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[RESEND PATCH v5] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 19 Mar '25

19 Mar '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convey 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> --- v5: - Instead of the lengthy approach, just replace if-continue with else if, has reduced this by 2/3 of LoCs - Use proper error code in fw_parse_fct_param_address() - return directly in tasdevice_dspfw_ready() - drop Unneeded assignment v4: - drop unnecceary "unsigned short size;" and "char *data;" in struct fct_param_address - replace strstr() with strncmp() in fct_param_address_parser() - replace hex_parse_u8() with memcpy() - replace hex_parse_u24() with direct assignment - drop hex_parse_u8() & hex_parse_u24() - Just return directly and use proper error codes and drop "goto out" and "out:" label in fw_parse_fct_param_address - fixed indentation issues in cali_reg_update - fixed the logical splits in tas2781_calib_stop_put() - return -ENOMEM if kmemdup fails in fw_parse_fct_param_address() - remove unnecessary "out:" label 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 | 30 ++++- include/sound/tas2781.h | 11 ++ sound/pci/hda/tas2781_spi_fwlib.c | 8 +- sound/soc/codecs/tas2781-fmwlib.c | 190 ++++++++++++++++++++++---- sound/soc/codecs/tas2781-i2c.c | 215 ++++++++++++++++++++++-------- 5 files changed, 368 insertions(+), 86 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..c3a9efa73d5d 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 +108,27 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + /* Thermal data for PG 1.0 device */ + unsigned char thr[3]; + /* Thermal data for PG 2.0 device */ + unsigned char thr2[3]; + /* Pilot tone enable flag, usually the sine wave */ + unsigned char plt_flg[3]; + /* Pilot tone gain for calibration */ + unsigned char sin_gn[3]; + /* Pilot tone gain for calibration */ + unsigned char sin_gn2[3]; + /* high 32-bit of 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]; + /* check spk resonant frequency */ + unsigned char a2_reg[3]; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +137,7 @@ 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 4d9a37544354..eff011444cc8 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -113,6 +113,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -127,6 +133,7 @@ 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 0e2acbc3c900..19a693d58ee8 100644 --- a/sound/pci/hda/tas2781_spi_fwlib.c +++ b/sound/pci/hda/tas2781_spi_fwlib.c @@ -2,7 +2,7 @@ // // TAS2781 HDA SPI driver // -// Copyright 2024 Texas Instruments, Inc. +// Copyright 2024 - 2025 Texas Instruments, Inc. // // Author: Baojun Xu <baojun.xu(a)ti.com> @@ -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-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index bc2f68bd9846..13a197468193 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -390,10 +390,10 @@ 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); } @@ -560,6 +560,124 @@ static int fw_parse_configuration_data_kernel( return offset; } +static void fct_param_address_parser(struct cali_reg *r, + struct tasdevice_fw *tas_fmw, const unsigned char *data) +{ + struct fct_param_address *p = &tas_fmw->fct_par_addr; + unsigned int i; + + /* + * Calibration parameters locations and data schema in dsp firmware. + * The number of items are flexible, but not more than 20. The dsp tool + * will reseve 20*24-byte space for fct params. In some cases, the + * number of fct param is less than 20, the data will be saved from the + * beginning, the rest part will be stuffed with zero. + * + * fct_param_num (not more than 20) + * for (i = 0; i < fct_param_num; i++) { + * Alias of fct param (20 bytes) + * Book (1 byte) + * Page (1 byte) + * Offset (1 byte) + * CoeffLength (1 byte) = 0x1 + * } + * if (20 - fct_param_num) + * 24*(20 - fct_param_num) pieces of '0' as stuffing + * + * As follow: + * umg_SsmKEGCye = Book, Page, Offset, CoeffLength + * iks_E0 = Book, Page, Offset, CoeffLength + * yep_LsqM0 = Book, Page, Offset, CoeffLength + * oyz_U0_ujx = Book, Page, Offset, CoeffLength + * iks_GC_GMgq = Book, Page, Offset, CoeffLength + * gou_Yao = Book, Page, Offset, CoeffLength + * kgd_Wsc_Qsbp = Book, Page, Offset, CoeffLength + * yec_CqseSsqs = Book, Page, Offset, CoeffLength + * iks_SogkGgog2 = Book, Page, Offset, CoeffLength + * yec_Sae_Y = Book, Page, Offset, CoeffLength + * Re_Int = Book, Page, Offset, CoeffLength + * SigFlag = Book, Page, Offset, CoeffLength + * a1_Int = Book, Page, Offset, CoeffLength + * a2_Int = Book, Page, Offset, CoeffLength + */ + for (i = 0; i < 20; i++) { + const unsigned char *dat = &data[24 * i]; + + /* + * check whether current fct param is empty. + */ + if (dat[23] != 1) + break; + + if (!strncmp(dat, "umg_SsmKEGCye", 20)) + r->pow_reg = TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* high 32-bit of real-time spk impedance */ + else if (!strncmp(dat, "iks_E0", 20)) + r->r0_reg = TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* inverse of real-time spk impedance */ + else if (!strncmp(dat, "yep_LsqM0", 20)) + r->invr0_reg = + TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* low 32-bit of real-time spk impedance */ + else if (!strncmp(dat, "oyz_U0_ujx", 20)) + r->r0_low_reg = + TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* Delta Thermal Limit */ + else if (!strncmp(dat, "iks_GC_GMgq", 20)) + r->tlimit_reg = + TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* Thermal data for PG 1.0 device */ + else if (!strncmp(dat, "gou_Yao", 20)) + memcpy(p->thr, &dat[20], 3); + /* Pilot tone enable flag, usually the sine wave */ + else if (!strncmp(dat, "kgd_Wsc_Qsbp", 20)) + memcpy(p->plt_flg, &dat[20], 3); + /* Pilot tone gain for calibration */ + else if (!strncmp(dat, "yec_CqseSsqs", 20)) + memcpy(p->sin_gn, &dat[20], 3); + /* Pilot tone gain for calibration, useless in PG 2.0 */ + else if (!strncmp(dat, "iks_SogkGgog2", 20)) + memcpy(p->sin_gn2, &dat[20], 3); + /* Thermal data for PG 2.0 device */ + else if (!strncmp(dat, "yec_Sae_Y", 20)) + memcpy(p->thr2, &dat[20], 3); + /* Spk Equivalent Resistance in fixed-point format */ + else if (!strncmp(dat, "Re_Int", 20)) + memcpy(p->r0_reg, &dat[20], 3); + /* Check whether the spk connection is open */ + else if (!strncmp(dat, "SigFlag", 20)) + memcpy(p->tf_reg, &dat[20], 3); + /* check spk resonant frequency */ + else if (!strncmp(dat, "a1_Int", 20)) + memcpy(p->a1_reg, &dat[20], 3); + /* check spk resonant frequency */ + else if (!strncmp(dat, "a2_Int", 20)) + memcpy(p->a2_reg, &dat[20], 3); + } +} + +static int fw_parse_fct_param_address(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + 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__); + return -EINVAL; + } + + /* skip reserved part */ + offset += 40; + + fct_param_address_parser(r, tas_fmw, &data[offset]); + + offset += 480; + + return offset; +} + static int fw_parse_variable_header_kernel( struct tasdevice_priv *tas_priv, const struct firmware *fmw, int offset) @@ -1687,13 +1805,29 @@ 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 = @@ -1702,6 +1836,7 @@ 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: @@ -1717,7 +1852,7 @@ 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; @@ -1953,28 +2088,25 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, struct tasdevice_fw_fixed_hdr *fw_fixed_hdr; struct tasdevice_fw *tas_fmw; int offset = 0; - int ret = 0; + int ret; if (!fmw || !fmw->data) { dev_err(tas_priv->dev, "%s: Failed to read firmware %s\n", __func__, tas_priv->coef_binaryname); - ret = -EINVAL; - goto out; + return -EINVAL; } tas_priv->fmw = kzalloc(sizeof(struct tasdevice_fw), GFP_KERNEL); - if (!tas_priv->fmw) { - ret = -ENOMEM; - goto out; - } + if (!tas_priv->fmw) + return -ENOMEM; + tas_fmw = tas_priv->fmw; tas_fmw->dev = tas_priv->dev; offset = fw_parse_header(tas_priv, tas_fmw, fmw, offset); - if (offset == -EINVAL) { - ret = -EINVAL; - goto out; - } + if (offset == -EINVAL) + return -EINVAL; + fw_fixed_hdr = &(tas_fmw->fw_hdr.fixed_hdr); /* Support different versions of firmware */ switch (fw_fixed_hdr->drv_ver) { @@ -2007,28 +2139,32 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, ret = dspfw_default_callback(tas_priv, fw_fixed_hdr->drv_ver, fw_fixed_hdr->ppcver); if (ret) - goto out; + return ret; break; } offset = tas_priv->fw_parse_variable_header(tas_priv, fmw, offset); - if (offset < 0) { - ret = offset; - goto out; - } + if (offset < 0) + return offset; + offset = tas_priv->fw_parse_program_data(tas_priv, tas_fmw, fmw, offset); - if (offset < 0) { - ret = offset; - goto out; - } + if (offset < 0) + return offset; + offset = tas_priv->fw_parse_configuration_data(tas_priv, tas_fmw, fmw, offset); if (offset < 0) - ret = offset; + return offset; -out: - return ret; + 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) + return offset; + } + + return 0; } int tasdevice_dsp_parser(void *context) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 90c5b2e74d12..b950914b7d48 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, @@ -452,14 +479,15 @@ static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, return 1; } -static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) +static void tas2781_calib_stop_put(struct tasdevice_priv *priv) { const int sum = ARRAY_SIZE(tas2781_cali_start_reg); int i, j; - for (i = 0; i < tas_priv->ndev; i++) { - struct tasdevice *tasdev = tas_priv->tasdevice; + for (i = 0; i < priv->ndev; i++) { + struct tasdevice *tasdev = 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; @@ -467,18 +495,21 @@ static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) for (j = 0; j < sum; j++) { if (p[j].val_len == 1) { if (p[j].is_locked) - tasdevice_dev_write(tas_priv, i, + tasdevice_dev_write(priv, i, TAS2781_TEST_UNLOCK_REG, TAS2781_TEST_PAGE_UNLOCK); - tasdevice_dev_write(tas_priv, i, p[j].reg, + tasdevice_dev_write(priv, i, p[j].reg, p[j].val[0]); } else { if (!p[j].reg) continue; - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + tasdevice_dev_bulk_write(priv, i, p[j].reg, p[j].val, 4); } } + + if (priv->dspbin_typ == TASDEV_ALPHA) + tasdevice_dev_bulk_write(priv, i, t->reg, t->val, 4); } } @@ -590,16 +621,20 @@ 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 +681,19 @@ 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 +708,19 @@ 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 +754,16 @@ 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 +774,16 @@ 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 +1231,51 @@ 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, 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

5 4

No Micrphone detected on ALC4080 - Gigabyte B850I Aorus Pro (rev 1.0)
by GitHub issues - edited 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf issue #528 was edited from chanster: The Rear Microphone is not detected when I plug in my mic. I don't use the Front Audio connection as my case doesn't have I/O for it, I just used the 2 audio ports in the rear. When I try recording directly I can get the mic to work, but it does not show up in the sound settings. `arecord --duration=3 -D hw:CARD=2,DEV=1 -f dat -vvv a.wav` ``` Hardware PCM card 2 'USB Audio' device 1 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Max peak (12000 samples): 0x00000167 # 1% Max peak (12000 samples): 0x000000da # 0% Max peak (12000 samples): 0x000006b1 ## 5% Max peak (12000 samples): 0x00000547 # 4% Max peak (12000 samples): 0x00000729 ## 5% Max peak (12000 samples): 0x00000a77 ## 8% Max peak (12000 samples): 0x000006d2 ## 5% Max peak (12000 samples): 0x000005ec # 4% Max peak (12000 samples): 0x00000627 # 4% Max peak (12000 samples): 0x00000d32 ### 10% Max peak (12000 samples): 0x00000e9f ### 11% Max peak (12000 samples): 0x00003813 ######### 43% Max peak (12000 samples): 0x00006278 ################ 76% Max peak (12000 samples): 0x00007fff #################### 99% Max peak (12000 samples): 0x00006694 ################# 80% Max peak (12000 samples): 0x000068ea ################# 81% Max peak (12000 samples): 0x00004ba0 ############ 59% Max peak (12000 samples): 0x0000207c ###### 25% Max peak (12000 samples): 0x000017e9 #### 18% Max peak (12000 samples): 0x00003938 ######### 44% Max peak (12000 samples): 0x00001f11 ##### 24% Max peak (12000 samples): 0x00000ee5 ### 11% Max peak (12000 samples): 0x00003944 ######### 44% Max peak (12000 samples): 0x00003942 ######### 44% ``` Motherboard: Gigabyte B850I Aorus Pro (rev 1.0) Audio Device: ALC4080 Distro: Fedora 41 Window Manager: Gnome-Shell Packages: ``` dnf list --installed | grep 'pipewire\|pulseaudio\|alsa' alsa-lib.i686 1.2.13-3.fc41 updates alsa-lib.x86_64 1.2.13-3.fc41 updates alsa-ucm.noarch 1.2.13-3.fc41 updates alsa-ucm-utils.x86_64 1.2.13-2.fc41 updates alsa-utils.x86_64 1.2.13-2.fc41 updates pipewire.i686 1.2.7-7.fc41 updates pipewire.x86_64 1.2.7-7.fc41 updates pipewire-alsa.i686 1.2.7-7.fc41 updates pipewire-alsa.x86_64 1.2.7-7.fc41 updates pipewire-gstreamer.x86_64 1.2.7-7.fc41 updates pipewire-libs.i686 1.2.7-7.fc41 updates pipewire-libs.x86_64 1.2.7-7.fc41 updates pipewire-pulseaudio.x86_64 1.2.7-7.fc41 updates pipewire-utils.x86_64 1.2.7-7.fc41 updates pulseaudio-libs.i686 17.0-2.fc41 fedora pulseaudio-libs.x86_64 17.0-2.fc41 fedora pulseaudio-libs-glib2.x86_64 17.0-2.fc41 fedora wine-alsa.i686 10.1-1.fc41 updates wine-alsa.x86_64 10.1-1.fc41 updates wine-pulseaudio.i686 10.1-1.fc41 updates wine-pulseaudio.x86_64 10.1-1.fc41 updates ``` `lsusb | grep -i audio` ``` Bus 001 Device 002: ID 0414:a014 Giga-Byte Technology Co., Ltd USB Audio ``` `alsactl monitor` (unplugging and plugging mic) ``` node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE ``` `amixer -c events` (unplugging and plugging mic) ``` event value: numid=9,iface=CARD,name='Mic - Input Jack',index=1 event value: numid=9,iface=CARD,name='Mic - Input Jack',index=1 ``` `amixer -c 2 contents` ``` numid=8,iface=CARD,name='Clock Source 3 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=12,iface=CARD,name='Clock Source 4 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=15,iface=CARD,name='Clock Source 5 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=19,iface=CARD,name='Clock Source 6 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=20,iface=CARD,name='Headphone - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=5,iface=CARD,name='Mic - Input Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=9,iface=CARD,name='Mic - Input Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on numid=16,iface=CARD,name='Speaker - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=13,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=17,iface=MIXER,name='PCM Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=off numid=14,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0 : values=62,62 | dBminmax-min=-65.25dB,max=0.00dB numid=18,iface=MIXER,name='PCM Playback Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0 : values=62,62 | dBminmax-min=-65.25dB,max=0.00dB numid=6,iface=MIXER,name='Mic Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=10,iface=MIXER,name='Mic Capture Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=7,iface=MIXER,name='Mic Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=39,39 | dBminmax-min=-17.25dB,max=12.00dB numid=11,iface=MIXER,name='Mic Capture Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=39,39 | dBminmax-min=-17.25dB,max=12.00dB numid=1,iface=PCM,name='Capture Channel Map' ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=3,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=2,iface=PCM,name='Capture Channel Map',device=1 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=4,iface=PCM,name='Playback Channel Map',device=1 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR ``` `arecord -l` ``` **** List of CAPTURE Hardware Devices **** card 2: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `alsaucm -c hw:2 dump text` ``` Verb.HiFi { Comment "HiFi 2.0 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 2 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,0" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 5+1" { Comment "HiFi 5.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 6 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,0" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 7+1" { Comment "HiFi 7.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 8 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,0" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/528 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No Micrphone detected on ALC4080 - Gigabyte B850I Aorus Pro (rev 1.0)
by GitHub issues - edited 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf issue #528 was edited from chanster: The Rear Microphone is not detected when I plug in my mic. I don't use the Front Audio connection as my case doesn't have I/O for it, I just used the 2 audio ports in the rear. When I try recording directly I can get the mic to work, but it does not show up in the sound settings. `arecord --duration=3 -D hw:CARD=2,DEV=1 -f dat -vvv a.wav` ``` Hardware PCM card 2 'USB Audio' device 1 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Max peak (12000 samples): 0x00000167 # 1% Max peak (12000 samples): 0x000000da # 0% Max peak (12000 samples): 0x000006b1 ## 5% Max peak (12000 samples): 0x00000547 # 4% Max peak (12000 samples): 0x00000729 ## 5% Max peak (12000 samples): 0x00000a77 ## 8% Max peak (12000 samples): 0x000006d2 ## 5% Max peak (12000 samples): 0x000005ec # 4% Max peak (12000 samples): 0x00000627 # 4% Max peak (12000 samples): 0x00000d32 ### 10% Max peak (12000 samples): 0x00000e9f ### 11% Max peak (12000 samples): 0x00003813 ######### 43% Max peak (12000 samples): 0x00006278 ################ 76% Max peak (12000 samples): 0x00007fff #################### 99% Max peak (12000 samples): 0x00006694 ################# 80% Max peak (12000 samples): 0x000068ea ################# 81% Max peak (12000 samples): 0x00004ba0 ############ 59% Max peak (12000 samples): 0x0000207c ###### 25% Max peak (12000 samples): 0x000017e9 #### 18% Max peak (12000 samples): 0x00003938 ######### 44% Max peak (12000 samples): 0x00001f11 ##### 24% Max peak (12000 samples): 0x00000ee5 ### 11% Max peak (12000 samples): 0x00003944 ######### 44% Max peak (12000 samples): 0x00003942 ######### 44% ``` Motherboard: Gigabyte B850I Aorus Pro (rev 1.0) Audio Device: ALC4080 Distro: Fedora 41 Window Manager: Gnome-Shell Packages: ``` dnf list --installed | grep 'pipewire\|pulseaudio\|alsa' alsa-lib.i686 1.2.13-3.fc41 updates alsa-lib.x86_64 1.2.13-3.fc41 updates alsa-ucm.noarch 1.2.13-3.fc41 updates alsa-ucm-utils.x86_64 1.2.13-2.fc41 updates alsa-utils.x86_64 1.2.13-2.fc41 updates pipewire.i686 1.2.7-7.fc41 updates pipewire.x86_64 1.2.7-7.fc41 updates pipewire-alsa.i686 1.2.7-7.fc41 updates pipewire-alsa.x86_64 1.2.7-7.fc41 updates pipewire-gstreamer.x86_64 1.2.7-7.fc41 updates pipewire-libs.i686 1.2.7-7.fc41 updates pipewire-libs.x86_64 1.2.7-7.fc41 updates pipewire-pulseaudio.x86_64 1.2.7-7.fc41 updates pipewire-utils.x86_64 1.2.7-7.fc41 updates pulseaudio-libs.i686 17.0-2.fc41 fedora pulseaudio-libs.x86_64 17.0-2.fc41 fedora pulseaudio-libs-glib2.x86_64 17.0-2.fc41 fedora wine-alsa.i686 10.1-1.fc41 updates wine-alsa.x86_64 10.1-1.fc41 updates wine-pulseaudio.i686 10.1-1.fc41 updates wine-pulseaudio.x86_64 10.1-1.fc41 updates ``` `lsusb | grep -i audio` ``` Bus 001 Device 002: ID 0414:a014 Giga-Byte Technology Co., Ltd USB Audio ``` `alsactl monitor` (unplugging and plugging mic) ``` node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE ``` `amixer -c events` (unplugging and plugging mic) ``` event value: numid=9,iface=CARD,name='Mic - Input Jack',index=1 event value: numid=9,iface=CARD,name='Mic - Input Jack',index=1 ``` `amixer -c 2 contents` ``` numid=8,iface=CARD,name='Clock Source 3 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=12,iface=CARD,name='Clock Source 4 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=15,iface=CARD,name='Clock Source 5 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=19,iface=CARD,name='Clock Source 6 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=20,iface=CARD,name='Headphone - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=5,iface=CARD,name='Mic - Input Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=9,iface=CARD,name='Mic - Input Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on numid=16,iface=CARD,name='Speaker - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=13,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=17,iface=MIXER,name='PCM Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=off numid=14,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0 : values=62,62 | dBminmax-min=-65.25dB,max=0.00dB numid=18,iface=MIXER,name='PCM Playback Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0 : values=62,62 | dBminmax-min=-65.25dB,max=0.00dB numid=6,iface=MIXER,name='Mic Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=10,iface=MIXER,name='Mic Capture Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=7,iface=MIXER,name='Mic Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=39,39 | dBminmax-min=-17.25dB,max=12.00dB numid=11,iface=MIXER,name='Mic Capture Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=39,39 | dBminmax-min=-17.25dB,max=12.00dB numid=1,iface=PCM,name='Capture Channel Map' ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=3,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=2,iface=PCM,name='Capture Channel Map',device=1 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=4,iface=PCM,name='Playback Channel Map',device=1 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR ``` `arecord -l` ``` **** List of CAPTURE Hardware Devices **** card 2: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `alsaucm -c hw:2 dump text` ``` Verb.HiFi { Comment "HiFi 2.0 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 2 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 5+1" { Comment "HiFi 5.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 6 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 7+1" { Comment "HiFi 7.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 8 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/528 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No Micrphone detected on ALC4080 - Gigabyte B850I Aorus Pro (rev 1.0)
by GitHub issues - edited 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf issue #528 was edited from chanster: The Rear Microphone is not detected when I plug in my mic. I don't use the Front Audio connection as my case doesn't have I/O for it, I just used the 2 audio ports in the rear. When I try recording directly I can get the mic to work, but it does not show up in the sound settings. `arecord --duration=3 -D hw:CARD=2,DEV=1 -f dat -vvv a.wav` ``` Hardware PCM card 2 'USB Audio' device 1 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Max peak (12000 samples): 0x00000167 # 1% Max peak (12000 samples): 0x000000da # 0% Max peak (12000 samples): 0x000006b1 ## 5% Max peak (12000 samples): 0x00000547 # 4% Max peak (12000 samples): 0x00000729 ## 5% Max peak (12000 samples): 0x00000a77 ## 8% Max peak (12000 samples): 0x000006d2 ## 5% Max peak (12000 samples): 0x000005ec # 4% Max peak (12000 samples): 0x00000627 # 4% Max peak (12000 samples): 0x00000d32 ### 10% Max peak (12000 samples): 0x00000e9f ### 11% Max peak (12000 samples): 0x00003813 ######### 43% Max peak (12000 samples): 0x00006278 ################ 76% Max peak (12000 samples): 0x00007fff #################### 99% Max peak (12000 samples): 0x00006694 ################# 80% Max peak (12000 samples): 0x000068ea ################# 81% Max peak (12000 samples): 0x00004ba0 ############ 59% Max peak (12000 samples): 0x0000207c ###### 25% Max peak (12000 samples): 0x000017e9 #### 18% Max peak (12000 samples): 0x00003938 ######### 44% Max peak (12000 samples): 0x00001f11 ##### 24% Max peak (12000 samples): 0x00000ee5 ### 11% Max peak (12000 samples): 0x00003944 ######### 44% Max peak (12000 samples): 0x00003942 ######### 44% ``` Motherboard: Gigabyte B850I Aorus Pro (rev 1.0) Audio Device: ALC4080 Distro: Fedora 41 Window Manager: Gnome-Shell Packages: ``` dnf list --installed | grep 'pipewire\|pulseaudio\|alsa' alsa-lib.i686 1.2.13-3.fc41 updates alsa-lib.x86_64 1.2.13-3.fc41 updates alsa-ucm.noarch 1.2.13-3.fc41 updates alsa-ucm-utils.x86_64 1.2.13-2.fc41 updates alsa-utils.x86_64 1.2.13-2.fc41 updates pipewire.i686 1.2.7-7.fc41 updates pipewire.x86_64 1.2.7-7.fc41 updates pipewire-alsa.i686 1.2.7-7.fc41 updates pipewire-alsa.x86_64 1.2.7-7.fc41 updates pipewire-gstreamer.x86_64 1.2.7-7.fc41 updates pipewire-libs.i686 1.2.7-7.fc41 updates pipewire-libs.x86_64 1.2.7-7.fc41 updates pipewire-pulseaudio.x86_64 1.2.7-7.fc41 updates pipewire-utils.x86_64 1.2.7-7.fc41 updates pulseaudio-libs.i686 17.0-2.fc41 fedora pulseaudio-libs.x86_64 17.0-2.fc41 fedora pulseaudio-libs-glib2.x86_64 17.0-2.fc41 fedora wine-alsa.i686 10.1-1.fc41 updates wine-alsa.x86_64 10.1-1.fc41 updates wine-pulseaudio.i686 10.1-1.fc41 updates wine-pulseaudio.x86_64 10.1-1.fc41 updates ``` `lsusb | grep -i audio` ``` Bus 001 Device 002: ID 0414:a014 Giga-Byte Technology Co., Ltd USB Audio ``` `alsactl monitor` (unplugging and plugging mic) ``` node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE ``` `arecord -l` ``` **** List of CAPTURE Hardware Devices **** card 2: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `alsaucm -c hw:2 dump text` ``` Verb.HiFi { Comment "HiFi 2.0 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 2 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 5+1" { Comment "HiFi 5.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 6 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 7+1" { Comment "HiFi 7.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 8 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/528 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No Micrphone detected on ALC4080
by GitHub issues - opened 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf issue #528 was opened from chanster: The Rear Microphone is not detected when I plug in my mic. I don't use the Front Audio connection as my case doesn't have I/O for it, I just used the 2 audio ports in the rear. When I try recording directly I can get the mic to work, but it does not show up in the sound settings. `arecord --duration=3 -D hw:CARD=2,DEV=1 -f dat -vvv a.wav` ``` Hardware PCM card 2 'USB Audio' device 1 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Max peak (12000 samples): 0x00000167 # 1% Max peak (12000 samples): 0x000000da # 0% Max peak (12000 samples): 0x000006b1 ## 5% Max peak (12000 samples): 0x00000547 # 4% Max peak (12000 samples): 0x00000729 ## 5% Max peak (12000 samples): 0x00000a77 ## 8% Max peak (12000 samples): 0x000006d2 ## 5% Max peak (12000 samples): 0x000005ec # 4% Max peak (12000 samples): 0x00000627 # 4% Max peak (12000 samples): 0x00000d32 ### 10% Max peak (12000 samples): 0x00000e9f ### 11% Max peak (12000 samples): 0x00003813 ######### 43% Max peak (12000 samples): 0x00006278 ################ 76% Max peak (12000 samples): 0x00007fff #################### 99% Max peak (12000 samples): 0x00006694 ################# 80% Max peak (12000 samples): 0x000068ea ################# 81% Max peak (12000 samples): 0x00004ba0 ############ 59% Max peak (12000 samples): 0x0000207c ###### 25% Max peak (12000 samples): 0x000017e9 #### 18% Max peak (12000 samples): 0x00003938 ######### 44% Max peak (12000 samples): 0x00001f11 ##### 24% Max peak (12000 samples): 0x00000ee5 ### 11% Max peak (12000 samples): 0x00003944 ######### 44% Max peak (12000 samples): 0x00003942 ######### 44% ``` Motherboard: Gigabyte B850I Aorus Pro (rev 1.0) Audio Device: ALC4080 Distro: Fedora 41 Window Manager: Gnome-Shell Packages: ``` dnf list --installed | grep 'pipewire\|pulseaudio\|alsa' alsa-lib.i686 1.2.13-3.fc41 updates alsa-lib.x86_64 1.2.13-3.fc41 updates alsa-ucm.noarch 1.2.13-3.fc41 updates alsa-ucm-utils.x86_64 1.2.13-2.fc41 updates alsa-utils.x86_64 1.2.13-2.fc41 updates pipewire.i686 1.2.7-7.fc41 updates pipewire.x86_64 1.2.7-7.fc41 updates pipewire-alsa.i686 1.2.7-7.fc41 updates pipewire-alsa.x86_64 1.2.7-7.fc41 updates pipewire-gstreamer.x86_64 1.2.7-7.fc41 updates pipewire-libs.i686 1.2.7-7.fc41 updates pipewire-libs.x86_64 1.2.7-7.fc41 updates pipewire-pulseaudio.x86_64 1.2.7-7.fc41 updates pipewire-utils.x86_64 1.2.7-7.fc41 updates pulseaudio-libs.i686 17.0-2.fc41 fedora pulseaudio-libs.x86_64 17.0-2.fc41 fedora pulseaudio-libs-glib2.x86_64 17.0-2.fc41 fedora wine-alsa.i686 10.1-1.fc41 updates wine-alsa.x86_64 10.1-1.fc41 updates wine-pulseaudio.i686 10.1-1.fc41 updates wine-pulseaudio.x86_64 10.1-1.fc41 updates ``` `lsusb | grep -i audio` ``` Bus 001 Device 002: ID 0414:a014 Giga-Byte Technology Co., Ltd USB Audio ``` `alsactl monitor` (unplugging and plugging mic) ``` node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE ``` `arecord -l` ``` **** List of CAPTURE Hardware Devices **** card 2: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `alsaucm -c hw:2 dump text` ``` Verb.HiFi { Comment "HiFi 2.0 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 2 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 5+1" { Comment "HiFi 5.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 6 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 7+1" { Comment "HiFi 7.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 8 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/528 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Unable to attach an LED to headphones without HW mute ALSA Control
by GitHub issues - opened 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf issue #527 was opened from stefdb: I have a laptop with a CS42L43 Codec which supplies the Headphone Jack. I'm trying to attach the mute state of the Headphones to the Speaker LED. However, the issue is that the CS42L43 does not possess a Hardware Mute ALSA Control which could be attached to the Speaker LED. I first tried to use the Volume Control to attach to the LED, however, that was a non-starter because the Mute is software controlled, and thus muting headphones did not touch the Volume Control. I then tried to add a new control using cset-new, and link it to the Headphones device ([0001-sof-soundwire-Attach-cs42l43-Headphone-Switch-to-spe.patch](https://gi…), and whilst that worked with the LED, the mute no longer actually muted the Headphone, since there was now a "real" control to set the mute which did not actually mute the Device. Other things I have tried include mapping the Volume control as a Switch, but that does not work, I assume, because the switch control needs to be Boolean. Is there a way to create a new control to track the LED status, whilst also ensuring that the Headphones still mute? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/527 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

UCM2 Conditional cset in BootSequence
by Giedrius Trainavičius 18 Mar '25

18 Mar '25

Hello, I'm writing a UCM2 .conf file, depending on Device Tree options, some kcontrols may or may not exist - how do I handle this in the BootSequence, so alsactl does not error out if some controls don't exist? I know there's a possible If condition to check if a control exists, but how do I use that to conditionally append to the BootSequence? Thank you! Giedrius.

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[PATCH] ALSA: hda/realtek: Support mute LED on HP Laptop 15s-du3xxx The mute LED on this HP laptop uses ALC236 and requires a quirk to function. This patch enables the existing quirk for the device.
by Dhruv Deshpande 18 Mar '25

18 Mar '25

Signed-off-by: Dhruv Deshpande <dhrv.d(a)proton.me> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a84857a3c2bf..c0b97c29bf89 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10519,6 +10519,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x881d, "HP 250 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), + SND_PCI_QUIRK(0x103c, 0x881e, "HP Laptop 15s-du3xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), -- 2.48.1

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[PATCH] ALSA: usb-audio: separate DJM-A9 cap lvl options
by Olivia Mackintosh 18 Mar '25

18 Mar '25

Mixer quicks for the Pioneer DJM-A9 mixer was added in 5289d00 with additional capture level values added to the common DJM array of values. This breaks the existing DJM mixers however as alsa-utils relies on enumeration of the actual mixer options based on the value array which results in error when storing state. This commit just separates the A9 values into a separate array and references them in the corresponding mixer control. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> --- sound/usb/mixer_quirks.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 79c8c1540ee2..49fb849f7d40 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3688,8 +3688,7 @@ static const char *snd_djm_get_label(u8 device_idx, u16 wvalue, u16 windex) // common DJM capture level option values static const u16 snd_djm_opts_cap_level[] = { - 0x0000, 0x0100, 0x0200, 0x0300, 0x400, 0x500 }; - + 0x0000, 0x0100, 0x0200, 0x0300 }; // DJM-250MK2 static const u16 snd_djm_opts_250mk2_cap1[] = { @@ -3831,6 +3830,8 @@ static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = { // DJM-A9 +static const u16 snd_djm_opts_a9_cap_level[] = { + 0x0000, 0x0100, 0x0200, 0x0300, 0x0400, 0x0500 }; static const u16 snd_djm_opts_a9_cap1[] = { 0x0107, 0x0108, 0x0109, 0x010a, 0x010e, 0x111, 0x112, 0x113, 0x114, 0x0131, 0x132, 0x133, 0x134 }; @@ -3844,7 +3845,7 @@ static const u16 snd_djm_opts_a9_cap5[] = { 0x0501, 0x0502, 0x0503, 0x0505, 0x0506, 0x0507, 0x0508, 0x0509, 0x050a, 0x050e }; static const struct snd_djm_ctl snd_djm_ctls_a9[] = { - SND_DJM_CTL("Capture Level", cap_level, 0, SND_DJM_WINDEX_CAPLVL), + SND_DJM_CTL("Capture Level", a9_cap_level, 0, SND_DJM_WINDEX_CAPLVL), SND_DJM_CTL("Master Input", a9_cap1, 3, SND_DJM_WINDEX_CAP), SND_DJM_CTL("Ch1 Input", a9_cap2, 2, SND_DJM_WINDEX_CAP), SND_DJM_CTL("Ch2 Input", a9_cap3, 2, SND_DJM_WINDEX_CAP), -- 2.48.1

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ucm2: HDA: HiFi-analog/mic: Refactor the analog mic discovery and pre…
by GitHub pull_request - opened 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf pull request #526 was opened from ujfalusi: …sentation The current mic device creation works on certain machines and fails on others. There are several places of conflicts and setups which can only just fail, but this is mostly not an issue if the user never uses the mic, only the speaker/headset - which, to be honest is what most of us do ;) As an example: The mic selection in most codecs are via enum and it is assumed to be named 'Input Source', which is not always the case as some device uses 'Capture Source' for the control's name. There is also different sets of mics that one can select from: Exhibit A numid=6,iface=MIXER,name='Input Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Headphone Mic' : values=1 Exhibit B numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Internal Mic' ; Item #1 'Headset Mic' : values=0 Exhibit C numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Internal Mic' ; Item #1 'Headset Mic' ; Item #2 'Headphone Mic' : values=0 Exhibit D (this pushes the limits... The patch will ignore item 1) numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Internal Mic' ; Item #1 'Internal Mic 1' ; Item #2 'Mic' : values=2 Other issue is that we have this 'Headphone Mic', which turned out to be a 'Stereo Microphone in Headphone Jack', so if it is selected then the Headphone cannot work, they conflict, they use the same rings for different direction and purpose. This patch aims to make the mic discovery a bit more deterministic and pragmatic. But even if the UCM creates the use case profiles correctly, it is still up to UIs (KDE/GNOME/etc) to misunderstand how UCM presents the profiles, what they mean and most of all what 'Mic1', `Mic2', etc is. KDE presents the profiles as they are and user can selct between them to pick the right combination of output and input. GNOME goes further with simplification (and fails with it) and presents 'random' Configuration profiles for Output and Input, plus a device selection and they do work in an interesting way. GNOME also have popup for specifying the type of the plugged accessory, which does not worl at all with UCM profiles. But, this patch is meant for a small step to have clear rules based mic presentation for HDA. The expectation is that what have worked will work as it used to and what did not worked should be detected and presented correctly. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/526 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/526.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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