February 2025 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/9] ALSA: usb-audio: More fixes / improvements on PCM
by Takashi Iwai 22 Apr '25

22 Apr '25

Hi, this is a patch set for USB-audio to fix and improve the PCM stream handling. The primary target is the enhancement of the low-latency playback that was introduced recently for dealing with the small period size (that is used typically by JACK). There are a few other fixes that are relevant with it. The current patches are found in topic/usb-audio branch of sound.git tree, too. Takashi === Takashi Iwai (9): ALSA: usb-audio: Restrict rates for the shared clocks ALSA: usb-audio: Fix possible race at sync of urb completions ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playback ALSA: usb-audio: Disable low-latency playback for free-wheel mode ALSA: usb-audio: Disable low-latency mode for implicit feedback sync ALSA: usb-audio: Check available frames for the next packet size ALSA: usb-audio: Add spinlock to stop_urbs() ALSA: usb-audio: Improved lowlatency playback support ALSA: usb-audio: Avoid killing in-flight URBs during draining sound/usb/card.h | 11 ++- sound/usb/endpoint.c | 230 +++++++++++++++++++++++++++++++------------ sound/usb/endpoint.h | 13 ++- sound/usb/pcm.c | 157 ++++++++++++++++++++++------- 4 files changed, 305 insertions(+), 106 deletions(-) -- 2.26.2

2 23

ALSA 1.2.8 release
by Jaroslav Kysela 15 Apr '25

15 Apr '25

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.

2 8

[PATCH] ALSA: hda: intel: Add Dell ALC3271 to power_save denylist
by Hoku Ishibe 05 Mar '25

05 Mar '25

Dell XPS 13 7390 with the Realtek ALC3271 codec experiences persistent humming noise when the power_save mode is enabled. This issue occurs when the codec enters power saving mode, leading to unwanted noise from the speakers. This patch adds the affected model (PCI ID 0x1028:0x0962) to the power_save denylist to ensure power_save is disabled by default, preventing power-off related noise issues. Steps to Reproduce 1. Boot the system with `snd_hda_intel` loaded. 2. Verify that `power_save` mode is enabled: ```sh cat /sys/module/snd_hda_intel/parameters/power_save ```` output: 10 (default power save timeout) 3. Wait for the power save timeout 4. Observe a persistent humming noise from the speakers 5. Disable `power_save` manually: ```sh echo 0 | sudo tee /sys/module/snd_hda_intel/parameters/power_save ```` 6. Confirm that the noise disappears immediately. This issue has been observed on my system, and this patch successfully eliminates the unwanted noise. If other users experience similar issues, additional reports would be helpful. Signed-off-by: Hoku Ishibe <me(a)hokuishi.be> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7d7f9aac50a9..ff536a0e480c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2232,6 +2232,8 @@ static const struct snd_pci_quirk power_save_denylist[] = { SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0), /* KONTRON SinglePC may cause a stall at runtime resume */ SND_PCI_QUIRK(0x1734, 0x1232, "KONTRON SinglePC", 0), + /* Dell ALC3271 */ + SND_PCI_QUIRK(0x1028, 0x0962, "Dell ALC3271", 0), {} }; -- 2.48.1

2 1

[PATCH v1 0/2] ASoC: codecs: add ES8389 codec driver
by Zhang Yi 03 Mar '25

03 Mar '25

The driver is for codec ES8389 of everest-semi. Zhang Yi (2): ASoC: codecs: add support for ES8389 ASoC: dt-bindings: Add Everest ES8389 audio CODEC .../bindings/sound/everest,es8389.yaml | 98 ++ sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8389.c | 985 ++++++++++++++++++ sound/soc/codecs/es8389.h | 139 +++ 5 files changed, 1229 insertions(+), 1 deletion(-) create mode 100755 Documentation/devicetree/bindings/sound/everest,es8389.yaml create mode 100644 sound/soc/codecs/es8389.c create mode 100644 sound/soc/codecs/es8389.h -- 2.17.1

2 3

Build with gettext-0.24 intermittently fails
by GitHub issues - edited 28 Feb '25

28 Feb '25

alsa-project/alsa-utils issue #294 was edited from heitbaum: After Autoconf the errors are as follows. The Makefile is not honouring / passing the builddir through. This is a build being done in a subdirectory. ``` Making all in po make[2]: Entering directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu/po' make[2]: *** No rule to make target '/config.status', needed by 'Makefile'. Stop. make[2]: Leaving directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu/po' make[1]: *** [Makefile:471: all-recursive] Error 1 make[1]: Leaving directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu' ``` - This is occurring just like https://github.com/zapping-vbi/zvbi/issues/57 - now testing to see if the change to `AM_GNU_GETTEXT_VERSION([0.24])` fixes the issue Issue URL : https://github.com/alsa-project/alsa-utils/issues/294 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Build with gettext-0.24 intermittently fails
by GitHub issues - opened 28 Feb '25

28 Feb '25

alsa-project/alsa-utils issue #294 was opened from heitbaum: After Autoconf the errors are as follows. The Makefile is not honouring / passing the builddir through. This is a build being done in a subdirectory. ``` Making all in po make[2]: Entering directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu/po' make[2]: *** No rule to make target '/config.status', needed by 'Makefile'. Stop. make[2]: Leaving directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu/po' make[1]: *** [Makefile:471: all-recursive] Error 1 make[1]: Leaving directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu' ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/294 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

How to clone from https://git.alsa-project.org/?
by GitHub issues - opened 28 Feb '25

28 Feb '25

alsa-project/alsa-utils issue #293 was opened from K0-RR: `git clone https://git.alsa-project.org/?p=alsa.git` gives me fatal: repository 'https://git.alsa-project.org/?p=alsa.git/' not found Issue URL : https://github.com/alsa-project/alsa-utils/issues/293 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v3] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 27 Feb '25

27 Feb '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convery the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - Merge the 4 patches into one patch v2: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/2025 | 02160812.P7szOygN-lkp(a)intel.com/__;!!G3vK!WriwhtbQGlg6p00bcWutlhb-y9vS | pDrSu5_g9oOzzGOoABdirEARTntFzyX6Ss3UPRDxi_E-IA$ - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/2025 | 02160632.u0zxG7yD-lkp(a)intel.com/__;!!G3vK!VtEdfur3fZklvPVfHFFfFP60Ub2E | JYvmjSUZ65lJ6BsQPJDl1fjYRDqh3c6Q5y6sA70ssXDG5g$ v1: - update the year from 2024 to 2025 in the files. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. - update the year from 2024 to 2025. - release the memory to save calibration addresses in tasdevice_dsp_remove(). - Add TASDEV_ALPHA and TASDEV_BETA firmware checking code. - Add fct_param_address_parser for Alpha and Beta firmware. - Add an extra branch to handle calibration address for TASDEV_ALPHA and TASDEV_BETA firmware. --- include/sound/tas2781-dsp.h | 26 +++- include/sound/tas2781.h | 13 +- sound/pci/hda/tas2781_spi_fwlib.c | 6 +- sound/soc/codecs/tas2781-comlib.c | 3 +- sound/soc/codecs/tas2781-fmwlib.c | 149 +++++++++++++++++++++- sound/soc/codecs/tas2781-i2c.c | 203 +++++++++++++++++++++++------- 6 files changed, 340 insertions(+), 60 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..77a240daaae0 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,23 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + unsigned char thr[3]; + unsigned char thr2[3]; + unsigned char plt_flg[3]; + unsigned char sin_gn[3]; + unsigned char sin_gn2[3]; + /* real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + unsigned char a2_reg[3]; + unsigned short size; + char *data; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +133,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 72d2060904f6..5cfbb9e0a626 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -112,6 +112,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -126,6 +132,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -171,6 +178,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -196,6 +204,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c index 131d9a77d140..c90641119509 100644 --- a/sound/pci/hda/tas2781_spi_fwlib.c +++ b/sound/pci/hda/tas2781_spi_fwlib.c @@ -370,10 +370,10 @@ static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, int i, n = ARRAY_SIZE(non_ppc3_mapping_table); unsigned char dev_idx = 0; - if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) { p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; n = ARRAY_SIZE(ppc3_tas2781_mapping_table); - } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION_BASE) { p = (struct blktyp_devidx_map *)ppc3_mapping_table; n = ARRAY_SIZE(ppc3_mapping_table); } @@ -1605,7 +1605,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION) { + if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 1e0b3aa95749..bba5abe4736f 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -2,7 +2,7 @@ // // TAS2563/TAS2781 Common functions for HDA and ASoC Audio drivers // -// Copyright 2023 - 2024 Texas Instruments, Inc. +// Copyright 2023 - 2025 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -429,6 +429,7 @@ void tasdevice_dsp_remove(void *context) if (tas_fmw->configs) tasdev_dsp_cfg_remove(tas_fmw->configs, tas_fmw->nr_configurations); + kfree(tas_fmw->fct_par_addr.data); kfree(tas_fmw); tas_dev->fmw = NULL; } diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 61d9c220b6a4..2fae7be99454 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2,7 +2,7 @@ // // tas2781-fmwlib.c -- TASDEVICE firmware support // -// Copyright 2023 - 2024 Texas Instruments, Inc. +// Copyright 2023 - 2025 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -389,10 +389,10 @@ static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, int i, n = ARRAY_SIZE(non_ppc3_mapping_table); unsigned char dev_idx = 0; - if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) { p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; n = ARRAY_SIZE(ppc3_tas2781_mapping_table); - } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION_BASE) { p = (struct blktyp_devidx_map *)ppc3_mapping_table; n = ARRAY_SIZE(ppc3_mapping_table); } @@ -559,6 +559,118 @@ static int fw_parse_configuration_data_kernel( return offset; } +static void hex_parse_u8(unsigned char *item, const unsigned char *val, + int len) +{ + for (int i = 0; i < len; i++) + item[i] = val[i]; +} + +static void hex_parse_u24(unsigned int *item, const unsigned char *val) +{ + *item = TASDEVICE_REG(val[0], val[1], val[2]); +} + +static void fct_param_address_parser(struct cali_reg *r, + struct tasdevice_fw *tas_fmw) +{ + struct fct_param_address *p = &(tas_fmw->fct_par_addr); + int i; + + for (i = 0; i < 20; i++) { + const unsigned char *dat = &p->data[24 * i]; + + if (dat[23] != 1) + break; + + if (strstr(dat, "umg_SsmKEGCye") != NULL) { + hex_parse_u24(&r->pow_reg, &dat[20]); + continue; + } + if (strstr(dat, "iks_E0") != NULL) { + hex_parse_u24(&r->r0_reg, &dat[20]); + continue; + } + if (strstr(dat, "yep_LsqM0") != NULL) { + hex_parse_u24(&r->invr0_reg, &dat[20]); + continue; + } + if (strstr(dat, "oyz_U0_ujx") != NULL) { + hex_parse_u24(&r->r0_low_reg, &dat[20]); + continue; + } + if (strstr(dat, "iks_GC_GMgq") != NULL) { + hex_parse_u24(&r->tlimit_reg, &dat[20]); + continue; + } + if (strstr(dat, "gou_Yao") != NULL) { + hex_parse_u8(p->thr, &dat[20], 3); + continue; + } + if (strstr(dat, "kgd_Wsc_Qsbp") != NULL) { + hex_parse_u8(p->plt_flg, &dat[20], 3); + continue; + } + if (strstr(dat, "yec_CqseSsqs") != NULL) { + hex_parse_u8(p->sin_gn, &dat[20], 3); + continue; + } + if (strstr(dat, "iks_SogkGgog2") != NULL) { + hex_parse_u8(p->sin_gn2, &dat[20], 3); + continue; + } + if (strstr(dat, "yec_Sae_Y") != NULL) { + hex_parse_u8(p->thr2, &dat[20], 3); + continue; + } + if (strstr(dat, "Re_Int") != NULL) { + hex_parse_u8(p->r0_reg, &dat[20], 3); + continue; + } + /* Check whether the spk connection is open */ + if (strstr(dat, "SigFlag") != NULL) { + hex_parse_u8(p->tf_reg, &dat[20], 3); + continue; + } + if (strstr(dat, "a1_Int") != NULL) { + hex_parse_u8(p->a1_reg, &dat[20], 3); + continue; + } + if (strstr(dat, "a2_Int") != NULL) { + hex_parse_u8(p->a2_reg, &dat[20], 3); + continue; + } + } +} + +static int fw_parse_fct_param_address(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct fct_param_address *p = &(tas_fmw->fct_par_addr); + struct calidata *cali_data = &tas_priv->cali_data; + struct cali_reg *r = &cali_data->cali_reg_array; + const unsigned char *data = fmw->data; + + if (offset + 520 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + + /* skip reserved part */ + offset += 40; + p->size = 480; + p->data = kmemdup(&data[offset], 480, GFP_KERNEL); + if (!p->data) { + offset = -1; + goto out; + } + offset += 480; + fct_param_address_parser(r, tas_fmw); +out: + return offset; +} + static int fw_parse_variable_header_kernel( struct tasdevice_priv *tas_priv, const struct firmware *fmw, int offset) @@ -1686,13 +1798,29 @@ static int tasdevice_load_block(struct tasdevice_priv *tas_priv, return rc; } +static void dspbin_type_check(struct tasdevice_priv *tas_priv, + unsigned int ppcver) +{ + if (ppcver >= PPC3_VERSION_TAS2781_ALPHA_MIN) { + if (ppcver >= PPC3_VERSION_TAS2781_BETA_MIN) + tas_priv->dspbin_typ = TASDEV_BETA; + else if (ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) + tas_priv->dspbin_typ = TASDEV_BASIC; + else + tas_priv->dspbin_typ = TASDEV_ALPHA; + } + if (tas_priv->dspbin_typ != TASDEV_BASIC) + tas_priv->fw_parse_fct_param_address = + fw_parse_fct_param_address; +} + static int dspfw_default_callback(struct tasdevice_priv *tas_priv, unsigned int drv_ver, unsigned int ppcver) { int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION) { + if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = @@ -1701,6 +1829,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, fw_parse_configuration_data_kernel; tas_priv->tasdevice_load_block = tasdevice_load_block_kernel; + dspbin_type_check(tas_priv, ppcver); } else { switch (ppcver) { case 0x00: @@ -1716,7 +1845,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, default: dev_err(tas_priv->dev, "%s: PPCVer must be 0x0 or 0x%02x", - __func__, PPC3_VERSION); + __func__, PPC3_VERSION_BASE); dev_err(tas_priv->dev, " Current:0x%02x\n", ppcver); rc = -EINVAL; @@ -2023,9 +2152,17 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, } offset = tas_priv->fw_parse_configuration_data(tas_priv, tas_fmw, fmw, offset); - if (offset < 0) + if (offset < 0) { ret = offset; + goto out; + } + if (tas_priv->fw_parse_fct_param_address) { + offset = tas_priv->fw_parse_fct_param_address(tas_priv, + tas_fmw, fmw, offset); + if (offset < 0) + ret = offset; + } out: return ret; } diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 90c5b2e74d12..8eb9fee27a42 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -349,13 +349,31 @@ static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, return rc; } +static int partial_cali_data_update(int *reg, int j) +{ + switch (tas2781_cali_start_reg[j].reg) { + case 0: + return reg[0]; + case TAS2781_PRM_PLT_FLAG_REG: + return reg[1]; + case TAS2781_PRM_SINEGAIN_REG: + return reg[2]; + case TAS2781_PRM_SINEGAIN2_REG: + return reg[3]; + default: + return 0; + } +} + static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, int *reg, unsigned char *dat) { struct tasdevice *tasdev = tas_priv->tasdevice; struct bulk_reg_val *p = tasdev[i].cali_data_backup; + struct bulk_reg_val *t = &tasdev[i].alp_cali_bckp; const int sum = ARRAY_SIZE(tas2781_cali_start_reg); - int j; + unsigned char val[4]; + int j, r; if (p == NULL) return; @@ -370,30 +388,23 @@ 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 (tas2781_cali_start_reg[j].reg) { - case 0: { - if (!reg[0]) - continue; - p[j].reg = reg[0]; + if (!tas_priv->dspbin_typ) { + r = partial_cali_data_update(reg, j); + if (r) + p[j].reg = r; } - break; - case TAS2781_PRM_PLT_FLAG_REG: - p[j].reg = reg[1]; - break; - case TAS2781_PRM_SINEGAIN_REG: - p[j].reg = reg[2]; - break; - case TAS2781_PRM_SINEGAIN2_REG: - p[j].reg = reg[3]; - break; - } - tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, - p[j].val, 4); + + if (p[j].reg) + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); } } + if (tas_priv->dspbin_typ == TASDEV_ALPHA) + tasdevice_dev_bulk_read(tas_priv, i, t->reg, t->val, 4); + /* Update the setting for calibration */ - for (j = 0; j < sum - 2; j++) { + for (j = 0; j < sum - 4; j++) { if (p[j].val_len == 1) { if (p[j].is_locked) tasdevice_dev_write(tas_priv, i, @@ -401,17 +412,33 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, TAS2781_TEST_PAGE_UNLOCK); tasdevice_dev_write(tas_priv, i, p[j].reg, tas2781_cali_start_reg[j].val[0]); - } else { - if (!p[j].reg) - continue; - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, - (unsigned char *) - tas2781_cali_start_reg[j].val, 4); } } - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, &dat[1], 4); - tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, &dat[5], 4); + if (tas_priv->dspbin_typ == TASDEV_ALPHA) { + val[0] = 0x00; + val[1] = 0x00; + val[2] = 0x21; + val[3] = 0x8e; + } else { + val[0] = tas2781_cali_start_reg[j].val[0]; + val[1] = tas2781_cali_start_reg[j].val[1]; + val[2] = tas2781_cali_start_reg[j].val[2]; + val[3] = tas2781_cali_start_reg[j].val[3]; + } + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, val, 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, + (unsigned char *)tas2781_cali_start_reg[j + 1].val, 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 2].reg, &dat[1], 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 3].reg, &dat[5], 4); + if (tas_priv->dspbin_typ == TASDEV_ALPHA) { + val[0] = 0x00; + val[1] = 0x00; + val[2] = 0x2a; + val[3] = 0x0b; + + tasdevice_dev_bulk_read(tas_priv, i, t->reg, val, 4); + } } static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, @@ -460,6 +487,7 @@ static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) for (i = 0; i < tas_priv->ndev; i++) { struct tasdevice *tasdev = tas_priv->tasdevice; struct bulk_reg_val *p = tasdev[i].cali_data_backup; + struct bulk_reg_val *t = &tasdev[i].alp_cali_bckp; if (p == NULL) continue; @@ -479,6 +507,10 @@ static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) p[j].val, 4); } } + + if (tas_priv->dspbin_typ == TASDEV_ALPHA) + tasdevice_dev_bulk_write(tas_priv, i, t->reg, + t->val, 4); } } @@ -590,16 +622,20 @@ static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, i += 2; priv->is_user_space_calidata = true; - p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; + if (priv->dspbin_typ == TASDEV_BASIC) { + p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + } else { + i += 15; + } memcpy(dst, &src[i], cali_data->total_sz); return 1; @@ -646,12 +682,19 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; - unsigned int reg; + unsigned int reg = TAS2781_RUNTIME_RE_REG_TF; + + if (tas_priv->chip_id == TAS2781) { + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); - if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG_TF; - else + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->tf_reg[0], p->tf_reg[1], + p->tf_reg[2]); + } else { reg = TAS2563_RUNTIME_RE_REG_TF; + } guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; @@ -666,12 +709,19 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; - unsigned int reg; + unsigned int reg = TAS2781_RUNTIME_RE_REG; - if (tas_priv->chip_id == TAS2781) - reg = TAS2781_RUNTIME_RE_REG; - else + if (tas_priv->chip_id == TAS2781) { + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); + + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->r0_reg[0], p->r0_reg[1], + p->r0_reg[2]); + } else { reg = TAS2563_RUNTIME_RE_REG; + } + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -705,11 +755,16 @@ static int tasdev_XMA1_data_get(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); + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A1_REG; + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->a1_reg[0], p->a1_reg[1], p->a1_reg[2]); + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -720,11 +775,16 @@ static int tasdev_XMA2_data_get(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); + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A2_REG; + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->a2_reg[0], p->a2_reg[1], p->a2_reg[2]); + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -1172,10 +1232,50 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) nr_controls < mix_index ? nr_controls : mix_index); } +static void cali_reg_update(struct bulk_reg_val *p, + struct fct_param_address *t) +{ + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int reg = 0; + int j; + + for (j = 0; j < sum; j++) { + switch (tas2781_cali_start_reg[j].reg) { + case 0: + reg = TASDEVICE_REG(t->thr[0], t->thr[1], t->thr[2]); + break; + case TAS2781_PRM_PLT_FLAG_REG: + reg = TASDEVICE_REG(t->plt_flg[0], t->plt_flg[1], + t->plt_flg[2]); + break; + case TAS2781_PRM_SINEGAIN_REG: + reg = TASDEVICE_REG(t->sin_gn[0], t->sin_gn[1], t->sin_gn[2]); + break; + case TAS2781_PRM_SINEGAIN2_REG: + reg = TASDEVICE_REG(t->sin_gn[0], t->sin_gn[1], t->sin_gn[2]); + break; + default: + reg = 0; + break; + } + if (reg) + p[j].reg = reg; + } +} + +static void alpa_cali_update(struct bulk_reg_val *p, + struct fct_param_address *t) +{ + p->is_locked = false; + p->reg = TASDEVICE_REG(t->thr2[0], t->thr2[1], t->thr2[2]); + p->val_len = 4; +} + static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) { struct calidata *cali_data = &priv->cali_data; struct tasdevice *tasdev = priv->tasdevice; + struct tasdevice_fw *fmw = priv->fmw; struct soc_bytes_ext *ext_cali_data; struct snd_kcontrol_new *cali_ctrls; unsigned int nctrls; @@ -1191,14 +1291,25 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) } if (priv->chip_id == TAS2781) { + struct fct_param_address *t = &(fmw->fct_par_addr); + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; nctrls = ARRAY_SIZE(tas2781_cali_controls); for (i = 0; i < priv->ndev; i++) { - tasdev[i].cali_data_backup = + struct bulk_reg_val *p; + + p = tasdev[i].cali_data_backup = kmemdup(tas2781_cali_start_reg, sizeof(tas2781_cali_start_reg), GFP_KERNEL); if (!tasdev[i].cali_data_backup) return -ENOMEM; + if (priv->dspbin_typ) { + cali_reg_update(p, t); + if (priv->dspbin_typ == TASDEV_ALPHA) { + p = &tasdev[i].alp_cali_bckp; + alpa_cali_update(p, t); + } + } } } else { cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; -- 2.34.1

3 4

[PATCH] soundwire: amd: change the log level for command response log
by Vijendar Mukunda 27 Feb '25

27 Feb '25

Change log level for command response log to dev_dbg_ratelimited when command is ignored. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index dcf85f94950a..a12c68b93b1c 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -346,7 +346,7 @@ static enum sdw_command_response amd_sdw_fill_msg_resp(struct amd_sdw_manager *a msg->dev_num); return SDW_CMD_FAIL; } - dev_err_ratelimited(amd_manager->dev, "command is ignored for Slave %d\n", + dev_dbg_ratelimited(amd_manager->dev, "command is ignored for Slave %d\n", msg->dev_num); return SDW_CMD_IGNORED; } -- 2.34.1

3 2

ALSA USB quirk mixer driver question
by Lena 27 Feb '25

27 Feb '25

Hi all! I'm currently trying to develop an usb quirk mixer driver for a desktop USB audio interface. The interface exposes the following: - interface 0-2 are USB UAC compliant, bound to by snd-usb-audio, and working flawlessly - interface 3 is vendor specific, and we can ignore it (used for firmware updates) - interface 4 exposes a non-standard mixer with a HID interface exposing 2 interrupt endpoints, and is used by the vendor windows app. I have studied the protocol with usbpcap and can successfully talk to the interface on the userspace, but i had a question regarding how to proceed in the kernel. This is because, after creating a new entry in `sound/usb/mixer_quirks.c` with all the needed controls, userspace can interact with them, etc. I can't figure out how to tell ALSA to bind to the correct interface for the mixer (and kick out usbhid). Looking at the other quirk mixer drivers it doesn't look (?) like any other quirk mixer driver is binding to a *different* interface, not even `sound/usb/mixer_scarlett.c`. Generally, in a standalone USB driver you'd have tracking/teardown handled easily with a callback, but i'm having trouble with the dependencies here. (simply using `usb_sndintpipe()` or `usb_interrupt_msg()` triggers kernel oopses, so that's probably not it) What i think i have to do is to use `usb_get_intf()`, then i have to somehow claim it (does `usb_driver_claim_interface()` work here?). But `usb_get_intf()` requires me to then release the interface with `usb_put_intf()`, and it doesn't look like there's anywhere for me to do that. (`snd_usb_mixer_elem_free()`? doesn't look like the order is correct) `iface_ref_find()` would sorta do what i need, but i cannot use it here. Could someone please point me in the right direction? I'm very stuck at the moment. Thanks, Lena

2 2

[PATCH v1] ASoC: tas2781: Clean up for some define
by Baojun Xu 26 Feb '25

26 Feb '25

Do some clean up for some define in header file. Like change lower case in value, up case in define, space add in recommends, change date of files and add author. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781.h | 21 +++++++++++---------- sound/soc/codecs/tas2781-fmwlib.c | 7 ++++--- 2 files changed, 15 insertions(+), 13 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 72d2060904f6..a2c8eb111e5f 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -11,6 +11,7 @@ // // Author: Shenghao Ding <shenghao-ding(a)ti.com> // Author: Kevin Lu <kevin-lu(a)ti.com> +// Author: Baojun Xu <baojun.xu(a)ti.com> // #ifndef __TAS2781_H__ @@ -31,7 +32,7 @@ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) -/*PAGE Control Register (available in page0 of each book) */ +/* PAGE Control Register (available in page0 of each book) */ #define TASDEVICE_PAGE_SELECT 0x00 #define TASDEVICE_BOOKCTL_PAGE 0x00 #define TASDEVICE_BOOKCTL_REG 127 @@ -42,12 +43,12 @@ #define TASDEVICE_REG(book, page, reg) (((book * 256 * 128) + \ (page * 128)) + reg) -/*Software Reset */ -#define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) +/* Software Reset */ +#define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) #define TASDEVICE_REG_SWRESET_RESET BIT(0) -/*I2C Checksum */ -#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) +/* I2C Checksum */ +#define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) /* XM_340 */ #define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x63, 0x3c) @@ -55,8 +56,8 @@ #define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x63, 0x38) /* Volume control */ -#define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) -#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) +#define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0c) +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1a) #define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) @@ -95,8 +96,8 @@ #define TAS2781_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x14, 0x40) #define TAS2781_PRM_SINEGAIN2_REG TASDEVICE_REG(0x00, 0x14, 0x44) -#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xFD, 0x0D) -#define TAS2781_TEST_PAGE_UNLOCK 0x0D +#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xfd, 0x0d) +#define TAS2781_TEST_PAGE_UNLOCK 0x0d #define TAS2781_RUNTIME_LATCH_RE_REG TASDEVICE_REG(0x00, 0x00, 0x49) #define TAS2781_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x62, 0x48) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 61d9c220b6a4..bc2f68bd9846 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2,9 +2,10 @@ // // tas2781-fmwlib.c -- TASDEVICE firmware support // -// Copyright 2023 - 2024 Texas Instruments, Inc. +// Copyright 2023 - 2025 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Baojun Xu <baojun.xu(a)ti.com> #include <linux/crc8.h> #include <linux/firmware.h> @@ -1496,7 +1497,7 @@ static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, unsigned int nr_value; int ret; - ret = tasdevice_dev_read(tas_priv, chn, TASDEVICE_I2CChecksum, + ret = tasdevice_dev_read(tas_priv, chn, TASDEVICE_CHECKSUM_REG, &nr_value); if (ret < 0) { dev_err(tas_priv->dev, "%s: Chn %d\n", __func__, chn); @@ -1540,7 +1541,7 @@ static int tasdev_load_blk(struct tasdevice_priv *tas_priv, while (block->nr_retry > 0) { if (block->is_pchksum_present) { ret = tasdevice_dev_write(tas_priv, chn, - TASDEVICE_I2CChecksum, 0); + TASDEVICE_CHECKSUM_REG, 0); if (ret < 0) break; } -- 2.34.1

1 0

ucm2: Add support for MSI PRO B650-A w/ Realtek ALC4080
by GitHub pull_request - opened 26 Feb '25

26 Feb '25

alsa-project/alsa-ucm-conf pull request #514 was opened from gbrener: Gets the front panel audio to work. [alsa-info.txt](https://github.com/user-attachments/files/18993322/alsa-info… Thanks in advance. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/514 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/514.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: MediaTek: mt8390-evk: Add support for SOF
by GitHub pull_request - opened 26 Feb '25

26 Feb '25

alsa-project/alsa-ucm-conf pull request #513 was opened from nfraprado: Add support for MT8390 EVK with the SOF enabled. The only differences compared to the UCM with SOF disabled is the device number for playback and capture, which uses the PCMs for the SOF instead, and the number of channels for capture, which is stereo rather than mono. Define those parameters as variables for each case, with and without SOF, and move the common intialization to an init.conf, so the UCM can be shared by both. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/513 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/513.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Potential incorrect argument to snd_seq_port_info_set_ump_group in seqmid.c
by GitHub issues - opened 26 Feb '25

26 Feb '25

alsa-project/alsa-lib issue #440 was opened from reuk: https://github.com/alsa-project/alsa-lib/blob/df8f1cc1ec9d9ee15be5e2c23ad25… I think this call in seqmid.c is incorrect. According to the docs, the second argument to `snd_seq_port_info_set_ump_group` should be a 1-based group number (probably 0 for the UMP endpoint). I'm testing on Fedora 41 with alsa-lib-devel 1.2.13-3.fc41, and I'm seeing the following printed to the console when I call `snd_seq_create_ump_endpoint`: ``` ALSA lib seqmid.c:592:(snd_seq_create_ump_endpoint) Failed to create MIDI 2.0 port ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/440 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v6.14-rc4
by Mark Brown 26 Feb '25

26 Feb '25

The following changes since commit 571b69f2f9b1ec7cf7d0e9b79e52115a87a869c4: ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device (2025-02-13 11:34:32 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc4 for you to fetch changes up to 9da0ed4a85027063441fa1c73967cafc38f0677c: ASoC: Intel: don't check number of sdw links when set (2025-02-25 17:47:04 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 More driver specific fixes, the firmware change is part of fixing the race conditions in the Cirrus driver. ---------------------------------------------------------------- Bard Liao (2): ASoC: SOF: Intel: don't check number of sdw links when set dmic_fixup ASoC: Intel: sof_sdw: warn both sdw and pch dmic are used Chancel Liu (1): ASoC: fsl: Rename stream name of SAI DAI driver Hector Martin (3): ASoC: tas2770: Fix volume scale ASoC: tas2764: Fix power control mask ASoC: tas2764: Set the SDOUT polarity correctly Mark Brown (1): ASoC: Intel: don't check number of sdw links when set Nicolas Frattaroli (2): ASoC: es8328: fix route from DAC to output ASoC: dapm-graph: set fill colour of turned on nodes Richard Fitzgerald (2): firmware: cs_dsp: Remove async regmap writes ASoC: cs35l56: Prevent races when soft-resetting using SPI control drivers/firmware/cirrus/cs_dsp.c | 24 +++--------- include/sound/cs35l56.h | 31 +++++++++++++++ sound/pci/hda/cs35l56_hda_spi.c | 3 ++ sound/soc/codecs/cs35l56-shared.c | 80 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l56-spi.c | 3 ++ sound/soc/codecs/es8328.c | 15 ++------ sound/soc/codecs/tas2764.c | 10 ++++- sound/soc/codecs/tas2764.h | 8 +++- sound/soc/codecs/tas2770.c | 2 +- sound/soc/fsl/fsl_sai.c | 6 +-- sound/soc/fsl/imx-audmix.c | 4 +- sound/soc/intel/boards/sof_sdw.c | 7 ++++ sound/soc/sof/intel/hda.c | 18 +-------- tools/sound/dapm-graph | 2 +- 14 files changed, 159 insertions(+), 54 deletions(-)

2 1

[GIT PULL] ASoC fixes for v6.14-rc3
by Mark Brown 26 Feb '25

26 Feb '25

The following changes since commit 571b69f2f9b1ec7cf7d0e9b79e52115a87a869c4: ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device (2025-02-13 11:34:32 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc3 for you to fetch changes up to f5468beeab1b1adfc63c2717b1f29ef3f49a5fab: ASoC: tas2764: Set the SDOUT polarity correctly (2025-02-18 20:08:43 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 A few fixes I and James Calligero picked out of the Asahi tree. ---------------------------------------------------------------- Hector Martin (3): ASoC: tas2770: Fix volume scale ASoC: tas2764: Fix power control mask ASoC: tas2764: Set the SDOUT polarity correctly sound/soc/codecs/tas2764.c | 10 +++++++++- sound/soc/codecs/tas2764.h | 8 +++++++- sound/soc/codecs/tas2770.c | 2 +- 3 files changed, 17 insertions(+), 3 deletions(-)

1 0

[PATCH 0/2] ASoC: codecs: add ES8389 codec driver
by Zhang Yi 26 Feb '25

26 Feb '25

The driver is for codec ES8389 of everest-semi. Zhang Yi (2): ASoC: codecs: add support for ES8389 ASoC: dt-bindings: Add Everest ES8389 audio CODEC .../bindings/sound/everest,es8389.txt | 68 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8389.c | 977 ++++++++++++++++++ sound/soc/codecs/es8389.h | 139 +++ 5 files changed, 1191 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/everest,es8389.txt create mode 100755 sound/soc/codecs/es8389.c create mode 100755 sound/soc/codecs/es8389.h -- 2.17.1

3 4

add rt713 & rt1318 LED control
by GitHub pull_request - opened 26 Feb '25

26 Feb '25

alsa-project/alsa-ucm-conf pull request #512 was opened from shumingfan: add rt713 & rt1318 LED control Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/512 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/512.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Request for Build Instructions for macOS and Ubuntu
by GitHub issues - opened 24 Feb '25

24 Feb '25

alsa-project/alsa-utils issue #292 was opened from sallavatsal: Hello, I'm having difficulty building alsa-lib on both macOS and Ubuntu. Specifically: macOS: After running autoreconf, I get warnings and issues generating the configure script. Ubuntu: The available libasound2-dev version seems to be older than 1.2.5, causing configuration errors. Could you please provide official build instructions or guidance for these platforms? Also, I want to become a member of mailing list if there's any. Thank you! Issue URL : https://github.com/alsa-project/alsa-utils/issues/292 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

alsa-info.8: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 21 Feb '25

21 Feb '25

Package: alsa-utils Version: 1.2.13-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use "groff -e ' $' -e '\\~$' <file>" to find obvious 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? an.tmac:<stdin>:1: style: .TH missing fourth argument; consider package/project name and version (e.g., "groff 1.23.0") troff:<stdin>:59: 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 6.12.12-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 1.2.13-1+b1 ii libatopology2t64 1.2.13-1+b1 ii libc6 2.40-6 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5+20250125-2 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5+20250125-2 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

FiiO K7 USB DAC no Inputs
by GitHub issues - opened 21 Feb '25

21 Feb '25

alsa-project/alsa-lib issue #439 was opened from Starfoxfs: I have problems with Fiio K7 USB DAC in Linux Mint Cinnamon 22.1 (6.8.0-56-generic) Alsa Version: 1.0.25+dfsg-0ubuntu7 Pipewire: 1.0.5-1ubuntu2 Wireplumber: 0.4.17-1ubuntu4 The USB Dac has this Ports: 2x Line IN (Cinch) 2x Line OUT or Pre OUT(Cinch) 1x Digital IN TOSLINK 1x COAX/RCA IN (Yellow Cinch) The Problem is: Alsa didnt find any of this IN Ports. There are 4 OUTPUTS: Analog Stereo works fine with Alsa Digital IEC958 (thats wrong Device has no Digital Output) Digital Surround 51 (thats wrong Device has no Digital Output) Pro Audio works fine with Alsa Would be nice to have the Inputs. Heres my Alsa-Info Output: [alsa-info.txt](https://github.com/user-attachments/files/18907991/alsa-info… Issue URL : https://github.com/alsa-project/alsa-lib/issues/439 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v6.14-rc3
by Mark Brown 21 Feb '25

21 Feb '25

The following changes since commit 571b69f2f9b1ec7cf7d0e9b79e52115a87a869c4: ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device (2025-02-13 11:34:32 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc3 for you to fetch changes up to f5468beeab1b1adfc63c2717b1f29ef3f49a5fab: ASoC: tas2764: Set the SDOUT polarity correctly (2025-02-18 20:08:43 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 A few fixes I and James Calligero picked out of the Asahi tree. ---------------------------------------------------------------- Hector Martin (3): ASoC: tas2770: Fix volume scale ASoC: tas2764: Fix power control mask ASoC: tas2764: Set the SDOUT polarity correctly sound/soc/codecs/tas2764.c | 10 +++++++++- sound/soc/codecs/tas2764.h | 8 +++++++- sound/soc/codecs/tas2770.c | 2 +- 3 files changed, 17 insertions(+), 3 deletions(-)

2 1

[PATCH] ALSA: usb-audio: Re-add sample rate quirk for Pioneer DJM-900NXS2
by Dmitry Panchenko | d-Systems 20 Feb '25

20 Feb '25

From: Dmitry Panchenko <dmitry(a)d-systems.ee> Re-add the sample-rate quirk for the Pioneer DJM-900NXS2. This device does not not work without setting sample-rate. Signed-off-by: Dmitry Panchenko <dmitry(a)d-systems.ee> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a97efb7b131e..09210fb4ac60 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1868,6 +1868,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; + case USB_ID(0x2b73, 0x000a): /* Pioneer DJM-900NXS2 */ case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */ pioneer_djm_set_format_quirk(subs, 0x0082); break; -- 2.43.0

2 1

[PATCH v2 1/4] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 20 Feb '25

20 Feb '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convery the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - No further update v1: - update the year from 2024 to 2025. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. --- include/sound/tas2781-dsp.h | 26 +++++++++++++++++++++++--- include/sound/tas2781.h | 13 ++++++++++++- 2 files changed, 35 insertions(+), 4 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..77a240daaae0 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,23 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + unsigned char thr[3]; + unsigned char thr2[3]; + unsigned char plt_flg[3]; + unsigned char sin_gn[3]; + unsigned char sin_gn2[3]; + /* real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + unsigned char a2_reg[3]; + unsigned short size; + char *data; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +133,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 72d2060904f6..5cfbb9e0a626 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -112,6 +112,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -126,6 +132,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -171,6 +178,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -196,6 +204,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); -- 2.34.1

2 5

Problems with async handler
by GitHub issues - edited 20 Feb '25

20 Feb '25

alsa-project/alsa-lib issue #438 was edited from m-paq: Hi, I'm currently working on a project and I'm trying to use the async handler through `snd_async_add_pcm_handler`. I don't really understand where the issue comes from and how to debug it but I keep getting an error on this function call in my project. To try to figure out what's wrong I went in the test files and tried to run the test file `pcm.c` and added a print of the error code (`snd_strerror`) Compiling with: `gcc pcm.c -o test_pcm -lasound -lm` Running it with `./test_pcm -m async` But it keeps giving me the following output: ``` Playback device is default Stream parameters are 44100Hz, S16_LE, 1 channels Sine wave rate is 440.0000Hz Using transfer method: async Unable to register async handler Function not implemented ``` from line 434. I am able to run it in write mode with the correct behaviour, so my question is: **Is this an issue with the device or is it an issue somewhere else?** If ever this is not the correct place to ask this question please refer me to a more adequate place. Thanks to anyone that can help me! Issue URL : https://github.com/alsa-project/alsa-lib/issues/438 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Problems with async handler
by GitHub issues - opened 20 Feb '25

20 Feb '25

alsa-project/alsa-lib issue #438 was opened from m-paq: Hi, I'm currently working on a project and I'm trying to use the async handler through `snd_async_add_pcm_handler`. I don't really understand where the issue comes from and how to debug it but I keep getting an error on this function call in my project. To try to figure out what's wrong I went in the test files and tried to run the test file `pcm.c` and added a print of the error code (`snd_strerror`) Compiling with: `gcc pcm.c -o test_pcm -lasound -lm` Running it with `./test_pcm -m async` But it keeps giving me the following output: ``` Playback device is default Stream parameters are 44100Hz, S16_LE, 1 channels Sine wave rate is 440.0000Hz Using transfer method: async Unable to register async handler Function not implemented ``` from line 441. I am able to run it in write mode with the correct behaviour, so my question is: **Is this an issue with the device or is it an issue somewhere else?** If ever this is not the correct place to ask this question please refer me to a more adequate place. Thanks to anyone that can help me! Issue URL : https://github.com/alsa-project/alsa-lib/issues/438 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSA: usb-audio: Add Pioneer DJ DJM-900NXS2 support
by Dmitry Panchenko | d-Systems 19 Feb '25

19 Feb '25

Hello, Pioneer DJ DJM-900NXS2 is a widely used DJ mixer with 2 audio USB interfaces. Both have a MIDI controller, 10 playback and 12 capture channels. I've submitted Pioneer DJM-900NXS2 support patches back in 2020, but someone has removed set_format_quirk function for DJM-900NXS2 model. The playback and capture do not work, quirks.c needs to be patched. The patch is attached, please add. Signed-off-by: Dmitry Panchenko <dmitry(a)d-systems.ee> -- Kind regards, Dmitry Panchenko

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related commands output: sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19 inxi -A Audio: Device-1: Intel Alder Lake PCH-P High Definition Audio driver: sof-audio-pci-intel-tgl Device-2: NVIDIA GA104 High Definition Audio driver: snd_hda_intel API: ALSA v: k6.13.2-arch1-1 status: kernel-api Server-1: PipeWire v: 1.2.7 status: active Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: `sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: `sudo dmesg | grep snd\n [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: ` sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - opened 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was opened from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH RESEND 0/7] Initial Support for CS40L26
by Fred Treven 19 Feb '25

19 Feb '25

Introduce driver for Cirrus Logic Device CS40L26: A boosted haptics driver with integrated DSP and waveform memory with advanced closed loop algorithms and LRA protection. The core CS40L26 driver is in MFD and touches the Input Force Feedback subsystem for haptics and the ASoC subsystem for audio to haptics streaming. This patchset includes changes to the CS DSP firmware driver which fixes two bugs and introduces support for multiple coefficient files. Fred Treven (7): firmware: cs_dsp: Fix error checking in wseq_write() firmware: cs_dsp: Check for valid num_regs in cs_dsp_wseq_multi_write() firmware: cs_dsp: Add ability to load multiple coefficient files dt-bindings: mfd: cirrus,cs40l26: Support for CS40L26 mfd: cs40l26: Add support for CS40L26 core driver ASoC: cs40l26: Support I2S streaming to CS40L26 Input: cs40l26 - Add support for CS40L26 haptic driver .../bindings/mfd/cirrus,cs40l26.yaml | 81 + MAINTAINERS | 4 +- drivers/firmware/cirrus/cs_dsp.c | 70 +- drivers/input/misc/Kconfig | 10 + drivers/input/misc/Makefile | 1 + drivers/input/misc/cs40l26-vibra.c | 669 ++++++++ drivers/mfd/Kconfig | 29 + drivers/mfd/Makefile | 4 + drivers/mfd/cs40l26-core.c | 1412 +++++++++++++++++ drivers/mfd/cs40l26-i2c.c | 63 + drivers/mfd/cs40l26-spi.c | 63 + include/linux/firmware/cirrus/cs_dsp.h | 14 + include/linux/mfd/cs40l26.h | 341 ++++ sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs40l26-codec.c | 523 ++++++ 16 files changed, 3281 insertions(+), 17 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/cirrus,cs40l26.yaml create mode 100644 drivers/input/misc/cs40l26-vibra.c create mode 100644 drivers/mfd/cs40l26-core.c create mode 100644 drivers/mfd/cs40l26-i2c.c create mode 100644 drivers/mfd/cs40l26-spi.c create mode 100644 include/linux/mfd/cs40l26.h create mode 100644 sound/soc/codecs/cs40l26-codec.c -- 2.34.1

6 17

[PATCH] ASoC: amd: ps: fix inconsistent indenting warning in check_and_handle_sdw_dma_irq()
by Vijendar Mukunda 18 Feb '25

18 Feb '25

Fix below inconsistent indenting smatch warning. smatch warnings: sound/soc/amd/ps/pci-ps.c:68 check_and_handle_sdw_dma_irq() warn: inconsistent indenting Fixes: 4516be370ced ("ASoC: amd: ps: refactor soundwire dma interrupt handling") Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202502132134.BlkNw1Iq-lkp@intel.com/ Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 2ff8e67c19bd..221c65ff03c9 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -129,7 +129,7 @@ static short int check_and_handle_sdw_dma_irq(struct acp63_dev_data *adata, u32 if (ext_intr_stat1 & ACP63_P1_AUDIO1_RX_THRESHOLD) { writel(ACP63_P1_AUDIO1_RX_THRESHOLD, adata->acp63_base + ACP_EXTERNAL_INTR_STAT1); - adata->acp63_sdw1_dma_intr_stat[ACP63_SDW1_AUDIO1_RX] = 1; + adata->acp63_sdw1_dma_intr_stat[ACP63_SDW1_AUDIO1_RX] = 1; sdw_dma_irq_flag = 1; } if (ext_intr_stat1 & ACP63_P1_AUDIO1_TX_THRESHOLD) { -- 2.34.1

2 1

ALC4080 5.1 don't work - rear speakers recognized as mic
by GitHub issues - edited 18 Feb '25

18 Feb '25

alsa-project/alsa-ucm-conf issue #510 was edited from atazx: kubuntu 24.10 MSI X870 Mag Tomahawk motherboard, with integrated ALC4080 (usb id `0db0:cd0e`) I tried to install the latest version of alsa-ucm-conf (1.2.13) On this card to have 5.1 audio you have to connect the rear speakers to the front panel, in what would normally be the microphone input, in fact when I connect the speakers they are recognized as a microphone. (only with ucm-conf 1.2.13) front speaker always works rear speakers only work after rebooting from windows center speakers do not work is there a way to fix this problem? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/510 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4080 5.1 don't work - rear speakers recognized as mic
by GitHub issues - opened 18 Feb '25

18 Feb '25

alsa-project/alsa-ucm-conf issue #510 was opened from atazx: kubuntu 24.10 MSI X870 Mag Tomahawk motherboard, with integrated ALC4080 (usb id 0db0:cd0e) I tried to install the latest version of alsa-ucm-conf (1.2.13) On this card to have 5.1 audio you have to connect the rear speakers to the front panel, in what would normally be the microphone input, in fact when I connect the speakers they are recognized as a microphone. (only with ucm-conf 1.2.13) front speaker always works rear speakers only work after rebooting from windows center speakers do not work is there a way to fix this problem? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/510 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1 1/3] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 17 Feb '25

17 Feb '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convery the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - update the year from 2024 to 2025. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. --- include/sound/tas2781-dsp.h | 26 +++++++++++++++++++++++--- include/sound/tas2781.h | 13 ++++++++++++- 2 files changed, 35 insertions(+), 4 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..77a240daaae0 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,23 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + unsigned char thr[3]; + unsigned char thr2[3]; + unsigned char plt_flg[3]; + unsigned char sin_gn[3]; + unsigned char sin_gn2[3]; + /* real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + unsigned char a2_reg[3]; + unsigned short size; + char *data; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +133,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 0f1e585635bb..f2eda8ac66e8 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -112,6 +112,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -126,6 +132,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -170,6 +177,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -195,6 +203,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); -- 2.34.1

2 3

[PATCH alsa-lib] seq: Fix bogus return of snd_seq_client_info_get_ump_conversion()
by Takashi Iwai 17 Feb '25

17 Feb '25

snd_seq_client_info_get_ump_conversion() should have returned the proper bit of group_filter bit field, but it just did return midi_version field -- a stupid copy & paste error. Let's fix it. Fixes: 2aefb5c41cc0 ("seq: Add UMP support") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/seq/seq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/seq/seq.c b/src/seq/seq.c index d3af93f58579..38cb1afd872f 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1870,7 +1870,7 @@ int snd_seq_client_info_get_ump_groupless_enabled(const snd_seq_client_info_t *i int snd_seq_client_info_get_ump_conversion(const snd_seq_client_info_t *info) { assert(info); - return info->midi_version; + return !(info->group_filter & SNDRV_SEQ_FILTER_NO_CONVERT); } /** -- 2.43.0

1 0

Ask snd_seq_set_client_midi_version
by Correo Alternativo 17 Feb '25

17 Feb '25

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.

3 3

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> ### Device interface The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring The device exposes all 20 channels in all three playback modes. No mixer controls are available with the current driver. ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic detection) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic detection) - XLR out: AES3 only ### Channel configurations The _HiFi_ use case only exposes the most common channel configurations to limit combinatorial complexity: - Analog line outputs in stereo as well as 4.0 and 5.1 surround, - analog microphone and instrument inputs in mono only, - ADAT Optical outputs and inputs in direct 8‐channel mapping, and - all other outputs and inputs in stereo. [alsa-info.txt](https://github.com/user-attachments/files/18814815/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> ### Device interface The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring The device exposes all 20 channels in all three playback modes. No mixer controls are available with the current driver. ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic detection) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic detection) - XLR out: AES3 only ### Channel configurations The _HiFi_ use case only exposes the most common channel configurations to limit combinatorial complexity: - Analog line outputs in stereo as well as 4.0 and 5.1 surround, - analog microphone and instrument inputs in mono only, - ADAT Optical outputs and inputs in direct 8‐channel mapping, and - all other outputs and inputs in stereo. [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> ### Device interface The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring The device exposes all 20 channels in all three playback modes. No hardware controls are available with the current driver. ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic detection) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic detection) - XLR out: AES3 only ### Channel configurations The _HiFi_ use case only exposes the most common channel configurations to limit combinatorial complexity: - Analog line outputs in stereo as well as 4.0 and 5.1 surround, - analog microphone and instrument inputs in mono only, - ADAT Optical outputs and inputs in direct 8‐channel mapping, and - all other outputs and inputs in stereo. [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring The device exposes all 20 channels in all three playback modes. No hardware controls are available with the current driver. ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic detection) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic detection) - XLR out: AES3 only ### Channel configurations The _HiFi_ use case only exposes the most common channel configurations to limit combinatorial complexity: - Analog line outputs in stereo as well as 4.0 and 5.1 surround, - analog microphone and instrument inputs in mono only, - ADAT Optical outputs and inputs in direct 8‐channel mapping, and - all other outputs and inputs in stereo. [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic) - XLR out: AES3 only [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 14 Feb '25

14 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are S/PDIF via coax - channels 10 and 11 are AES/EBU or S/PDIF via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK - _ADAT OUT_: TOSLINK - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[GIT PULL] ASoC fixes for v6.14-rc2
by Mark Brown 14 Feb '25

14 Feb '25

The following changes since commit 2014c95afecee3e76ca4a56956a936e23283f05b: Linux 6.14-rc1 (2025-02-02 15:39:26 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc2 for you to fetch changes up to 571b69f2f9b1ec7cf7d0e9b79e52115a87a869c4: ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device (2025-02-13 11:34:32 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 More fixes and deviec quirks, most of them driver specific including a few SOF robustness fixes. Nothing super remarkable individually. ---------------------------------------------------------------- Bard Liao (2): ASoC: Intel: soc-acpi-intel-tgl-match: declare adr as ull ASoC: Intel: soc-acpi-intel-mtl-match: declare adr as ull Cristian Ciocaltea (4): ASoC: SOF: amd: Add post_fw_run_delay ACP quirk ASoC: SOF: amd: Drop unused includes from Vangogh driver ASoC: SOF: amd: Handle IPC replies before FW_BOOT_COMPLETE ASoC: SOF: amd: Add branch prediction hint in ACP IRQ handler John Keeping (1): ASoC: rockchip: i2s-tdm: fix shift config for SND_SOC_DAIFMT_DSP_[AB] Kuninori Morimoto (4): ASoC: simple-card-utils.c: add missing dlc->of_node ASoC: rsnd: indicate unsupported clock rate ASoC: rsnd: don't indicate warning on rsnd_kctrl_accept_runtime() ASoC: rsnd: adjust convert rate limitation Mark Brown (5): ASoC: Intel: soc-acpi-intel-tgl/mtl-match: declare adr ASoC: rsnd: adjust convert rate in 1% ASoC: SOF: Correct sps->stream and cstream nullity ASoC: Intel: sof_sdw: Add support for new devices ASoC: Intel: soc-acpi-intel-ptl-match typo fixups Nikita Zhandarovich (1): ASoC: fsl_micfil: Enable default case in micfil_set_quality() Peter Ujfalusi (5): ASoC: SOF: stream-ipc: Check for cstream nullity in sof_ipc_msg_data() ASoC: SOF: pcm: Clear the susbstream pointer to NULL on close ASoC: SOF: ipc4-topology: Harden loops for looking up ALH copiers ASoC: Intel: soc-acpi-intel-ptl-match: revise typo of rt712_vb + rt1320 support ASoC: Intel: soc-acpi-intel-ptl-match: revise typo of rt713_vb_l2_rt1320_l13 Richard Fitzgerald (2): ASoC: Intel: sof_sdw: Add lookup of quirk using PCI subsystem ID ASoC: Intel: sof_sdw: Add quirk for Asus Zenbook S14 Shenghao Ding (1): ASoC: tas2781: drop a redundant code Shengjiu Wang (1): ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device Stefan Binding (2): ASoC: cs35l41: Fallback to using HID for system_name if no SUB is available ASoC: cs35l41: Fix acpi_device_hid() not found Terry Cheong (1): ASoC: SOF: Intel: hda: add softdep pre to snd-hda-codec-hdmi module Uday M Bhat (1): ASoC: Intel: sof_sdw: Add support for Fatcat board with BT offload enabled in PTL platform Vitaly Rodionov (1): ASoC: arizona/madera: use fsleep() in up/down DAPM event delays. sound/soc/codecs/arizona.c | 14 +-- sound/soc/codecs/cs35l41.c | 30 ++++-- sound/soc/codecs/madera.c | 10 +- sound/soc/codecs/tas2781-i2c.c | 4 +- sound/soc/codecs/wm5110.c | 8 +- sound/soc/fsl/fsl_micfil.c | 2 + sound/soc/fsl/imx-audmix.c | 31 ------ sound/soc/generic/simple-card-utils.c | 1 + sound/soc/intel/boards/sof_sdw.c | 41 ++++++-- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 12 +-- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 16 +-- sound/soc/renesas/rcar/core.c | 14 --- sound/soc/renesas/rcar/rsnd.h | 1 - sound/soc/renesas/rcar/src.c | 116 +++++++++++++++++----- sound/soc/renesas/rcar/ssi.c | 3 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 4 +- sound/soc/sof/amd/acp-ipc.c | 25 +++-- sound/soc/sof/amd/acp.c | 1 + sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-vangogh.c | 2 - sound/soc/sof/amd/vangogh.c | 22 +++- sound/soc/sof/intel/hda-codec.c | 1 + sound/soc/sof/ipc4-topology.c | 12 ++- sound/soc/sof/pcm.c | 2 + sound/soc/sof/stream-ipc.c | 6 +- 26 files changed, 245 insertions(+), 136 deletions(-)

2 1

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 14 Feb '25

14 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are S/PDIF via coax - channels 10 and 11 are AES/EBU or S/PDIF via XLR - channels 12 to 19 are ADAT (8 channels) or S/PDIF (12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring ### Physical connector layout #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK - _ADAT OUT_: TOSLINK - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 14 Feb '25

14 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are S/PDIF via coax - channels 10 and 11 are AES/EBU or S/PDIF via XLR - channels 12 to 19 are ADAT (8 channels) or S/PDIF (12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring ### Physical connector layout #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK - _ADAT OUT_: TOSLINK - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - opened 14 Feb '25

14 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was opened from illdefined: [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH V4 0/6] soundwire: amd: code improvements and new platform support
by Vijendar Mukunda 13 Feb '25

13 Feb '25

This patch series consists of code improvements and new platform support for ACP7.0 & ACP7.1 platforms. Changes since v3: - Remove extra white line in amd_sdw_manager_start() function. Changes since v2: - Rebase the patch series after dropping the patch which adds conditional check for SoundWire manager device resume when wake event is asserted. Changes since v1: - Drop white line insertion in the code. - Add a conditional check for SoundWire manager device resume when wake event is asserted. - Add new patch to set ACP_PME_EN during runtime suspend sequence - Split the existing implementation of host wake interrupt mask enablement and setting device power state in to two separate patches and refactor set device power state logic. - Drop PME status clear logic. Vijendar Mukunda (6): 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: set device power state during suspend/resume sequence soundwire: amd: set ACP_PME_EN during runtime suspend sequence soundwire: amd: add soundwire host wake interrupt enable/disable sequence drivers/soundwire/amd_manager.c | 149 +++++++++++++++++++++++++++++- drivers/soundwire/amd_manager.h | 26 ++++++ include/linux/soundwire/sdw_amd.h | 2 + 3 files changed, 172 insertions(+), 5 deletions(-) -- 2.34.1

3 8

[PATCH] ASoC: amd: ps: use switch statements for acp pci revision id check
by Vijendar Mukunda 13 Feb '25

13 Feb '25

Use switch statements for acp pci revision id check in SoundWire dma irq handling. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 220dca8cba85..2ff8e67c19bd 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -111,16 +111,21 @@ static short int check_and_handle_sdw_dma_irq(struct acp63_dev_data *adata, u32 stream_id = ACP63_SDW0_AUDIO2_RX; break; } - if (adata->acp_rev >= ACP70_PCI_REV) - adata->acp70_sdw0_dma_intr_stat[stream_id] = 1; - else + switch (adata->acp_rev) { + case ACP63_PCI_REV: adata->acp63_sdw0_dma_intr_stat[stream_id] = 1; - + break; + case ACP70_PCI_REV: + case ACP71_PCI_REV: + adata->acp70_sdw0_dma_intr_stat[stream_id] = 1; + break; + } sdw_dma_irq_flag = 1; } } } - if (adata->acp_rev == ACP63_PCI_REV) { + switch (adata->acp_rev) { + case ACP63_PCI_REV: if (ext_intr_stat1 & ACP63_P1_AUDIO1_RX_THRESHOLD) { writel(ACP63_P1_AUDIO1_RX_THRESHOLD, adata->acp63_base + ACP_EXTERNAL_INTR_STAT1); @@ -133,7 +138,9 @@ static short int check_and_handle_sdw_dma_irq(struct acp63_dev_data *adata, u32 adata->acp63_sdw1_dma_intr_stat[ACP63_SDW1_AUDIO1_TX] = 1; sdw_dma_irq_flag = 1; } - } else { + break; + case ACP70_PCI_REV: + case ACP71_PCI_REV: if (ext_intr_stat1 & ACP70_P1_SDW_DMA_IRQ_MASK) { for (index = ACP70_P1_AUDIO2_RX_THRESHOLD; index <= ACP70_P1_AUDIO0_TX_THRESHOLD; index++) { @@ -166,6 +173,7 @@ static short int check_and_handle_sdw_dma_irq(struct acp63_dev_data *adata, u32 } } } + break; } return sdw_dma_irq_flag; } -- 2.34.1

2 1

Re: [PATCH v14 29/53] usb: host: xhci-plat: Set XHCI max interrupters if property is present
by Dennis Babcock 12 Feb '25

12 Feb '25

Hi Dennis Louis Babcock jr

1 0

Add HIFI Config support for QCM6490-IDP and QCS6490-RB3Gen2 Qualcomm boards
by GitHub pull_request - opened 12 Feb '25

12 Feb '25

alsa-project/alsa-ucm-conf pull request #508 was opened from mohsRafi: Add HIFI Config support for QCM6490-IDP and QCS6490-RB3Gen2 Qualcomm boards Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/508 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/508.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add HIFI config support for QCM6490-IDP, QCS6490-RB3Gen2 Qualcomm boards
by GitHub pull_request - opened 12 Feb '25

12 Feb '25

alsa-project/alsa-ucm-conf pull request #507 was opened from mohsRafi: Add HIFI config support for QCM6490-IDP, QCS6490-RB3Gen2 Qualcomm boards Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/507 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/507.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

RE: [Bug 219743] [REGRESSION, BISECTED] ASUS GA402XY Headset mic stopped working after 6.12.2
by Kailang 12 Feb '25

12 Feb '25

Fixup headset MIC no function.

1 0

[PATCH V2 00/23] ASoC: amd: acp70: add soundwire and acp pdm support
by Vijendar Mukunda 11 Feb '25

11 Feb '25

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. Changes since v1: - Change the code sequence for handling SoundWire wake interrupts. - Add new patch to restore host wake interrupt mask - Add new patch to enable ACP PME during ACP init sequence. - Change ACP PCI driver PM ops code sequence. Vijendar Mukunda (23): 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 wake interrupt handling ASoC: amd: acp70: enable soundwire host wake irq mask ASoC: amd: acp70: enable acp pme enable during acp init sequence 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 | 288 ++++++++ sound/soc/amd/acp70/pci-acp70.c | 833 +++++++++++++++++++++++ sound/soc/amd/mach-config.h | 1 + 14 files changed, 2871 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

4 35

[PATCH V3 00/25] ASoC: amd: Add support for ACP7.0 & ACP7.1 platforms
by Vijendar Mukunda 10 Feb '25

10 Feb '25

This patch series includes the below changes - Refactor existing ACP6.3 platform ACP PCI driver, SoundWire DMA driver code. - Add Audio IO support for ACP7.0 and ACP7.1 platforms for SoundWire IO and ACP PDM controller combination. - 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. Changes since v2: - As per Mark's comment, to avoid code duplication drop separate folder(acp70) for ACP7.0 & ACP7.1 platforms and refactor existing ACP6.3 platform ACP PCI driver, SoundWire DMA driver code and add support for ACP7.0 & ACP7.1 platforms. - Add ACP PCI driver hw_ops for ACP6.3, ACP7.0 & ACP7.1 platforms. - Implement platform specific code in Soundwire DMA driver by adding acp pci revision id checks for ACP6.3/ACP7.0/ACP7.1 platforms. - Refactor existing ACP PCI driver Interrupt handler code. - Add new Soundwire machine for RT1320 Speaker + RT722 multi function codec combination. - Add machine driver quirks for Dell SKU's. Changes since v1: - Change the code sequence for handling SoundWire wake interrupts. - Add new patch to restore host wake interrupt mask - Add new patch to enable ACP PME during ACP init sequence. - Change ACP PCI driver PM ops code sequence. Vijendar Mukunda (25): ASoC: amd: ps: rename structure names, variable and other macros ASoC: amd: ps: use macro for ACP6.3 pci revision id ASoC: amd: ps: add acp pci driver hw_ops for acp6.3 platform ASoC: amd: ps: add callback functions for acp pci driver pm ops ASoC: amd: ps: add callback to read acp pin configuration ASoC: amd: ps: add soundwire dma irq thread callback ASoC: amd: ps: refactor soundwire dma interrupt handling ASoC: amd: ps: store acp revision id in SoundWire dma driver private data ASoC: amd: ps: refactor soundwire dma driver code ASoC: amd: ps: refactor soundwire dma interrupts enable/disable sequence ASoC: amd: ps: rename acp_restore_sdw_dma_config() function ASoC: amd: ps: add pci driver hw_ops for ACP7.0 & ACP7.1 variants ASoC: amd: ps: add pm ops related hw_ops for ACP7.0 & ACP7.1 platforms ASoC: amd: ps: add ACP7.0 & ACP7.1 specific soundwire dma driver changes ASoC: amd: ps: implement function to restore dma config for ACP7.0 platform ASoC: amd: ps: add soundwire dma interrupts handling for ACP7.0 platform ASoC: amd: ps: add soundwire wake interrupt handling ASoC: amd: ps: update module description ASoC: amd: ps: update file description and copyright year ASoC: amd: update Pink Sardine platform Kconfig description 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 ASoC: amd: acp: amd-acp70-acpi-match: Add RT1320 & RT722 combination soundwire machine ASoC: amd: amd_sdw: Add quirks for Dell SKU's sound/soc/amd/Kconfig | 10 +- sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/Makefile | 2 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 45 +++ sound/soc/amd/acp/acp-sdw-mach-common.c | 34 ++ sound/soc/amd/acp/amd-acp70-acpi-match.c | 160 ++++++++ sound/soc/amd/acp/soc_amd_sdw_common.h | 12 + sound/soc/amd/mach-config.h | 1 + sound/soc/amd/ps/Makefile | 2 +- sound/soc/amd/ps/acp63.h | 246 ++++++++++-- sound/soc/amd/ps/pci-ps.c | 422 +++++++++----------- sound/soc/amd/ps/ps-common.c | 475 +++++++++++++++++++++++ sound/soc/amd/ps/ps-pdm-dma.c | 6 +- sound/soc/amd/ps/ps-sdw-dma.c | 375 ++++++++++++++---- 14 files changed, 1440 insertions(+), 351 deletions(-) create mode 100644 sound/soc/amd/acp/amd-acp70-acpi-match.c create mode 100644 sound/soc/amd/ps/ps-common.c -- 2.34.1

3 28

[PATCH] ALSA: hda/conexant: Add quirk for HP ProBook 450 G4 mute LED
by John Veness 10 Feb '25

10 Feb '25

Allows the LED on the dedicated mute button on the HP ProBook 450 G4 laptop to change colour correctly. Signed-off-by: John Veness <john-linux(a)pelago.org.uk> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 538c37a78..84ab357b8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1080,6 +1080,7 @@ static const struct hda_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x8231, "HP ProBook 450 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), -- 2.48.1

1 0

alsaloop doesn't pass sound with libsamplerate resampling on raspberry pi 5
by GitHub issues - edited 10 Feb '25

10 Feb '25

alsa-project/alsa-utils issue #291 was edited from willangley: Hi alsa-utils team! I'm using `alsaloop`s on a Raspberry Pi 5 as part of a slightly ridiculous signal chain: * Yamaha P-525 keyboard * USB C-to-A cable * Raspberry Pi 5 * USB A-to-B cable * ESI U24XL * SPDIF in/out * Focusrite Clarett+ 4Pre * ... * MacBook Pro [Why? The analog Aux In/Aux Out on the keyboard have an always on loopback connection, which is undesirable when playing to a video meeting. The USB port has user controllable loopback.] Two alsaloop processes run on the Pi, one moving sound from the computer to the piano, one moving sound from the piano to the computer. Both run under `supervisord`. I've got them running at acceptable latency — RTL Utility says 40ms from my MacBook Pro. But I haven't been able to get any of the high quality resampling options working -- everything other than add/drop stops passing sound within a few seconds. I'm surprised; this Pi should be powerful enough to handle this. Any ideas how to troubleshoot it? I didn't see errors in logs at the default log level. Thank you! Will Issue URL : https://github.com/alsa-project/alsa-utils/issues/291 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

alsaloop doesn't pass sound with libsamplerate resampling on raspberry pi 5
by GitHub issues - edited 10 Feb '25

10 Feb '25

alsa-project/alsa-utils issue #291 was edited from willangley: Hi alsa-utils team! I'm using `alsaloop`s on a Raspberry Pi 5 as part of a slightly ridiculous signal chain: * Yamaha P-525 keyboard * USB C-to-A cable * Raspberry Pi 5 * USB A-to-B cable * ESI U24XL * SPDIF in/out * Focusrite Clarett+ 4Pre * ... * MacBook Pro [Why? The analog Aux In/Aux Out on the keyboard have an always on loopback connection, which is undesirable when playing to a video meeting. The USB port has user controllable loopback.] Two alsaloop processes run on the Pi, one moving sound from the computer to the piano, one moving sound from the piano to the computer. Both run under `supervisord`. I've got them running at acceptable latency — RTL Utility says 40ms from my MacBook Pro. But I haven't been able to get any of the high quality resampling options working -- everything other than add/drop stops passing sound within a few seconds. I'm surprised; this Pi should be powerful enough to handle this. Any ideas how to troubleshoot it? Thank you! Will Issue URL : https://github.com/alsa-project/alsa-utils/issues/291 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

alsaloop doesn't pass sound with libsamplerate on raspberry pi 5
by GitHub issues - opened 10 Feb '25

10 Feb '25

alsa-project/alsa-utils issue #291 was opened from willangley: Hi alsa-utils team! I'm using `alsaloop`s on a Raspberry Pi 5 as part of a slightly ridiculous signal chain: * Yamaha P-525 keyboard * USB C-to-A cable * Raspberry Pi 5 * USB A-to-B cable * ESI U24XL * SPDIF in/out * Focusrite Clarett+ 4Pre * ... * MacBook Pro [Why? The analog Aux In/Aux Out on the keyboard have an always on loopback connection, which is undesirable when playing to a video meeting. The USB port has user controllable loopback.] Two alsaloop processes run on the Pi, one moving sound from the computer to the piano, one moving sound from the piano to the computer. Both run under `supervisord`. I've got them running at acceptable latency — RTL Utility says 40ms from my MacBook Pro. But I haven't been able to get any of the high quality resampling options working -- everything other than add/drop stops passing sound within a few seconds. I'm surprised; this Pi should be powerful enough to handle this. Any ideas how to troubleshoot it? Thank you! Will Issue URL : https://github.com/alsa-project/alsa-utils/issues/291 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Soundblaster G8 USB Audio not working correctly
by GitHub issues - opened 07 Feb '25

07 Feb '25

alsa-project/alsa-lib issue #437 was opened from Starfoxfs: Soundblaster G8 works partly under Linux Mint 22.1 Cinnamon. Audio playback works in Analog stereo or Duplex mode with headphones and these are displayed as Speakers. Digital SPDIF Out works and also the switch button between headphones and Digital out to external Speakers (only with pipewire not with Pulseaudio). What doesn't work is the Direct Mode and all the Crystalizer surround settings as well as output in virtual 5.1 or 7.1 sound. Microphone doesn´t work, if it plugged in i have no sound at all. [alsa-info.txt](https://github.com/user-attachments/files/18710112/alsa-info… Issue URL : https://github.com/alsa-project/alsa-lib/issues/437 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

rt722: change output volume of speaker to 0dB
by GitHub pull_request - opened 07 Feb '25

07 Feb '25

alsa-project/alsa-ucm-conf pull request #506 was opened from shumingfan: This patch also changes hardware gain to software gain when using volume bar. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/506 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/506.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

soundwire : LnK SoundWire Protocol Analyzer on Linux
by Muni Sekhar 06 Feb '25

06 Feb '25

Hi all, I hope this message finds you well. I am planning to use the LnK SoundWire Protocol Analyzer on a Linux platform. However, I am uncertain whether the current Linux kernel includes the necessary SoundWire drivers to support this . I would appreciate any guidance on how to use the LnK SoundWire Protocol Analyzer with the Linux kernel, as well as information on the specific device drivers required. Could you please provide details on: The compatibility of the LnK SoundWire Protocol Analyzer with the Linux kernel. The steps to set up and use the analyzer on a Linux platform. The necessary device drivers and any additional configurations needed. Thank you for your assistance. -- Thanks, Sekhar

1 0

[PATCH v1] ASoC: tas2781: drop a redundant code
by Shenghao Ding 06 Feb '25

06 Feb '25

Report from internal ticket, priv->cali_data.data devm_kzalloc twice, drop the first one, it is the unnecessary one. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - update the year from 2024 to 2025. - drop a redundant code. --- sound/soc/codecs/tas2781-i2c.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index a730ab6ad4e3..90c5b2e74d12 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -1260,8 +1260,6 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) (cali_data->cali_dat_sz_per_dev + 1) + 1 + 15 + 1; priv->cali_data.total_sz = priv->ndev * (cali_data->cali_dat_sz_per_dev + 1); - priv->cali_data.data = devm_kzalloc(priv->dev, - ext_cali_data->max, GFP_KERNEL); cali_ctrls[i].name = cali_name; cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; cali_ctrls[i].info = snd_soc_bytes_info_ext; -- 2.34.1

2 1

[PATCH v1] MAINTAINERS: remove Kevin Lu from TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS
by Shenghao Ding 04 Feb '25

04 Feb '25

Due to internal re-org, Kevin Lu has no longer maintained audio driver. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 896a307fa065..bc0bf402818d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23370,7 +23370,6 @@ F: sound/soc/ti/ TEXAS INSTRUMENTS AUDIO (ASoC/HDA) 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 -- 2.34.1

1 0

[PATCH alsa-utils] amixer: implement handling of TLV bytes kcontrols
by Jan Dakinevich 03 Feb '25

03 Feb '25

SND_SOC_BYTES_EXT kcontrols are deprecated and SND_SOC_BYTES_TLV should be used instead. However, there is no support of these kcontrols in amixer. This commit introduces a mechanism to show and write the content of SND_SOC_BYTES_TLV-based kcontrols. Example: $ amixer cset name="Test" 0x10,0x20,0x30,0x40,0x50,0x60,0x70,0x80,0x90,0xa0 $ amixer cget name="Test" numid=42,iface=MIXER,name='Test' ; type=BYTES,access=-----RW-,values=10 : values=0x10,0x20,0x30,0x40,0x50,0x60,0x70,0x80,0x90,0xa0 Signed-off-by: Jan Dakinevich <jan.dakinevich(a)salutedevices.com> --- amixer/amixer.c | 180 +++++++++++++++++++++++++++++++++++++----------- 1 file changed, 141 insertions(+), 39 deletions(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index 07e9819e5cdb..09f774fe2885 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -609,6 +609,8 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, { int err; unsigned int item, idx, count, *tlv; + uint8_t *tlv_bytes = NULL; + unsigned int tlv_bytes_count = 0; snd_ctl_elem_type_t type; snd_ctl_elem_id_t *id; snd_ctl_elem_info_t *info; @@ -628,6 +630,18 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, } count = snd_ctl_elem_info_get_count(info); type = snd_ctl_elem_info_get_type(info); + if (type == SND_CTL_ELEM_TYPE_BYTES && + !snd_ctl_elem_info_is_readable(info) && + !snd_ctl_elem_info_is_writable(info)) { + if (count <= 2 * sizeof(unsigned int)) + return -EINVAL; + tlv_bytes_count = count; + count -= 2 * sizeof(unsigned int); + + tlv_bytes = malloc(tlv_bytes_count); + if (!tlv_bytes) + return -ENOMEM; + } printf("%s; type=%s,access=%s,values=%u", space, control_type(info), control_access(info), count); switch (type) { case SND_CTL_ELEM_TYPE_INTEGER: @@ -661,16 +675,30 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, break; } if (level & LEVEL_BASIC) { - if (!snd_ctl_elem_info_is_readable(info)) + if (snd_ctl_elem_info_is_readable(info)) { + err = snd_hctl_elem_read(elem, control); + if (err < 0) { + error("Control %s element read error: %s\n", card, snd_strerror(err)); + return err; + } + } else if (tlv_bytes) { + err = snd_hctl_elem_tlv_read(elem, (void *)tlv_bytes, tlv_bytes_count); + if (err < 0) { + error("Control %s element TLV read error: %s\n", card, snd_strerror(err)); + free(tlv_bytes); + return err; + } + } else goto __skip_read; - if ((err = snd_hctl_elem_read(elem, control)) < 0) { - error("Control %s element read error: %s\n", card, snd_strerror(err)); - return err; - } + printf("%s: values=", space); for (idx = 0; idx < count; idx++) { if (idx > 0) printf(","); + if (tlv_bytes) { + printf("0x%02x", tlv_bytes[idx + 2 * sizeof(unsigned int)]); + continue; + } switch (type) { case SND_CTL_ELEM_TYPE_BOOLEAN: printf("%s", snd_ctl_elem_value_get_boolean(control, idx) ? "on" : "off"); @@ -699,17 +727,11 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, } } printf("\n"); + if (tlv_bytes) + goto __skip_tlv; __skip_read: if (!snd_ctl_elem_info_is_tlv_readable(info)) goto __skip_tlv; - /* skip ASoC ext bytes controls that may have huge binary TLV data */ - if (type == SND_CTL_ELEM_TYPE_BYTES && - !snd_ctl_elem_info_is_readable(info) && - !snd_ctl_elem_info_is_writable(info)) { - printf("%s; ASoC TLV Byte control, skipping bytes dump\n", space); - goto __skip_tlv; - } - tlv = malloc(4096); if ((err = snd_hctl_elem_tlv_read(elem, tlv, 4096)) < 0) { error("Control %s element TLV read error: %s\n", card, snd_strerror(err)); @@ -720,6 +742,7 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, free(tlv); } __skip_tlv: + free(tlv_bytes); return 0; } @@ -1133,16 +1156,110 @@ static int parse_simple_id(const char *str, snd_mixer_selem_id_t *sid) return 0; } +static int parse_bytes(const char *str, void *ptr, unsigned int len) +{ + char *p; + unsigned long tmp; + unsigned int idx = 0; + unsigned char *data = ptr; + + for (;;) { + if (!*str) + break; + if (idx == len) + return -EINVAL; + + tmp = strtoul(str, &p, 0); + if (p - str < 1) + return -EINVAL; + if (*p == ',') + str = p + 1; + else if (*p == '\0') + str = p; + else + return -EINVAL; + + if (tmp > 255) + return -EINVAL; + + data[idx] = tmp; + idx++; + } + + if (!idx) + return -EINVAL; + + return idx; +} + +static int cset_tlv(int argc, char *argv[], snd_ctl_t *handle, + snd_ctl_elem_info_t *info, snd_ctl_elem_id_t *id) +{ + int err; + unsigned int count; + unsigned int *tlv = NULL; + + count = snd_ctl_elem_info_get_count(info); + if (count <= 2 * sizeof(unsigned int)) { + err = -EINVAL; + goto out; + } + tlv = malloc(count); + if (!tlv) { + err = -ENOMEM; + goto out; + } + err = parse_bytes(argv[1], tlv + 2, + count - 2 * sizeof(unsigned int)); + if (err < 0) + goto out; + tlv[0] = 0; + tlv[1] = err; + + err = snd_ctl_elem_tlv_write(handle, id, tlv); + +out: + free(tlv); + return err; +} + +static int cset_elem(int argc, char *argv[], snd_ctl_t *handle, + snd_ctl_elem_info_t *info, snd_ctl_elem_id_t *id) +{ + int err; + snd_ctl_elem_value_t *control; + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_value_set_id(control, id); + if ((err = snd_ctl_elem_read(handle, control)) < 0) { + if (!ignore_error) + error("Cannot read the given element from control %s\n", card); + return err; + } + err = snd_ctl_ascii_value_parse(handle, control, info, argv[1]); + if (err < 0) { + if (!ignore_error) + error("Control %s parse error: %s\n", card, snd_strerror(err)); + return err; + } + if ((err = snd_ctl_elem_write(handle, control)) < 0) { + if (!ignore_error) + error("Control %s element write error: %s\n", card, snd_strerror(err)); + return err; + } + + return 0; +} + static int cset(int argc, char *argv[], int roflag, int keep_handle) { int err; static snd_ctl_t *handle = NULL; snd_ctl_elem_info_t *info; snd_ctl_elem_id_t *id; - snd_ctl_elem_value_t *control; + snd_ctl_elem_type_t type; snd_ctl_elem_info_alloca(&info); snd_ctl_elem_id_alloca(&id); - snd_ctl_elem_value_alloca(&control); if (argc < 1) { fprintf(stderr, "Specify a full control identifier: [[iface=<iface>,][name='name',][index=<index>,][device=<device>,][subdevice=<subdevice>]]|[numid=<numid>]\n"); @@ -1175,30 +1292,15 @@ static int cset(int argc, char *argv[], int roflag, int keep_handle) } snd_ctl_elem_info_get_id(info, id); /* FIXME: Remove it when hctl find works ok !!! */ if (!roflag) { - snd_ctl_elem_value_set_id(control, id); - if ((err = snd_ctl_elem_read(handle, control)) < 0) { - if (ignore_error) - return 0; - error("Cannot read the given element from control %s\n", card); - if (! keep_handle) { - snd_ctl_close(handle); - handle = NULL; - } - return err; - } - err = snd_ctl_ascii_value_parse(handle, control, info, argv[1]); - if (err < 0) { - if (!ignore_error) - error("Control %s parse error: %s\n", card, snd_strerror(err)); - if (!keep_handle) { - snd_ctl_close(handle); - handle = NULL; - } - return ignore_error ? 0 : err; - } - if ((err = snd_ctl_elem_write(handle, control)) < 0) { - if (!ignore_error) - error("Control %s element write error: %s\n", card, snd_strerror(err)); + type = snd_ctl_elem_info_get_type(info); + if (type == SND_CTL_ELEM_TYPE_BYTES && + !snd_ctl_elem_info_is_readable(info) && + !snd_ctl_elem_info_is_writable(info)) { + err = cset_tlv(argc, argv, handle, info, id); + } else + err = cset_elem(argc, argv, handle, info, id); + + if (err) { if (!keep_handle) { snd_ctl_close(handle); handle = NULL; -- 2.34.1

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Compilation failure with Clang-21 due to macro conflicts
by GitHub issues - opened 02 Feb '25

02 Feb '25

alsa-project/alsa-lib issue #436 was opened from ms178: **Description:** Attempting to compile alsa-lib (both the `git` version and version 1.2.13) with Clang-21 (commit `bfa7edcc6652bdb37d53e0cec64926aab3f280eb`) results in multiple compilation errors. These errors stem from conflicts between macros defined in `midifile.h` and member names within structures defined in `include/alsa/ump_msg.h`. **Environment:** * **alsa-lib version:** `git` (and 1.2.13) * **Compiler:** Clang-21 (`bfa7edcc6652bdb37d53e0cec64926aab3f280eb`) * **Operating System:** CachyOS * **Build System:** Autotools **Steps to Reproduce:** 1. Obtain the PKGBUILD and the other files from https://github.com/ms178/archpkgbuilds/blob/main/packages/alsa-lib-main/PKG… 2. Uncomment the patch that I apply to workaround the issue. 3. Configure the build to use Clang-21 (`bfa7edcc6652bdb37d53e0cec64926aab3f280eb`). 4. Attempt to compile the library. **Expected Behavior:** The compilation should complete successfully without errors. **Actual Behavior:** The compilation fails with multiple errors similar to the following: ``` In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:170:27: error: expected member name or ';' after declaration specifiers 170 | snd_ump_msg_midi1_note_t note_on; /**< MIDI1 note-on message */ | ~~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:55:27: note: expanded from macro 'note_on' 55 | #define note_on 0x90 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:170:26: error: expected ';' at end of declaration list 170 | snd_ump_msg_midi1_note_t note_on; /**< MIDI1 note-on message */ | ^ | ; ../include/alsa/ump_msg.h:171:27: error: expected member name or ';' after declaration specifiers 171 | snd_ump_msg_midi1_note_t note_off; /**< MIDI1 note-off message */ | ~~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:54:27: note: expanded from macro 'note_off' 54 | #define note_off 0x80 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:171:26: error: expected ';' at end of declaration list 171 | snd_ump_msg_midi1_note_t note_off; /**< MIDI1 note-off message */ | ^ | ; ../include/alsa/ump_msg.h:173:26: error: expected member name or ';' after declaration specifiers 173 | snd_ump_msg_midi1_cc CCLD rawmidi _t control_change; /**< MIDI1 control-change message */ | ~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:57:28: note: expanded from macro 'control_change' 57 | #define control_change 0xb0 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h CCLD oldapi :31: ../include/alsa/ump_msg.h:173:24: error: expected ';' at end of declaration list 173 | snd_ump_msg_midi1_cc_t control_change; /**< MIDI1 control-change message */ | ^ | ; ../include/alsa/ump_msg.h:432:27: error: expected member name or ';' after declaration specifiers 432 | snd_ump_msg_midi2_note_t note_on; /**< MIDI2 note-on message */ | ~~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:55:27: note: expanded from macro 'note_on' 55 | #define note_on 0x90 | ^ CCLD pcm_min In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:432:26: error: expected ';' at end of declaration list 432 | snd_ump_msg_midi2_note_t note_on; /**< MIDI2 note-on message */ | ^ | ; ../include/alsa/ump_msg.h:433:27: error: expected member name or ';' after declaration specifiers 433 | snd_ump_msg_midi2_note_t note_off; /**< MIDI2 note-off message */ | ~~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:54:27: note: expanded from macro 'note_off' 54 | #define note_off 0x80 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:433:26: error: expected ';' at end of declaration list 433 | snd_ump_msg_midi2_note_t note_off; /**< MIDI2 note-off message */ | ^ | ; ../include/alsa/ump_msg.h:438:26: error: expected member name or ';' after declaration specifiers 438 | snd_ump_msg_midi2_cc_t control_change; /**< MIDI2 control-change message */ | ~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:57:28: note: expanded from macro 'control_change' 57 | #define control_change 0xb0 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:438:24: error: expected ';' at end of declaration list 438 | snd_ump_msg_midi2_cc_t control_change; /**< MIDI2 control-change message */ | ^ | ; ../include/alsa/ump_msg.h:711:27: error: expected member name or ';' after declaration specifiers 711 | snd_ump_msg_set_tempo_t set_tempo; /**< Set Tempo */ | ~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.c:57:20: note: expanded from macro 'set_tempo' 57 | #define set_tempo 0x51 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:711:25: error: expected ';' at end of declaration list 711 | snd_ump_msg_set_tempo_t set_tempo; /**< Set Tempo */ | ^ | ; CCLD control CCLD chmap 14 errors generated. ``` **Root Cause analysis by AI:** The macros `note_on`, `note_off`, `control_change`, and `set_tempo` defined in `midifile.h` are conflicting with member names used in structures within `include/alsa/ump_msg.h`. This leads to incorrect code expansion and subsequent compilation errors. **Workaround:** The following patch applied to `include/asoundlib-head.h` allows the compilation to succeed: ```diff --- a/include/asoundlib-head.h +++ b/include/asoundlib-head.h @@ -28,6 +28,11 @@ #ifndef __ASOUNDLIB_H #define __ASOUNDLIB_H +#undef note_on +#undef note_off +#undef control_change +#undef set_tempo + #include <unistd.h> #include <stdio.h> #include <stdlib.h> ``` **Suggested Solutions by AI:** While the provided workaround resolves the immediate compilation issue, a more robust solution might involve: 1. **Renaming conflicting members:** Consider renaming the members in `ump_msg.h` (e.g., `midi1_note_on`, `midi2_note_off`, etc.) to avoid clashes with commonly used MIDI event names. 2. **Namespace/prefix:** If renaming is not feasible, consider using a namespace or prefix for the `ump_msg.h` members to prevent conflicts. 3. **Conditional undef:** Instead of unconditionally undefining the macros in `asoundlib-head.h`, a more targeted approach could be to undefine them only within the scope of `ump_msg.h`'s inclusion. 4. **Review macro definitions:** Investigate if the macros in `midifile.h` are strictly necessary or if they can be replaced with constants or enums within a namespace. **Additional Notes:** * I am not a programmer and cannot provide a definitive fix. The provided workaround is based on AI feedback. Issue URL : https://github.com/alsa-project/alsa-lib/issues/436 Repository URL: https://github.com/alsa-project/alsa-lib

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alsaucm fails with unknown If.Condition.Type
by GitHub issues - opened 02 Feb '25

02 Feb '25

alsa-project/alsa-utils issue #290 was opened from abewieland: I get the following when running `listcards`: ``` $ alsaucm listcards ALSA lib ucm_cond.c:367:(if_eval) unknown If.Condition.Type ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -22 ALSA lib parser.c:2978:(uc_mgr_scan_master_configs) Unable to open '-hw:0': Invalid argument alsaucm: error failed to get card list: Invalid argument ``` Let me know if I can provide any better debugging data. My system is a Dell laptop with Intel Tiger Lake and Realtek ALC3204, if that's helpful. Issue URL : https://github.com/alsa-project/alsa-utils/issues/290 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

ALC4082 - Asrock B850 Steel Legend
by GitHub issues - opened 01 Feb '25

01 Feb '25

alsa-project/alsa-ucm-conf issue #505 was opened from ct85msi: Hello, ALC4082 isn`t functioning properly, please add the needed support. Here is the output of alsa-info.sh: http://alsa-project.org/db/?f=212e4da6e5c545eeb6d105b17a769202aff651a7 Thanks in advance, Sorin Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/505 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0