1 + .../bindings/sound/qcom,lpass-va-macro.yaml | 1 + .../bindings/sound/qcom,lpass-wsa-macro.yaml | 1 + .../devicetree/bindings/sound/qcom,sm8250.yaml | 1 + .../devicetree/bindings/sound/realtek,rt5640.yaml | 146 + .../devicetree/bindings/sound/renesas,rsnd.txt | 2 +- .../bindings/sound/rockchip,rk3036-codec.yaml | 58 + 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| 5 +- Documentation/networking/tcp_ao.rst | 20 +- Documentation/process/maintainer-netdev.rst | 17 + Documentation/process/maintainer-soc.rst | 42 +- Documentation/rust/arch-support.rst | 2 +- Documentation/scheduler/sched-ext.rst | 2 +- Documentation/sound/soc/clocking.rst | 12 + Documentation/sound/soc/dpcm.rst | 11 +- Documentation/sound/soc/machine.rst | 26 + Documentation/userspace-api/mseal.rst | 307 +- Documentation/virt/kvm/api.rst | 16 +- Documentation/virt/kvm/locking.rst | 2 +- MAINTAINERS | 247 +- Makefile | 2 +- arch/Kconfig | 26 +- arch/arm/boot/dts/broadcom/bcm2837-rpi-cm3-io3.dts | 2 +- arch/arm/boot/dts/rockchip/rk3036-kylin.dts | 4 +- arch/arm/boot/dts/rockchip/rk3036.dtsi | 14 +- arch/arm64/Kconfig | 1 + arch/arm64/boot/dts/freescale/imx8-ss-lvds0.dtsi | 12 +- arch/arm64/boot/dts/freescale/imx8-ss-vpu.dtsi | 4 +- .../dts/freescale/imx8mp-phyboard-pollux-rdk.dts | 12 + .../freescale/imx8mp-skov-revb-mi1010ait-1cp1.dts | 1 + arch/arm64/boot/dts/freescale/imx8mp.dtsi | 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265 +- kernel/sched/ext.h | 2 +- kernel/sched/fair.c | 52 +- kernel/sched/sched.h | 39 +- kernel/sched/stats.h | 48 +- kernel/sched/syscalls.c | 13 +- kernel/signal.c | 3 +- kernel/task_work.c | 15 +- kernel/time/posix-clock.c | 3 + kernel/time/tick-sched.c | 6 + kernel/trace/bpf_trace.c | 42 +- kernel/trace/fgraph.c | 41 +- kernel/trace/ring_buffer.c | 53 +- kernel/trace/trace.c | 10 +- kernel/trace/trace_eprobe.c | 7 +- kernel/trace/trace_fprobe.c | 6 +- kernel/trace/trace_kprobe.c | 6 +- kernel/trace/trace_probe.c | 2 +- kernel/trace/trace_selftest.c | 2 +- kernel/trace/trace_uprobe.c | 13 +- kernel/ucount.c | 9 +- lib/Kconfig.debug | 2 +- lib/buildid.c | 5 + lib/codetag.c | 3 + lib/crypto/mpi/mpi-mul.c | 2 +- lib/iov_iter.c | 25 +- lib/maple_tree.c | 14 +- lib/objpool.c | 18 +- lib/slub_kunit.c | 2 +- mm/Kconfig | 1 - mm/damon/core.c | 42 +- mm/damon/tests/sysfs-kunit.h | 1 + mm/filemap.c | 2 +- mm/gup.c | 33 +- mm/huge_memory.c | 71 +- mm/internal.h | 55 +- mm/kasan/init.c | 8 +- mm/kasan/kasan_test_c.c | 27 - mm/khugepaged.c | 6 +- mm/memcontrol-v1.c | 25 + mm/memcontrol.c | 9 +- mm/memory.c | 32 +- mm/migrate.c | 9 +- mm/mlock.c | 9 +- mm/mmap.c | 229 +- mm/mprotect.c | 2 +- mm/mremap.c | 11 +- mm/nommu.c | 9 +- mm/numa_memblks.c | 2 +- mm/page_alloc.c | 26 +- mm/page_io.c | 4 +- mm/pagewalk.c | 16 +- mm/rmap.c | 9 +- mm/secretmem.c | 4 +- mm/shmem.c | 12 +- mm/shrinker.c | 8 +- mm/slab_common.c | 33 +- mm/sparse-vmemmap.c | 5 + mm/swap.c | 4 +- mm/swapfile.c | 58 +- mm/vma.c | 37 +- mm/vma.h | 32 +- mm/vmscan.c | 110 +- mm/zswap.c | 1 - net/9p/Kconfig | 2 + net/9p/client.c | 12 +- net/bluetooth/af_bluetooth.c | 25 + net/bluetooth/bnep/core.c | 3 +- net/bluetooth/hci_conn.c | 3 + net/bluetooth/hci_core.c | 24 +- net/bluetooth/hci_sync.c | 30 +- net/bluetooth/iso.c | 24 +- net/bluetooth/rfcomm/sock.c | 2 - net/bluetooth/sco.c | 18 +- net/bpf/test_run.c | 1 + net/bridge/br_netfilter_hooks.c | 5 + net/bridge/br_netlink.c | 6 +- net/bridge/br_private.h | 5 +- net/bridge/br_vlan.c | 19 +- net/core/dev.c | 4 + net/core/dst.c | 17 +- net/core/filter.c | 50 +- net/core/rtnetlink.c | 33 +- net/core/sock_map.c | 12 + net/dsa/user.c | 11 +- net/ipv4/devinet.c | 35 +- net/ipv4/inet_connection_sock.c | 21 +- net/ipv4/ip_tunnel.c | 2 +- net/ipv4/netfilter/nft_fib_ipv4.c | 4 +- net/ipv4/tcp_bpf.c | 7 +- net/ipv4/tcp_input.c | 42 +- net/ipv4/tcp_output.c | 5 +- net/ipv4/udp.c | 4 +- net/ipv4/xfrm4_policy.c | 40 +- net/ipv6/netfilter/nf_reject_ipv6.c | 15 +- net/ipv6/netfilter/nft_fib_ipv6.c | 5 +- net/ipv6/udp.c | 4 +- net/ipv6/xfrm6_policy.c | 31 +- net/l2tp/l2tp_netlink.c | 4 +- net/mac80211/Kconfig | 2 +- net/mac80211/cfg.c | 25 +- net/mac80211/ieee80211_i.h | 10 +- net/mac80211/key.c | 42 +- net/mac80211/link.c | 7 +- net/mac80211/main.c | 2 + net/mctp/af_mctp.c | 6 +- net/mctp/device.c | 30 +- net/mctp/neigh.c | 31 +- net/mctp/route.c | 33 +- net/mpls/af_mpls.c | 32 +- net/mptcp/mib.c | 3 + net/mptcp/mib.h | 3 + net/mptcp/mptcp_pm_gen.c | 1 - net/mptcp/pm_netlink.c | 6 +- net/mptcp/pm_userspace.c | 3 +- net/mptcp/protocol.c | 26 +- net/mptcp/protocol.h | 1 + net/mptcp/subflow.c | 17 +- net/netfilter/nf_bpf_link.c | 7 +- net/netfilter/nf_tables_api.c | 41 +- net/netfilter/nft_payload.c | 3 + net/netfilter/x_tables.c | 2 +- net/netfilter/xt_CHECKSUM.c | 33 +- net/netfilter/xt_CLASSIFY.c | 16 +- net/netfilter/xt_CONNSECMARK.c | 36 +- net/netfilter/xt_CT.c | 106 +- net/netfilter/xt_IDLETIMER.c | 59 +- net/netfilter/xt_LED.c | 39 +- net/netfilter/xt_NFLOG.c | 36 +- net/netfilter/xt_RATEEST.c | 39 +- net/netfilter/xt_SECMARK.c | 27 +- net/netfilter/xt_TRACE.c | 36 +- net/netfilter/xt_addrtype.c | 15 +- net/netfilter/xt_cluster.c | 33 +- net/netfilter/xt_connbytes.c | 4 +- net/netfilter/xt_connlimit.c | 39 +- net/netfilter/xt_connmark.c | 28 +- net/netfilter/xt_mark.c | 42 +- net/netlink/af_netlink.c | 3 +- net/netlink/genetlink.c | 28 +- net/phonet/pn_netlink.c | 28 +- net/rxrpc/ar-internal.h | 2 +- net/rxrpc/conn_client.c | 4 + net/rxrpc/io_thread.c | 10 +- net/rxrpc/local_object.c | 2 +- net/rxrpc/sendmsg.c | 10 +- net/sched/act_api.c | 23 +- net/sched/cls_api.c | 1 + net/sched/sch_api.c | 9 +- net/sched/sch_generic.c | 8 +- net/sched/sch_taprio.c | 21 +- net/sctp/sm_statefuns.c | 2 +- net/sctp/socket.c | 18 +- net/smc/af_smc.c | 4 +- net/smc/smc_ib.c | 8 +- net/smc/smc_inet.c | 11 + net/smc/smc_pnet.c | 6 +- net/smc/smc_wr.c | 6 +- net/socket.c | 7 +- net/sunrpc/svc.c | 11 +- net/sunrpc/xprtrdma/ib_client.c | 1 + net/sunrpc/xprtsock.c | 1 + net/vmw_vsock/virtio_transport.c | 8 +- net/vmw_vsock/virtio_transport_common.c | 14 +- net/vmw_vsock/vsock_bpf.c | 8 - net/wireless/core.c | 8 + net/wireless/nl80211.c | 8 +- net/wireless/scan.c | 4 + net/xfrm/xfrm_device.c | 11 +- net/xfrm/xfrm_policy.c | 53 +- net/xfrm/xfrm_user.c | 10 +- rust/kernel/device.rs | 15 +- rust/kernel/firmware.rs | 2 +- scripts/Kconfig.include | 3 + scripts/Makefile.compiler | 14 +- scripts/Makefile.package | 7 +- scripts/kconfig/menu.c | 13 +- scripts/mod/file2alias.c | 12 +- scripts/mod/sumversion.c | 5 +- scripts/package/builddeb | 2 +- scripts/package/install-extmod-build | 6 +- scripts/package/mkdebian | 10 +- scripts/rustc-llvm-version.sh | 22 + security/ipe/Kconfig | 19 + security/ipe/policy.c | 18 +- security/keys/keyring.c | 7 +- security/keys/trusted-keys/trusted_dcp.c | 9 +- security/security.c | 10 +- sound/Kconfig | 2 +- sound/core/ump.c | 2 +- sound/firewire/amdtp-stream.c | 3 + sound/firewire/tascam/amdtp-tascam.c | 2 +- sound/hda/intel-sdw-acpi.c | 33 +- sound/pci/hda/Kconfig | 2 +- sound/pci/hda/patch_conexant.c | 21 +- sound/pci/hda/patch_cs8409.c | 5 +- sound/pci/hda/patch_realtek.c | 162 +- sound/soc/Kconfig | 3 +- sound/soc/Makefile | 3 +- sound/soc/amd/acp-da7219-max98357a.c | 20 +- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/Kconfig | 29 + sound/soc/amd/acp/Makefile | 4 + sound/soc/amd/acp/acp-i2s.c | 38 +- sound/soc/amd/acp/acp-legacy-common.c | 24 +- sound/soc/amd/acp/acp-legacy-mach.c | 8 +- sound/soc/amd/acp/acp-mach-common.c | 63 +- sound/soc/amd/acp/acp-mach.h | 12 +- sound/soc/amd/acp/acp-pci.c | 7 +- sound/soc/amd/acp/acp-pdm.c | 2 +- sound/soc/amd/acp/acp-platform.c | 18 +- sound/soc/amd/acp/acp-rembrandt.c | 4 +- sound/soc/amd/acp/acp-renoir.c | 4 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 486 ++ sound/soc/amd/acp/acp-sdw-mach-common.c | 64 + sound/soc/amd/acp/acp-sdw-sof-mach.c | 104 +- sound/soc/amd/acp/acp-sof-mach.c | 6 +- sound/soc/amd/acp/acp63.c | 4 +- sound/soc/amd/acp/acp70.c | 14 +- sound/soc/amd/acp/acp_common.h | 19 + sound/soc/amd/acp/amd-acp63-acpi-match.c | 54 + sound/soc/amd/acp/amd.h | 9 +- sound/soc/amd/acp/soc_amd_sdw_common.h | 4 + sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/mach-config.h | 1 + sound/soc/amd/ps/pci-ps.c | 4 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 + sound/soc/amd/vangogh/acp5x-mach.c | 6 - sound/soc/atmel/atmel_ssc_dai.c | 5 +- sound/soc/atmel/mchp-spdifrx.c | 2 +- sound/soc/atmel/mchp-spdiftx.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/bcm/bcm2835-i2s.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 6 +- sound/soc/bcm/cygnus-pcm.c | 2 +- sound/soc/codecs/Kconfig | 69 +- sound/soc/codecs/Makefile | 20 + sound/soc/codecs/adau1372-i2c.c | 1 + sound/soc/codecs/adau1372-spi.c | 1 + sound/soc/codecs/adau1372.c | 8 + sound/soc/codecs/adau1372.h | 1 + sound/soc/codecs/adau1373.c | 200 +- sound/soc/codecs/aw88081.c | 1087 ++++ sound/soc/codecs/aw88081.h | 286 + sound/soc/codecs/aw88395/aw88395_device.c | 2 +- sound/soc/codecs/aw88395/aw88395_lib.c | 2 +- sound/soc/codecs/aw88399.c | 2 +- sound/soc/codecs/cpcap.c | 2 +- sound/soc/codecs/cs42l84.c | 1111 ++++ sound/soc/codecs/cs42l84.h | 210 + sound/soc/codecs/da7213.c | 27 +- sound/soc/codecs/da7213.h | 1 + sound/soc/codecs/da7219.c | 9 +- sound/soc/codecs/es8323.c | 792 +++ sound/soc/codecs/es8323.h | 78 + sound/soc/codecs/es8326.c | 20 +- sound/soc/codecs/hdmi-codec.c | 140 +- sound/soc/codecs/max98088.c | 86 +- sound/soc/codecs/nau8821.c | 9 +- sound/soc/codecs/ntp8835.c | 480 ++ sound/soc/codecs/ntp8918.c | 397 ++ sound/soc/codecs/ntpfw.c | 137 + sound/soc/codecs/ntpfw.h | 23 + sound/soc/codecs/pcm186x.c | 4 +- sound/soc/codecs/pcm5102a.c | 2 +- sound/soc/codecs/rt-sdw-common.c | 238 + sound/soc/codecs/rt-sdw-common.h | 66 + sound/soc/codecs/rt1320-sdw.c | 3668 ++--------- sound/soc/codecs/rt1320-sdw.h | 6 + sound/soc/codecs/rt712-sdca-sdw.c | 1 + sound/soc/codecs/rt712-sdca.c | 38 +- sound/soc/codecs/rt712-sdca.h | 1 + sound/soc/codecs/rt721-sdca-sdw.c | 546 ++ sound/soc/codecs/rt721-sdca-sdw.h | 150 + sound/soc/codecs/rt721-sdca.c | 1545 +++++ sound/soc/codecs/rt721-sdca.h | 269 + sound/soc/codecs/rt722-sdca-sdw.c | 12 +- sound/soc/codecs/rt722-sdca.c | 15 +- sound/soc/codecs/simple-mux.c | 39 +- sound/soc/codecs/sma1307.c | 2049 ++++++ sound/soc/codecs/sma1307.h | 444 ++ sound/soc/codecs/spdif_receiver.c | 2 +- sound/soc/codecs/spdif_transmitter.c | 2 +- sound/soc/codecs/tas2781-i2c.c | 35 +- sound/soc/codecs/tas5805m.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/codecs/tlv320adc3xxx.c | 2 +- sound/soc/codecs/twl4030.c | 12 +- sound/soc/codecs/uda1342.c | 347 + sound/soc/codecs/uda1342.h | 78 + sound/soc/codecs/wcd9335.c | 1 - sound/soc/codecs/wcd937x.c | 1 - sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl-asoc-card.c | 32 +- sound/soc/fsl/fsl_aud2htx.c | 2 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_micfil.c | 51 +- sound/soc/fsl/fsl_mqs.c | 41 + sound/soc/fsl/fsl_qmc_audio.c | 2 - sound/soc/fsl/fsl_xcvr.c | 94 +- sound/soc/fsl/fsl_xcvr.h | 5 + sound/soc/fsl/imx-audmix.c | 18 +- sound/soc/fsl/imx-card.c | 69 +- sound/soc/generic/audio-graph-card2.c | 2 +- sound/soc/generic/simple-card-utils.c | 16 +- sound/soc/generic/test-component.c | 2 +- sound/soc/intel/Kconfig | 8 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/avs/pcm.c | 2 +- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/ehl_rt5660.c | 14 +- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/sof_board_helpers.c | 15 +- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 9 +- sound/soc/intel/boards/sof_rt5682.c | 15 + sound/soc/intel/boards/sof_sdw.c | 117 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/intel/common/Makefile | 6 +- sound/soc/intel/common/soc-acpi-intel-arl-match.c | 63 + sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 65 + sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 58 + sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 75 +- .../soc/intel/common/soc-acpi-intel-sdca-quirks.c | 42 + .../soc/intel/common/soc-acpi-intel-sdca-quirks.h | 14 + sound/soc/intel/common/sst-dsp-priv.h | 101 - sound/soc/intel/common/sst-dsp.c | 250 - sound/soc/intel/common/sst-dsp.h | 61 - sound/soc/intel/common/sst-ipc.c | 294 - sound/soc/intel/common/sst-ipc.h | 86 - sound/soc/loongson/Kconfig | 32 +- sound/soc/loongson/Makefile | 9 +- sound/soc/loongson/loongson_i2s.c | 5 + sound/soc/loongson/loongson_i2s_plat.c | 185 + sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 +- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 34 +- sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 7 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 86 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 1 - sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 5 +- sound/soc/mediatek/mt8188/mt8188-dai-pcm.c | 2 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 +- .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 +- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 2 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 +- sound/soc/mediatek/mt8365/mt8365-dai-dmic.c | 6 +- sound/soc/mediatek/mt8365/mt8365-dai-pcm.c | 2 +- sound/soc/mediatek/mt8365/mt8365-mt6357.c | 14 +- sound/soc/meson/axg-card.c | 6 +- sound/soc/meson/axg-tdm-interface.c | 12 +- sound/soc/meson/axg-tdm.h | 2 - sound/soc/meson/gx-card.c | 2 +- sound/soc/qcom/sc8280xp.c | 1 + sound/soc/qcom/sm8250.c | 12 +- sound/soc/qcom/x1e80100.c | 40 +- sound/soc/{sh => renesas}/Kconfig | 0 sound/soc/{sh => renesas}/Makefile | 0 sound/soc/{sh => renesas}/dma-sh7760.c | 0 sound/soc/{sh => renesas}/fsi.c | 0 sound/soc/{sh => renesas}/hac.c | 0 sound/soc/{sh => renesas}/migor.c | 0 sound/soc/{sh => renesas}/rcar/Makefile | 0 sound/soc/{sh => renesas}/rcar/adg.c | 0 sound/soc/{sh => renesas}/rcar/cmd.c | 0 sound/soc/{sh => renesas}/rcar/core.c | 32 +- sound/soc/{sh => renesas}/rcar/ctu.c | 0 sound/soc/{sh => renesas}/rcar/debugfs.c | 0 sound/soc/{sh => renesas}/rcar/dma.c | 0 sound/soc/{sh => renesas}/rcar/dvc.c | 0 sound/soc/{sh => renesas}/rcar/gen.c | 0 sound/soc/{sh => renesas}/rcar/mix.c | 0 sound/soc/{sh => renesas}/rcar/rsnd.h | 0 sound/soc/{sh => renesas}/rcar/src.c | 0 sound/soc/{sh => renesas}/rcar/ssi.c | 0 sound/soc/{sh => renesas}/rcar/ssiu.c | 0 sound/soc/{sh => renesas}/rz-ssi.c | 6 +- sound/soc/{sh => renesas}/sh7760-ac97.c | 0 sound/soc/{sh => renesas}/siu.h | 0 sound/soc/{sh => renesas}/siu_dai.c | 0 sound/soc/{sh => renesas}/siu_pcm.c | 0 sound/soc/{sh => renesas}/ssi.c | 0 sound/soc/samsung/odroid.c | 11 +- sound/soc/sdca/Kconfig | 11 + sound/soc/sdca/Makefile | 5 + sound/soc/sdca/sdca_device.c | 67 + sound/soc/sdca/sdca_functions.c | 177 + sound/soc/sdw_utils/Makefile | 3 +- sound/soc/sdw_utils/soc_sdw_rt712_sdca.c | 48 - sound/soc/sdw_utils/soc_sdw_rt722_sdca.c | 41 - sound/soc/sdw_utils/soc_sdw_rt_mf_sdca.c | 90 + sound/soc/sdw_utils/soc_sdw_rt_sdca_jack_common.c | 8 + sound/soc/sdw_utils/soc_sdw_utils.c | 80 +- sound/soc/soc-acpi.c | 30 +- sound/soc/soc-component.c | 14 +- sound/soc/soc-compress.c | 28 +- sound/soc/soc-core.c | 70 +- sound/soc/soc-dai.c | 14 +- sound/soc/soc-devres.c | 37 - sound/soc/soc-link.c | 10 +- sound/soc/soc-pcm.c | 152 +- sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/soc-utils.c | 4 +- sound/soc/sof/amd/acp-common.c | 3 +- sound/soc/sof/core.c | 64 + sound/soc/sof/intel/hda-dai.c | 4 + sound/soc/sof/intel/hda-dsp.c | 5 - sound/soc/sof/intel/hda-loader.c | 100 +- sound/soc/sof/intel/hda-mlink.c | 18 + sound/soc/sof/intel/hda-stream.c | 32 +- sound/soc/sof/intel/hda.c | 27 +- sound/soc/sof/intel/hda.h | 14 +- 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2 1

[PATCH v3] ASoC: hdmi-codec: reorder channel allocation list
by Christian Hewitt 15 Nov '24

15 Nov '24

From: Jonas Karlman <jonas(a)kwiboo.se> The ordering in hdmi_codec_get_ch_alloc_table_idx() results in wrong channel allocation for a number of cases, e.g. when ELD reports FL|FR|LFE|FC|RL|RR or FL|FR|LFE|FC|RL|RR|RC|RLC|RRC: ca_id 0x01 with speaker mask FL|FR|LFE is selected instead of ca_id 0x03 with speaker mask FL|FR|LFE|FC for 4 channels and ca_id 0x04 with speaker mask FL|FR|RC gets selected instead of ca_id 0x0b with speaker mask FL|FR|LFE|FC|RL|RR for 6 channels Fix this by reordering the channel allocation list with most specific speaker masks at the top. Signed-off-by: Jonas Karlman <jonas(a)kwiboo.se> Signed-off-by: Christian Hewitt <christianshewitt(a)gmail.com> --- Changes since v2: - No changes, resending to ensure patch formatting is good Changes since v1: - Squash content from an additional related patch authored by Jonas since the original 2019 v1 submission [0] - Minor rewording in the description (nothing creditworthy) [0] https://patchwork.kernel.org/project/alsa-devel/patch/HE1PR06MB4011885AED9F… NB: This patch has been included with most Allwinner, Amlogic and Rockchip LibreELEC distro images since 2019 but has been somewhat forgotten about (despite the occasional nag from me to Jonas who is super-busy). However in the last month or so adding it to LibreELEC x86-64 and Rasbperry Pi images has also solved several 3.1/4.0 speaker placement problems reported in forums so it's time to send v2 on Jonas' behalf. sound/soc/codecs/hdmi-codec.c | 140 +++++++++++++++++++--------------- 1 file changed, 77 insertions(+), 63 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 74caae52e127..d9df29a26f4f 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -185,84 +185,97 @@ static const struct snd_pcm_chmap_elem hdmi_codec_8ch_chmaps[] = { /* * hdmi_codec_channel_alloc: speaker configuration available for CEA * - * This is an ordered list that must match with hdmi_codec_8ch_chmaps struct + * This is an ordered list where ca_id must exist in hdmi_codec_8ch_chmaps * The preceding ones have better chances to be selected by * hdmi_codec_get_ch_alloc_table_idx(). */ static const struct hdmi_codec_cea_spk_alloc hdmi_codec_channel_alloc[] = { { .ca_id = 0x00, .n_ch = 2, - .mask = FL | FR}, - /* 2.1 */ - { .ca_id = 0x01, .n_ch = 4, - .mask = FL | FR | LFE}, - /* Dolby Surround */ + .mask = FL | FR }, + { .ca_id = 0x03, .n_ch = 4, + .mask = FL | FR | LFE | FC }, { .ca_id = 0x02, .n_ch = 4, .mask = FL | FR | FC }, - /* surround51 */ + { .ca_id = 0x01, .n_ch = 4, + .mask = FL | FR | LFE }, { .ca_id = 0x0b, .n_ch = 6, - .mask = FL | FR | LFE | FC | RL | RR}, - /* surround40 */ - { .ca_id = 0x08, .n_ch = 6, - .mask = FL | FR | RL | RR }, - /* surround41 */ - { .ca_id = 0x09, .n_ch = 6, - .mask = FL | FR | LFE | RL | RR }, - /* surround50 */ + .mask = FL | FR | LFE | FC | RL | RR }, { .ca_id = 0x0a, .n_ch = 6, .mask = FL | FR | FC | RL | RR }, - /* 6.1 */ - { .ca_id = 0x0f, .n_ch = 8, - .mask = FL | FR | LFE | FC | RL | RR | RC }, - /* surround71 */ + { .ca_id = 0x09, .n_ch = 6, + .mask = FL | FR | LFE | RL | RR }, + { .ca_id = 0x08, .n_ch = 6, + .mask = FL | FR | RL | RR }, + { .ca_id = 0x07, .n_ch = 6, + .mask = FL | FR | LFE | FC | RC }, + { .ca_id = 0x06, .n_ch = 6, + .mask = FL | FR | FC | RC }, + { .ca_id = 0x05, .n_ch = 6, + .mask = FL | FR | LFE | RC }, + { .ca_id = 0x04, .n_ch = 6, + .mask = FL | FR | RC }, { .ca_id = 0x13, .n_ch = 8, .mask = FL | FR | LFE | FC | RL | RR | RLC | RRC }, - /* others */ - { .ca_id = 0x03, .n_ch = 8, - .mask = FL | FR | LFE | FC }, - { .ca_id = 0x04, .n_ch = 8, - .mask = FL | FR | RC}, - { .ca_id = 0x05, .n_ch = 8, - .mask = FL | FR | LFE | RC }, - { .ca_id = 0x06, .n_ch = 8, - .mask = FL | FR | FC | RC }, - { .ca_id = 0x07, .n_ch = 8, - .mask = FL | FR | LFE | FC | RC }, - { .ca_id = 0x0c, .n_ch = 8, - .mask = FL | FR | RC | RL | RR }, - { .ca_id = 0x0d, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | RC }, - { .ca_id = 0x0e, .n_ch = 8, - .mask = FL | FR | FC | RL | RR | RC }, - { .ca_id = 0x10, .n_ch = 8, - .mask = FL | FR | RL | RR | RLC | RRC }, - { .ca_id = 0x11, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | RLC | RRC }, + { .ca_id = 0x1f, .n_ch = 8, + .mask = FL | FR | LFE | FC | RL | RR | FLC | FRC }, { .ca_id = 0x12, .n_ch = 8, .mask = FL | FR | FC | RL | RR | RLC | RRC }, - { .ca_id = 0x14, .n_ch = 8, - .mask = FL | FR | FLC | FRC }, - { .ca_id = 0x15, .n_ch = 8, - .mask = FL | FR | LFE | FLC | FRC }, - { .ca_id = 0x16, .n_ch = 8, - .mask = FL | FR | FC | FLC | FRC }, - { .ca_id = 0x17, .n_ch = 8, - .mask = FL | FR | LFE | FC | FLC | FRC }, - { .ca_id = 0x18, .n_ch = 8, - .mask = FL | FR | RC | FLC | FRC }, - { .ca_id = 0x19, .n_ch = 8, - .mask = FL | FR | LFE | RC | FLC | FRC }, - { .ca_id = 0x1a, .n_ch = 8, - .mask = FL | FR | RC | FC | FLC | FRC }, - { .ca_id = 0x1b, .n_ch = 8, - .mask = FL | FR | LFE | RC | FC | FLC | FRC }, - { .ca_id = 0x1c, .n_ch = 8, - .mask = FL | FR | RL | RR | FLC | FRC }, - { .ca_id = 0x1d, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | FLC | FRC }, { .ca_id = 0x1e, .n_ch = 8, .mask = FL | FR | FC | RL | RR | FLC | FRC }, - { .ca_id = 0x1f, .n_ch = 8, - .mask = FL | FR | LFE | FC | RL | RR | FLC | FRC }, + { .ca_id = 0x11, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | RLC | RRC }, + { .ca_id = 0x1d, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | FLC | FRC }, + { .ca_id = 0x10, .n_ch = 8, + .mask = FL | FR | RL | RR | RLC | RRC }, + { .ca_id = 0x1c, .n_ch = 8, + .mask = FL | FR | RL | RR | FLC | FRC }, + { .ca_id = 0x0f, .n_ch = 8, + .mask = FL | FR | LFE | FC | RL | RR | RC }, + { .ca_id = 0x1b, .n_ch = 8, + .mask = FL | FR | LFE | RC | FC | FLC | FRC }, + { .ca_id = 0x0e, .n_ch = 8, + .mask = FL | FR | FC | RL | RR | RC }, + { .ca_id = 0x1a, .n_ch = 8, + .mask = FL | FR | RC | FC | FLC | FRC }, + { .ca_id = 0x0d, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | RC }, + { .ca_id = 0x19, .n_ch = 8, + .mask = FL | FR | LFE | RC | FLC | FRC }, + { .ca_id = 0x0c, .n_ch = 8, + .mask = FL | FR | RC | RL | RR }, + { .ca_id = 0x18, .n_ch = 8, + .mask = FL | FR | RC | FLC | FRC }, + { .ca_id = 0x17, .n_ch = 8, + .mask = FL | FR | LFE | FC | FLC | FRC }, + { .ca_id = 0x16, .n_ch = 8, + .mask = FL | FR | FC | FLC | FRC }, + { .ca_id = 0x15, .n_ch = 8, + .mask = FL | FR | LFE | FLC | FRC }, + { .ca_id = 0x14, .n_ch = 8, + .mask = FL | FR | FLC | FRC }, + { .ca_id = 0x0b, .n_ch = 8, + .mask = FL | FR | LFE | FC | RL | RR }, + { .ca_id = 0x0a, .n_ch = 8, + .mask = FL | FR | FC | RL | RR }, + { .ca_id = 0x09, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR }, + { .ca_id = 0x08, .n_ch = 8, + .mask = FL | FR | RL | RR }, + { .ca_id = 0x07, .n_ch = 8, + .mask = FL | FR | LFE | FC | RC }, + { .ca_id = 0x06, .n_ch = 8, + .mask = FL | FR | FC | RC }, + { .ca_id = 0x05, .n_ch = 8, + .mask = FL | FR | LFE | RC }, + { .ca_id = 0x04, .n_ch = 8, + .mask = FL | FR | RC }, + { .ca_id = 0x03, .n_ch = 8, + .mask = FL | FR | LFE | FC }, + { .ca_id = 0x02, .n_ch = 8, + .mask = FL | FR | FC }, + { .ca_id = 0x01, .n_ch = 8, + .mask = FL | FR | LFE }, }; struct hdmi_codec_priv { @@ -371,7 +384,8 @@ static int hdmi_codec_chmap_ctl_get(struct snd_kcontrol *kcontrol, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hdmi_codec_priv *hcp = info->private_data; - map = info->chmap[hcp->chmap_idx].map; + if (hcp->chmap_idx != HDMI_CODEC_CHMAP_IDX_UNKNOWN) + map = info->chmap[hcp->chmap_idx].map; for (i = 0; i < info->max_channels; i++) { if (hcp->chmap_idx == HDMI_CODEC_CHMAP_IDX_UNKNOWN) -- 2.34.1

1 0

Fix JackControl name
by GitHub pull_request - opened 15 Nov '24

15 Nov '24

alsa-project/alsa-ucm-conf pull request #464 was opened from lbilli: Not sure whether it makes a difference but that is the name reported by `amixer events`. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/464 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/464.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/6] Add generic AMD Soundwire machine driver for Legacy(No
by Vijendar Mukunda 14 Nov '24

14 Nov '24

This patch series add SoundWire machines for RT711, RT714, RT1316 and RT722 codecs for ACP 6.3 platform. Also, it adds a generic SoundWire machine driver code for legacy(No DSP) stack. Vijendar Mukunda (6): ASoC: amd: acp: add rt722 based soundwire machines ASoC: amd: acp: add RT711, RT714 & RT1316 support for acp 6.3 platform ASoC: amd: ps: add soundwire machines for acp6.3 platform ASoC: amd: acp: move get_acp63_cpu_pin_id() to common file ASoC: amd: acp: add soundwire machine driver for legacy stack ASoC: amd: ps: fix the pcm device numbering for acp 6.3 platform sound/soc/amd/acp/Kconfig | 29 ++ sound/soc/amd/acp/Makefile | 4 + sound/soc/amd/acp/acp-sdw-legacy-mach.c | 486 +++++++++++++++++++++++ sound/soc/amd/acp/acp-sdw-mach-common.c | 64 +++ sound/soc/amd/acp/acp-sdw-sof-mach.c | 49 +-- sound/soc/amd/acp/amd-acp63-acpi-match.c | 54 +++ sound/soc/amd/acp/soc_amd_sdw_common.h | 4 + sound/soc/amd/mach-config.h | 1 + sound/soc/amd/ps/pci-ps.c | 1 + sound/soc/amd/ps/ps-sdw-dma.c | 2 + 10 files changed, 646 insertions(+), 48 deletions(-) create mode 100644 sound/soc/amd/acp/acp-sdw-legacy-mach.c create mode 100644 sound/soc/amd/acp/acp-sdw-mach-common.c -- 2.34.1

2 7

[GIT PULL] ASoC fixes for v6.12-rc7
by Mark Brown 14 Nov '24

14 Nov '24

The following changes since commit 48b86532c10128cf50c854a90c2d5b1410f4012d: ASoC: SOF: sof-client-probes-ipc4: Set param_size extension bits (2024-11-07 15:35:52 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.12-rc7 for you to fetch changes up to 2ae6da569e34e1d26c5275442d17ffd75fd343b3: ASoC: max9768: Fix event generation for playback mute (2024-11-12 13:16:37 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.12 Some last updates for v6.12, one quirk plus a couple of fixes. One is a minor fix for a relatively obscure driver and the other is a relatively important fix for boot hangs with some audio graph based cards. ---------------------------------------------------------------- Deep Harsora (1): ASoC: intel: sof_sdw: add quirk for Dell SKU John Watts (1): ASoC: audio-graph-card2: Purge absent supplies for device tree nodes Mark Brown (1): ASoC: max9768: Fix event generation for playback mute sound/soc/codecs/max9768.c | 11 +++++++++-- sound/soc/generic/audio-graph-card2.c | 3 +++ sound/soc/intel/boards/sof_sdw.c | 8 ++++++++ 3 files changed, 20 insertions(+), 2 deletions(-)

2 1

src/Versions.in.in: Update *_tempo_base name
by GitHub pull_request - edited 14 Nov '24

14 Nov '24

alsa-project/alsa-lib pull request #421 was edited from nvinson: Change @SYMBOL_PREFIX@snd_has_tempo_base to @SYMBOL_PREFIX@snd_has_queue_tempo_base. Starting with version 1.2.13, alsa-lib fails to link with ld.lld-19 due to "version script assignment of 'ALSA_1.2.13' to symbol 'snd_seq_has_tempo_base' failed: symbol not defined". Per commit 769d1db1b0a213a39c7e59c0d1d724e7f45b1ac3 the correct name for the symbol is @SYMBOL_PREFIX@snd_has_queue_tempo_base; therefore, update src/Vesions.in.in to match. Fixes bug #420 Fixes Gentoo bug 943399 (https://bugs.gentoo.org/943399) Request URL : https://github.com/alsa-project/alsa-lib/pull/421 Patch URL : https://github.com/alsa-project/alsa-lib/pull/421.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

src/Versions.in.in: Update *_tempo_base name
by GitHub pull_request - opened 14 Nov '24

14 Nov '24

alsa-project/alsa-lib pull request #421 was opened from nvinson: Change @SYMBOL_PREFIX@snd_has_tempo_base to @SYMBOL_PREFIX@snd_has_queue_tempo_base. Starting with version 1.2.13, alsa-lib fails to link with ld.lld-19 due to "version script assignment of 'ALSA_1.2.13' to symbol 'snd_seq_has_tempo_base' failed: symbol not defined". Per commit 769d1db1b0a213a39c7e59c0d1d724e7f45b1ac3 the correct name for the symbol is @SYMBOL_PREFIX@snd_has_queue_tempo_base; therefore, update src/Vesions.in.in to match. Fixes bug 420 Fixes Gentoo bug 943399 (https://bugs.gentoo.org/943399) Request URL : https://github.com/alsa-project/alsa-lib/pull/421 Patch URL : https://github.com/alsa-project/alsa-lib/pull/421.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsa-lib-1.2.13: ld.lld: error: version script assignment of 'ALSA_1.2.13' to symbol 'snd_seq_has_tempo_base' failed: symbol not defined
by GitHub issues - opened 14 Nov '24

14 Nov '24

alsa-project/alsa-lib issue #420 was opened from nvinson: libasound.so fails to link with ld.lld with error ```ld.lld: error: version script assignment of 'ALSA_1.2.13' to symbol 'snd_seq_has_tempo_base' failed: symbol not defined``` Bug: [Gentoo bug 943399](https://bugs.gentoo.org/9433399) Issue URL : https://github.com/alsa-project/alsa-lib/issues/420 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

udevd[142]: GOTO 'alsa_restore_std' has no matching label in: '/usr/lib/udev/rules.d/90-alsa-restore.rules'
by GitHub issues - opened 14 Nov '24

14 Nov '24

alsa-project/alsa-utils issue #280 was opened from sxzzsf: There is no 'alsa_restore_std' label in the udev rule file, is it misspelled alsa_restore_go? Issue URL : https://github.com/alsa-project/alsa-utils/issues/280 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

UCM2: Intel: sof-hda-dsp: Fix handling of empty sys_vendor string
by GitHub pull_request - opened 13 Nov '24

13 Nov '24

alsa-project/alsa-ucm-conf pull request #463 was opened from singalsu: The mistake in UCM syntax caused in alsaucm start error: ALSA lib ucm_cond.c:367:(if_eval) unknown If.Condition.Type Fixes: 13022a97711d ("sof-hda-dsp: Fix the case where sysfs dmi sys_vendor attribute is not set") Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/463 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/463.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

memory leakage detected in alsa-lib-1.2.9 in Linux-5.10.24
by GitHub issues - edited 13 Nov '24

13 Nov '24

alsa-project/alsa-lib issue #419 was edited from wangtao13: I wrote an ALSA application using alsa-lib-1.2.9, the main logic of the application is ``` static int init_pcm_play(snd_pcm_t **playback_handle...) { snd_pcm_hw_params_t *hw_params; snd_pcm_format_t format; snd_pcm_uframes_t frames, periodsize; int ret; //1. openPCM， if (0 > snd_pcm_open(playback_handle, "default", SND_PCM_STREAM_PLAYBACK, 0)) { return -1; } //2. snd_pcm_hw_params_t if (0 > snd_pcm_hw_params_malloc (&hw_params)) { return -1; } //3. hw_params if (0 > snd_pcm_hw_params_any (*playback_handle, hw_params)) { return -1; } //4. if (0 > snd_pcm_hw_params_set_access (*playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) { return -1; } //5. SND_PCM_FORMAT_U8,8 if (8 == bits_per_sample) { format = SND_PCM_FORMAT_U8; } if (16 == bits_per_sample) { format = SND_PCM_FORMAT_S16_LE; } if (0 > snd_pcm_hw_params_set_format (*playback_handle, hw_params, format)) { return -1; } //6. if (0 > snd_pcm_hw_params_set_rate_near (*playback_handle, hw_params, &rate, 0)) { return -1; } //7. if (0 > snd_pcm_hw_params_set_channels(*playback_handle, hw_params, channels)) { return -1; } frames = 64; periodsize = frames; ret = snd_pcm_hw_params_set_buffer_size_near(*playback_handle, hw_params, &periodsize); if (ret < 0) { printf("Unable to set buffer size %li : %s\n", frames * 2, snd_strerror(ret)); } periodsize /= channels; ret = snd_pcm_hw_params_set_period_size_near(*playback_handle, hw_params, &periodsize, 0); if (ret < 0) { printf("Unable to set period size %li : %s\n", periodsize, snd_strerror(ret)); } //8. set hw_params if (0 > snd_pcm_hw_params (*playback_handle, hw_params)) { return -1; } snd_pcm_uframes_t temp; snd_pcm_hw_params_get_buffer_size(hw_params, &temp); printf("Got buffer size: %ld\n", temp); snd_pcm_hw_params_get_period_size(hw_params, &temp, 0); printf("Got period size: %ld\n", temp); snd_pcm_hw_params_free(hw_params); ret = snd_pcm_prepare(playback_handle); if (ret < 0) { printf("snd_pcm_prepare err\n"); } return ret; } int sound_play(playback_handle, ...) { rc = snd_pcm_writei(playback_handle, buf, buflen); if (rc < 0) { if (rc == -EAGAIN) { usleep(1000); return -1; } snd_pcm_prepare(playback_handle); return -1; } else if (rc == 0) { usleep(10 * 1000); } return 0; } int app_exit(....) { snd_pcm_close(*playback_handle); snd_config_update_free_global(); } #define TEST_LOOPS 2 /* > 1, can trigger memory leakage */ int main_test() { init_pcm_play(&playback_handle, ...., ..); for (int i=0; i<2; i++) { pcm_exit(); init_pcm_play(&playback_handle); sound_play(playback_handle) } return 0; } ``` ``` ==5176==ERROR: LeakSanitizer: detected memory leaks Direct leak of 12792 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x7626c458 (/usr/lib/libasound.so.2+0x4c458) Direct leak of 12 byte(s) in 1 object(s) allocated from: #0 0x77901ad8 in __interceptor_posix_memalign /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:155 #1 0x7758b3f4 in av_malloc (/usr/lib/libavutil.so.56+0x3b3f4) Indirect leak of 151552 byte(s) in 37 object(s) allocated from: #0 0x77900acc in __interceptor_malloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:86 #1 0x76281230 in snd_pcm_mmap (/usr/lib/libasound.so.2+0x61230) Indirect leak of 37392 byte(s) in 114 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x7626c458 (/usr/lib/libasound.so.2+0x4c458) Indirect leak of 31200 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762b91b8 in snd_pcm_softvol_open (/usr/lib/libasound.so.2+0x991b8) Indirect leak of 15808 byte(s) in 76 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x76280d1c in snd_pcm_mmap (/usr/lib/libasound.so.2+0x60d1c) Indirect leak of 10032 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762b4e28 (/usr/lib/libasound.so.2+0x94e28) Indirect leak of 5168 byte(s) in 38 object(s) allocated from: #0 0x77900acc in __interceptor_malloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:86 #1 0x7627e644 (/usr/lib/libasound.so.2+0x5e644) Indirect leak of 3344 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x7627e4e0 (/usr/lib/libasound.so.2+0x5e4e0) Indirect leak of 1976 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762939ac in snd_pcm_plug_open (/usr/lib/libasound.so.2+0x739ac) Indirect leak of 1824 byte(s) in 76 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x76280d3c in snd_pcm_mmap (/usr/lib/libasound.so.2+0x60d3c) Indirect leak of 1596 byte(s) in 38 object(s) allocated from: #0 0x7786d584 in __interceptor_strdup /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_interceptors.cc:405 #1 0x762c09b8 in snd_timer_hw_open (/usr/lib/libasound.so.2+0xa09b8) Indirect leak of 1560 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762b7c58 (/usr/lib/libasound.so.2+0x97c58) Indirect leak of 1560 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762512d4 (/usr/lib/libasound.so.2+0x312d4) Indirect leak of 1520 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762c0978 in snd_timer_hw_open (/usr/lib/libasound.so.2+0xa0978) Indirect leak of 1419 byte(s) in 153 object(s) allocated from: #0 0x7786d584 in __interceptor_strdup /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_interceptors.cc:405 #1 0x7626c47c (/usr/lib/libasound.so.2+0x4c47c) Indirect leak of 468 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762572e8 in snd_ctl_hw_open (/usr/lib/libasound.so.2+0x372e8) Indirect leak of 304 byte(s) in 76 object(s) allocated from: #0 0x779011a4 in __interceptor_realloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:105 #1 0x762690c4 (/usr/lib/libasound.so.2+0x490c4) Indirect leak of 273 byte(s) in 1 object(s) allocated from: #0 0x779011a4 in __interceptor_realloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:105 #1 0x77569f3c in av_buffer_realloc (/usr/lib/libavutil.so.56+0x19f3c) Indirect leak of 195 byte(s) in 39 object(s) allocated from: #0 0x7786d584 in __interceptor_strdup /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_interceptors.cc:405 #1 0x762512fc (/usr/lib/libasound.so.2+0x312fc) Indirect leak of 28 byte(s) in 1 object(s) allocated from: #0 0x77901ad8 in __interceptor_posix_memalign /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:155 #1 0x7758b3f4 in av_malloc (/usr/lib/libavutil.so.56+0x3b3f4) SUMMARY: AddressSanitizer: 280023 byte(s) leaked in 997 allocation(s). ``` If the `TEST_LOOPS` is defined as 1, there is NO memory leakage reported by ASAN. Issue URL : https://github.com/alsa-project/alsa-lib/issues/419 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

memory leakage detected in alsa-lib-1.2.9 in Linux-5.10.24
by GitHub issues - opened 13 Nov '24

13 Nov '24

alsa-project/alsa-lib issue #419 was opened from wangtao13: I wrote an ALSA application using alsa-lib-1.2.9, the main logic of the application is ``` static int init_pcm_play(snd_pcm_t **playback_handle...) { snd_pcm_hw_params_t *hw_params; snd_pcm_format_t format; snd_pcm_uframes_t frames, periodsize; int ret; //1. openPCM， if (0 > snd_pcm_open(playback_handle, "default", SND_PCM_STREAM_PLAYBACK, 0)) { return -1; } //2. snd_pcm_hw_params_t if (0 > snd_pcm_hw_params_malloc (&hw_params)) { return -1; } //3. hw_params if (0 > snd_pcm_hw_params_any (*playback_handle, hw_params)) { return -1; } //4. if (0 > snd_pcm_hw_params_set_access (*playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) { return -1; } //5. SND_PCM_FORMAT_U8,8 if (8 == bits_per_sample) { format = SND_PCM_FORMAT_U8; } if (16 == bits_per_sample) { format = SND_PCM_FORMAT_S16_LE; } if (0 > snd_pcm_hw_params_set_format (*playback_handle, hw_params, format)) { return -1; } //6. if (0 > snd_pcm_hw_params_set_rate_near (*playback_handle, hw_params, &rate, 0)) { return -1; } //7. if (0 > snd_pcm_hw_params_set_channels(*playback_handle, hw_params, channels)) { return -1; } frames = 64; periodsize = frames; ret = snd_pcm_hw_params_set_buffer_size_near(*playback_handle, hw_params, &periodsize); if (ret < 0) { printf("Unable to set buffer size %li : %s\n", frames * 2, snd_strerror(ret)); } periodsize /= channels; ret = snd_pcm_hw_params_set_period_size_near(*playback_handle, hw_params, &periodsize, 0); if (ret < 0) { printf("Unable to set period size %li : %s\n", periodsize, snd_strerror(ret)); } //8. set hw_params if (0 > snd_pcm_hw_params (*playback_handle, hw_params)) { return -1; } snd_pcm_uframes_t temp; snd_pcm_hw_params_get_buffer_size(hw_params, &temp); printf("Got buffer size: %ld\n", temp); snd_pcm_hw_params_get_period_size(hw_params, &temp, 0); printf("Got period size: %ld\n", temp); snd_pcm_hw_params_free(hw_params); ret = snd_pcm_prepare(playback_handle); if (ret < 0) { printf("snd_pcm_prepare err\n"); } return ret; } int sound_play(playback_handle, ...) { rc = snd_pcm_writei(playback_handle, buf, buflen); if (rc < 0) { if (rc == -EAGAIN) { usleep(1000); return -1; } snd_pcm_prepare(playback_handle); return -1; } else if (rc == 0) { usleep(10 * 1000); } return 0; } int app_exit(....) { snd_pcm_close(*playback_handle); snd_config_update_free_global(); } int main_test() { init_pcm_play(&playback_handle, ...., ..); for (int i=0; i<2; i++) { pcm_exit(); init_pcm_play(&playback_handle); sound_play(playback_handle) } return 0; } ``` ``` ==5176==ERROR: LeakSanitizer: detected memory leaks Direct leak of 12792 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x7626c458 (/usr/lib/libasound.so.2+0x4c458) Direct leak of 12 byte(s) in 1 object(s) allocated from: #0 0x77901ad8 in __interceptor_posix_memalign /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:155 #1 0x7758b3f4 in av_malloc (/usr/lib/libavutil.so.56+0x3b3f4) Indirect leak of 151552 byte(s) in 37 object(s) allocated from: #0 0x77900acc in __interceptor_malloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:86 #1 0x76281230 in snd_pcm_mmap (/usr/lib/libasound.so.2+0x61230) Indirect leak of 37392 byte(s) in 114 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x7626c458 (/usr/lib/libasound.so.2+0x4c458) Indirect leak of 31200 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762b91b8 in snd_pcm_softvol_open (/usr/lib/libasound.so.2+0x991b8) Indirect leak of 15808 byte(s) in 76 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x76280d1c in snd_pcm_mmap (/usr/lib/libasound.so.2+0x60d1c) Indirect leak of 10032 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762b4e28 (/usr/lib/libasound.so.2+0x94e28) Indirect leak of 5168 byte(s) in 38 object(s) allocated from: #0 0x77900acc in __interceptor_malloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:86 #1 0x7627e644 (/usr/lib/libasound.so.2+0x5e644) Indirect leak of 3344 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x7627e4e0 (/usr/lib/libasound.so.2+0x5e4e0) Indirect leak of 1976 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762939ac in snd_pcm_plug_open (/usr/lib/libasound.so.2+0x739ac) Indirect leak of 1824 byte(s) in 76 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x76280d3c in snd_pcm_mmap (/usr/lib/libasound.so.2+0x60d3c) Indirect leak of 1596 byte(s) in 38 object(s) allocated from: #0 0x7786d584 in __interceptor_strdup /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_interceptors.cc:405 #1 0x762c09b8 in snd_timer_hw_open (/usr/lib/libasound.so.2+0xa09b8) Indirect leak of 1560 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762b7c58 (/usr/lib/libasound.so.2+0x97c58) Indirect leak of 1560 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762512d4 (/usr/lib/libasound.so.2+0x312d4) Indirect leak of 1520 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762c0978 in snd_timer_hw_open (/usr/lib/libasound.so.2+0xa0978) Indirect leak of 1419 byte(s) in 153 object(s) allocated from: #0 0x7786d584 in __interceptor_strdup /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_interceptors.cc:405 #1 0x7626c47c (/usr/lib/libasound.so.2+0x4c47c) Indirect leak of 468 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762572e8 in snd_ctl_hw_open (/usr/lib/libasound.so.2+0x372e8) Indirect leak of 304 byte(s) in 76 object(s) allocated from: #0 0x779011a4 in __interceptor_realloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:105 #1 0x762690c4 (/usr/lib/libasound.so.2+0x490c4) Indirect leak of 273 byte(s) in 1 object(s) allocated from: #0 0x779011a4 in __interceptor_realloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:105 #1 0x77569f3c in av_buffer_realloc (/usr/lib/libavutil.so.56+0x19f3c) Indirect leak of 195 byte(s) in 39 object(s) allocated from: #0 0x7786d584 in __interceptor_strdup /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_interceptors.cc:405 #1 0x762512fc (/usr/lib/libasound.so.2+0x312fc) Indirect leak of 28 byte(s) in 1 object(s) allocated from: #0 0x77901ad8 in __interceptor_posix_memalign /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:155 #1 0x7758b3f4 in av_malloc (/usr/lib/libavutil.so.56+0x3b3f4) SUMMARY: AddressSanitizer: 280023 byte(s) leaked in 997 allocation(s). ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/419 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] Revert "aplay: fix S24_LE wav header"
by Shengjiu Wang 13 Nov '24

13 Nov '24

S24_LE means the lower 24bit in 32bit space has valid data. So previous code is correct. The commit causes regression, then revert it. This reverts commit 9cdcbe0e932b97fac88a6195f87647159a6c7194. Fixes: 9cdcbe0e932b ("aplay: fix S24_LE wav header") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- aplay/aplay.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index c58c2e9..d11e63a 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -2719,11 +2719,11 @@ static void begin_wave(int fd, size_t cnt) case SND_PCM_FORMAT_S16_LE: bits = 16; break; - case SND_PCM_FORMAT_S24_LE: /* S24_LE is 24 bits stored in 32 bit width with 8 bit padding */ case SND_PCM_FORMAT_S32_LE: - case SND_PCM_FORMAT_FLOAT_LE: + case SND_PCM_FORMAT_FLOAT_LE: bits = 32; break; + case SND_PCM_FORMAT_S24_LE: case SND_PCM_FORMAT_S24_3LE: bits = 24; break; -- 2.34.1

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up on | Relevant PRs | Related issues | | ------------------| ---------------- | ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | #434 | ### Problem statement The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The "USB" audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ### Relevant files & the such * [alsa-info info dump](https://github.com/user-attachments/files/17721762/alsa-info.txt) * Screenshot of the other USB device * ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up on | Relevant PRs | Related issues | | ------------------| ---------------- | ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | #434 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The "USB" audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) ### Relevant files & the such [alsa-info.txt](https://github.com/user-attachments/files/17721762/alsa-info… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up on | Relevant PRs | Related issues | | ------------------| ---------------- | ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | #434 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The "USB" audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up | Relevant PR | Related issues | | ------------------| ---------------- | ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | #434 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The "USB" audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up | Relevant PR | | ------------------| ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The "USB" audio outs (edit: *for the ALC408x*) are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Clevo Platfrom with ALC255
by Kailang 12 Nov '24

12 Nov '24

Hi Takashi, Attach was let Clevo platform with ALC255 enable headset Mic. BR, Kailang

2 1

ALSA 1.2.8 release
by Jaroslav Kysela 12 Nov '24

12 Nov '24

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-lib alsa-utils alsa-ucm-conf tinycompress Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.7.2_v1.2.8 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 5

[PATCH AUTOSEL 5.4 4/5] ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div()
by Sasha Levin 12 Nov '24

12 Nov '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 23569c8b314925bdb70dd1a7b63cfe6100868315 ] 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 is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Reviewed-by: Olivier Moysan <olivier.moysan(a)foss.st.com> Link: https://patch.msgid.link/20241107015936.211902-1-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 | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index b7dc9d3192597..e8cd58c0838ee 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -318,7 +318,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); - if (div > SAI_XCR1_MCKDIV_MAX(version)) { + if (div > SAI_XCR1_MCKDIV_MAX(version) || div <= 0) { dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); return -EINVAL; } -- 2.43.0

1 0

[PATCH AUTOSEL 5.4 3/5] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate()
by Sasha Levin 12 Nov '24

12 Nov '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 7e965848796c3..b7dc9d3192597 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -379,8 +379,8 @@ 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

1 0

[PATCH AUTOSEL 5.10 5/6] ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div()
by Sasha Levin 12 Nov '24

12 Nov '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 23569c8b314925bdb70dd1a7b63cfe6100868315 ] 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 is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Reviewed-by: Olivier Moysan <olivier.moysan(a)foss.st.com> Link: https://patch.msgid.link/20241107015936.211902-1-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 | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3a7f0102b4c5c..90e4757f76b0f 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -319,7 +319,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); - if (div > SAI_XCR1_MCKDIV_MAX(version)) { + if (div > SAI_XCR1_MCKDIV_MAX(version) || div <= 0) { dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); return -EINVAL; } -- 2.43.0

1 0

[PATCH AUTOSEL 5.15 7/8] ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div()
by Sasha Levin 12 Nov '24

12 Nov '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 23569c8b314925bdb70dd1a7b63cfe6100868315 ] 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 is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Reviewed-by: Olivier Moysan <olivier.moysan(a)foss.st.com> Link: https://patch.msgid.link/20241107015936.211902-1-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 | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index aa9cdd93b5778..d71b4aecd269f 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -319,7 +319,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); - if (div > SAI_XCR1_MCKDIV_MAX(version)) { + if (div > SAI_XCR1_MCKDIV_MAX(version) || div <= 0) { dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); return -EINVAL; } -- 2.43.0

1 0

[PATCH AUTOSEL 5.15 6/8] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate()
by Sasha Levin 12 Nov '24

12 Nov '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 9c3b8e2096565..aa9cdd93b5778 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

1 0

[PATCH AUTOSEL 6.1 10/12] ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div()
by Sasha Levin 12 Nov '24

12 Nov '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 23569c8b314925bdb70dd1a7b63cfe6100868315 ] 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 is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Reviewed-by: Olivier Moysan <olivier.moysan(a)foss.st.com> Link: https://patch.msgid.link/20241107015936.211902-1-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 | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3d237f75e81f5..0629aa5f2fe4b 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -317,7 +317,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); - if (div > SAI_XCR1_MCKDIV_MAX(version)) { + if (div > SAI_XCR1_MCKDIV_MAX(version) || div <= 0) { dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); return -EINVAL; } -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 09/12] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate()
by Sasha Levin 12 Nov '24

12 Nov '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 eb31b49e65978..3d237f75e81f5 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -378,8 +378,8 @@ 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

1 0

[PATCH AUTOSEL 6.6 13/15] ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div()
by Sasha Levin 12 Nov '24

12 Nov '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 23569c8b314925bdb70dd1a7b63cfe6100868315 ] 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 is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Reviewed-by: Olivier Moysan <olivier.moysan(a)foss.st.com> Link: https://patch.msgid.link/20241107015936.211902-1-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 | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 1b61110cb9174..dcbcd1a59a3aa 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -317,7 +317,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); - if (div > SAI_XCR1_MCKDIV_MAX(version)) { + if (div > SAI_XCR1_MCKDIV_MAX(version) || div <= 0) { dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); return -EINVAL; } -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 12/15] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate()
by Sasha Levin 12 Nov '24

12 Nov '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 0acc848c1f004..1b61110cb9174 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -378,8 +378,8 @@ 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

1 0

[PATCH AUTOSEL 6.6 06/15] ASoC: tas2781: Add new driver version for tas2563 & tas2781 qfn chip
by Sasha Levin 12 Nov '24

12 Nov '24

From: Shenghao Ding <shenghao-ding(a)ti.com> [ Upstream commit fe09de2db2365eed8b44b572cff7d421eaf1754a ] Add new driver version to support tas2563 & tas2781 qfn chip Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> Link: https://patch.msgid.link/20241104100055.48-1-shenghao-ding@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tas2781-fmwlib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 629e2195a890b..1cc64ed8de6da 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2022,6 +2022,7 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, break; case 0x202: case 0x400: + case 0x401: tas_priv->fw_parse_variable_header = fw_parse_variable_header_git; tas_priv->fw_parse_program_data = -- 2.43.0

1 0

[PATCH AUTOSEL 6.11 14/16] ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div()
by Sasha Levin 12 Nov '24

12 Nov '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 23569c8b314925bdb70dd1a7b63cfe6100868315 ] 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 is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Reviewed-by: Olivier Moysan <olivier.moysan(a)foss.st.com> Link: https://patch.msgid.link/20241107015936.211902-1-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 | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 19307812ec765..64f52c75e2aa8 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -317,7 +317,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); - if (div > SAI_XCR1_MCKDIV_MAX(version)) { + if (div > SAI_XCR1_MCKDIV_MAX(version) || div <= 0) { dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); return -EINVAL; } -- 2.43.0

1 0

[PATCH AUTOSEL 6.11 13/16] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate()
by Sasha Levin 12 Nov '24

12 Nov '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 ad2492efb1cdc..19307812ec765 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -378,8 +378,8 @@ 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

1 0

[PATCH AUTOSEL 6.11 07/16] ASoC: tas2781: Add new driver version for tas2563 & tas2781 qfn chip
by Sasha Levin 12 Nov '24

12 Nov '24

From: Shenghao Ding <shenghao-ding(a)ti.com> [ Upstream commit fe09de2db2365eed8b44b572cff7d421eaf1754a ] Add new driver version to support tas2563 & tas2781 qfn chip Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> Link: https://patch.msgid.link/20241104100055.48-1-shenghao-ding@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tas2781-fmwlib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index f3a7605f07104..6474cc551d551 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1992,6 +1992,7 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, break; case 0x202: case 0x400: + case 0x401: tas_priv->fw_parse_variable_header = fw_parse_variable_header_git; tas_priv->fw_parse_program_data = -- 2.43.0

1 0

[RESEND PATCH v5 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 12 Nov '24

12 Nov '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20241002093904.180979… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

1 6

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up | Relevant PR | | ------------------| ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The "USB" audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up | Relevant PR | | ------------------| ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The USB audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up | Relevant PR | | ------------------| ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The USB audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: Following up on https://github.com/alsa-project/alsa-ucm-conf/issues/334 | Relevant PR: https://github.com/alsa-project/alsa-ucm-conf/pull/461 The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The USB audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ALC4082 - ASRock X870 Steel Legend
by GitHub issues - opened 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was opened from cybik: Following up on https://github.com/alsa-project/alsa-ucm-conf/issues/334 The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The USB audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Relevant PR: https://github.com/alsa-project/alsa-ucm-conf/pull/461 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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`amixer sset Master` doesnt increase and decrease both sides equally
by GitHub issues - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-utils issue #279 was opened from lquenti: I look through the issues, man pages and internet and hope I didn't miss anything. Please tell me if its a user error. Whenever I use `amixer sset Master` to increase volume, it will not increase them evenly. Here is a before ![before](https://github.com/user-attachments/assets/4ccfa220-608c-4407-9f6e-fc1d27f1a91b) and after ![after](https://github.com/user-attachments/assets/250ab254-d46a-44e4-a201-58f9224b6293) Interestingly amixer seem to think that it is still evenly balanced, although alsamixer shows that it is unbalanced afterwards. --- Interestingly, when decreasing, the unevenness flips to the other side. [Other people also seem to struggle with it](https://stackoverflow.com/questions/69826139/using-amixer-to-set-audio-… Thank you so much. If it is something somewhat simple to fix, I can also write a fix myself, but for now I am unsure what the problem is. Issue URL : https://github.com/alsa-project/alsa-utils/issues/279 Repository URL: https://github.com/alsa-project/alsa-utils

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Change 26ce:0a08 to list multiple motherboards
by GitHub pull_request - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-ucm-conf pull request #461 was opened from cybik: The hardware id `26ce:0a08` seems to be re-used by multiple motherboards at ASRock. Identify them within `[brackets]` to point out the specific boards in question. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/461 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/461.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ThinkPad P14s Gen 5 AMD micmute LED is not working sometimes.
by GitHub issues - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-utils issue #278 was opened from pma1: The mic and sound output works fine. The mute LED works fine. But when press Fn+F4, the mic-mute led is not working sometimes after bootup. The failrate is like 1/10. Tested with latest alsa-ucm from git, it's reproduced. /sys/class/sound/ctl-led/mic/card1# cat list 7 # amixer -c 1 controls numid=12,iface=CARD,name='Headphone Jack' numid=11,iface=CARD,name='Mic Jack' numid=13,iface=CARD,name='Speaker Phantom Jack' numid=10,iface=MIXER,name='Master Playback Switch' numid=9,iface=MIXER,name='Master Playback Volume' numid=2,iface=MIXER,name='Headphone Playback Switch' numid=1,iface=MIXER,name='Headphone Playback Volume' numid=16,iface=MIXER,name='PCM Playback Volume' numid=17,iface=MIXER,name='Mic ACP LED Capture Switch' numid=8,iface=MIXER,name='Mic Boost Volume' numid=7,iface=MIXER,name='Capture Switch' numid=6,iface=MIXER,name='Capture Volume' numid=5,iface=MIXER,name='Auto-Mute Mode' numid=4,iface=MIXER,name='Speaker Playback Switch' numid=3,iface=MIXER,name='Speaker Playback Volume' numid=15,iface=PCM,name='Capture Channel Map' numid=14,iface=PCM,name='Playback Channel Map' # echo 'Capture Switch' > detach # echo 'Mic ACP LED Capture Switch' > attach /sys/class/sound/ctl-led/mic/card1# cat list 17 Then the micmute LED will work. [alsa-info-amd-micmutled.txt](https://github.com/user-attachments/files/1627… Issue URL : https://github.com/alsa-project/alsa-utils/issues/278 Repository URL: https://github.com/alsa-project/alsa-utils

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USB-Audio: ALC4080: Add support for MSI MPG X870E CARBON WIFI (0db0:0b58)
by GitHub pull_request - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-ucm-conf pull request #460 was opened from xatier: This PR updates the regex for MSI MPG X870E CARBON WIFI motherboard. ``` $ lsusb | ag audio Bus 003 Device 007: ID 0db0:0b58 Micro Star International USB Audio ``` ``` Audio Realtek® ALC4080 Codec ``` https://www.msi.com/Motherboard/MPG-X870E-CARBON-WIFI/Specification Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/460 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/460.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 6.1.y] ALSA: hda/realtek: Fix headset mic on TUXEDO Gemini 17 Gen3
by Werner Sembach 09 Nov '24

09 Nov '24

From: Christoffer Sandberg <cs(a)tuxedo.de> Quirk is needed to enable headset microphone on missing pin 0x19. Signed-off-by: Christoffer Sandberg <cs(a)tuxedo.de> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> Link: https://patch.msgid.link/20241029151653.80726-1-wse@tuxedocomputers.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 a8bc95ffa41a3..da14ab97783f8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10001,6 +10001,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x1403, "Clevo N140CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1404, "Clevo N150CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x28c1, "Clevo V370VND", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.43.0

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Odd behavior with Tempotec Serenade PCIe (CM8888)
by GitHub issues - opened 09 Nov '24

09 Nov '24

alsa-project/alsa-lib issue #418 was opened from lss4: This card is powered by CM8888, using `snd-hda-intel` driver, and it does work in Linux, but its behavior is quite odd. 1. Both the RCA connection and 6.35 headphone jack on the card are connected at the moment. Preferably I'd like to be able to switch to use either RCA (to speakers) or headphone, but I can't -- Audio always goes through both outputs, and there's only one analog output from this card. 2. This is the odd behavior I'm talking about -- the card's outputs are muted by default. Playing an audio file will trigger the card to unmute the output after 2-3 seconds followed by an audible click from the relay on the sound card. When the output is active, if no audio is being actively played for at least 10 seconds, the output will be muted again on its own, and this is again followed by an audible click from the relay. Honestly I've never seen an audio card capable of doing something that "intelligent" before. `alsa-info.sh` output attached here. The sound card in question is `card1` (`hwC1D1`). [alsa-info-20241109.txt](https://github.com/user-attachments/files/17687665/… I'm currently using Garuda Linux on this system. Previously I was using Manjaro. Both are Arch-based, and the behaviors are the same. I'm not sure about the support of this chip (CM8888) in general, as the information I could find online about CM8888 on Linux is limited and mostly outdated (several years ago). Issue URL : https://github.com/alsa-project/alsa-lib/issues/418 Repository URL: https://github.com/alsa-project/alsa-lib

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[Question] About Roland Rubix series, alsa-ucm-conf and snd_usb_audio
by GitHub issues - opened 08 Nov '24

08 Nov '24

alsa-project/alsa-ucm-conf issue #459 was opened from sim590: I want to understand what `alsa_ucm-conf` is about exactly regarding the support for specific devices. I'm thinking to buy a Roland Rubix series audio interface (either 22 or 24) and I'm wondering if both my XLR inputs are gonna appear in alsa/pulseaudio/pipewire as separate input lines. Does that require a potential change in `snd_usb_audio` kernel module or only a configuration file on this repository? I've looked at the configuration files on this repository and I have found a Roland directory, but nothing about the Rubix series explicitly. Presently, I have a Focusrite Scarlett 2i2 *fake clone* (unfortunately) which I'm gonna return shortly because it's counterfeit. As of right now, I don't see the two separate outputs, but only the following input: ![image](https://github.com/user-attachments/assets/7bf58e54-67d1-4e20-989c-5e1ec0828850) which takes audio from my first XLR input port. I guess that's the class compliant interface behaviour. Correct me if I'm wrong, I'm trying to learn about all of this. So, I just wanna be sure that if I go and buy my Roland Rubix, that the device is actually gonna show up the two input lines, and if not that I will know what I will have to do to make it so. Is there gonna be a configuration file required in `alsa-ucm-conf`? Any potentiel change in `snd_usb_audio` kernel module? I'm a bit confused about the details regarding the support of the audio interface devices. Thanks. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/459 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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