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

3 4

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 3

[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

2 1

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 5

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(-)

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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 (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

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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 (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

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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

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