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[PATCH RESEND] ASoC: rt5650: Eliminate the high frequency glitch
by derek.fang＠realtek.com 21 Jul '25

21 Jul '25

From: Derek Fang <derek.fang(a)realtek.com> The glitch was detected in the high frequency of the HP playback. This patch adjusts the DAC dither setting to avoid this situation for almost all cases. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 08df87238..29a403526 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -82,6 +82,7 @@ static const struct reg_sequence rt5650_init_list[] = { {0xf6, 0x0100}, {RT5645_PWR_ANLG1, 0x02}, {RT5645_IL_CMD3, 0x6728}, + {RT5645_PR_BASE + 0x3a, 0x0000}, }; static const struct reg_default rt5645_reg[] = { -- 2.34.1

2 1

Rename qcs6490-rb3gen2 and qcs9075-iq-evk ucm2 conf
by GitHub pull_request - opened 20 Jul '25

20 Jul '25

alsa-project/alsa-ucm-conf pull request #596 was opened from mohsRafi: Rename the ucm2 conf for Qualcomm qcs6490-rb3gen2 and qcs9075-iq-evk qcs6490-rb3gen2-snd-card.conf -> QCS6490-RB3Gen2.conf qcs9075-iq-evk-snd-card.conf -> QCS9075-IQ-EVK.conf Removed snd-card tags from conf files Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/596 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/596.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1] ALSA: hda: Add TAS2770 support
by Baojun Xu 19 Jul '25

19 Jul '25

Add TAS2770 support in TI's HDA driver. And add hda_chip_id for more product support in the future. Separated DSP and non-DSP in firmware load function. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2770-tlv.h | 23 +++++ sound/pci/hda/tas2781_hda_i2c.c | 157 ++++++++++++++++++++++---------- 2 files changed, 133 insertions(+), 47 deletions(-) create mode 100644 include/sound/tas2770-tlv.h diff --git a/include/sound/tas2770-tlv.h b/include/sound/tas2770-tlv.h new file mode 100644 index 000000000000..c0bd495b4a07 --- /dev/null +++ b/include/sound/tas2770-tlv.h @@ -0,0 +1,23 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2770 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2770 hda driver implements for one, two, or even multiple +// TAS2770 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2770_TLV_H__ +#define __TAS2770_TLV_H__ + +#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x17) +#define TAS2770_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_amp_tlv, 1100, 50, 0); + +#endif diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index d91eed9f7804..2fdff0f88af7 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -24,6 +24,7 @@ #include <sound/tas2781.h> #include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> +#include <sound/tas2770-tlv.h> #include <sound/tas2781-tlv.h> #include "hda_local.h" @@ -45,9 +46,18 @@ #define TAS2563_CAL_TLIM TASDEVICE_REG(0, 0x10, 0x14) #define TAS2563_CAL_R0 TASDEVICE_REG(0, 0x0f, 0x34) +enum device_chip_id { + HDA_TAS2563, + HDA_TAS2770, + HDA_TAS2781, + HDA_OTHERS +}; + struct tas2781_hda_i2c_priv { struct snd_kcontrol *snd_ctls[2]; int (*save_calibration)(struct tas2781_hda *h); + + int hda_chip_id; }; static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) @@ -246,6 +256,15 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static const struct snd_kcontrol_new tas2770_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2770_AMP_LEVEL, + 0, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, tas2770_amp_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2770_DVC_LEVEL, + 0, 0, 31, 0, tas2781_amp_getvol, + tas2781_amp_putvol, tas2770_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -254,7 +273,7 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_force_fwload_get, tas2781_force_fwload_put), }; -static const struct snd_kcontrol_new tas2781_prof_ctrl = { +static const struct snd_kcontrol_new tasdevice_prof_ctrl = { .name = "Speaker Profile Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = tasdevice_info_profile, @@ -262,7 +281,7 @@ static const struct snd_kcontrol_new tas2781_prof_ctrl = { .put = tasdevice_set_profile_id, }; -static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl = { +static const struct snd_kcontrol_new tasdevice_dsp_prog_ctrl = { .name = "Speaker Program Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = tasdevice_info_programs, @@ -270,7 +289,7 @@ static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl = { .put = tasdevice_program_put, }; -static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl = { +static const struct snd_kcontrol_new tasdevice_dsp_conf_ctrl = { .name = "Speaker Config Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = tasdevice_info_config, @@ -379,44 +398,15 @@ static void tas2781_hda_remove_controls(struct tas2781_hda *tas_hda) snd_ctl_remove(codec->card, tas_hda->prof_ctl); } -static void tasdev_fw_ready(const struct firmware *fmw, void *context) +static void tasdevice_dspfw_init(void *context) { struct tasdevice_priv *tas_priv = context; struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); struct tas2781_hda_i2c_priv *hda_priv = tas_hda->hda_priv; struct hda_codec *codec = tas_priv->codec; - int i, ret, spk_id; - - pm_runtime_get_sync(tas_priv->dev); - mutex_lock(&tas_priv->codec_lock); - - ret = tasdevice_rca_parser(tas_priv, fmw); - if (ret) - goto out; - - tas_hda->prof_ctl = snd_ctl_new1(&tas2781_prof_ctrl, tas_priv); - ret = snd_ctl_add(codec->card, tas_hda->prof_ctl); - if (ret) { - dev_err(tas_priv->dev, - "Failed to add KControl %s = %d\n", - tas2781_prof_ctrl.name, ret); - goto out; - } - - for (i = 0; i < ARRAY_SIZE(tas2781_snd_controls); i++) { - hda_priv->snd_ctls[i] = snd_ctl_new1(&tas2781_snd_controls[i], - tas_priv); - ret = snd_ctl_add(codec->card, hda_priv->snd_ctls[i]); - if (ret) { - dev_err(tas_priv->dev, - "Failed to add KControl %s = %d\n", - tas2781_snd_controls[i].name, ret); - goto out; - } - } + int ret, spk_id; tasdevice_dsp_remove(tas_priv); - tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; if (tas_priv->speaker_id != NULL) { // Speaker id need to be checked for ASUS only. @@ -442,48 +432,101 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) dev_err(tas_priv->dev, "dspfw load %s error\n", tas_priv->coef_binaryname); tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - goto out; + return; } - tas_hda->dsp_prog_ctl = snd_ctl_new1(&tas2781_dsp_prog_ctrl, + tas_hda->dsp_prog_ctl = snd_ctl_new1(&tasdevice_dsp_prog_ctrl, tas_priv); ret = snd_ctl_add(codec->card, tas_hda->dsp_prog_ctl); if (ret) { dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", - tas2781_dsp_prog_ctrl.name, ret); - goto out; + tasdevice_dsp_prog_ctrl.name, ret); + return; } - tas_hda->dsp_conf_ctl = snd_ctl_new1(&tas2781_dsp_conf_ctrl, + tas_hda->dsp_conf_ctl = snd_ctl_new1(&tasdevice_dsp_conf_ctrl, tas_priv); ret = snd_ctl_add(codec->card, tas_hda->dsp_conf_ctl); if (ret) { dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", - tas2781_dsp_conf_ctrl.name, ret); - goto out; + tasdevice_dsp_conf_ctrl.name, ret); + return; } - tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + tasdevice_prmg_load(tas_priv, 0); if (tas_priv->fmw->nr_programs > 0) tas_priv->cur_prog = 0; if (tas_priv->fmw->nr_configurations > 0) tas_priv->cur_conf = 0; - /* If calibrated data occurs error, dsp will still works with default * calibrated data inside algo. */ hda_priv->save_calibration(tas_hda); +} + +static void tasdev_add_kcontrols(struct tasdevice_priv *tas_priv, + struct snd_kcontrol **ctls, struct hda_codec *codec, + const struct snd_kcontrol_new *tas_snd_ctrls, int num_ctls) +{ + int i, ret; + + for (i = 0; i < num_ctls; i++) { + ctls[i] = snd_ctl_new1( + &tas_snd_ctrls[i], tas_priv); + ret = snd_ctl_add(codec->card, ctls[i]); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas_snd_ctrls[i].name, ret); + break; + } + } +} + +static void tasdev_fw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); + struct tas2781_hda_i2c_priv *hda_priv = tas_hda->hda_priv; + struct hda_codec *codec = tas_priv->codec; + int ret; + + pm_runtime_get_sync(tas_priv->dev); + mutex_lock(&tas_priv->codec_lock); - tasdevice_tuning_switch(tas_hda->priv, 0); - tas_hda->priv->playback_started = true; + ret = tasdevice_rca_parser(tas_priv, fmw); + if (ret) + goto out; + + tas_priv->fw_state = TASDEVICE_RCA_FW_OK; + tasdev_add_kcontrols(tas_priv, &tas_hda->prof_ctl, codec, + &tasdevice_prof_ctrl, 1); + + switch (hda_priv->hda_chip_id) { + case HDA_TAS2770: + tasdev_add_kcontrols(tas_priv, hda_priv->snd_ctls, codec, + &tas2770_snd_controls[0], + ARRAY_SIZE(tas2770_snd_controls)); + break; + case HDA_TAS2781: + tasdev_add_kcontrols(tas_priv, hda_priv->snd_ctls, codec, + &tas2781_snd_controls[0], + ARRAY_SIZE(tas2781_snd_controls)); + tasdevice_dspfw_init(context); + break; + case HDA_TAS2563: + tasdevice_dspfw_init(context); + break; + default: + break; + } out: mutex_unlock(&tas_hda->priv->codec_lock); release_firmware(fmw); - pm_runtime_mark_last_busy(tas_hda->dev); pm_runtime_put_autosuspend(tas_hda->dev); } @@ -584,15 +627,34 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) return -ENOMEM; if (strstr(dev_name(&clt->dev), "TIAS2781")) { + /* + * TAS2781, integrated on-chip DSP with + * global I2C address supported. + */ device_name = "TIAS2781"; + hda_priv->hda_chip_id = HDA_TAS2781; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; + } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW2770")) { + /* + * TAS2770, has no on-chip DSP, so no calibration data + * required; has no global I2C address supported. + */ + device_name = "TXNW2770"; + hda_priv->hda_chip_id = HDA_TAS2770; } else if (strstr(dev_name(&clt->dev), "INT8866")) { + /* + * TAS2563, integrated on-chip DSP with + * global I2C address supported. + */ device_name = "INT8866"; + hda_priv->hda_chip_id = HDA_TAS2563; hda_priv->save_calibration = tas2563_save_calibration; tas_hda->priv->global_addr = TAS2563_GLOBAL_ADDR; - } else + } else { return -ENODEV; + } + tas_hda->priv->irq = clt->irq; ret = tas2781_read_acpi(tas_hda->priv, device_name); @@ -724,6 +786,7 @@ static const struct i2c_device_id tas2781_hda_i2c_id[] = { static const struct acpi_device_id tas2781_acpi_hda_match[] = { {"TIAS2781", 0 }, {"INT8866", 0 }, + {"TXNW2770", 0 }, {} }; MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); -- 2.34.1

3 2

Internal Microphone not working
by GitHub issues - opened 19 Jul '25

19 Jul '25

alsa-project/alsa-lib issue #464 was opened from crashthepose: After updating my computer my internal microphone is not working. Originally when I went to my Audio settings it would say under Input "headset unplugged". I was able to go into the HDAjackretask interface and override one of the pins where now the internal microphone is showing on the input list but now I only hear static noise. Info on my computer: OS: Pop!_OS 22.04 LTS x86_64 Host: ASUSLaptop_Q530VJ 1.0 Kernal: 6.12.10-76061203-generic Desktop Environment: GNOME 42.9 Terminal: gnome-terminal CPU: 13th Gen Intel i7-13620H (16) @ 4.700GHz GPU: Intel Device a7a8 GPU: NVIDIA 01:00.0 NVIDIA Corporation Device 25ac Memory: 7828MiB / 15610MiB Here is my alsa info: http://alsa-project.org/db/?f=ed8089460c52b3b91175d52dfd51c4bad1f7388f I've never submitted a bug before so apologies if i'm not doing this the correct way. Issue URL : https://github.com/alsa-project/alsa-lib/issues/464 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

speaker-test 1.2.13 dose not produce 20kHz sine wav
by GitHub issues - opened 17 Jul '25

17 Jul '25

alsa-project/alsa-utils issue #304 was opened from gharig: Hello Alsa-Dev, I hope that I’m writing to the proper place to report a problem, please advise if not… I’m not able to produce 20kHz sine wav with speaker-test 1.2.13 Running on Tinycore 16.0 . Below is what I submitted to Tinycore Team, and they suggested that I contact you with my findings. The hardware I’m using is Winsystems E3825 dual-core, 1.33GHz SATA 2.0 channel mSATA socket (shared with miniPCIe) 2 GB of Soldered Down DDR3 [https://winsystems.com/product/ppm-c407/](url) Tinycore Team; My problem today is that speaker-test does not produce a 20kHz sine wave on Tinycore 16.0 to the line out. speaker-test does produce something just not a 20kHz sine wave. Code: ` speaker-test -t sine -f 20000 -X -l 0` However, using the exact same hardware running Ubuntu 20.24 or 22.04 the following code works and a 20kHz sine wave is produced. The difference being that the alsa versions are different between the 3 Operating Systems. Tinycore 16.0 Alsa version is 1.2.13 - failure Ubuntu 20.04 Alsa version is 1.2.2 - success Ubuntu 22.04 Alsa version is 1.2.6 - success Through my testing, starting at 8000 Hz, and incrementing by 1000 Hz, speaker-test does produce a sine wave, until I went over 15000 Hz then no more sine wave. :) :) Code: ``` speaker-test -t sine -f 8000 -X -l 0 speaker-test -t sine -f 15000 -X -l 0 ``` :( Code: ` speaker-test -t sine -f 16000 -X -l 0` The work around for the time being is to configure speaker-test to play a 20kHz wav file, this dose work and produces a beautiful sine wav :) Code: ` speaker-test -t wav -f /opt/20kHz.wav -l 0` Now for the people who will say why on earth do you want to produce a sound that no one can hear, and my reply is "Exactly". I send the 20kHz sine wav across the audio line as a way to supervise that the audio line is fully functioning. The second the 20kHz tone stops we are notified and send someone out to fix the problem, whether it be hardware or someone accidentality cuts the wire, we want to know. Basically the 20 kHz sine wave does not interferer with paging in the system. My first thoughts on why the 20 kHz sine is not produced, is because of missing driver. But I made sure that every sound driver that was loaded on Ubuntu was loaded on Tinycore. Code: ` sudo modprobe snd_seq_midi snd_seq_midi_event snd_intel_dspcfg` I also loaded AlsaPlugins.tcz just incase, but that did not make a difference. Code: ` tceload -i alsa-plugins.tcz` The last thing I tried after was to set the format sample -F parameter to 96000 Code: `speaker-test -t sine -f 20000 -X -l 0 -F 96000` At first when I did this I thought I was so smart. A sine wave appeared, however I did not have a speaker hooked up, but when the speaker got plugged in I heard a high pitched noise. So the parameter affected the output but not the desired output that I need. Another thought that pass through my mind is could this be a configuration setting during the build of alsa that need to be turned on. After all 15000 Hz does work. This is slightly related to a previous post linked below. [https://forum.tinycorelinux.net/index.php/topic,27673.0.html](url) Thank you Any help is greatly appreciated gharig Issue URL : https://github.com/alsa-project/alsa-utils/issues/304 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[GIT PULL] ASoC fixes for v6.16-rc6
by Mark Brown 17 Jul '25

17 Jul '25

The following changes since commit 3b3312f28ee2d9c386602f8521e419cfc69f4823: ASoC: cs35l56: probe() should fail if the device ID is not recognized (2025-07-03 12:22:27 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.16-rc6 for you to fetch changes up to 7bab1bd9fdf15b9fa7e6a4b0151deab93df3c80d: ASoC: amd: yc: Add DMI quirk for HP Laptop 17 cp-2033dx (2025-07-16 11:50:33 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.16 A relatively large set of changes, but most of them are quirk information for various x86 systems. There is one more substantial fix for a NULL pointer dereference when removing the AVS driver, plus one for Kconfig dependencies. ---------------------------------------------------------------- Adam Queler (1): ASoC: amd: yc: Add DMI entries to support HP 15-fb1xxx Alexandru Andries (1): ASoC: amd: yc: add DMI quirk for ASUS M6501RM Amadeusz Sławiński (1): ASoC: Intel: avs: Fix NULL ptr deref on rmmod Arnd Bergmann (1): ASoC: Intel: fix SND_SOC_SOF dependencies Bard Liao (1): ASoC: Intel: soc-acpi: add support for HP Omen14 ARL Lane Odenbach (1): ASoC: amd: yc: Add DMI quirk for HP Laptop 17 cp-2033dx Oder Chiou (1): ASoC: rt5660: Fix the dmic data source from GPIO2 sound/soc/amd/yc/acp6x-mach.c | 21 ++++++++++++++++ sound/soc/codecs/rt5660.c | 7 ++++-- sound/soc/intel/avs/pcm.c | 4 +++- sound/soc/intel/boards/Kconfig | 2 +- sound/soc/intel/common/soc-acpi-intel-arl-match.c | 29 +++++++++++++++++++++++ 5 files changed, 59 insertions(+), 4 deletions(-)

2 1

[GIT PULL] ASoC fixes for v6.16-rc5
by Mark Brown 17 Jul '25

17 Jul '25

The following changes since commit 3b3312f28ee2d9c386602f8521e419cfc69f4823: ASoC: cs35l56: probe() should fail if the device ID is not recognized (2025-07-03 12:22:27 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.16-rc5 for you to fetch changes up to a609bd74b8680dba62c44f7e6d00d381ddb2d3c0: ASoC: Intel: avs: Fix NULL ptr deref on rmmod (2025-07-08 14:54:40 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.16 Two patches here, one quirk for an AMD system and a fix for an issue on remove of the AVS driver. ---------------------------------------------------------------- Alexandru Andries (1): ASoC: amd: yc: add DMI quirk for ASUS M6501RM Amadeusz Sławiński (1): ASoC: Intel: avs: Fix NULL ptr deref on rmmod sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/intel/avs/pcm.c | 4 +++- 2 files changed, 10 insertions(+), 1 deletion(-)

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[PATCH] ASoC: qcom: x1e80100: Support boards with two speakers
by Krzysztof Kozlowski 17 Jul '25

17 Jul '25

Some Qualcomm X1E laptops have only two speakers. Regardless whether this sound card driver is suitable for them (we could re-use one for some older SoC), we should set reasonable channel map depending on the number of channels, not always 4-speaker setup. This change is necessary for bringing audio support on Lenovo Thinkpad T14s with Qualcomm X1E78100 and only two speakers. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/x1e80100.c | 40 ++++++++++++++++++++++++++++++++++----- 1 file changed, 35 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/x1e80100.c b/sound/soc/qcom/x1e80100.c index 898b5c26bf1e..8eb57fc12f0d 100644 --- a/sound/soc/qcom/x1e80100.c +++ b/sound/soc/qcom/x1e80100.c @@ -95,23 +95,53 @@ static int x1e80100_snd_hw_params(struct snd_pcm_substream *substream, return qcom_snd_sdw_hw_params(substream, params, &data->sruntime[cpu_dai->id]); } +static int x1e80100_snd_hw_map_channels(unsigned int *ch_map, int num) +{ + switch (num) { + case 1: + ch_map[0] = PCM_CHANNEL_FC; + break; + case 2: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + break; + case 3: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + ch_map[2] = PCM_CHANNEL_FC; + break; + case 4: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_LB; + ch_map[2] = PCM_CHANNEL_FR; + ch_map[3] = PCM_CHANNEL_RB; + break; + default: + return -EINVAL; + } + + return 0; +} + static int x1e80100_snd_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; - const unsigned int rx_slot[4] = { PCM_CHANNEL_FL, - PCM_CHANNEL_LB, - PCM_CHANNEL_FR, - PCM_CHANNEL_RB }; + unsigned int channels = substream->runtime->channels; + unsigned int rx_slot[4]; int ret; switch (cpu_dai->id) { case WSA_CODEC_DMA_RX_0: case WSA_CODEC_DMA_RX_1: + ret = x1e80100_snd_hw_map_channels(rx_slot, channels); + if (ret) + return ret; + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, - ARRAY_SIZE(rx_slot), rx_slot); + channels, rx_slot); if (ret) return ret; break; -- 2.43.0

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qcom-lpass: RX HPH Mode is set to a ULP mode which causes distortion
by GitHub issues - opened 17 Jul '25

17 Jul '25

alsa-project/alsa-ucm-conf issue #595 was opened from valpackett: The headphone enable sequence for all Qualcomm devices sets `CLS_H_ULP` mode: https://github.com/alsa-project/alsa-ucm-conf/blob/1b69ade9b6d7ee37a87c08b1… I was trying to figure out why I was getting horrendous distortion on the headphone jack of my Dell Latitude 7455 laptop above a certain relatively-quiet volume level. After lots of fiddling and searching (thanks, [XDA forum audio modders](https://xdaforums.com/t/mod-audio-biquads-utilizing-qualcomms-audi… from several years ago!) I figured it out. Changing that to `CLS_AB_HIFI` has completely fixed the distortion! Should the default be changed to a HIFI mode for everyone, i.e. right in that file? ping: @Srinivas-Kandagatla Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/595 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Alsa speaker-test dose not generate 20kHz Sine Wav
by Harig, George William 16 Jul '25

16 Jul '25

Hello Alsa-Dev, I hope that I'm writing to the proper place to report a problem, please advise if not... I'm not able to produce 20kHz sine wav with speaker-test 1.2.13 Running on Tinycore 16.0 . Below is what I submitted to Tinycore Team, and they suggested that I contact you with my findings. The hardware I'm using is Winsystems E3825 dual-core, 1.33GHz SATA 2.0 channel mSATA socket (shared with miniPCIe) 2 GB of Soldered Down DDR3 Link to product: https://winsystems.com/product/ppm-c407/ Tinycore Team; My problem today is that speaker-test does not produce a 20kHz sine wave on Tinycore 16.0 to the line out. speaker-test does produce something just not a 20kHz sine wave. Code: speaker-test -t sine -f 20000 -X -l 0 However, using the exact same hardware running Ubuntu 20.24 or 22.04 the following code works and a 20kHz sine wave is produced. The difference being that the alsa versions are different between the 3 Operating Systems. Tinycore 16.0 Alsa version is 1.2.13 - failure Ubuntu 20.04 Alsa version is 1.2.2 - success Ubuntu 22.04 Alsa version is 1.2.6 - success Through my testing, starting at 8000 Hz, and incrementing by 1000 Hz, speaker-test does produce a sine wave, until I went over 15000 Hz then no more sine wave. :) :) Code: speaker-test -t sine -f 8000 -X -l 0 speaker-test -t sine -f 15000 -X -l 0 :( Code: speaker-test -t sine -f 16000 -X -l 0 The work around for the time being is to configure speaker-test to play a 20kHz wav file, this dose work and produces a beautiful sine wav :) Code: speaker-test -t wav -f /opt/20kHz.wav -l 0 Now for the people who will say why on earth do you want to produce a sound that no one can hear, and my reply is "Exactly". I send the 20kHz sine wav across the audio line as a way to supervise that the audio line is fully functioning. The second the 20kHz tone stops we are notified and send someone out to fix the problem, whether it be hardware or someone accidentality cuts the wire, we want to know. Basically the 20 kHz sine wave does not interferer with paging in the system. My first thoughts on why the 20 kHz sine is not produced, is because of missing driver. But I made sure that every sound driver that was loaded on Ubuntu was loaded on Tinycore. Code: sudo modprobe snd_seq_midi snd_seq_midi_event snd_intel_dspcfg I also loaded AlsaPlugins.tcz just incase, but that did not make a difference. Code: tceload -i alsa-plugins.tcz The last thing I tried after was to set the format sample -F parameter to 96000 Code: speaker-test -t sine -f 20000 -X -l 0 -F 96000 At first when I did this I thought I was so smart. A sine wave appeared, however I did not have a speaker hooked up, but when the speaker got plugged in I heard a high pitched noise. So the parameter affected the output but not the desired output that I need. Another thought that pass through my mind is could this be a configuration setting during the build of alsa that need to be turned on. After all 15000 Hz does work. This is slightly related to a previous post linked below. https://forum.tinycorelinux.net/index.php/topic,27673.0.html Thank you Any help is greatly appreciated gharig

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