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[RESEND PATCH v5] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 13 Mar '25

13 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

3 2

Split the GoXLR configuration into Full Device and Mini
by GitHub pull_request - opened 13 Mar '25

13 Mar '25

alsa-project/alsa-ucm-conf pull request #520 was opened from FrostyCoolSlug: This commit splits the GoXLR and GoXLR Mini into separate configurations. The GoXLR Mini will always (at this point) have 21 channels defined, where as the full sized GoXLR will have either 23 or 25 depending on Firmware (23 is FAR more common at this point, as the firmware with 25 channels is still in beta). Separating out the two devices should at least partially solve the recent issue with Pipewire. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/520 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/520.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Need to edit Steinberg configuration files
by GitHub issues - opened 13 Mar '25

13 Mar '25

alsa-project/alsa-ucm-conf issue #519 was opened from AdrianusWest: I faced a problem with disappearance of HiFi profile in Steinberg UR22C audio card settings. Attempting to fix it myself did not lead to anything. I addressed the issue to pipewire developers, who pointed out that the existing configuration file needs to be corrected: "The problem here is that previously, the channel splitting was done with ALSA dshare plugin which as it turns out does not require the exact channel numbers to match. As this introduces additional latency, in 1.4.0 the splitting is now done on Pipewire side. But of course there was a nasty surprise (and affects multiple devices), so probably there will be 1.4.1 with a workaround for wrong channel numbers in profiles." The link to the thread is attached: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4609 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/519 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 00/14] Implement acp_common_hw_ops support for all acp platforms
by Venkata Prasad Potturu 13 Mar '25

13 Mar '25

This patch series is to implement acp_common_hw_ops support and refactor interrupt handler registration and interrupt handling. Venkata Prasad Potturu (14): ASoC: amd: acp: Remove redundant acp70 chip->name ASoC: amd: acp: Implement acp_common_hw_ops support for acp platforms ASoC: amd: acp: Refactor dmic-codec platform device creation ASoC: amd: acp: Refactor acp platform device creation ASoC: amd: acp: Refactor acp machine select ASoC: amd: acp: Add new interrupt handle callbacks in acp_common_hw_ops ASoC: amd: acp: Remove redundant acp_dev_data structure ASoC: amd: acp: Move spin_lock and list initialization to acp-pci driver ASoC: amd: acp: Remove white line ASoC: amd: acp: Refactor acp70 platform resource structure ASoC: amd: acp: Refactor acp63 platform resource structure ASoC: amd: acp: Refactor rembrant platform resource structure ASoC: amd: acp: Refactor renoir platform resource structure ASoC: amd: acp: Fix for enabling DMIC on acp platforms via _DSD entry sound/soc/amd/acp/acp-i2s.c | 185 +++++++++--------- sound/soc/amd/acp/acp-legacy-common.c | 254 ++++++++++++++++++------ sound/soc/amd/acp/acp-pci.c | 210 ++++++++++++-------- sound/soc/amd/acp/acp-pdm.c | 53 +++-- sound/soc/amd/acp/acp-platform.c | 145 ++++---------- sound/soc/amd/acp/acp-rembrandt.c | 111 +++-------- sound/soc/amd/acp/acp-renoir.c | 110 ++--------- sound/soc/amd/acp/acp63.c | 94 +++------ sound/soc/amd/acp/acp70.c | 84 +++----- sound/soc/amd/acp/amd.h | 270 +++++++++++++++++++++----- sound/soc/amd/acp/chip_offset_byte.h | 12 +- 11 files changed, 792 insertions(+), 736 deletions(-) -- 2.39.2

3 16

Dynamic PCM switching in ALSA with multi room audio sync support
by GitHub issues - opened 13 Mar '25

13 Mar '25

alsa-project/alsa-lib issue #443 was opened from Asikbct: **Challenges:** The PCM device is already opened by applications, making it difficult to switch to a new output seamlessly. Since most Music Service SDKs initialize the PCM device at startup, re-initializing it would require users to manually reopen their streaming app and re-select the device, negatively impacting the user experience. **Requirement:** User can switch between different audio output options (Line Out, Optical, USB Out, Bluetooth, etc.) at any time. The transition to a new output should be seamless, whether the user is currently playing a song from any music sources. Playback should resume on the newly selected output without requiring the user to manually reopen the content provider app and restart playback from the beginning. **Problems:** 1. **Reopening the PCM** Most music services open and configure the PCM device during initialization. Changing the PCM device (e.g., switching from Line Out to Optical Out) requires reopening it, which forces music services to reinitialize. This may require the end user to relaunch the respective app, reselect the device, and restart playback, impacting the user experience. 2. **Multiroom Sync** Many music services that support multiroom synchronization rely on Linux ALSA APIs like snd_pcm_delay() to estimate the kernel buffer fill level and synchronize audio playback across devices. Any new approach to meet the requirements, should not disrupt this synchronization. I've tried virtual loopback, PulseAudio, and a customized dswitch plugin. While all these methods enable dynamic PCM switching, I couldn't achieve stable long-term latency on the same PCM. Any guidance or suggestions on achieving stable long-term latency along with dynamic PCM switching would be greatly appreciated. Issue URL : https://github.com/alsa-project/alsa-lib/issues/443 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v6.14-rc6
by Mark Brown 13 Mar '25

13 Mar '25

The following changes since commit 7eb172143d5508b4da468ed59ee857c6e5e01da6: Linux 6.14-rc5 (2025-03-02 11:48:20 -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-rc6 for you to fetch changes up to 658fb7fe8e7f4014ea17a4da0e0c1d9bc319fa35: ASoC: cs42l43: convert to SYSTEM_SLEEP_PM_OPS (2025-03-12 16:12:24 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 The bulk of this is driver specific fixes, mostly unremarkable. There's also one core fix from Charles, fixing up confusion around the limiting of maximum control values. ---------------------------------------------------------------- Alexey Klimov (1): ASoC: codecs: wsa884x: report temps to hwmon in millidegree of Celsius Arnd Bergmann (1): ASoC: cs42l43: convert to SYSTEM_SLEEP_PM_OPS Bard Liao (2): ASoC: rt1320: set wake_capable = 0 explicitly ASoC: rt722-sdca: add missing readable registers Charles Keepax (2): ASoC: ops: Consistently treat platform_max as control value ASoC: cs42l43: Fix maximum ADC Volume Christophe JAILLET (1): ASoC: codecs: wm0010: Fix error handling path in wm0010_spi_probe() Maciej Strozek (1): ASoC: cs42l43: Add jack delay debounce after suspend Peter Ujfalusi (1): ASoC: Intel: sof_sdw: Fix unlikely uninitialized variable use in create_sdw_dailinks() Thomas Mizrahi (1): ASoC: amd: yc: Support mic on another Lenovo ThinkPad E16 Gen 2 model Thorsten Blum (1): ASoC: tegra: Fix ADX S24_LE audio format include/sound/soc.h | 5 ++++- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/codecs/cs42l43-jack.c | 13 ++++++++++--- sound/soc/codecs/cs42l43.c | 17 +++++++++++++++-- sound/soc/codecs/cs42l43.h | 3 +++ sound/soc/codecs/rt1320-sdw.c | 3 +++ sound/soc/codecs/rt722-sdca-sdw.c | 4 ++++ sound/soc/codecs/wm0010.c | 13 +++++++++++-- sound/soc/codecs/wsa884x.c | 4 ++-- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/soc-ops.c | 15 +++++++-------- sound/soc/tegra/tegra210_adx.c | 4 ++-- 12 files changed, 69 insertions(+), 21 deletions(-)

2 1

Channel count in GoXLR SplitPCM configuration
by GitHub issues - edited 12 Mar '25

12 Mar '25

alsa-project/alsa-ucm-conf issue #518 was edited from pv: Similar issue as #515, and related to #444: We have a report https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4602 that GoXLR UCM profile with SplitPCM=1 specifies always 23 channels, but the device can't be opened with exactly 23 channels. According to https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4602#note_2819696 the device may have 21, 23, or 25 depending on firmware/etc details. In Pipewire, I think we add a workaround that just uses the available channel count even if it is different from what UCM profile specifies, but logs warnings if the counts are different so these issues in the profiles can be caught. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/518 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Channel count in GoXLR SplitPCM configuration
by GitHub issues - opened 12 Mar '25

12 Mar '25

alsa-project/alsa-ucm-conf issue #518 was opened from pv: Similar issue as #515, and related to #444: We have a report https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4602#note_2819696 that GoXLR UCM profile with SplitPCM=1 specifies always 23 channels, but the device can't be opened with exactly 23 channels. According to https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4602#note_2819696 the device may have 21, 23, or 25 depending on firmware/etc details. In Pipewire, I think we add a workaround that just uses the available channel count even if it is different from what UCM profile specifies, but logs warnings if the counts are different so these issues in the profiles can be caught. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/518 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC fixes for v6.14-rc6
by Mark Brown 12 Mar '25

12 Mar '25

The following changes since commit 7eb172143d5508b4da468ed59ee857c6e5e01da6: Linux 6.14-rc5 (2025-03-02 11:48:20 -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-rc6 for you to fetch changes up to ed92bc5264c4357d4fca292c769ea9967cd3d3b6: ASoC: codecs: wm0010: Fix error handling path in wm0010_spi_probe() (2025-03-11 13:51:52 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 The bulk of this is driver specific fixes, mostly unremarkable. There's also one core fix from Charles, fixing up confusion around the limiting of maximum control values. ---------------------------------------------------------------- Alexey Klimov (1): ASoC: codecs: wsa884x: report temps to hwmon in millidegree of Celsius Bard Liao (2): ASoC: rt1320: set wake_capable = 0 explicitly ASoC: rt722-sdca: add missing readable registers Charles Keepax (2): ASoC: ops: Consistently treat platform_max as control value ASoC: cs42l43: Fix maximum ADC Volume Christophe JAILLET (1): ASoC: codecs: wm0010: Fix error handling path in wm0010_spi_probe() Maciej Strozek (1): ASoC: cs42l43: Add jack delay debounce after suspend Peter Ujfalusi (1): ASoC: Intel: sof_sdw: Fix unlikely uninitialized variable use in create_sdw_dailinks() Thomas Mizrahi (1): ASoC: amd: yc: Support mic on another Lenovo ThinkPad E16 Gen 2 model Thorsten Blum (1): ASoC: tegra: Fix ADX S24_LE audio format include/sound/soc.h | 5 ++++- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/codecs/cs42l43-jack.c | 13 ++++++++++--- sound/soc/codecs/cs42l43.c | 17 +++++++++++++++-- sound/soc/codecs/cs42l43.h | 3 +++ sound/soc/codecs/rt1320-sdw.c | 3 +++ sound/soc/codecs/rt722-sdca-sdw.c | 4 ++++ sound/soc/codecs/wm0010.c | 13 +++++++++++-- sound/soc/codecs/wsa884x.c | 4 ++-- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/soc-ops.c | 15 +++++++-------- sound/soc/tegra/tegra210_adx.c | 4 ++-- 12 files changed, 69 insertions(+), 21 deletions(-)

2 1

Add SetLED macro to hide implementation details for LED handling
by GitHub pull_request - opened 11 Mar '25

11 Mar '25

alsa-project/alsa-ucm-conf pull request #517 was opened from perexg: This change also reduces modprobe calls when settings is split to multiple configuration files. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/517 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/517.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0