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[PATCH v3] ASoC: tas2781: Add Calibration Kcontrols for Chromebook
by Shenghao Ding 13 Sep '24

13 Sep '24

Add calibration related kcontrol for speaker impedance calibration and speaker leakage check for Chromebook. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - Add missing grammatical period - Refer to the functions as func() in comments. - Create a macro for calibration register setting to reduce the line - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202409030627.SFdj22Xs-lkp@intel.com/ v2: - Add channel no into the log for error debug - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/202407131… - renamae tasdevice_chn_switch to put it in a single line - With inverted check this entire function becomes neater in tasdev_chn_switch. - Add registers for TAS2563 and TAS2781 calibration. - Add cali_data_backup for regsiter restore after calibration. - Add is_user_space_calidata to store the flag where the calibrated from, user space or bin file from driver parsing. - tasdevice_chn_switch for chip switch among multiple chips(tas2563 or tas2781) - Add loading the calibrated values from user space in tasdev_load_calibrated_data - Correct no dsp no work, it can still work in bypass mode. - Add calibrated register setting for tas2563&tas2781. - Define tas2781_cali_control and tas2563_cali_control for calibration --- include/sound/tas2781.h | 68 +++ sound/soc/codecs/tas2781-comlib.c | 26 + sound/soc/codecs/tas2781-fmwlib.c | 60 +- sound/soc/codecs/tas2781-i2c.c | 887 +++++++++++++++++++++++++++++- 4 files changed, 1030 insertions(+), 11 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index dbda552398b5..8cd6da0480b7 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -49,12 +49,59 @@ /*I2C Checksum */ #define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) +/* XM_340 */ +#define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x63, 0x3c) +/* XM_341 */ +#define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x63, 0x38) + /* Volume control */ #define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) #define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) #define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) +#define TAS2563_IDLE TASDEVICE_REG(0x00, 0x00, 0x3e) +#define TAS2563_PRM_R0_REG TASDEVICE_REG(0x00, 0x0f, 0x34) + +#define TAS2563_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x02, 0x70) +#define TAS2563_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x02, 0x48) + +#define TAS2563_PRM_ENFF_REG TASDEVICE_REG(0x00, 0x0d, 0x54) +#define TAS2563_PRM_DISTCK_REG TASDEVICE_REG(0x00, 0x0d, 0x58) +#define TAS2563_PRM_TE_SCTHR_REG TASDEVICE_REG(0x00, 0x0f, 0x60) +#define TAS2563_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x0d, 0x74) +#define TAS2563_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x0d, 0x7c) +/* prm_Int_B0 */ +#define TAS2563_TE_TA1_REG TASDEVICE_REG(0x00, 0x10, 0x0c) +/* prm_Int_A1 */ +#define TAS2563_TE_TA1_AT_REG TASDEVICE_REG(0x00, 0x10, 0x10) +/* prm_TE_Beta */ +#define TAS2563_TE_TA2_REG TASDEVICE_REG(0x00, 0x0f, 0x64) +/* prm_TE_Beta1 */ +#define TAS2563_TE_AT_REG TASDEVICE_REG(0x00, 0x0f, 0x68) +/* prm_TE_1_Beta1 */ +#define TAS2563_TE_DT_REG TASDEVICE_REG(0x00, 0x0f, 0x70) + +#define TAS2781_PRM_INT_MASK_REG TASDEVICE_REG(0x00, 0x00, 0x3b) +#define TAS2781_PRM_CLK_CFG_REG TASDEVICE_REG(0x00, 0x00, 0x5c) +#define TAS2781_PRM_RSVD_REG TASDEVICE_REG(0x00, 0x01, 0x19) +#define TAS2781_PRM_TEST_57_REG TASDEVICE_REG(0x00, 0xfd, 0x39) +#define TAS2781_PRM_TEST_62_REG TASDEVICE_REG(0x00, 0xfd, 0x3e) +#define TAS2781_PRM_PVDD_UVLO_REG TASDEVICE_REG(0x00, 0x00, 0x71) +#define TAS2781_PRM_CHNL_0_REG TASDEVICE_REG(0x00, 0x00, 0x03) +#define TAS2781_PRM_NG_CFG0_REG TASDEVICE_REG(0x00, 0x00, 0x35) +#define TAS2781_PRM_IDLE_CH_DET_REG TASDEVICE_REG(0x00, 0x00, 0x66) +#define TAS2781_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x14, 0x38) +#define TAS2781_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x14, 0x40) +#define TAS2781_PRM_SINEGAIN2_REG TASDEVICE_REG(0x00, 0x14, 0x44) + +#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xFD, 0x0D) +#define TAS2781_TEST_PAGE_UNLOCK 0x0D + +#define TAS2781_RUNTIME_LATCH_RE_REG TASDEVICE_REG(0x00, 0x00, 0x49) +#define TAS2781_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x62, 0x48) +#define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) + #define TASDEVICE_CMD_SING_W 0x1 #define TASDEVICE_CMD_BURST 0x2 #define TASDEVICE_CMD_DELAY 0x3 @@ -70,7 +117,15 @@ enum device_catlog_id { OTHERS }; +struct bulk_reg_val { + int reg; + unsigned char val[4]; + unsigned char val_len; + bool is_locked; +}; + struct tasdevice { + struct bulk_reg_val *cali_data_backup; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -81,9 +136,19 @@ struct tasdevice { bool is_loaderr; }; +struct cali_reg { + unsigned int r0_reg; + unsigned int r0_low_reg; + unsigned int invr0_reg; + unsigned int pow_reg; + unsigned int tlimit_reg; +}; + struct calidata { unsigned char *data; unsigned long total_sz; + struct cali_reg cali_reg_array; + unsigned int cali_dat_sz_per_dev; }; struct tasdevice_priv { @@ -119,6 +184,7 @@ struct tasdevice_priv { bool force_fwload_status; bool playback_started; bool isacpi; + bool is_user_space_calidata; unsigned int global_addr; int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, @@ -145,6 +211,8 @@ int tasdevice_init(struct tasdevice_priv *tas_priv); void tasdevice_remove(struct tasdevice_priv *tas_priv); int tasdevice_save_calibration(struct tasdevice_priv *tas_priv); void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv); +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn); int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *value); int tasdevice_dev_write(struct tasdevice_priv *tas_priv, diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 664c371796d6..1e0b3aa95749 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -88,6 +88,32 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, return ret; } +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + struct regmap *map = tas_priv->regmap; + int ret; + + if (client->addr != tasdev->dev_addr) { + client->addr = tasdev->dev_addr; + /* All devices share the same regmap, clear the page + * inside regmap once switching to another device. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + return ret; + } + return 1; + } + return 0; +} +EXPORT_SYMBOL_GPL(tasdev_chn_switch); + int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *val) { diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index f3a7605f0710..3de0132c345d 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2151,20 +2151,61 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) { + struct tasdevice_fw *cal_fmw = priv->tasdevice[i].cali_data_fmw; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *data = cali_data->data; struct tasdevice_calibration *cal; - struct tasdevice_fw *cal_fmw; + int k = i * (cali_data->cali_dat_sz_per_dev + 1); + int rc; - cal_fmw = priv->tasdevice[i].cali_data_fmw; + /* Load the calibrated data from cal bin file */ + if (!priv->is_user_space_calidata && cal_fmw) { + cal = cal_fmw->calibrations; - /* No calibrated data for current devices, playback will go ahead. */ - if (!cal_fmw) + if (cal) + load_calib_data(priv, &cal->dev_data); return; - - cal = cal_fmw->calibrations; - if (!cal) + } + if (!priv->is_user_space_calidata) + return; + /* load calibrated data from user space */ + if (data[k] != i) { + dev_err(priv->dev, "%s: no cal-data for dev %d from usr-spc\n", + __func__, i); return; + } + k++; - load_calib_data(priv, &cal->dev_data); + rc = tasdevice_dev_bulk_write(priv, i, p->r0_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_reg bulk_wr err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->r0_low_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_low_reg err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->invr0_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d invr0_reg err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->pow_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d pow_reg bulk_wr err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->tlimit_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d tlimit_reg err = %d\n", i, rc); + return; + } } int tasdevice_select_tuningprm_cfg(void *context, int prm_no, @@ -2259,9 +2300,10 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, tas_priv->tasdevice[i].cur_conf = cfg_no; } } - } else + } else { dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", __func__, cfg_no); + } status |= cfg_info[rca_conf_no]->active_dev; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 3bc10d615fd3..d0ba7cbe03a8 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -33,6 +33,69 @@ #include <sound/tas2781-tlv.h> #include <asm/unaligned.h> +#define X2563_CL_STT_VAL(xreg, xval) \ +{ .reg = xreg, \ + .val = { xval }, \ + .val_len = 1, } + +#define X2563_CL_STT_4BYTS(xreg, byte0, byte1, byte2, byte3) \ +{ .reg = xreg, \ + .val = { byte0, byte1, byte2, byte3 }, \ + .val_len = 4, } + +static const struct bulk_reg_val tas2563_cali_start_reg[] = { + X2563_CL_STT_VAL(TAS2563_IDLE, 0x00), + X2563_CL_STT_4BYTS(TAS2563_PRM_ENFF_REG, 0x40, 0x00, 0x00, 0x00), + X2563_CL_STT_4BYTS(TAS2563_PRM_DISTCK_REG, 0x40, 0x00, 0x00, 0x00), + X2563_CL_STT_4BYTS(TAS2563_PRM_TE_SCTHR_REG, 0x7f, 0xff, 0xff, 0xff), + X2563_CL_STT_4BYTS(TAS2563_PRM_PLT_FLAG_REG, 0x40, 0x00, 0x00, 0x00), + X2563_CL_STT_4BYTS(TAS2563_PRM_SINEGAIN_REG, 0x0a, 0x3d, 0x70, 0xa4), + X2563_CL_STT_4BYTS(TAS2563_TE_TA1_REG, 0x00, 0x36, 0x91, 0x5e), + X2563_CL_STT_4BYTS(TAS2563_TE_TA1_AT_REG, 0x00, 0x36, 0x91, 0x5e), + X2563_CL_STT_4BYTS(TAS2563_TE_TA2_REG, 0x00, 0x06, 0xd3, 0x72), + X2563_CL_STT_4BYTS(TAS2563_TE_AT_REG, 0x00, 0x36, 0x91, 0x5e), + X2563_CL_STT_4BYTS(TAS2563_TE_DT_REG, 0x00, 0x36, 0x91, 0x5e), +}; + +#define X2781_CL_STT_VAL(xreg, xval, xlocked) \ +{ .reg = xreg, \ + .val = { xval }, \ + .val_len = 1, \ + .is_locked = xlocked, } + +#define X2781_CL_STT_4BYTS_UNLOCKED(xreg, byte0, byte1, byte2, byte3) \ +{ .reg = xreg, \ + .val = { byte0, byte1, byte2, byte3 }, \ + .val_len = 4, \ + .is_locked = false, } + +#define X2781_CL_STT_LEN_UNLOCKED(xreg) \ +{ .reg = xreg, \ + .val_len = 4, \ + .is_locked = false, } + +static const struct bulk_reg_val tas2781_cali_start_reg[] = { + X2781_CL_STT_VAL(TAS2781_PRM_INT_MASK_REG, 0xfe, false), + X2781_CL_STT_VAL(TAS2781_PRM_CLK_CFG_REG, 0xdd, false), + X2781_CL_STT_VAL(TAS2781_PRM_RSVD_REG, 0x20, false), + X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, false), + X2781_CL_STT_VAL(TAS2781_PRM_TEST_62_REG, 0x45, true), + X2781_CL_STT_VAL(TAS2781_PRM_PVDD_UVLO_REG, 0x03, false), + X2781_CL_STT_VAL(TAS2781_PRM_CHNL_0_REG, 0xa8, false), + X2781_CL_STT_VAL(TAS2781_PRM_NG_CFG0_REG, 0xb9, false), + X2781_CL_STT_VAL(TAS2781_PRM_IDLE_CH_DET_REG, 0x92, false), + /* + * This register is pilot tone threshold, different with the + * calibration tool version, it will be updated in + * tas2781_calib_start_put(), set to 1mA. + */ + X2781_CL_STT_4BYTS_UNLOCKED(0, 0x00, 0x00, 0x00, 0x56), + X2781_CL_STT_4BYTS_UNLOCKED(TAS2781_PRM_PLT_FLAG_REG, + 0x40, 0x00, 0x00, 0x00), + X2781_CL_STT_LEN_UNLOCKED(TAS2781_PRM_SINEGAIN_REG), + X2781_CL_STT_LEN_UNLOCKED(TAS2781_PRM_SINEGAIN2_REG), +}; + static const struct i2c_device_id tasdevice_id[] = { { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, @@ -141,6 +204,557 @@ static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static int tasdev_cali_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned char *data = cali_data->data; + unsigned int i = 0; + unsigned int j, k; + int rc; + + guard(mutex)(&priv->codec_lock); + if (!priv->is_user_space_calidata) + return -1; + + if (!p->r0_reg) + return -1; + + dst[i++] = bytes_ext->max; + dst[i++] = 'r'; + + dst[i++] = TASDEVICE_BOOK_ID(p->r0_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->r0_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->r0_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->r0_low_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->r0_low_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->r0_low_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->invr0_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->invr0_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->invr0_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->pow_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->pow_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->pow_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->tlimit_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->tlimit_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->tlimit_reg); + + for (j = 0, k = 0; j < priv->ndev; j++) { + if (j == data[k]) { + dst[i++] = j; + k++; + } else { + dev_err(priv->dev, "chn %d device %u not match\n", + j, data[k]); + k += 21; + continue; + } + rc = tasdevice_dev_bulk_read(priv, j, p->r0_reg, &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_reg bulk_rd err = %d\n", + j, rc); + i += 20; + k += 20; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d r0_data is not same\n", j); + k += 4; + i += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->r0_low_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_low bulk_rd err = %d\n", + j, rc); + i += 16; + k += 16; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d r0_low is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->invr0_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d invr0 bulk_rd err = %d\n", + j, rc); + i += 12; + k += 12; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d invr0 is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->pow_reg, &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d pow_reg bulk_rd err = %d\n", + j, rc); + i += 8; + k += 8; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d pow_reg is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->tlimit_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d tlimit bulk_rd err = %d\n", + j, rc); + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d tlimit is not same\n", j); + i += 4; + k += 4; + } + return 0; +} + +static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, + unsigned char *dst) +{ + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int rc = -1; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + rc = tasdevice_dev_bulk_read(tas_priv, i, reg, &dst[1], + 4); + break; + } + } + + return rc; +} + +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; + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int j; + + if (p == NULL) + return; + + /* Store the current setting from the chip */ + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_read(tas_priv, i, p[j].reg, + (int *)&p[j].val[0]); + } else { + switch (p[j].reg) { + case 0: { + if (!reg[0]) + continue; + p[j].reg = reg[0]; + } + 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); + } + } + + /* Update the setting for calibration */ + for (j = 0; j < sum - 2; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + 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); +} + +static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dat = ucontrol->value.bytes.data; + int i, reg[4]; + int j = 0; + + guard(mutex)(&priv->codec_lock); + if (priv->chip_id != TAS2781 || bytes_ext->max != dat[0] || + dat[1] != 'r') { + dev_err(priv->dev, "%s: package fmt or chipid incorrect\n", + __func__); + return 0; + } + j += 2; + /* refresh pilot tone and SineGain register */ + for (i = 0; i < ARRAY_SIZE(reg); i++) { + reg[i] = TASDEVICE_REG(dat[j], dat[j + 1], dat[j + 2]); + j += 3; + } + + for (i = 0; i < priv->ndev; i++) { + int k = i * 9 + j; + + if (dat[k] != i) { + dev_err(priv->dev, "%s:no cal-setting for dev %d\n", + __func__, i); + continue; + } + sngl_calib_start(priv, i, reg, dat + k); + } + return 1; +} + +static void tas2781_calib_stop_put(struct tasdevice_priv *tas_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; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_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, + p[j].val, 4); + } + } + } +} + +static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct bulk_reg_val *q = (struct bulk_reg_val *)tas2563_cali_start_reg; + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int rc = 1; + int i, j; + + guard(mutex)(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2563) { + rc = -1; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_read(tas_priv, + i, p[j].reg, + (unsigned int *)&p[j].val[0]); + else + tasdevice_dev_bulk_read(tas_priv, + i, p[j].reg, p[j].val, 4); + } + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_write(tas_priv, i, p[j].reg, + q[j].val[0]); + else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + q[j].val, 4); + } + } +out: + return rc; +} + +static void tas2563_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tasdev_calib_stop_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + + guard(mutex)(&priv->codec_lock); + if (priv->chip_id == TAS2563) + tas2563_calib_stop_put(priv); + else + tas2781_calib_stop_put(priv); + + return 1; +} + +static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *src = ucontrol->value.bytes.data; + unsigned char *dst = cali_data->data; + int rc = 1, i = 0; + int j; + + guard(mutex)(&priv->codec_lock); + if (src[0] != bytes_ext->max || src[1] != 'r') { + dev_err(priv->dev, "%s: pkg fmt invalid\n", __func__); + return 0; + } + for (j = 0; j < priv->ndev; j++) { + if (src[17 + j * 21] != j) { + dev_err(priv->dev, "%s: pkg fmt invalid\n", __func__); + return 0; + } + } + 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; + + memcpy(dst, &src[i], cali_data->total_sz); + return rc; +} + +static int tas2781_latch_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct tasdevice *tasdev = tas_priv->tasdevice; + unsigned char *dst = ucontrol->value.bytes.data; + int i, val, rc = -1; + + dst[0] = bytes_ext->max; + guard(mutex)(&tas_priv->codec_lock); + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[1] = i; + rc = tasdevice_dev_read(tas_priv, i, + TAS2781_RUNTIME_LATCH_RE_REG, &val); + if (rc < 0) + dev_err(tas_priv->dev, "%s, get value error\n", + __func__); + else + dst[2] = val; + + break; + } + } + + return rc; +} + +static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG_TF; + else + reg = TAS2563_RUNTIME_RE_REG_TF; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG; + else + reg = TAS2563_RUNTIME_RE_REG; + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct calidata *cali_data = &tas_priv->cali_data; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + + if (tas_priv->chip_id == TAS2563) + reg = TAS2563_PRM_R0_REG; + else if (cali_data->cali_reg_array.r0_reg) + reg = cali_data->cali_reg_array.r0_reg; + else + return -1; + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + 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; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + 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; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_nop_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + static int tas2563_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -241,6 +855,16 @@ static const struct snd_kcontrol_new tasdevice_snd_controls[] = { tasdev_force_fwload_get, tasdev_force_fwload_put), }; +static const struct snd_kcontrol_new tasdevice_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tasdev_calib_stop_put), + SND_SOC_BYTES_EXT("Amp TF Data", 6, tasdev_tf_data_get, NULL), + SND_SOC_BYTES_EXT("Amp RE Data", 6, tasdev_re_data_get, NULL), + SND_SOC_BYTES_EXT("Amp R0 Data", 6, tasdev_r0_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA1 Data", 6, tasdev_XMA1_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA2 Data", 6, tasdev_XMA2_data_get, NULL), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -250,6 +874,10 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, dvc_tlv), }; +static const struct snd_kcontrol_new tas2781_cali_controls[] = { + SND_SOC_BYTES_EXT("Amp Latch Data", 3, tas2781_latch_reg_get, NULL), +}; + static const struct snd_kcontrol_new tas2563_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2563_DVC_LVL, 0, 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, @@ -257,6 +885,11 @@ static const struct snd_kcontrol_new tas2563_snd_controls[] = { tas2563_dvc_tlv), }; +static const struct snd_kcontrol_new tas2563_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Start", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tas2563_calib_start_put), +}; + static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -274,6 +907,31 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, return ret; } +static int tasdevice_info_active_num(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->ndev - 1; + + return 0; +} + +static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = TAS2563; + uinfo->value.integer.max = TAS2781; + + return 0; +} + static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -330,6 +988,17 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, return 0; } +static int tasdevice_get_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->chip_id; + + return 0; +} + static int tasdevice_create_control(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *prof_ctrls; @@ -421,11 +1090,47 @@ static int tasdevice_configuration_put( return ret; } +static int tasdevice_active_num_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + ucontrol->value.integer.value[0] = i; + return 0; + } + } + + return -1; +} + +static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int dev_id = ucontrol->value.integer.value[0]; + int max = tas_priv->ndev - 1, rc; + + dev_id = clamp(dev_id, 0, max); + + guard(mutex)(&tas_priv->codec_lock); + rc = tasdev_chn_switch(tas_priv, dev_id); + + return rc; +} + static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *prog_name, *conf_name; - int nr_controls = 2; + char *active_dev_num, *chip_id; + char *conf_name, *prog_name; + int nr_controls = 4; int mix_index = 0; int ret; @@ -466,6 +1171,30 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].put = tasdevice_configuration_put; mix_index++; + active_dev_num = devm_kstrdup(tas_priv->dev, "Activate Tasdevice Num", + GFP_KERNEL); + if (!active_dev_num) { + ret = -ENOMEM; + goto out; + } + dsp_ctrls[mix_index].name = active_dev_num; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_active_num; + dsp_ctrls[mix_index].get = tasdevice_active_num_get; + dsp_ctrls[mix_index].put = tasdevice_active_num_put; + mix_index++; + + chip_id = devm_kstrdup(tas_priv->dev, "Tasdevice Chip Id", GFP_KERNEL); + if (!chip_id) { + ret = -ENOMEM; + goto out; + } + dsp_ctrls[mix_index].name = chip_id; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_chip_id; + dsp_ctrls[mix_index].get = tasdevice_get_chip_id; + mix_index++; + ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); @@ -473,6 +1202,149 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) return ret; } +static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) +{ + struct calidata *cali_data = &priv->cali_data; + struct tasdevice *tasdev = priv->tasdevice; + struct soc_bytes_ext *ext_cali_data; + struct snd_kcontrol_new *cali_ctrls; + unsigned int nctrls; + char *cali_name; + int rc, i; + + rc = snd_soc_add_component_controls(priv->codec, + tasdevice_cali_controls, ARRAY_SIZE(tasdevice_cali_controls)); + if (rc < 0) { + dev_err(priv->dev, "%s: Add cali controls err rc = %d", + __func__, rc); + return rc; + } + + if (priv->chip_id == TAS2781) { + 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 = + kmemdup(tas2781_cali_start_reg, + sizeof(tas2781_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_backup) + return -ENOMEM; + } + } else { + cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; + nctrls = ARRAY_SIZE(tas2563_cali_controls); + for (i = 0; i < priv->ndev; i++) { + tasdev[i].cali_data_backup = + kmemdup(tas2563_cali_start_reg, + sizeof(tas2563_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_backup) + return -ENOMEM; + } + } + + rc = snd_soc_add_component_controls(priv->codec, cali_ctrls, nctrls); + if (rc < 0) { + dev_err(priv->dev, "%s: Add chip cali ctrls err rc = %d", + __func__, rc); + return rc; + } + + /* index for cali_ctrls */ + i = 0; + if (priv->chip_id == TAS2781) + nctrls = 2; + else + nctrls = 1; + + /* + * Alloc kcontrol via devm_kzalloc(), which don't manually + * free the kcontrol。 + */ + cali_ctrls = devm_kcalloc(priv->dev, nctrls, + sizeof(cali_ctrls[0]), GFP_KERNEL); + if (!cali_ctrls) + return -ENOMEM; + + ext_cali_data = devm_kzalloc(priv->dev, sizeof(*ext_cali_data), + GFP_KERNEL); + if (!ext_cali_data) + return -ENOMEM; + + cali_name = devm_kstrdup(priv->dev, "Speaker Calibrated Data", + GFP_KERNEL); + if (!cali_name) + return -ENOMEM; + /* the number of calibrated data per tas2563/tas2781 */ + cali_data->cali_dat_sz_per_dev = 20; + /* + * Data structure for tas2563/tas2781 calibrated data: + * Pkg len (1 byte) + * Reg id (1 byte, constant 'r') + * book, page, register array for calibrated data (15 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info (1 byte) + * Calibrated data for Device #i (20 bytes) + * } + */ + ext_cali_data->max = priv->ndev * + (cali_data->cali_dat_sz_per_dev + 1) + 1 + 15 + 1; + priv->cali_data.total_sz = priv->ndev * + (cali_data->cali_dat_sz_per_dev + 1); + priv->cali_data.data = devm_kzalloc(priv->dev, + ext_cali_data->max, GFP_KERNEL); + cali_ctrls[i].name = cali_name; + cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cali_ctrls[i].info = snd_soc_bytes_info_ext; + cali_ctrls[i].get = tasdev_cali_data_get; + cali_ctrls[i].put = tasdev_cali_data_put; + cali_ctrls[i].private_value = (unsigned long)ext_cali_data; + i++; + + cali_data->data = devm_kzalloc(priv->dev, cali_data->total_sz, + GFP_KERNEL); + if (!cali_data->data) + return -ENOMEM; + + if (priv->chip_id == TAS2781) { + struct soc_bytes_ext *ext_cali_start; + char *cali_start_name; + + ext_cali_start = devm_kzalloc(priv->dev, + sizeof(*ext_cali_start), GFP_KERNEL); + if (!ext_cali_start) + return -ENOMEM; + + cali_start_name = devm_kstrdup(priv->dev, + "Calibration Start", GFP_KERNEL); + if (!cali_start_name) + return -ENOMEM; + /* + * package structure for tas2781 ftc start: + * Pkg len (1 byte) + * Reg id (1 byte, constant 'r') + * book, page, register for pilot threshold, pilot tone + * and sine gain (12 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info (1 byte) + * Sine gain for Device #i (8 bytes) + * } + */ + ext_cali_start->max = 14 + priv->ndev * 9; + cali_ctrls[i].name = cali_start_name; + cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cali_ctrls[i].info = snd_soc_bytes_info_ext; + cali_ctrls[i].put = tas2781_calib_start_put; + cali_ctrls[i].get = tasdev_nop_get; + cali_ctrls[i].private_value = (unsigned long)ext_cali_start; + i++; + } + + rc = snd_soc_add_component_controls(priv->codec, cali_ctrls, + nctrls < i ? nctrls : i); + + return rc; +} + static void tasdevice_fw_ready(const struct firmware *fmw, void *context) { @@ -519,6 +1391,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, goto out; } + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; /* If calibrated data occurs error, dsp will still works with default @@ -720,6 +1598,11 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) static void tasdevice_deinit(void *context) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) + kfree(tasdev[i].cali_data_backup); tasdevice_config_info_remove(tas_priv); tasdevice_dsp_remove(tas_priv); -- 2.34.1

2 1

[PATCH 0/5] AMD SoundWire machine driver code refactor
by Vijendar Mukunda 13 Sep '24

13 Sep '24

This patch series moves common Soundwire endpoint parsing and dai creation logic to common placeholder from Intel generic SoundWire machine driver code to make it generic. AMD SoundWire machine driver code is refactored to use these functions for SoundWire endpoint parsing and dai creation logic. Link: https://github.com/thesofproject/linux/pull/5171 Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Vijendar Mukunda (5): ASoC: intel: sof_sdw: rename soundwire endpoint and dailink structures ASoC: intel: sof_sdw: rename soundwire parsing helper functions ASoC: sdw_util/intel: move soundwire endpoint and dai link structures ASoC: sdw_utils/intel: move soundwire endpoint parsing helper functions ASoC: amd: acp: refactor SoundWire machine driver code include/sound/soc_sdw_utils.h | 31 ++ sound/soc/amd/acp/acp-sdw-sof-mach.c | 540 ++++++++------------------- sound/soc/intel/boards/sof_sdw.c | 195 +--------- sound/soc/sdw_utils/soc_sdw_utils.c | 161 ++++++++ 4 files changed, 354 insertions(+), 573 deletions(-) -- 2.34.1

2 6

[PATCH v2] ASoC: atmel: mchp-pdmc: Retain Non-Runtime Controls
by Andrei Simion 13 Sep '24

13 Sep '24

From: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Avoid removing these controls, as doing so can cause issues if the stream is initiated from another control. Ensure these controls remain intact when the stream is started or finished. Instead of removing them, return an -EBUSY error code to indicate that the controller is busy, especially when the audio filter and the SINC filter are in use. [andrei.simion(a)microchip.com: Reword the commit title and the commit message. Replace spinlock and busy variable with atomic_t busy_stream.] Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Signed-off-by: Andrei Simion <andrei.simion(a)microchip.com> --- v1 -> v2: - drop spinlock and u8 busy:1; - use atomic_t variable busy_stream - use atomic_read, atomic_set --- sound/soc/atmel/mchp-pdmc.c | 57 ++++++++++--------------------------- 1 file changed, 15 insertions(+), 42 deletions(-) diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index d97d153ee375..939cd44ebc8a 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -124,6 +124,7 @@ struct mchp_pdmc { int mic_no; int sinc_order; bool audio_filter_en; + atomic_t busy_stream; }; static const char *const mchp_pdmc_sinc_filter_order_text[] = { @@ -167,6 +168,10 @@ static int mchp_pdmc_sinc_order_put(struct snd_kcontrol *kcontrol, return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + + if (atomic_read(&dd->busy_stream)) + return -EBUSY; + if (val == dd->sinc_order) return 0; @@ -193,6 +198,9 @@ static int mchp_pdmc_af_put(struct snd_kcontrol *kcontrol, struct mchp_pdmc *dd = snd_soc_component_get_drvdata(component); bool af = uvalue->value.integer.value[0] ? true : false; + if (atomic_read(&dd->busy_stream)) + return -EBUSY; + if (dd->audio_filter_en == af) return 0; @@ -379,52 +387,10 @@ static const struct snd_kcontrol_new mchp_pdmc_snd_controls[] = { }, }; -static int mchp_pdmc_close(struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - return snd_soc_add_component_controls(component, mchp_pdmc_snd_controls, - ARRAY_SIZE(mchp_pdmc_snd_controls)); -} - -static int mchp_pdmc_open(struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - int i; - - /* remove controls that can't be changed at runtime */ - for (i = 0; i < ARRAY_SIZE(mchp_pdmc_snd_controls); i++) { - const struct snd_kcontrol_new *control = &mchp_pdmc_snd_controls[i]; - struct snd_ctl_elem_id id; - int err; - - if (component->name_prefix) - snprintf(id.name, sizeof(id.name), "%s %s", component->name_prefix, - control->name); - else - strscpy(id.name, control->name, sizeof(id.name)); - - id.numid = 0; - id.iface = control->iface; - id.device = control->device; - id.subdevice = control->subdevice; - id.index = control->index; - err = snd_ctl_remove_id(component->card->snd_card, &id); - if (err < 0) - dev_err(component->dev, "%d: Failed to remove %s\n", err, - control->name); - } - - return 0; -} - static const struct snd_soc_component_driver mchp_pdmc_dai_component = { .name = "mchp-pdmc", .controls = mchp_pdmc_snd_controls, .num_controls = ARRAY_SIZE(mchp_pdmc_snd_controls), - .open = &mchp_pdmc_open, - .close = &mchp_pdmc_close, - .legacy_dai_naming = 1, - .trigger_start = SND_SOC_TRIGGER_ORDER_LDC, }; static const unsigned int mchp_pdmc_1mic[] = {1}; @@ -587,6 +553,11 @@ static int mchp_pdmc_hw_params(struct snd_pcm_substream *substream, cfgr_val |= MCHP_PDMC_CFGR_BSSEL(i); } + /* + * from these point forward, we consider the controller busy, so the + * audio filter and SINC order can't be changed + */ + atomic_set(&dd->busy_stream, 1); for (osr_start = dd->audio_filter_en ? 64 : 8; osr_start <= 256 && best_diff_rate; osr_start *= 2) { long round_rate; @@ -1143,6 +1114,8 @@ static void mchp_pdmc_remove(struct platform_device *pdev) { struct mchp_pdmc *dd = platform_get_drvdata(pdev); + atomic_set(&dd->busy_stream, 0); + if (!pm_runtime_status_suspended(dd->dev)) mchp_pdmc_runtime_suspend(dd->dev); base-commit: 5acd9952f95fb4b7da6d09a3be39195a80845eb6 -- 2.34.1

1 0

change getopt_long return variable from char to int
by GitHub pull_request - opened 12 Sep '24

12 Sep '24

alsa-project/alsa-utils pull request #275 was opened from sadguitarius: fixes #274 Request URL : https://github.com/alsa-project/alsa-utils/pull/275 Patch URL : https://github.com/alsa-project/alsa-utils/pull/275.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aseqsend getopt_long variable should be int instead of char
by GitHub issues - opened 12 Sep '24

12 Sep '24

alsa-project/alsa-utils issue #274 was opened from sadguitarius: Hi, just wanted to point out that aseqsend is failing because the value returned by getopt_long can never be -1 since it's a char. Issue URL : https://github.com/alsa-project/alsa-utils/issues/274 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: usb-audio: Add logitech Audio profile quirk
by Joshua Pius 12 Sep '24

12 Sep '24

Specify shortnames for the following Logitech Devices: Rally bar, Rally bar mini, Tap, MeetUp and Huddle. Signed-off-by: Joshua Pius <joshuapius(a)chromium.org> --- sound/usb/card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 778de9244f1e..9c411b82a218 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -384,6 +384,12 @@ static const struct usb_audio_device_name usb_audio_names[] = { /* Creative/Toshiba Multimedia Center SB-0500 */ DEVICE_NAME(0x041e, 0x3048, "Toshiba", "SB-0500"), + /* Logitech Audio Devices */ + DEVICE_NAME(0x046d, 0x0867, "Logitech, Inc.", "Logi-MeetUp"), + DEVICE_NAME(0x046d, 0x0874, "Logitech, Inc.", "Logi-Tap-Audio"), + DEVICE_NAME(0x046d, 0x087c, "Logitech, Inc.", "Logi-Huddle"), + DEVICE_NAME(0x046d, 0x0898, "Logitech, Inc.", "Logi-RB-Audio"), + DEVICE_NAME(0x046d, 0x08d2, "Logitech, Inc.", "Logi-RBM-Audio"), DEVICE_NAME(0x046d, 0x0990, "Logitech, Inc.", "QuickCam Pro 9000"), DEVICE_NAME(0x05e1, 0x0408, "Syntek", "STK1160"), -- 2.46.0.598.g6f2099f65c-goog

1 0

[PATCH v2 0/3] Add a driver for the Iron Device SMA1307 Amp
by Kiseok Jo 12 Sep '24

12 Sep '24

This adds basic audio support for the Iron Device SMA1307 amplifier Kiseok Jo (3): ASoC: sma1307: Add driver for Iron Device SMA1307 ASoC: dt-bindings: irondevice,sma1307: Add initial DT binding doc: ABI: testing: sma1307: Add support for SMA1307 .../ABI/testing/sysfs-bus-i2c-devices-sma1307 | 17 + .../bindings/sound/irondevice,sma1307.yaml | 54 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2191 +++++++++++++++++ sound/soc/codecs/sma1307.h | 454 ++++ 6 files changed, 2726 insertions(+) create mode 100644 Documentation/ABI/testing/sysfs-bus-i2c-devices-sma1307 create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h -- 2.39.2

6 22

[PATCH 0/3] Improvements for mchp-pdmc
by Andrei Simion 12 Sep '24

12 Sep '24

This patch set is intended to enhance the functionality and maintainability of the mchp-pdmc driver: - Enhances performance by refining maxburst logic. - Introduces a name for better identification and management. - Ensures controls remain intact when streams start/finish, returning -EBUSY if the controller is busy. Codrin Ciubotariu (3): ASoC: atmel: mchp-pdmc: Improve maxburst calculation for better performance ASoC: atmel: mchp-pdmc: Add snd_soc_dai_driver name ASoC: atmel: mchp-pdmc: Retain Non-Runtime Controls sound/soc/atmel/mchp-pdmc.c | 118 +++++++++++++++++++----------------- 1 file changed, 63 insertions(+), 55 deletions(-) base-commit: 6708132e80a2ced620bde9b9c36e426183544a23 -- 2.34.1

3 6

ucm2: soundwire: rt722: add ConflictingDevice in each device
by GitHub pull_request - opened 12 Sep '24

12 Sep '24

alsa-project/alsa-ucm-conf pull request #441 was opened from shumingfan: This patch allows only one input and one output to be opened simultaneously. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/441 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/441.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: usb-audio: Add logitech Audio profile quirk
by Joshua Pius 12 Sep '24

12 Sep '24

Specify shortnames for the following Logitech Devices: Rally bar, Rally bar mini, Tap, MeetUp and Huddle. Signed-off-by: Joshua Pius <joshuapius(a)google.com> --- sound/usb/card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 778de9244f1e..9c411b82a218 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -384,6 +384,12 @@ static const struct usb_audio_device_name usb_audio_names[] = { /* Creative/Toshiba Multimedia Center SB-0500 */ DEVICE_NAME(0x041e, 0x3048, "Toshiba", "SB-0500"), + /* Logitech Audio Devices */ + DEVICE_NAME(0x046d, 0x0867, "Logitech, Inc.", "Logi-MeetUp"), + DEVICE_NAME(0x046d, 0x0874, "Logitech, Inc.", "Logi-Tap-Audio"), + DEVICE_NAME(0x046d, 0x087c, "Logitech, Inc.", "Logi-Huddle"), + DEVICE_NAME(0x046d, 0x0898, "Logitech, Inc.", "Logi-RB-Audio"), + DEVICE_NAME(0x046d, 0x08d2, "Logitech, Inc.", "Logi-RBM-Audio"), DEVICE_NAME(0x046d, 0x0990, "Logitech, Inc.", "QuickCam Pro 9000"), DEVICE_NAME(0x05e1, 0x0408, "Syntek", "STK1160"), -- 2.46.0.598.g6f2099f65c-goog

3 4

Mismatched volume controls of Sound Blaster X3
by GitHub issues - opened 12 Sep '24

12 Sep '24

alsa-project/alsa-utils issue #273 was opened from buboleck: Hello, Recently I got Creative Sound Blaster X3 (SB1810) as it seemed it works fine with linux. I'm using Gentoo with Plasma desktop (ALSA + Pipewire) and the onboard sound is OK but the SB X3 seems is misbehaving, I found that the volume controls for RearLeft is swapped with Center and RearRight is swapped with the LFE. (Also tried live Ubuntu 24.04 same issue) On the other hand maybe related, I was not able to get stereo upmix working in any way, ALSA upmix does not want to work as also the Pipiwire upmix. Gentoo kernel 6.10.9 alsa-lib-1.2.12 alsa-tools-1.2.11 alsa-utils-1.2.12 alsa-plugins-1.2.12 pyalsa-1.2.12 Issue URL : https://github.com/alsa-project/alsa-utils/issues/273 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Mic mute LED only follows internal rather than selected
by GitHub issues - opened 12 Sep '24

12 Sep '24

alsa-project/alsa-ucm-conf issue #440 was opened from JL2210: I have a laptop keyboard with a key to mute the microphone. Using this key mutes whichever mic is selected as default. However, the light on the key only changes with the internal microphone rather than the active mic. Pressing the key when a headset is plugged in doesn't change the LED, but pressing it without a headset does. [alsa-info.txt](https://github.com/user-attachments/files/16971038/alsa-info… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/440 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1] ASoC: meson: axg-card: fix 'use-after-free'
by Arseniy Krasnov 11 Sep '24

11 Sep '24

Buffer 'card->dai_link' is reallocated in 'meson_card_reallocate_links()', so move 'pad' pointer initialization after this function when memory is already reallocated. Kasan bug report: ================================================================== BUG: KASAN: slab-use-after-free in axg_card_add_link+0x76c/0x9bc Read of size 8 at addr ffff000000e8b260 by task modprobe/356 CPU: 0 PID: 356 Comm: modprobe Tainted: G O 6.9.12-sdkernel #1 Call trace: dump_backtrace+0x94/0xec show_stack+0x18/0x24 dump_stack_lvl+0x78/0x90 print_report+0xfc/0x5c0 kasan_report+0xb8/0xfc __asan_load8+0x9c/0xb8 axg_card_add_link+0x76c/0x9bc [snd_soc_meson_axg_sound_card] meson_card_probe+0x344/0x3b8 [snd_soc_meson_card_utils] platform_probe+0x8c/0xf4 really_probe+0x110/0x39c __driver_probe_device+0xb8/0x18c driver_probe_device+0x108/0x1d8 __driver_attach+0xd0/0x25c bus_for_each_dev+0xe0/0x154 driver_attach+0x34/0x44 bus_add_driver+0x134/0x294 driver_register+0xa8/0x1e8 __platform_driver_register+0x44/0x54 axg_card_pdrv_init+0x20/0x1000 [snd_soc_meson_axg_sound_card] do_one_initcall+0xdc/0x25c do_init_module+0x10c/0x334 load_module+0x24c4/0x26cc init_module_from_file+0xd4/0x128 __arm64_sys_finit_module+0x1f4/0x41c invoke_syscall+0x60/0x188 el0_svc_common.constprop.0+0x78/0x13c do_el0_svc+0x30/0x40 el0_svc+0x38/0x78 el0t_64_sync_handler+0x100/0x12c el0t_64_sync+0x190/0x194 Fixes: 7864a79f37b5 ("ASoC: meson: add axg sound card support") Cc: <Stable(a)vger.kernel.org> Signed-off-by: Arseniy Krasnov <avkrasnov(a)salutedevices.com> --- sound/soc/meson/axg-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 8c5605c1e34e8..eb0302f207407 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -104,7 +104,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, int *index) { struct meson_card *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_dai_link *pad = &card->dai_link[*index]; + struct snd_soc_dai_link *pad; struct snd_soc_dai_link *lb; struct snd_soc_dai_link_component *dlc; int ret; @@ -114,6 +114,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, if (ret) return ret; + pad = &card->dai_link[*index]; lb = &card->dai_link[*index + 1]; lb->name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-lb", pad->name); -- 2.30.1

1 0

[PATCH v3] ASoC: dt-bindings: renesas,rsnd: add post-init-providers property
by Kuninori Morimoto 11 Sep '24

11 Sep '24

At least if rsnd is using DPCM connection on Audio-Graph-Card2, fw_devlink might doesn't have enough information to break the cycle (Same problem might occur with Multi-CPU/Codec or Codec2Codec). In such case, rsnd driver will not be probed. Add post-init-providers support to break the link cycle. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 - remove "|" from description - removed common post-init-providers description - 100 char to 80 char for 1 line Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 07ec6247d9def..3bc93c59535e9 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -112,6 +112,12 @@ properties: description: List of necessary clock names. # details are defined below + post-init-providers: + description: At least if rsnd is using DPCM connection on Audio-Graph-Card2, + fw_devlink might doesn't have enough information to break the cycle. rsnd + driver will not be probed in such case. Same problem might occur with + Multi-CPU/Codec or Codec2Codec. + # ports is below port: $ref: audio-graph-port.yaml#/definitions/port-base -- 2.43.0

3 2

[PATCH] ASoC: dt-bindings: microchip,sama7g5-spdifrx: Add common DAI reference
by Andrei Simion 11 Sep '24

11 Sep '24

Update the spdifrx yaml file to reference the dai-common.yaml schema, enabling the use of the 'sound-name-prefix' property Signed-off-by: Andrei Simion <andrei.simion(a)microchip.com> --- .../devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml index 2f43c684ab88..7fbab5871be4 100644 --- a/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml @@ -13,6 +13,9 @@ description: The Microchip Sony/Philips Digital Interface Receiver is a serial port compliant with the IEC-60958 standard. +allOf: + - $ref: dai-common.yaml# + properties: "#sound-dai-cells": const: 0 @@ -53,7 +56,7 @@ required: - dmas - dma-names -additionalProperties: false +unevaluatedProperties: false examples: - | base-commit: 6708132e80a2ced620bde9b9c36e426183544a23 -- 2.34.1

1 0

[PATCH] ASoC: codecs: avoid possible garbage value in peb2466_reg_read()
by Su Hui 11 Sep '24

11 Sep '24

Clang static checker (scan-build) warning: sound/soc/codecs/peb2466.c:232:8: Assigned value is garbage or undefined [core.uninitialized.Assign] 232 | *val = tmp; | ^ ~~~ When peb2466_read_byte() fails, 'tmp' will have a garbage value. Add a judgemnet to avoid this problem. Fixes: 227f609c7c0e ("ASoC: codecs: Add support for the Infineon PEB2466 codec") Signed-off-by: Su Hui <suhui(a)nfschina.com> --- sound/soc/codecs/peb2466.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/peb2466.c b/sound/soc/codecs/peb2466.c index 76ee7e3f4d9b..67ea70cef0c7 100644 --- a/sound/soc/codecs/peb2466.c +++ b/sound/soc/codecs/peb2466.c @@ -229,7 +229,8 @@ static int peb2466_reg_read(void *context, unsigned int reg, unsigned int *val) case PEB2466_CMD_XOP: case PEB2466_CMD_SOP: ret = peb2466_read_byte(peb2466, reg, &tmp); - *val = tmp; + if (!ret) + *val = tmp; break; default: dev_err(&peb2466->spi->dev, "Not a XOP or SOP command\n"); -- 2.30.2

1 0

[PATCH 0/1] Add support for primary mi2s on SM8250
by Jens Reidel 11 Sep '24

11 Sep '24

This patch adds support for the primary mi2s interface on devices using SM8250 audio drivers. Tested on SM7150 (xiaomi-davinci). SM7150 sound is close to SM8250 and we intend to use it as a fallback in the future. To: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> To: Liam Girdwood <lgirdwood(a)gmail.com> To: Mark Brown <broonie(a)kernel.org> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-arm-msm(a)vger.kernel.org Cc: linux-sound(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linux(a)mainlining.org Jens Reidel (1): ASoC: qcom: sm8250: enable primary mi2s sound/soc/qcom/sm8250.c | 8 ++++++++ 1 file changed, 8 insertions(+) -- 2.46.0

4 5

[PATCH 00/13] ALSA: update sample rate definitions
by Jerome Brunet 11 Sep '24

11 Sep '24

This patchset adds rate definitions for 12kHz, 24kHz and 128kHz. It is follow-up on the series/discussion [0] about adding 128kHz for spdif/eARC support. The outcome was to add 12kHz and 24kHz as well and clean up the drivers that no longer require custom rules to allow these rates. Identifying these drivers is not straight forward, I tried my best but I may have missed some. Those will continue to work anyway so it is not critical. Some drivers using these rates have not been changed on purpose. The reason for this may be: * The driver used other uncommon rates that still don't have a definition so a custom rule is still required. * The constraint structure is used in some other way by the driver and removing it would not help the readability or maintainability. This is the case the freescale asrc drivers for example. There is one change per driver so, if there is a problem later on, it will easier to properly add Fixes tag. The series has been tested with * ARM64 defconfig - spdif 128kHz at runtime. * x86_64 allmodconfig - compile test only Last, the change adding IEC958 definitions has been dropped for this patchset and will be resent separately [0]: https://lore.kernel.org/all/20240628122429.2018059-1-jbrunet@baylibre.com/ --- Jerome Brunet (13): ALSA: pcm: add more sample rate definitions ALSA: cmipci: drop SNDRV_PCM_RATE_KNOT ALSA: emu10k1: drop SNDRV_PCM_RATE_KNOT ALSA: hdsp: drop SNDRV_PCM_RATE_KNOT ALSA: hdspm: drop SNDRV_PCM_RATE_KNOT ASoC: cs35l36: drop SNDRV_PCM_RATE_KNOT ASoC: cs35l41: drop SNDRV_PCM_RATE_KNOT ASoC: cs53l30: drop SNDRV_PCM_RATE_KNOT ASoC: Intel: avs: drop SNDRV_PCM_RATE_KNOT ASoC: qcom: q6asm-dai: drop SNDRV_PCM_RATE_KNOT ASoC: sunxi: sun4i-codec: drop SNDRV_PCM_RATE_KNOT ASoC: cs35l34: drop useless rate contraint ASoC: spdif: extend supported rates to 768kHz include/sound/pcm.h | 31 +++++++++++++++++-------------- sound/core/pcm_native.c | 6 +++--- sound/pci/cmipci.c | 32 +++++++++----------------------- sound/pci/emu10k1/emupcm.c | 31 +++++-------------------------- sound/pci/rme9652/hdsp.c | 18 ++++++------------ sound/pci/rme9652/hdspm.c | 16 +--------------- sound/soc/codecs/cs35l34.c | 21 --------------------- sound/soc/codecs/cs35l36.c | 34 ++++++++++++---------------------- sound/soc/codecs/cs35l41.c | 34 +++++++++++----------------------- sound/soc/codecs/cs53l30.c | 24 +++--------------------- sound/soc/codecs/spdif_receiver.c | 3 ++- sound/soc/codecs/spdif_transmitter.c | 3 ++- sound/soc/intel/avs/pcm.c | 22 ++++++---------------- sound/soc/qcom/qdsp6/q6asm-dai.c | 31 ++++++++++--------------------- sound/soc/sunxi/sun4i-codec.c | 28 +++++++++------------------- 15 files changed, 96 insertions(+), 238 deletions(-) --- base-commit: 785f4052380684dbcc156203c537c799e2f4be09 change-id: 20240905-alsa-12-24-128-8edab4c08dd4 Best regards, -- Jerome

11 35

[PATCH 1/2] ASoC: amd: acp: remove MODULE_ALIAS for legacy machine driver
by Vijendar Mukunda 11 Sep '24

11 Sep '24

As module device table added for AMD legacy machine driver, MODULE_ALIAS is not required. Remove MODULE_ALIAS for AMD legacy machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-legacy-mach.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-mach.c b/sound/soc/amd/acp/acp-legacy-mach.c index 0d529e32e552..d104f7e8fdcd 100644 --- a/sound/soc/amd/acp/acp-legacy-mach.c +++ b/sound/soc/amd/acp/acp-legacy-mach.c @@ -242,11 +242,4 @@ module_platform_driver(acp_asoc_audio); MODULE_IMPORT_NS(SND_SOC_AMD_MACH); MODULE_DESCRIPTION("ACP chrome audio support"); -MODULE_ALIAS("platform:acp3xalc56821019"); -MODULE_ALIAS("platform:acp3xalc5682sm98360"); -MODULE_ALIAS("platform:acp3xalc5682s1019"); -MODULE_ALIAS("platform:acp3x-es83xx"); -MODULE_ALIAS("platform:rmb-nau8825-max"); -MODULE_ALIAS("platform:rmb-rt5682s-rt1019"); -MODULE_ALIAS("platform:acp-pdm-mach"); MODULE_LICENSE("GPL v2"); -- 2.34.1

2 2

[PATCH v1 0/4] Add static channel mapping between soundwire master and slave
by Mohammad Rafi Shaik 11 Sep '24

11 Sep '24

Add static channel map support between soundwire master and slave. This patch series will resolve channel mask mismatch between master and slave. Scenario: wcd937x AMIC2 usecase Master Slave (wcd937x) +--------------+ +--------------+ | +--------+ | | +--------+ | AMIC1 ----->| | PORT1 | | | | TX1 | |<-----------AMIC1 AMIC2 ----->| | | | | | | | | +--------+ | | +--------+ | | | | | AMIC3 ----->| +--------+ | | +--------+ | | | PORT2 | | | | TX2 | |<-----------AMIC2 | | | | | | | |<-----------AMIC3 | +--------+ | | +--------+ | | | | | | +--------+ | | +--------+ | DMIC0...DMIC3------>| | PORT3 | | | | TX3 | |<-----------DMIC0...DMIC3 | | | | | | | |<-----------MBHC | +--------+ | | +--------+ | | | | | | +--------+ | | +--------+ | DMIC4...DMIC37----->| | PORT4 | | | | TX4 | |<-----------DMIC4...DMIC7 | | | | | | | | | +--------+ | | +--------+ | | | | | +------------- + +--------------+ For AMIC2 usecase, The Slave need to configure TX2 Port with channel mask 1 and for Master required PORT1 with channel mask 2, In existing design master and slave configured with same channel mask, it will fail AMIC2 usecase. The New design will help to configure channel mapping between master and slave from device tree. Mohammad Rafi Shaik (4): ASoC: dt-bindings: wcd938x-sdw: Add static channel mapping support soundwire: stream: Add set_master_channel_map() to set static channel mapping soundwire: qcom: Add static channel mapping support in soundwire master ASoC: codecs: wcd937x: Add static channel mapping support in wcd937x-sdw .../bindings/sound/qcom,wcd937x-sdw.yaml | 28 ++++++++++ drivers/soundwire/qcom.c | 18 +++++++ drivers/soundwire/stream.c | 16 ++++++ include/linux/soundwire/sdw.h | 5 ++ sound/soc/codecs/wcd937x-sdw.c | 52 ++++++++++++++++--- sound/soc/codecs/wcd937x.c | 12 ++++- sound/soc/codecs/wcd937x.h | 6 ++- 7 files changed, 126 insertions(+), 11 deletions(-) base-commit: 9aaeb87ce1e966169a57f53a02ba05b30880ffb8 -- 2.25.1

6 10

[PATCH RESEND] soundwire: stream: Revert "soundwire: stream: fix programming slave ports for non-continous port maps"
by Krzysztof Kozlowski 11 Sep '24

11 Sep '24

This reverts commit ab8d66d132bc8f1992d3eb6cab8d32dda6733c84 because it breaks codecs using non-continuous masks in source and sink ports. The commit missed the point that port numbers are not used as indices for iterating over prop.sink_ports or prop.source_ports. Soundwire core and existing codecs expect that the array passed as prop.sink_ports and prop.source_ports is continuous. The port mask still might be non-continuous, but that's unrelated. Reported-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Closes: https://lore.kernel.org/all/b6c75eee-761d-44c8-8413-2a5b34ee2f98@linux.inte… Fixes: ab8d66d132bc ("soundwire: stream: fix programming slave ports for non-continous port maps") Acked-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Resending with Ack/Rb tags and missing Cc-stable. --- drivers/soundwire/stream.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index f275143d7b18..7aa4900dcf31 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1291,18 +1291,18 @@ struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, unsigned int port_num) { struct sdw_dpn_prop *dpn_prop; - unsigned long mask; + u8 num_ports; int i; if (direction == SDW_DATA_DIR_TX) { - mask = slave->prop.source_ports; + num_ports = hweight32(slave->prop.source_ports); dpn_prop = slave->prop.src_dpn_prop; } else { - mask = slave->prop.sink_ports; + num_ports = hweight32(slave->prop.sink_ports); dpn_prop = slave->prop.sink_dpn_prop; } - for_each_set_bit(i, &mask, 32) { + for (i = 0; i < num_ports; i++) { if (dpn_prop[i].num == port_num) return &dpn_prop[i]; } -- 2.43.0

3 4

[PATCH] ASoC: tlv320aic31xx: Fix typos
by Andrew Kreimer 11 Sep '24

11 Sep '24

Fix typos in comments. Reported-by: Matthew Wilcox <willy(a)infradead.org> Signed-off-by: Andrew Kreimer <algonell(a)gmail.com> --- sound/soc/codecs/tlv320aic31xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 7e624c4b77b6..187d68e8688c 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -895,7 +895,7 @@ static int aic31xx_setup_pll(struct snd_soc_component *component, dev_err(component->dev, "%s: Sample rate (%u) and format not supported\n", __func__, params_rate(params)); - /* See bellow for details how fix this. */ + /* See below for details on how to fix this. */ return -EINVAL; } if (bclk_score != 0) { -- 2.46.0

2 3

[Patch v5 00/12] Add audio support for LPC32XX CPUs
by Piotr Wojtaszczyk 11 Sep '24

11 Sep '24

This pach set is to bring back audio to machines with a LPC32XX CPU. The legacy LPC32XX SoC used to have audio spport in linux 2.6.27. The support was dropped due to lack of interest from mainaeners. Piotr Wojtaszczyk (12): dt-bindings: dma: pl08x: Add dma-cells description dt-bindings: dma: Add lpc32xx DMA mux binding ASoC: dt-bindings: lpc32xx: Add lpc32xx i2s DT binding ARM: dts: lpc32xx: Use simple-mfd for clock control block ARM: dts: lpc32xx: Add missing dma properties ARM: dts: lpc32xx: Add missing i2s properties clk: lpc32xx: initialize regmap using parent syscon dmaengine: Add dma router for pl08x in LPC32XX SoC ARM: lpc32xx: Remove pl08x platform data in favor for device tree mtd: rawnand: lpx32xx: Request DMA channels using DT entries ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs i2x: pnx: Fix potential deadlock warning from del_timer_sync() call in isr .../devicetree/bindings/dma/arm-pl08x.yaml | 7 + .../bindings/dma/nxp,lpc3220-dmamux.yaml | 49 +++ .../bindings/sound/nxp,lpc3220-i2s.yaml | 73 ++++ MAINTAINERS | 20 + arch/arm/boot/dts/nxp/lpc/lpc32xx.dtsi | 53 ++- arch/arm/mach-lpc32xx/phy3250.c | 54 --- drivers/clk/Kconfig | 1 + drivers/clk/nxp/clk-lpc32xx.c | 26 +- drivers/dma/Kconfig | 9 + drivers/dma/Makefile | 1 + drivers/dma/lpc32xx-dmamux.c | 195 +++++++++ drivers/i2c/busses/i2c-pnx.c | 48 +-- drivers/mtd/nand/raw/lpc32xx_mlc.c | 26 +- drivers/mtd/nand/raw/lpc32xx_slc.c | 26 +- sound/soc/fsl/Kconfig | 7 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/lpc3xxx-i2s.c | 375 ++++++++++++++++++ sound/soc/fsl/lpc3xxx-i2s.h | 79 ++++ sound/soc/fsl/lpc3xxx-pcm.c | 72 ++++ 19 files changed, 993 insertions(+), 130 deletions(-) create mode 100644 Documentation/devicetree/bindings/dma/nxp,lpc3220-dmamux.yaml create mode 100644 Documentation/devicetree/bindings/sound/nxp,lpc3220-i2s.yaml create mode 100644 drivers/dma/lpc32xx-dmamux.c create mode 100644 sound/soc/fsl/lpc3xxx-i2s.c create mode 100644 sound/soc/fsl/lpc3xxx-i2s.h create mode 100644 sound/soc/fsl/lpc3xxx-pcm.c -- 2.25.1

7 19

[PATCH AUTOSEL 5.10 1/4] ASoC: intel: fix module autoloading
by Sasha Levin 10 Sep '24

10 Sep '24

From: Liao Chen <liaochen4(a)huawei.com> [ Upstream commit ae61a3391088d29aa8605c9f2db84295ab993a49 ] Add MODULE_DEVICE_TABLE(), so modules could be properly autoloaded based on the alias from of_device_id table. Signed-off-by: Liao Chen <liaochen4(a)huawei.com> Link: https://patch.msgid.link/20240826084924.368387-2-liaochen4@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/keembay/kmb_platform.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index 291a686568c2..c7b754034d24 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -634,6 +634,7 @@ static const struct of_device_id kmb_plat_of_match[] = { { .compatible = "intel,keembay-tdm", .data = &intel_kmb_tdm_dai}, {} }; +MODULE_DEVICE_TABLE(of, kmb_plat_of_match); static int kmb_plat_dai_probe(struct platform_device *pdev) { -- 2.43.0

1 0

[PATCH AUTOSEL 5.15 1/5] ASoC: intel: fix module autoloading
by Sasha Levin 10 Sep '24

10 Sep '24

From: Liao Chen <liaochen4(a)huawei.com> [ Upstream commit ae61a3391088d29aa8605c9f2db84295ab993a49 ] Add MODULE_DEVICE_TABLE(), so modules could be properly autoloaded based on the alias from of_device_id table. Signed-off-by: Liao Chen <liaochen4(a)huawei.com> Link: https://patch.msgid.link/20240826084924.368387-2-liaochen4@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/keembay/kmb_platform.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index a6fb74ba1c42..86a4c32686e7 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -815,6 +815,7 @@ static const struct of_device_id kmb_plat_of_match[] = { { .compatible = "intel,keembay-tdm", .data = &intel_kmb_tdm_dai}, {} }; +MODULE_DEVICE_TABLE(of, kmb_plat_of_match); static int kmb_plat_dai_probe(struct platform_device *pdev) { -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 3/8] ASoC: intel: fix module autoloading
by Sasha Levin 10 Sep '24

10 Sep '24

From: Liao Chen <liaochen4(a)huawei.com> [ Upstream commit ae61a3391088d29aa8605c9f2db84295ab993a49 ] Add MODULE_DEVICE_TABLE(), so modules could be properly autoloaded based on the alias from of_device_id table. Signed-off-by: Liao Chen <liaochen4(a)huawei.com> Link: https://patch.msgid.link/20240826084924.368387-2-liaochen4@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/keembay/kmb_platform.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index b4893365d01d..d5c48bed7a25 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -817,6 +817,7 @@ static const struct of_device_id kmb_plat_of_match[] = { { .compatible = "intel,keembay-tdm", .data = &intel_kmb_tdm_dai}, {} }; +MODULE_DEVICE_TABLE(of, kmb_plat_of_match); static int kmb_plat_dai_probe(struct platform_device *pdev) { -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 2/8] ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict
by Sasha Levin 10 Sep '24

10 Sep '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 839a4ec06f75cec8fec2cc5fc14e921d0c3f7369 ] There are 2G and 4G RAM versions of the Lenovo Yoga Tab 3 X90F and it turns out that the 2G version has a DMI product name of "CHERRYVIEW D1 PLATFORM" where as the 4G version has "CHERRYVIEW C0 PLATFORM". The sys-vendor + product-version check are unique enough that the product-name check is not necessary. Drop the product-name check so that the existing DMI match for the 4G RAM version also matches the 2G RAM version. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240823074305.16873-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 5e2ec60e2954..e4c3492a0c28 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -84,7 +84,6 @@ static const struct dmi_system_id lenovo_yoga_tab3_x90[] = { /* Lenovo Yoga Tab 3 Pro YT3-X90, codec missing from DSDT */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), - DMI_MATCH(DMI_PRODUCT_NAME, "CHERRYVIEW D1 PLATFORM"), DMI_MATCH(DMI_PRODUCT_VERSION, "Blade3-10A-001"), }, }, -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 03/12] ASoC: intel: fix module autoloading
by Sasha Levin 10 Sep '24

10 Sep '24

From: Liao Chen <liaochen4(a)huawei.com> [ Upstream commit ae61a3391088d29aa8605c9f2db84295ab993a49 ] Add MODULE_DEVICE_TABLE(), so modules could be properly autoloaded based on the alias from of_device_id table. Signed-off-by: Liao Chen <liaochen4(a)huawei.com> Link: https://patch.msgid.link/20240826084924.368387-2-liaochen4@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/keembay/kmb_platform.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index 6b06b7b5ede8..ffe558ef4922 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -815,6 +815,7 @@ static const struct of_device_id kmb_plat_of_match[] = { { .compatible = "intel,keembay-tdm", .data = &intel_kmb_tdm_dai}, {} }; +MODULE_DEVICE_TABLE(of, kmb_plat_of_match); static int kmb_plat_dai_probe(struct platform_device *pdev) { -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 02/12] ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict
by Sasha Levin 10 Sep '24

10 Sep '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 839a4ec06f75cec8fec2cc5fc14e921d0c3f7369 ] There are 2G and 4G RAM versions of the Lenovo Yoga Tab 3 X90F and it turns out that the 2G version has a DMI product name of "CHERRYVIEW D1 PLATFORM" where as the 4G version has "CHERRYVIEW C0 PLATFORM". The sys-vendor + product-version check are unique enough that the product-name check is not necessary. Drop the product-name check so that the existing DMI match for the 4G RAM version also matches the 2G RAM version. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240823074305.16873-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 5e2ec60e2954..e4c3492a0c28 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -84,7 +84,6 @@ static const struct dmi_system_id lenovo_yoga_tab3_x90[] = { /* Lenovo Yoga Tab 3 Pro YT3-X90, codec missing from DSDT */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), - DMI_MATCH(DMI_PRODUCT_NAME, "CHERRYVIEW D1 PLATFORM"), DMI_MATCH(DMI_PRODUCT_VERSION, "Blade3-10A-001"), }, }, -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 04/18] ASoC: intel: fix module autoloading
by Sasha Levin 10 Sep '24

10 Sep '24

From: Liao Chen <liaochen4(a)huawei.com> [ Upstream commit ae61a3391088d29aa8605c9f2db84295ab993a49 ] Add MODULE_DEVICE_TABLE(), so modules could be properly autoloaded based on the alias from of_device_id table. Signed-off-by: Liao Chen <liaochen4(a)huawei.com> Link: https://patch.msgid.link/20240826084924.368387-2-liaochen4@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/keembay/kmb_platform.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index 37ea2e1d2e92..aa5de167e790 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -814,6 +814,7 @@ static const struct of_device_id kmb_plat_of_match[] = { { .compatible = "intel,keembay-tdm", .data = &intel_kmb_tdm_dai}, {} }; +MODULE_DEVICE_TABLE(of, kmb_plat_of_match); static int kmb_plat_dai_probe(struct platform_device *pdev) { -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 03/18] ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict
by Sasha Levin 10 Sep '24

10 Sep '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 839a4ec06f75cec8fec2cc5fc14e921d0c3f7369 ] There are 2G and 4G RAM versions of the Lenovo Yoga Tab 3 X90F and it turns out that the 2G version has a DMI product name of "CHERRYVIEW D1 PLATFORM" where as the 4G version has "CHERRYVIEW C0 PLATFORM". The sys-vendor + product-version check are unique enough that the product-name check is not necessary. Drop the product-name check so that the existing DMI match for the 4G RAM version also matches the 2G RAM version. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240823074305.16873-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 5e2ec60e2954..e4c3492a0c28 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -84,7 +84,6 @@ static const struct dmi_system_id lenovo_yoga_tab3_x90[] = { /* Lenovo Yoga Tab 3 Pro YT3-X90, codec missing from DSDT */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), - DMI_MATCH(DMI_PRODUCT_NAME, "CHERRYVIEW D1 PLATFORM"), DMI_MATCH(DMI_PRODUCT_VERSION, "Blade3-10A-001"), }, }, -- 2.43.0

1 0

[PATCH 0/2] Adjust Stream Name and DT Bindings Updates
by Andrei Simion 10 Sep '24

10 Sep '24

This patch set proposes the following changes to improve the flexibility and configurability of the mchp-i2s-mcc driver by allowing the interface name to be set through the device tree and by introducing a new property to better manage multiple interfaces. Codrin Ciubotariu (2): ASoC: atmel: mchp-i2s-mcc: Remove interface name from stream_name ASoC: dt-bindings: microchip,sama7g5-i2smcc: Add 'sound-name-prefix' property .../bindings/sound/microchip,sama7g5-i2smcc.yaml | 7 +++++++ sound/soc/atmel/mchp-i2s-mcc.c | 4 ++-- 2 files changed, 9 insertions(+), 2 deletions(-) base-commit: 9aaeb87ce1e966169a57f53a02ba05b30880ffb8 -- 2.34.1

4 8

[PATCH] ASoC: Switch back to struct platform_driver::remove()
by Uwe Kleine-König 09 Sep '24

09 Sep '24

After commit 0edb555a65d1 ("platform: Make platform_driver::remove() return void") .remove() is (again) the right callback to implement for platform drivers. Convert all drivers below sound/soc to use .remove(), with the eventual goal to drop struct platform_driver::remove_new(). As .remove() and .remove_new() have the same prototypes, conversion is done by just changing the structure member name in the driver initializer. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)baylibre.com> --- Hello, this is a single patch for all of sound/soc. If you prefer, I can split it. If so, just tell how (per subdir? per driver? somehow else?) Also I trimmed the list of recipents compared to what scripts/get_maintainer.sh emits. I dropped all the maintainers for the individual drivers and only kept the list. Otherwise the set of recipents is too huge. Best regards Uwe sound/soc/adi/axi-i2s.c | 2 +- sound/soc/adi/axi-spdif.c | 2 +- sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/amd/acp/acp-rembrandt.c | 2 +- sound/soc/amd/acp/acp-renoir.c | 2 +- sound/soc/amd/acp/acp-sdw-sof-mach.c | 2 +- sound/soc/amd/acp/acp63.c | 2 +- sound/soc/amd/acp/acp70.c | 2 +- sound/soc/amd/ps/ps-pdm-dma.c | 2 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 +- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 2 +- sound/soc/amd/yc/acp6x-pdm-dma.c | 2 +- sound/soc/apple/mca.c | 2 +- sound/soc/atmel/atmel-i2s.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 2 +- sound/soc/atmel/mchp-pdmc.c | 2 +- sound/soc/atmel/mchp-spdifrx.c | 2 +- sound/soc/atmel/mchp-spdiftx.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/atmel/tse850-pcm5142.c | 2 +- sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/i2sc.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/au1x/psc-i2s.c | 2 +- sound/soc/bcm/bcm63xx-i2s-whistler.c | 2 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-i2s.c | 2 +- sound/soc/codecs/cs42l43.c | 2 +- sound/soc/codecs/cs47l15.c | 2 +- sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/cs47l35.c | 2 +- sound/soc/codecs/cs47l85.c | 2 +- sound/soc/codecs/cs47l90.c | 2 +- sound/soc/codecs/cs47l92.c | 2 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/lpass-rx-macro.c | 2 +- sound/soc/codecs/lpass-tx-macro.c | 2 +- sound/soc/codecs/lpass-va-macro.c | 2 +- sound/soc/codecs/lpass-wsa-macro.c | 2 +- sound/soc/codecs/msm8916-wcd-digital.c | 2 +- sound/soc/codecs/rk817_codec.c | 2 +- sound/soc/codecs/wcd937x.c | 2 +- sound/soc/codecs/wcd938x.c | 2 +- sound/soc/codecs/wcd939x.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- sound/soc/codecs/wm8998.c | 2 +- sound/soc/dwc/dwc-i2s.c | 2 +- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_aud2htx.c | 2 +- sound/soc/fsl/fsl_audmix.c | 2 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_micfil.c | 2 +- sound/soc/fsl/fsl_mqs.c | 2 +- sound/soc/fsl/fsl_rpmsg.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 2 +- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-pcm-rpmsg.c | 2 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/mpc5200_psc_ac97.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/fsl/p1022_rdk.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 2 +- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/audio-graph-card2-custom-sample.c | 2 +- sound/soc/generic/audio-graph-card2.c | 2 +- sound/soc/generic/simple-card.c | 2 +- sound/soc/generic/test-component.c | 2 +- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/img/img-parallel-out.c | 2 +- sound/soc/img/img-spdif-in.c | 2 +- sound/soc/img/img-spdif-out.c | 2 +- sound/soc/img/pistachio-internal-dac.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/sof_es8336.c | 2 +- sound/soc/intel/boards/sof_pcm512x.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 2 +- sound/soc/intel/catpt/device.c | 2 +- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/mediatek/common/mtk-btcvsd.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 2 +- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 2 +- sound/soc/mediatek/mt8365/mt8365-afe-pcm.c | 2 +- sound/soc/meson/aiu.c | 2 +- sound/soc/meson/axg-card.c | 2 +- sound/soc/meson/gx-card.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/pxa2xx-ac97.c | 2 +- sound/soc/qcom/lpass-apq8016.c | 2 +- sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/qcom/lpass-sc7180.c | 2 +- sound/soc/qcom/lpass-sc7280.c | 2 +- sound/soc/qcom/qdsp6/q6routing.c | 2 +- sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_rt5645.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/samsung/arndale.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/odroid.c | 2 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/spdif.c | 2 +- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/rz-ssi.c | 2 +- sound/soc/sh/siu_dai.c | 2 +- sound/soc/sof/imx/imx8.c | 2 +- sound/soc/sof/imx/imx8m.c | 2 +- sound/soc/sof/imx/imx8ulp.c | 2 +- sound/soc/sof/intel/bdw.c | 2 +- sound/soc/sof/intel/byt.c | 2 +- sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- sound/soc/sprd/sprd-mcdt.c | 2 +- sound/soc/starfive/jh7110_pwmdac.c | 2 +- sound/soc/starfive/jh7110_tdm.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/stm/stm32_i2s.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/stm/stm32_spdifrx.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 2 +- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/sunxi/sun50i-dmic.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 2 +- sound/soc/tegra/tegra186_asrc.c | 2 +- sound/soc/tegra/tegra186_dspk.c | 2 +- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra210_admaif.c | 2 +- sound/soc/tegra/tegra210_adx.c | 2 +- sound/soc/tegra/tegra210_ahub.c | 2 +- sound/soc/tegra/tegra210_amx.c | 2 +- sound/soc/tegra/tegra210_dmic.c | 2 +- sound/soc/tegra/tegra210_i2s.c | 2 +- sound/soc/tegra/tegra210_mixer.c | 2 +- sound/soc/tegra/tegra210_mvc.c | 2 +- sound/soc/tegra/tegra210_ope.c | 2 +- sound/soc/tegra/tegra210_sfc.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 2 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/tegra/tegra_audio_graph_card.c | 2 +- sound/soc/ti/ams-delta.c | 2 +- sound/soc/ti/davinci-i2s.c | 2 +- sound/soc/ti/davinci-mcasp.c | 2 +- sound/soc/ti/omap-mcbsp.c | 2 +- sound/soc/uniphier/aio-ld11.c | 2 +- sound/soc/uniphier/aio-pxs2.c | 2 +- sound/soc/uniphier/evea.c | 2 +- sound/soc/ux500/mop500.c | 2 +- sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 +- sound/soc/xilinx/xlnx_spdif.c | 2 +- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 184 files changed, 184 insertions(+), 184 deletions(-) diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 4107eddc9ca8..41f89384f8fd 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -293,7 +293,7 @@ static struct platform_driver axi_i2s_driver = { .of_match_table = axi_i2s_of_match, }, .probe = axi_i2s_probe, - .remove_new = axi_i2s_dev_remove, + .remove = axi_i2s_dev_remove, }; module_platform_driver(axi_i2s_driver); diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c index 10545bd99704..5581134201a3 100644 --- a/sound/soc/adi/axi-spdif.c +++ b/sound/soc/adi/axi-spdif.c @@ -258,7 +258,7 @@ static struct platform_driver axi_spdif_driver = { .of_match_table = axi_spdif_of_match, }, .probe = axi_spdif_probe, - .remove_new = axi_spdif_dev_remove, + .remove = axi_spdif_dev_remove, }; module_platform_driver(axi_spdif_driver); diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index b857e2676fe8..897dde630022 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1426,7 +1426,7 @@ static const struct dev_pm_ops acp_pm_ops = { static struct platform_driver acp_dma_driver = { .probe = acp_audio_probe, - .remove_new = acp_audio_remove, + .remove = acp_audio_remove, .driver = { .name = DRV_NAME, .pm = &acp_pm_ops, diff --git a/sound/soc/amd/acp/acp-rembrandt.c b/sound/soc/amd/acp/acp-rembrandt.c index e19981c7d65a..396434a45eea 100644 --- a/sound/soc/amd/acp/acp-rembrandt.c +++ b/sound/soc/amd/acp/acp-rembrandt.c @@ -295,7 +295,7 @@ static const struct dev_pm_ops rmb_dma_pm_ops = { static struct platform_driver rembrandt_driver = { .probe = rembrandt_audio_probe, - .remove_new = rembrandt_audio_remove, + .remove = rembrandt_audio_remove, .driver = { .name = "acp_asoc_rembrandt", .pm = &rmb_dma_pm_ops, diff --git a/sound/soc/amd/acp/acp-renoir.c b/sound/soc/amd/acp/acp-renoir.c index db835ed7c208..5e3f730aa6bf 100644 --- a/sound/soc/amd/acp/acp-renoir.c +++ b/sound/soc/amd/acp/acp-renoir.c @@ -244,7 +244,7 @@ static const struct dev_pm_ops rn_dma_pm_ops = { static struct platform_driver renoir_driver = { .probe = renoir_audio_probe, - .remove_new = renoir_audio_remove, + .remove = renoir_audio_remove, .driver = { .name = "acp_asoc_renoir", .pm = &rn_dma_pm_ops, diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index 08f368b3bbc8..b1cd173f607d 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -729,7 +729,7 @@ static struct platform_driver sof_sdw_driver = { .pm = &snd_soc_pm_ops, }, .probe = mc_probe, - .remove_new = mc_remove, + .remove = mc_remove, .id_table = mc_id_table, }; diff --git a/sound/soc/amd/acp/acp63.c b/sound/soc/amd/acp/acp63.c index f340920b3289..f325c374f228 100644 --- a/sound/soc/amd/acp/acp63.c +++ b/sound/soc/amd/acp/acp63.c @@ -304,7 +304,7 @@ static const struct dev_pm_ops acp63_dma_pm_ops = { static struct platform_driver acp63_driver = { .probe = acp63_audio_probe, - .remove_new = acp63_audio_remove, + .remove = acp63_audio_remove, .driver = { .name = "acp_asoc_acp63", .pm = &acp63_dma_pm_ops, diff --git a/sound/soc/amd/acp/acp70.c b/sound/soc/amd/acp/acp70.c index 1b0b59a22924..68d2590e1a4e 100644 --- a/sound/soc/amd/acp/acp70.c +++ b/sound/soc/amd/acp/acp70.c @@ -278,7 +278,7 @@ static const struct dev_pm_ops acp70_dma_pm_ops = { static struct platform_driver acp70_driver = { .probe = acp_acp70_audio_probe, - .remove_new = acp_acp70_audio_remove, + .remove = acp_acp70_audio_remove, .driver = { .name = "acp_asoc_acp70", .pm = &acp70_dma_pm_ops, diff --git a/sound/soc/amd/ps/ps-pdm-dma.c b/sound/soc/amd/ps/ps-pdm-dma.c index 7bbacbab1095..318fc260f293 100644 --- a/sound/soc/amd/ps/ps-pdm-dma.c +++ b/sound/soc/amd/ps/ps-pdm-dma.c @@ -448,7 +448,7 @@ static const struct dev_pm_ops acp63_pdm_pm_ops = { static struct platform_driver acp63_pdm_dma_driver = { .probe = acp63_pdm_audio_probe, - .remove_new = acp63_pdm_audio_remove, + .remove = acp63_pdm_audio_remove, .driver = { .name = "acp_ps_pdm_dma", .pm = &acp63_pdm_pm_ops, diff --git a/sound/soc/amd/ps/ps-sdw-dma.c b/sound/soc/amd/ps/ps-sdw-dma.c index 2f630753278d..3b4b9c6b3171 100644 --- a/sound/soc/amd/ps/ps-sdw-dma.c +++ b/sound/soc/amd/ps/ps-sdw-dma.c @@ -551,7 +551,7 @@ static const struct dev_pm_ops acp63_pm_ops = { static struct platform_driver acp63_sdw_dma_driver = { .probe = acp63_sdw_platform_probe, - .remove_new = acp63_sdw_platform_remove, + .remove = acp63_sdw_platform_remove, .driver = { .name = "amd_ps_sdw_dma", .pm = &acp63_pm_ops, diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 3a50558f6751..bb9ed52d744d 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -509,7 +509,7 @@ static const struct dev_pm_ops acp3x_pm_ops = { static struct platform_driver acp3x_dma_driver = { .probe = acp3x_audio_probe, - .remove_new = acp3x_audio_remove, + .remove = acp3x_audio_remove, .driver = { .name = "acp3x_rv_i2s_dma", .pm = &acp3x_pm_ops, diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index c3b47e9bd239..95ac8c680037 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -489,7 +489,7 @@ static const struct dev_pm_ops acp_pdm_pm_ops = { static struct platform_driver acp_pdm_dma_driver = { .probe = acp_pdm_audio_probe, - .remove_new = acp_pdm_audio_remove, + .remove = acp_pdm_audio_remove, .driver = { .name = "acp_rn_pdm_dma", .pm = &acp_pdm_pm_ops, diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index 491b16e52a72..d5965f2b09bc 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -499,7 +499,7 @@ static const struct dev_pm_ops acp5x_pm_ops = { static struct platform_driver acp5x_dma_driver = { .probe = acp5x_audio_probe, - .remove_new = acp5x_audio_remove, + .remove = acp5x_audio_remove, .driver = { .name = "acp5x_i2s_dma", .pm = &acp5x_pm_ops, diff --git a/sound/soc/amd/yc/acp6x-pdm-dma.c b/sound/soc/amd/yc/acp6x-pdm-dma.c index 72c4591e451b..3eb3e82efb10 100644 --- a/sound/soc/amd/yc/acp6x-pdm-dma.c +++ b/sound/soc/amd/yc/acp6x-pdm-dma.c @@ -440,7 +440,7 @@ static const struct dev_pm_ops acp6x_pdm_pm_ops = { static struct platform_driver acp6x_pdm_dma_driver = { .probe = acp6x_pdm_audio_probe, - .remove_new = acp6x_pdm_audio_remove, + .remove = acp6x_pdm_audio_remove, .driver = { .name = "acp_yc_pdm_dma", .pm = &acp6x_pdm_pm_ops, diff --git a/sound/soc/apple/mca.c b/sound/soc/apple/mca.c index 3780aca71076..c9e7d40c47cc 100644 --- a/sound/soc/apple/mca.c +++ b/sound/soc/apple/mca.c @@ -1179,7 +1179,7 @@ static struct platform_driver apple_mca_driver = { .of_match_table = apple_mca_of_match, }, .probe = apple_mca_probe, - .remove_new = apple_mca_remove, + .remove = apple_mca_remove, }; module_platform_driver(apple_mca_driver); diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 6c20c643f321..762199faf872 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -733,7 +733,7 @@ static struct platform_driver atmel_i2s_driver = { .of_match_table = atmel_i2s_dt_ids, }, .probe = atmel_i2s_probe, - .remove_new = atmel_i2s_remove, + .remove = atmel_i2s_remove, }; module_platform_driver(atmel_i2s_driver); diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index b7f16ea0cdfc..0f4021c6c588 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -187,7 +187,7 @@ static struct platform_driver atmel_asoc_wm8904_driver = { .pm = &snd_soc_pm_ops, }, .probe = atmel_asoc_wm8904_probe, - .remove_new = atmel_asoc_wm8904_remove, + .remove = atmel_asoc_wm8904_remove, }; module_platform_driver(atmel_asoc_wm8904_driver); diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 35a56b266b8d..fbcb0070c188 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -1129,7 +1129,7 @@ static struct platform_driver mchp_i2s_mcc_driver = { .of_match_table = mchp_i2s_mcc_dt_ids, }, .probe = mchp_i2s_mcc_probe, - .remove_new = mchp_i2s_mcc_remove, + .remove = mchp_i2s_mcc_remove, }; module_platform_driver(mchp_i2s_mcc_driver); diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index dcc4e14b3dde..260074018da9 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -1153,7 +1153,7 @@ static struct platform_driver mchp_pdmc_driver = { .pm = pm_ptr(&mchp_pdmc_pm_ops), }, .probe = mchp_pdmc_probe, - .remove_new = mchp_pdmc_remove, + .remove = mchp_pdmc_remove, }; module_platform_driver(mchp_pdmc_driver); diff --git a/sound/soc/atmel/mchp-spdifrx.c b/sound/soc/atmel/mchp-spdifrx.c index 33ce5e54482b..b2507a1491b7 100644 --- a/sound/soc/atmel/mchp-spdifrx.c +++ b/sound/soc/atmel/mchp-spdifrx.c @@ -1194,7 +1194,7 @@ static void mchp_spdifrx_remove(struct platform_device *pdev) static struct platform_driver mchp_spdifrx_driver = { .probe = mchp_spdifrx_probe, - .remove_new = mchp_spdifrx_remove, + .remove = mchp_spdifrx_remove, .driver = { .name = "mchp_spdifrx", .of_match_table = mchp_spdifrx_dt_ids, diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index a201a96fa690..4c60ea652896 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -888,7 +888,7 @@ static void mchp_spdiftx_remove(struct platform_device *pdev) static struct platform_driver mchp_spdiftx_driver = { .probe = mchp_spdiftx_probe, - .remove_new = mchp_spdiftx_remove, + .remove = mchp_spdiftx_remove, .driver = { .name = "mchp_spdiftx", .of_match_table = mchp_spdiftx_dt_ids, diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index d3ec9826d505..335e216ea7b4 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -207,7 +207,7 @@ static struct platform_driver at91sam9g20ek_audio_driver = { .of_match_table = of_match_ptr(at91sam9g20ek_wm8731_dt_ids), }, .probe = at91sam9g20ek_audio_probe, - .remove_new = at91sam9g20ek_audio_remove, + .remove = at91sam9g20ek_audio_remove, }; module_platform_driver(at91sam9g20ek_audio_driver); diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index d1c1f370a9cd..1b5ef4e9d2b8 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -196,7 +196,7 @@ static struct platform_driver sam9x5_wm8731_driver = { .of_match_table = of_match_ptr(sam9x5_wm8731_of_match), }, .probe = sam9x5_wm8731_driver_probe, - .remove_new = sam9x5_wm8731_driver_remove, + .remove = sam9x5_wm8731_driver_remove, }; module_platform_driver(sam9x5_wm8731_driver); diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 5d208e0b4b90..0a9efd5f2861 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -431,7 +431,7 @@ static struct platform_driver tse850_driver = { .of_match_table = tse850_dt_ids, }, .probe = tse850_probe, - .remove_new = tse850_remove, + .remove = tse850_remove, }; module_platform_driver(tse850_driver); diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index b0e1a1253e10..f8ab936250dc 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -336,7 +336,7 @@ static struct platform_driver au1xac97c_driver = { .pm = AU1XPSCAC97_PMOPS, }, .probe = au1xac97c_drvprobe, - .remove_new = au1xac97c_drvremove, + .remove = au1xac97c_drvremove, }; module_platform_driver(au1xac97c_driver); diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 064406080d72..7d296f29dade 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -313,7 +313,7 @@ static struct platform_driver au1xi2s_driver = { .pm = AU1XI2SC_PMOPS, }, .probe = au1xi2s_drvprobe, - .remove_new = au1xi2s_drvremove, + .remove = au1xi2s_drvremove, }; module_platform_driver(au1xi2s_driver); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 1727eeb12b64..8a59a50978b9 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -486,7 +486,7 @@ static struct platform_driver au1xpsc_ac97_driver = { .pm = AU1XPSCAC97_PMOPS, }, .probe = au1xpsc_ac97_drvprobe, - .remove_new = au1xpsc_ac97_drvremove, + .remove = au1xpsc_ac97_drvremove, }; module_platform_driver(au1xpsc_ac97_driver); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 52734dec8247..bee013555e7a 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -404,7 +404,7 @@ static struct platform_driver au1xpsc_i2s_driver = { .pm = AU1XPSCI2S_PMOPS, }, .probe = au1xpsc_i2s_drvprobe, - .remove_new = au1xpsc_i2s_drvremove, + .remove = au1xpsc_i2s_drvremove, }; module_platform_driver(au1xpsc_i2s_driver); diff --git a/sound/soc/bcm/bcm63xx-i2s-whistler.c b/sound/soc/bcm/bcm63xx-i2s-whistler.c index c64609718738..c47ed1e6ea2b 100644 --- a/sound/soc/bcm/bcm63xx-i2s-whistler.c +++ b/sound/soc/bcm/bcm63xx-i2s-whistler.c @@ -293,7 +293,7 @@ static struct platform_driver bcm63xx_i2s_driver = { .of_match_table = of_match_ptr(snd_soc_bcm_audio_match), }, .probe = bcm63xx_i2s_dev_probe, - .remove_new = bcm63xx_i2s_dev_remove, + .remove = bcm63xx_i2s_dev_remove, }; module_platform_driver(bcm63xx_i2s_driver); diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index 90088516fed0..e0ce0232eb1e 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1390,7 +1390,7 @@ MODULE_DEVICE_TABLE(of, cygnus_ssp_of_match); static struct platform_driver cygnus_ssp_driver = { .probe = cygnus_ssp_probe, - .remove_new = cygnus_ssp_remove, + .remove = cygnus_ssp_remove, .driver = { .name = "cygnus-ssp", .of_match_table = cygnus_ssp_of_match, diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c index 8bb67d7d2b4b..8dac754ddb0d 100644 --- a/sound/soc/cirrus/edb93xx.c +++ b/sound/soc/cirrus/edb93xx.c @@ -105,7 +105,7 @@ static struct platform_driver edb93xx_driver = { .name = "edb93xx-audio", }, .probe = edb93xx_probe, - .remove_new = edb93xx_remove, + .remove = edb93xx_remove, }; module_platform_driver(edb93xx_driver); diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 522de4b80293..d45862ceb0c9 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -523,7 +523,7 @@ MODULE_DEVICE_TABLE(of, ep93xx_i2s_of_ids); static struct platform_driver ep93xx_i2s_driver = { .probe = ep93xx_i2s_probe, - .remove_new = ep93xx_i2s_remove, + .remove = ep93xx_i2s_remove, .driver = { .name = "ep93xx-i2s", .of_match_table = ep93xx_i2s_of_ids, diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 5183b4586424..d0098b4558b5 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2461,7 +2461,7 @@ static struct platform_driver cs42l43_codec_driver = { }, .probe = cs42l43_codec_probe, - .remove_new = cs42l43_codec_remove, + .remove = cs42l43_codec_remove, .id_table = cs42l43_codec_id_table, }; module_platform_driver(cs42l43_codec_driver); diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c index ab6e7cd99733..29a2bcfb3048 100644 --- a/sound/soc/codecs/cs47l15.c +++ b/sound/soc/codecs/cs47l15.c @@ -1493,7 +1493,7 @@ static struct platform_driver cs47l15_codec_driver = { .name = "cs47l15-codec", }, .probe = &cs47l15_probe, - .remove_new = cs47l15_remove, + .remove = cs47l15_remove, }; module_platform_driver(cs47l15_codec_driver); diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index ec405ef66a8e..e2a839fae4fc 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1344,7 +1344,7 @@ static struct platform_driver cs47l24_codec_driver = { .name = "cs47l24-codec", }, .probe = cs47l24_probe, - .remove_new = cs47l24_remove, + .remove = cs47l24_remove, }; module_platform_driver(cs47l24_codec_driver); diff --git a/sound/soc/codecs/cs47l35.c b/sound/soc/codecs/cs47l35.c index 0d7ee7ea6257..85555c7a2e4b 100644 --- a/sound/soc/codecs/cs47l35.c +++ b/sound/soc/codecs/cs47l35.c @@ -1769,7 +1769,7 @@ static struct platform_driver cs47l35_codec_driver = { .name = "cs47l35-codec", }, .probe = &cs47l35_probe, - .remove_new = cs47l35_remove, + .remove = cs47l35_remove, }; module_platform_driver(cs47l35_codec_driver); diff --git a/sound/soc/codecs/cs47l85.c b/sound/soc/codecs/cs47l85.c index 2dfb867e6edd..d34f4e8c26d3 100644 --- a/sound/soc/codecs/cs47l85.c +++ b/sound/soc/codecs/cs47l85.c @@ -2720,7 +2720,7 @@ static struct platform_driver cs47l85_codec_driver = { .name = "cs47l85-codec", }, .probe = &cs47l85_probe, - .remove_new = cs47l85_remove, + .remove = cs47l85_remove, }; module_platform_driver(cs47l85_codec_driver); diff --git a/sound/soc/codecs/cs47l90.c b/sound/soc/codecs/cs47l90.c index 2549cb1fc121..a9e703981f37 100644 --- a/sound/soc/codecs/cs47l90.c +++ b/sound/soc/codecs/cs47l90.c @@ -2644,7 +2644,7 @@ static struct platform_driver cs47l90_codec_driver = { .name = "cs47l90-codec", }, .probe = &cs47l90_probe, - .remove_new = cs47l90_remove, + .remove = cs47l90_remove, }; module_platform_driver(cs47l90_codec_driver); diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c index 0c05ae0b09fb..2c355c61acd8 100644 --- a/sound/soc/codecs/cs47l92.c +++ b/sound/soc/codecs/cs47l92.c @@ -2092,7 +2092,7 @@ static struct platform_driver cs47l92_codec_driver = { .name = "cs47l92-codec", }, .probe = &cs47l92_probe, - .remove_new = cs47l92_remove, + .remove = cs47l92_remove, }; module_platform_driver(cs47l92_codec_driver); diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c index 11320423c69c..fdd19f8e8864 100644 --- a/sound/soc/codecs/inno_rk3036.c +++ b/sound/soc/codecs/inno_rk3036.c @@ -476,7 +476,7 @@ static struct platform_driver rk3036_codec_platform_driver = { .of_match_table = of_match_ptr(rk3036_codec_of_match), }, .probe = rk3036_codec_platform_probe, - .remove_new = rk3036_codec_platform_remove, + .remove = rk3036_codec_platform_remove, }; module_platform_driver(rk3036_codec_platform_driver); diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index ce42749660c8..71e0d3bffd3f 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -4024,7 +4024,7 @@ static struct platform_driver rx_macro_driver = { .pm = &rx_macro_pm_ops, }, .probe = rx_macro_probe, - .remove_new = rx_macro_remove, + .remove = rx_macro_remove, }; module_platform_driver(rx_macro_driver); diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 209c12ec16dd..a134584acf90 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -2534,7 +2534,7 @@ static struct platform_driver tx_macro_driver = { .pm = &tx_macro_pm_ops, }, .probe = tx_macro_probe, - .remove_new = tx_macro_remove, + .remove = tx_macro_remove, }; module_platform_driver(tx_macro_driver); diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index e95d1f29ef18..c781da476240 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1738,7 +1738,7 @@ static struct platform_driver va_macro_driver = { .pm = &va_macro_pm_ops, }, .probe = va_macro_probe, - .remove_new = va_macro_remove, + .remove = va_macro_remove, }; module_platform_driver(va_macro_driver); diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index 76900274acf3..c989d82d1d3c 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -2980,7 +2980,7 @@ static struct platform_driver wsa_macro_driver = { .pm = &wsa_macro_pm_ops, }, .probe = wsa_macro_probe, - .remove_new = wsa_macro_remove, + .remove = wsa_macro_remove, }; module_platform_driver(wsa_macro_driver); diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 978c4d056e81..ebb6f2e84818 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -1241,7 +1241,7 @@ static struct platform_driver msm8916_wcd_digital_driver = { .of_match_table = msm8916_wcd_digital_match_table, }, .probe = msm8916_wcd_digital_probe, - .remove_new = msm8916_wcd_digital_remove, + .remove = msm8916_wcd_digital_remove, }; module_platform_driver(msm8916_wcd_digital_driver); diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 5fea600bc3a4..3c5b66357661 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -529,7 +529,7 @@ static struct platform_driver rk817_codec_driver = { .name = "rk817-codec", }, .probe = rk817_platform_probe, - .remove_new = rk817_platform_remove, + .remove = rk817_platform_remove, }; module_platform_driver(rk817_codec_driver); diff --git a/sound/soc/codecs/wcd937x.c b/sound/soc/codecs/wcd937x.c index af296b77a723..45f32d281908 100644 --- a/sound/soc/codecs/wcd937x.c +++ b/sound/soc/codecs/wcd937x.c @@ -2957,7 +2957,7 @@ MODULE_DEVICE_TABLE(of, wcd937x_of_match); static struct platform_driver wcd937x_codec_driver = { .probe = wcd937x_probe, - .remove_new = wcd937x_remove, + .remove = wcd937x_remove, .driver = { .name = "wcd937x_codec", .of_match_table = of_match_ptr(wcd937x_of_match), diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index aa92f1bfc921..f2a4f3262bdb 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3596,7 +3596,7 @@ MODULE_DEVICE_TABLE(of, wcd938x_dt_match); static struct platform_driver wcd938x_codec_driver = { .probe = wcd938x_probe, - .remove_new = wcd938x_remove, + .remove = wcd938x_remove, .driver = { .name = "wcd938x_codec", .of_match_table = of_match_ptr(wcd938x_dt_match), diff --git a/sound/soc/codecs/wcd939x.c b/sound/soc/codecs/wcd939x.c index 68fc591670dc..4a417a92514d 100644 --- a/sound/soc/codecs/wcd939x.c +++ b/sound/soc/codecs/wcd939x.c @@ -3683,7 +3683,7 @@ MODULE_DEVICE_TABLE(of, wcd939x_dt_match); static struct platform_driver wcd939x_codec_driver = { .probe = wcd939x_probe, - .remove_new = wcd939x_remove, + .remove = wcd939x_remove, .driver = { .name = "wcd939x_codec", .of_match_table = of_match_ptr(wcd939x_dt_match), diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 4ecf07c7448c..651f1319204d 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2174,7 +2174,7 @@ static struct platform_driver wm5102_codec_driver = { .name = "wm5102-codec", }, .probe = wm5102_probe, - .remove_new = wm5102_remove, + .remove = wm5102_remove, }; module_platform_driver(wm5102_codec_driver); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 0f299cd07b2e..502196253d42 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2534,7 +2534,7 @@ static struct platform_driver wm5110_codec_driver = { .name = "wm5110-codec", }, .probe = wm5110_probe, - .remove_new = wm5110_remove, + .remove = wm5110_remove, }; module_platform_driver(wm5110_codec_driver); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a99908582a50..a4abe6e53bfc 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4699,7 +4699,7 @@ static struct platform_driver wm8994_codec_driver = { .pm = &wm8994_pm_ops, }, .probe = wm8994_probe, - .remove_new = wm8994_remove, + .remove = wm8994_remove, }; module_platform_driver(wm8994_codec_driver); diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 87442840f0af..5389c363b14e 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1210,7 +1210,7 @@ static struct platform_driver wm8997_codec_driver = { .name = "wm8997-codec", }, .probe = wm8997_probe, - .remove_new = wm8997_remove, + .remove = wm8997_remove, }; module_platform_driver(wm8997_codec_driver); diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 3c2c4d12c08e..b72b8a64be8f 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1426,7 +1426,7 @@ static struct platform_driver wm8998_codec_driver = { .name = "wm8998-codec", }, .probe = wm8998_probe, - .remove_new = wm8998_remove, + .remove = wm8998_remove, }; module_platform_driver(wm8998_codec_driver); diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index e6a5eebe5bc9..57b789d7fbed 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -1089,7 +1089,7 @@ static const struct dev_pm_ops dwc_pm_ops = { static struct platform_driver dw_i2s_driver = { .probe = dw_i2s_probe, - .remove_new = dw_i2s_remove, + .remove = dw_i2s_remove, .driver = { .name = "designware-i2s", .of_match_table = of_match_ptr(dw_i2s_of_match), diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index b793263291dc..bd5c46d763c0 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -1392,7 +1392,7 @@ MODULE_DEVICE_TABLE(of, fsl_asrc_ids); static struct platform_driver fsl_asrc_driver = { .probe = fsl_asrc_probe, - .remove_new = fsl_asrc_remove, + .remove = fsl_asrc_remove, .driver = { .name = "fsl-asrc", .of_match_table = fsl_asrc_ids, diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index a6cbaa6364c7..021d73e409aa 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -296,7 +296,7 @@ static const struct dev_pm_ops fsl_aud2htx_pm_ops = { static struct platform_driver fsl_aud2htx_driver = { .probe = fsl_aud2htx_probe, - .remove_new = fsl_aud2htx_remove, + .remove = fsl_aud2htx_remove, .driver = { .name = "fsl-aud2htx", .pm = pm_ptr(&fsl_aud2htx_pm_ops), diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index f3a24758aedb..3cd9a66b70a1 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -548,7 +548,7 @@ static const struct dev_pm_ops fsl_audmix_pm = { static struct platform_driver fsl_audmix_driver = { .probe = fsl_audmix_probe, - .remove_new = fsl_audmix_remove, + .remove = fsl_audmix_remove, .driver = { .name = "fsl-audmix", .of_match_table = fsl_audmix_ids, diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index c4bc9395dff7..aca066b5a43c 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -911,7 +911,7 @@ static struct platform_driver fsl_soc_dma_driver = { .of_match_table = fsl_soc_dma_ids, }, .probe = fsl_soc_dma_probe, - .remove_new = fsl_soc_dma_remove, + .remove = fsl_soc_dma_remove, }; module_platform_driver(fsl_soc_dma_driver); diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 962f30912091..82359edd6a8b 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -2093,7 +2093,7 @@ static const struct dev_pm_ops fsl_easrc_pm_ops = { static struct platform_driver fsl_easrc_driver = { .probe = fsl_easrc_probe, - .remove_new = fsl_easrc_remove, + .remove = fsl_easrc_remove, .driver = { .name = "fsl-easrc", .pm = pm_ptr(&fsl_easrc_pm_ops), diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d0d8a01da9bd..a65f5b9935a2 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -1198,7 +1198,7 @@ static const struct dev_pm_ops fsl_esai_pm_ops = { static struct platform_driver fsl_esai_driver = { .probe = fsl_esai_probe, - .remove_new = fsl_esai_remove, + .remove = fsl_esai_remove, .driver = { .name = "fsl-esai-dai", .pm = &fsl_esai_pm_ops, diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 22b240a70ad4..193be098fa5e 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -1317,7 +1317,7 @@ static const struct dev_pm_ops fsl_micfil_pm_ops = { static struct platform_driver fsl_micfil_driver = { .probe = fsl_micfil_probe, - .remove_new = fsl_micfil_remove, + .remove = fsl_micfil_remove, .driver = { .name = "fsl-micfil-dai", .pm = &fsl_micfil_pm_ops, diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index df160834c81b..145f9ca15e43 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -381,7 +381,7 @@ MODULE_DEVICE_TABLE(of, fsl_mqs_dt_ids); static struct platform_driver fsl_mqs_driver = { .probe = fsl_mqs_probe, - .remove_new = fsl_mqs_remove, + .remove = fsl_mqs_remove, .driver = { .name = "fsl-mqs", .of_match_table = fsl_mqs_dt_ids, diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index be46fbfd487a..0a551be3053b 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -328,7 +328,7 @@ static const struct dev_pm_ops fsl_rpmsg_pm_ops = { static struct platform_driver fsl_rpmsg_driver = { .probe = fsl_rpmsg_probe, - .remove_new = fsl_rpmsg_remove, + .remove = fsl_rpmsg_remove, .driver = { .name = "fsl_rpmsg", .pm = pm_ptr(&fsl_rpmsg_pm_ops), diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d03b0172b8ad..ab58a4461073 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1817,7 +1817,7 @@ static const struct dev_pm_ops fsl_sai_pm_ops = { static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, - .remove_new = fsl_sai_remove, + .remove = fsl_sai_remove, .driver = { .name = "fsl-sai", .pm = &fsl_sai_pm_ops, diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index eace399cb064..b6ff04f7138a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1763,7 +1763,7 @@ static struct platform_driver fsl_spdif_driver = { .pm = pm_ptr(&fsl_spdif_pm), }, .probe = fsl_spdif_probe, - .remove_new = fsl_spdif_remove, + .remove = fsl_spdif_remove, }; module_platform_driver(fsl_spdif_driver); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c4c1d9c44056..320108bebf30 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1734,7 +1734,7 @@ static struct platform_driver fsl_ssi_driver = { .pm = pm_sleep_ptr(&fsl_ssi_pm), }, .probe = fsl_ssi_probe, - .remove_new = fsl_ssi_remove, + .remove = fsl_ssi_remove, }; module_platform_driver(fsl_ssi_driver); diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index c98e3db74fe7..beede7344efd 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1540,7 +1540,7 @@ static struct platform_driver fsl_xcvr_driver = { .pm = pm_ptr(&fsl_xcvr_pm_ops), .of_match_table = fsl_xcvr_dt_ids, }, - .remove_new = fsl_xcvr_remove, + .remove = fsl_xcvr_remove, }; module_platform_driver(fsl_xcvr_driver); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index f97ae14dc452..43e14f2eca8d 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -354,7 +354,7 @@ static const struct dev_pm_ops imx_audmux_pm = { static struct platform_driver imx_audmux_driver = { .probe = imx_audmux_probe, - .remove_new = imx_audmux_remove, + .remove = imx_audmux_remove, .driver = { .name = DRIVER_NAME, .pm = pm_sleep_ptr(&imx_audmux_pm), diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index 22156f99dcee..1daf0be3d100 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -822,7 +822,7 @@ MODULE_DEVICE_TABLE(platform, imx_rpmsg_pcm_id_table); static struct platform_driver imx_pcm_rpmsg_driver = { .probe = imx_rpmsg_pcm_probe, - .remove_new = imx_rpmsg_pcm_remove, + .remove = imx_rpmsg_pcm_remove, .id_table = imx_rpmsg_pcm_id_table, .driver = { .name = IMX_PCM_DRV_NAME, diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 3c1e69092d2f..8bcf54ef709e 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -214,7 +214,7 @@ static struct platform_driver imx_sgtl5000_driver = { .of_match_table = imx_sgtl5000_dt_ids, }, .probe = imx_sgtl5000_probe, - .remove_new = imx_sgtl5000_remove, + .remove = imx_sgtl5000_remove, }; module_platform_driver(imx_sgtl5000_driver); diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 0423cf43c7a0..8554fb690772 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -327,7 +327,7 @@ MODULE_DEVICE_TABLE(of, psc_ac97_match); static struct platform_driver psc_ac97_driver = { .probe = psc_ac97_of_probe, - .remove_new = psc_ac97_of_remove, + .remove = psc_ac97_of_remove, .driver = { .name = "mpc5200-psc-ac97", .of_match_table = psc_ac97_match, diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 931b430fa983..9ad44eeed6ad 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -225,7 +225,7 @@ MODULE_DEVICE_TABLE(of, psc_i2s_match); static struct platform_driver psc_i2s_driver = { .probe = psc_i2s_of_probe, - .remove_new = psc_i2s_of_remove, + .remove = psc_i2s_of_remove, .driver = { .name = "mpc5200-psc-i2s", .of_match_table = psc_i2s_match, diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 6f5eecf6d88c..66db05970d82 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -408,7 +408,7 @@ static void p1022_ds_remove(struct platform_device *pdev) static struct platform_driver p1022_ds_driver = { .probe = p1022_ds_probe, - .remove_new = p1022_ds_remove, + .remove = p1022_ds_remove, .driver = { /* * The name must match 'compatible' property in the device tree, diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c index a42149311c8b..d4568346714f 100644 --- a/sound/soc/fsl/p1022_rdk.c +++ b/sound/soc/fsl/p1022_rdk.c @@ -370,7 +370,7 @@ static void p1022_rdk_remove(struct platform_device *pdev) static struct platform_driver p1022_rdk_driver = { .probe = p1022_rdk_probe, - .remove_new = p1022_rdk_remove, + .remove = p1022_rdk_remove, .driver = { /* * The name must match 'compatible' property in the device tree, diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 2bab0fc1de59..5542c4ee6d12 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -124,7 +124,7 @@ MODULE_DEVICE_TABLE(of, pcm030_audio_match); static struct platform_driver pcm030_fabric_driver = { .probe = pcm030_fabric_probe, - .remove_new = pcm030_fabric_remove, + .remove = pcm030_fabric_remove, .driver = { .name = DRV_NAME, .of_match_table = pcm030_audio_match, diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 03add80e75f3..4b384475b972 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -664,7 +664,7 @@ static struct platform_driver graph_card = { .of_match_table = graph_of_match, }, .probe = graph_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(graph_card); diff --git a/sound/soc/generic/audio-graph-card2-custom-sample.c b/sound/soc/generic/audio-graph-card2-custom-sample.c index 8e5a51098490..7151d426bee9 100644 --- a/sound/soc/generic/audio-graph-card2-custom-sample.c +++ b/sound/soc/generic/audio-graph-card2-custom-sample.c @@ -177,7 +177,7 @@ static struct platform_driver custom_card = { .of_match_table = custom_of_match, }, .probe = custom_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(custom_card); diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index bf4c1ab4ec58..4ad3d1b0714f 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -1438,7 +1438,7 @@ static struct platform_driver graph_card = { .of_match_table = graph_of_match, }, .probe = graph_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(graph_card); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 42c60b92cca5..76a1d05e2ebe 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -838,7 +838,7 @@ static struct platform_driver simple_card = { .of_match_table = simple_of_match, }, .probe = simple_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(simple_card); diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index df2487b700cc..407288055741 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -637,7 +637,7 @@ static struct platform_driver test_driver = { .of_match_table = test_of_match, }, .probe = test_driver_probe, - .remove_new = test_driver_remove, + .remove = test_driver_remove, }; module_platform_driver(test_driver); diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index b69a364d619e..6a988976fb0d 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -607,7 +607,7 @@ static struct platform_driver img_i2s_in_driver = { .pm = &img_i2s_in_pm_ops }, .probe = img_i2s_in_probe, - .remove_new = img_i2s_in_dev_remove + .remove = img_i2s_in_dev_remove }; module_platform_driver(img_i2s_in_driver); diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index 6f9831c6d6e0..1211e6184d97 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -607,7 +607,7 @@ static struct platform_driver img_i2s_out_driver = { .pm = &img_i2s_out_pm_ops }, .probe = img_i2s_out_probe, - .remove_new = img_i2s_out_dev_remove + .remove = img_i2s_out_dev_remove }; module_platform_driver(img_i2s_out_driver); diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c index 815e68a7048c..4ec63119d67c 100644 --- a/sound/soc/img/img-parallel-out.c +++ b/sound/soc/img/img-parallel-out.c @@ -311,7 +311,7 @@ static struct platform_driver img_prl_out_driver = { .pm = &img_prl_out_pm_ops }, .probe = img_prl_out_probe, - .remove_new = img_prl_out_dev_remove + .remove = img_prl_out_dev_remove }; module_platform_driver(img_prl_out_driver); diff --git a/sound/soc/img/img-spdif-in.c b/sound/soc/img/img-spdif-in.c index 9646e9d3f0bc..3c513f5b8c54 100644 --- a/sound/soc/img/img-spdif-in.c +++ b/sound/soc/img/img-spdif-in.c @@ -878,7 +878,7 @@ static struct platform_driver img_spdif_in_driver = { .pm = &img_spdif_in_pm_ops }, .probe = img_spdif_in_probe, - .remove_new = img_spdif_in_dev_remove + .remove = img_spdif_in_dev_remove }; module_platform_driver(img_spdif_in_driver); diff --git a/sound/soc/img/img-spdif-out.c b/sound/soc/img/img-spdif-out.c index dfa72afa946e..402695b5fc41 100644 --- a/sound/soc/img/img-spdif-out.c +++ b/sound/soc/img/img-spdif-out.c @@ -468,7 +468,7 @@ static struct platform_driver img_spdif_out_driver = { .pm = &img_spdif_out_pm_ops }, .probe = img_spdif_out_probe, - .remove_new = img_spdif_out_dev_remove + .remove = img_spdif_out_dev_remove }; module_platform_driver(img_spdif_out_driver); diff --git a/sound/soc/img/pistachio-internal-dac.c b/sound/soc/img/pistachio-internal-dac.c index da6251680e41..fdeceb271e7f 100644 --- a/sound/soc/img/pistachio-internal-dac.c +++ b/sound/soc/img/pistachio-internal-dac.c @@ -271,7 +271,7 @@ static struct platform_driver pistachio_internal_dac_plat_driver = { .pm = &pistachio_internal_dac_pm_ops }, .probe = pistachio_internal_dac_probe, - .remove_new = pistachio_internal_dac_remove + .remove = pistachio_internal_dac_remove }; module_platform_driver(pistachio_internal_dac_plat_driver); diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 8652b4a20020..373d68b4cf88 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -812,7 +812,7 @@ static struct platform_driver sst_platform_driver = { .pm = &sst_platform_pm, }, .probe = sst_platform_probe, - .remove_new = sst_platform_remove, + .remove = sst_platform_remove, }; module_platform_driver(sst_platform_driver); diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 29d44c989e5f..9956dc63db74 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -358,7 +358,7 @@ static struct platform_driver sst_acpi_driver = { .pm = &intel_sst_pm, }, .probe = sst_acpi_probe, - .remove_new = sst_acpi_remove, + .remove = sst_acpi_remove, }; module_platform_driver(sst_acpi_driver); diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 3539c9ff0fd2..d3327bc237b5 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -709,7 +709,7 @@ static struct platform_driver snd_byt_cht_es8316_mc_driver = { .name = "bytcht_es8316", }, .probe = snd_byt_cht_es8316_mc_probe, - .remove_new = snd_byt_cht_es8316_mc_remove, + .remove = snd_byt_cht_es8316_mc_remove, }; module_platform_driver(snd_byt_cht_es8316_mc_driver); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 4479825c08b5..2ed49acb4e36 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1918,7 +1918,7 @@ static struct platform_driver snd_byt_rt5640_mc_driver = { .name = "bytcr_rt5640", }, .probe = snd_byt_rt5640_mc_probe, - .remove_new = snd_byt_rt5640_mc_remove, + .remove = snd_byt_rt5640_mc_remove, }; module_platform_driver(snd_byt_rt5640_mc_driver); diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 1f54da98aacf..8e4b821efe92 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -1142,7 +1142,7 @@ static struct platform_driver snd_byt_rt5651_mc_driver = { .name = "bytcr_rt5651", }, .probe = snd_byt_rt5651_mc_probe, - .remove_new = snd_byt_rt5651_mc_remove, + .remove = snd_byt_rt5651_mc_remove, }; module_platform_driver(snd_byt_rt5651_mc_driver); diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index e5a7cc606aa9..0b10d89cb189 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -663,7 +663,7 @@ static struct platform_driver snd_byt_wm5102_mc_driver = { .name = "bytcr_wm5102", }, .probe = snd_byt_wm5102_mc_probe, - .remove_new = snd_byt_wm5102_mc_remove, + .remove = snd_byt_wm5102_mc_remove, }; module_platform_driver(snd_byt_wm5102_mc_driver); diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index f43bc20d6aae..d7c114858833 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -637,7 +637,7 @@ static struct platform_driver snd_cht_mc_driver = { .name = "cht-bsw-max98090", }, .probe = snd_cht_mc_probe, - .remove_new = snd_cht_mc_remove, + .remove = snd_cht_mc_remove, }; module_platform_driver(snd_cht_mc_driver) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 578271b4230a..fc998fe4b196 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -838,7 +838,7 @@ static struct platform_driver sof_es8336_driver = { .pm = &snd_soc_pm_ops, }, .probe = sof_es8336_probe, - .remove_new = sof_es8336_remove, + .remove = sof_es8336_remove, .id_table = board_ids, }; module_platform_driver(sof_es8336_driver); diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c index b01cb2329542..5de883b9563e 100644 --- a/sound/soc/intel/boards/sof_pcm512x.c +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -430,7 +430,7 @@ static void sof_pcm512x_remove(struct platform_device *pdev) static struct platform_driver sof_audio = { .probe = sof_audio_probe, - .remove_new = sof_pcm512x_remove, + .remove = sof_pcm512x_remove, .driver = { .name = "sof_pcm512x", .pm = &snd_soc_pm_ops, diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 855109d22131..f10ed95770ea 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1368,7 +1368,7 @@ static struct platform_driver sof_sdw_driver = { .pm = &snd_soc_pm_ops, }, .probe = mc_probe, - .remove_new = mc_remove, + .remove = mc_remove, .id_table = mc_id_table, }; diff --git a/sound/soc/intel/boards/sof_wm8804.c b/sound/soc/intel/boards/sof_wm8804.c index 0a5ce34d7f7b..facc6c32cbfe 100644 --- a/sound/soc/intel/boards/sof_wm8804.c +++ b/sound/soc/intel/boards/sof_wm8804.c @@ -294,7 +294,7 @@ static struct platform_driver sof_wm8804_driver = { .pm = &snd_soc_pm_ops, }, .probe = sof_wm8804_probe, - .remove_new = sof_wm8804_remove, + .remove = sof_wm8804_remove, }; module_platform_driver(sof_wm8804_driver); diff --git a/sound/soc/intel/catpt/device.c b/sound/soc/intel/catpt/device.c index 2e1fa79a04d4..2aa637124bec 100644 --- a/sound/soc/intel/catpt/device.c +++ b/sound/soc/intel/catpt/device.c @@ -374,7 +374,7 @@ MODULE_DEVICE_TABLE(acpi, catpt_ids); static struct platform_driver catpt_acpi_driver = { .probe = catpt_acpi_probe, - .remove_new = catpt_acpi_remove, + .remove = catpt_acpi_remove, .driver = { .name = "intel_catpt", .acpi_match_table = catpt_ids, diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d1eb90310afa..99bd066c7309 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -759,7 +759,7 @@ MODULE_DEVICE_TABLE(of, mvebu_audio_of_match); static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, - .remove_new = kirkwood_i2s_dev_remove, + .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .of_match_table = of_match_ptr(mvebu_audio_of_match), diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index c12d170fa1de..d07f288f9752 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -1400,7 +1400,7 @@ static struct platform_driver mtk_btcvsd_snd_driver = { .of_match_table = mtk_btcvsd_snd_dt_match, }, .probe = mtk_btcvsd_snd_probe, - .remove_new = mtk_btcvsd_snd_remove, + .remove = mtk_btcvsd_snd_remove, }; module_platform_driver(mtk_btcvsd_snd_driver); diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index 6a17deb874df..5f11bc5438bd 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -1473,7 +1473,7 @@ static struct platform_driver mt2701_afe_pcm_driver = { .pm = &mt2701_afe_pm_ops, }, .probe = mt2701_afe_pcm_dev_probe, - .remove_new = mt2701_afe_pcm_dev_remove, + .remove = mt2701_afe_pcm_dev_remove, }; module_platform_driver(mt2701_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index c1dee119e93a..9159b42adf6a 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -890,7 +890,7 @@ static struct platform_driver mt6797_afe_pcm_driver = { .pm = &mt6797_afe_pm_ops, }, .probe = mt6797_afe_pcm_dev_probe, - .remove_new = mt6797_afe_pcm_dev_remove, + .remove = mt6797_afe_pcm_dev_remove, }; module_platform_driver(mt6797_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c index 572ded279b53..d400bea0ef9c 100644 --- a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c +++ b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c @@ -601,7 +601,7 @@ static struct platform_driver mt7986_afe_pcm_driver = { .pm = &mt7986_afe_pm_ops, }, .probe = mt7986_afe_pcm_dev_probe, - .remove_new = mt7986_afe_pcm_dev_remove, + .remove = mt7986_afe_pcm_dev_remove, }; module_platform_driver(mt7986_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index b6291b7c811e..03250273ea9c 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1223,7 +1223,7 @@ static struct platform_driver mt8173_afe_pcm_driver = { .pm = &mt8173_afe_pm_ops, }, .probe = mt8173_afe_pcm_dev_probe, - .remove_new = mt8173_afe_pcm_dev_remove, + .remove = mt8173_afe_pcm_dev_remove, }; module_platform_driver(mt8173_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index 25348fdf75fa..3f377ba4ad53 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -1268,7 +1268,7 @@ static struct platform_driver mt8183_afe_pcm_driver = { .pm = &mt8183_afe_pm_ops, }, .probe = mt8183_afe_pcm_dev_probe, - .remove_new = mt8183_afe_pcm_dev_remove, + .remove = mt8183_afe_pcm_dev_remove, }; module_platform_driver(mt8183_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c index 424c5c68f78a..9b502f4cd6ea 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c @@ -2325,7 +2325,7 @@ static struct platform_driver mt8192_afe_pcm_driver = { .pm = &mt8192_afe_pm_ops, }, .probe = mt8192_afe_pcm_dev_probe, - .remove_new = mt8192_afe_pcm_dev_remove, + .remove = mt8192_afe_pcm_dev_remove, }; module_platform_driver(mt8192_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c index 38891d1bd18a..8016bfb35015 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c @@ -3199,7 +3199,7 @@ static struct platform_driver mt8195_afe_pcm_driver = { .pm = &mt8195_afe_pm_ops, }, .probe = mt8195_afe_pcm_dev_probe, - .remove_new = mt8195_afe_pcm_dev_remove, + .remove = mt8195_afe_pcm_dev_remove, }; module_platform_driver(mt8195_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c index df6dd8c5bbe5..49efcdea975e 100644 --- a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c +++ b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c @@ -2264,7 +2264,7 @@ static struct platform_driver mt8365_afe_pcm_driver = { .pm = &mt8365_afe_pm_ops, }, .probe = mt8365_afe_pcm_dev_probe, - .remove_new = mt8365_afe_pcm_dev_remove, + .remove = mt8365_afe_pcm_dev_remove, }; module_platform_driver(mt8365_afe_pcm_driver); diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index 5d1419ed7a62..f2890111c1d2 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -345,7 +345,7 @@ MODULE_DEVICE_TABLE(of, aiu_of_match); static struct platform_driver aiu_pdrv = { .probe = aiu_probe, - .remove_new = aiu_remove, + .remove = aiu_remove, .driver = { .name = "meson-aiu", .of_match_table = aiu_of_match, diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 646ab87afac2..59deb332bd35 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -360,7 +360,7 @@ MODULE_DEVICE_TABLE(of, axg_card_of_match); static struct platform_driver axg_card_pdrv = { .probe = meson_card_probe, - .remove_new = meson_card_remove, + .remove = meson_card_remove, .driver = { .name = "axg-sound-card", .of_match_table = axg_card_of_match, diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c index 7edca3e49c8f..455f6bfc9f8f 100644 --- a/sound/soc/meson/gx-card.c +++ b/sound/soc/meson/gx-card.c @@ -129,7 +129,7 @@ MODULE_DEVICE_TABLE(of, gx_card_of_match); static struct platform_driver gx_card_pdrv = { .probe = meson_card_probe, - .remove_new = meson_card_remove, + .remove = meson_card_remove, .driver = { .name = "gx-sound-card", .of_match_table = gx_card_of_match, diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 310e3ac77424..a41a13ae38a5 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -185,7 +185,7 @@ static struct platform_driver mxs_sgtl5000_audio_driver = { .of_match_table = mxs_sgtl5000_dt_ids, }, .probe = mxs_sgtl5000_probe, - .remove_new = mxs_sgtl5000_remove, + .remove = mxs_sgtl5000_remove, }; module_platform_driver(mxs_sgtl5000_audio_driver); diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index abfaf3cdf5bb..73f36c9dd35c 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -574,7 +574,7 @@ static struct platform_driver asoc_mmp_sspa_driver = { .of_match_table = of_match_ptr(mmp_sspa_of_match), }, .probe = asoc_mmp_sspa_probe, - .remove_new = asoc_mmp_sspa_remove, + .remove = asoc_mmp_sspa_remove, }; module_platform_driver(asoc_mmp_sspa_driver); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 80e0ea0ec9fb..78f50032afc5 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -286,7 +286,7 @@ static DEFINE_SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, static struct platform_driver pxa2xx_ac97_driver = { .probe = pxa2xx_ac97_dev_probe, - .remove_new = pxa2xx_ac97_dev_remove, + .remove = pxa2xx_ac97_dev_remove, .driver = { .name = "pxa2xx-ac97", .pm = &pxa2xx_ac97_pm_ops, diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 9005c85f8c54..b8f23414eb77 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -300,7 +300,7 @@ static struct platform_driver apq8016_lpass_cpu_platform_driver = { .of_match_table = of_match_ptr(apq8016_lpass_cpu_device_id), }, .probe = asoc_qcom_lpass_cpu_platform_probe, - .remove_new = asoc_qcom_lpass_cpu_platform_remove, + .remove = asoc_qcom_lpass_cpu_platform_remove, }; module_platform_driver(apq8016_lpass_cpu_platform_driver); diff --git a/sound/soc/qcom/lpass-ipq806x.c b/sound/soc/qcom/lpass-ipq806x.c index 5c874139f39d..e57d29ea4ce7 100644 --- a/sound/soc/qcom/lpass-ipq806x.c +++ b/sound/soc/qcom/lpass-ipq806x.c @@ -172,7 +172,7 @@ static struct platform_driver ipq806x_lpass_cpu_platform_driver = { .of_match_table = of_match_ptr(ipq806x_lpass_cpu_device_id), }, .probe = asoc_qcom_lpass_cpu_platform_probe, - .remove_new = asoc_qcom_lpass_cpu_platform_remove, + .remove = asoc_qcom_lpass_cpu_platform_remove, }; module_platform_driver(ipq806x_lpass_cpu_platform_driver); diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index e6bcdf6ed796..fbead6af3d95 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -315,7 +315,7 @@ static struct platform_driver sc7180_lpass_cpu_platform_driver = { .pm = &sc7180_lpass_pm_ops, }, .probe = asoc_qcom_lpass_cpu_platform_probe, - .remove_new = asoc_qcom_lpass_cpu_platform_remove, + .remove = asoc_qcom_lpass_cpu_platform_remove, .shutdown = asoc_qcom_lpass_cpu_platform_shutdown, }; diff --git a/sound/soc/qcom/lpass-sc7280.c b/sound/soc/qcom/lpass-sc7280.c index 47c622327a8d..7cd3e291382a 100644 --- a/sound/soc/qcom/lpass-sc7280.c +++ b/sound/soc/qcom/lpass-sc7280.c @@ -445,7 +445,7 @@ static struct platform_driver sc7280_lpass_cpu_platform_driver = { .pm = &sc7280_lpass_pm_ops, }, .probe = asoc_qcom_lpass_cpu_platform_probe, - .remove_new = asoc_qcom_lpass_cpu_platform_remove, + .remove = asoc_qcom_lpass_cpu_platform_remove, .shutdown = asoc_qcom_lpass_cpu_platform_shutdown, }; diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 81fde0681f95..90228699ba7d 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -1161,7 +1161,7 @@ static struct platform_driver q6pcm_routing_platform_driver = { .of_match_table = of_match_ptr(q6pcm_routing_device_id), }, .probe = q6pcm_routing_probe, - .remove_new = q6pcm_routing_remove, + .remove = q6pcm_routing_remove, }; module_platform_driver(q6pcm_routing_platform_driver); diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index b378f870b3ad..4315da4a47c1 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -866,7 +866,7 @@ static const struct dev_pm_ops rockchip_i2s_pm_ops = { static struct platform_driver rockchip_i2s_driver = { .probe = rockchip_i2s_probe, - .remove_new = rockchip_i2s_remove, + .remove = rockchip_i2s_remove, .driver = { .name = DRV_NAME, .of_match_table = of_match_ptr(rockchip_i2s_match), diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index ee517d7b5b7b..d1f28699652f 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -1423,7 +1423,7 @@ static const struct dev_pm_ops rockchip_i2s_tdm_pm_ops = { static struct platform_driver rockchip_i2s_tdm_driver = { .probe = rockchip_i2s_tdm_probe, - .remove_new = rockchip_i2s_tdm_remove, + .remove = rockchip_i2s_tdm_remove, .driver = { .name = DRV_NAME, .of_match_table = rockchip_i2s_tdm_match, diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index d16a4a67a6a2..cae91108f7a8 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -703,7 +703,7 @@ static const struct dev_pm_ops rockchip_pdm_pm_ops = { static struct platform_driver rockchip_pdm_driver = { .probe = rockchip_pdm_probe, - .remove_new = rockchip_pdm_remove, + .remove = rockchip_pdm_remove, .driver = { .name = "rockchip-pdm", .of_match_table = of_match_ptr(rockchip_pdm_match), diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 449f62820045..b085d80ea2e4 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -233,7 +233,7 @@ MODULE_DEVICE_TABLE(of, rockchip_rt5645_of_match); static struct platform_driver snd_rk_mc_driver = { .probe = snd_rk_mc_probe, - .remove_new = snd_rk_mc_remove, + .remove = snd_rk_mc_remove, .driver = { .name = DRV_NAME, .pm = &snd_soc_pm_ops, diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index eb9d5dee196e..d87c0e4f6f91 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -380,7 +380,7 @@ static const struct dev_pm_ops rk_spdif_pm_ops = { static struct platform_driver rk_spdif_driver = { .probe = rk_spdif_probe, - .remove_new = rk_spdif_remove, + .remove = rk_spdif_remove, .driver = { .name = "rockchip-spdif", .of_match_table = of_match_ptr(rk_spdif_match), diff --git a/sound/soc/samsung/arndale.c b/sound/soc/samsung/arndale.c index f02873b6ce7f..9619f550608c 100644 --- a/sound/soc/samsung/arndale.c +++ b/sound/soc/samsung/arndale.c @@ -207,7 +207,7 @@ static struct platform_driver arndale_audio_driver = { .of_match_table = arndale_audio_of_match, }, .probe = arndale_audio_probe, - .remove_new = arndale_audio_remove, + .remove = arndale_audio_remove, }; module_platform_driver(arndale_audio_driver); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 1bcabb114e29..8f6deb06e234 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1741,7 +1741,7 @@ static const struct dev_pm_ops samsung_i2s_pm = { static struct platform_driver samsung_i2s_driver = { .probe = samsung_i2s_probe, - .remove_new = samsung_i2s_remove, + .remove = samsung_i2s_remove, .id_table = samsung_i2s_driver_ids, .driver = { .name = "samsung-i2s", diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 110ae14dd7ea..ed865cc07e2e 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -341,7 +341,7 @@ static struct platform_driver odroid_audio_driver = { .pm = &snd_soc_pm_ops, }, .probe = odroid_audio_probe, - .remove_new = odroid_audio_remove, + .remove = odroid_audio_remove, }; module_platform_driver(odroid_audio_driver); diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 573b2dee7f07..a03ba9374c2e 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -590,7 +590,7 @@ static void s3c_pcm_dev_remove(struct platform_device *pdev) static struct platform_driver s3c_pcm_driver = { .probe = s3c_pcm_dev_probe, - .remove_new = s3c_pcm_dev_remove, + .remove = s3c_pcm_dev_remove, .driver = { .name = "samsung-pcm", }, diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index aad0f9b4d4fc..4bbe7bcdb845 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -245,7 +245,7 @@ static struct platform_driver snow_driver = { .of_match_table = snow_of_match, }, .probe = snow_probe, - .remove_new = snow_remove, + .remove = snow_remove, }; module_platform_driver(snow_driver); diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index f44e3180e8d3..235d0063d1b3 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -476,7 +476,7 @@ static void spdif_remove(struct platform_device *pdev) static struct platform_driver samsung_spdif_driver = { .probe = spdif_probe, - .remove_new = spdif_remove, + .remove = spdif_remove, .driver = { .name = "samsung-spdif", }, diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 087e379aa3bc..221ce91f1950 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -2107,7 +2107,7 @@ static struct platform_driver fsi_driver = { .of_match_table = fsi_of_match, }, .probe = fsi_probe, - .remove_new = fsi_remove, + .remove = fsi_remove, .id_table = fsi_id_table, }; diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index cc200f45826c..db618c09d1e0 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -334,7 +334,7 @@ static struct platform_driver hac_pcm_driver = { }, .probe = hac_soc_platform_probe, - .remove_new = hac_soc_platform_remove, + .remove = hac_soc_platform_remove, }; module_platform_driver(hac_pcm_driver); diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 15cb5e7008f9..9784718a2b6f 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -2087,7 +2087,7 @@ static struct platform_driver rsnd_driver = { .of_match_table = rsnd_of_match, }, .probe = rsnd_probe, - .remove_new = rsnd_remove, + .remove = rsnd_remove, }; module_platform_driver(rsnd_driver); diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index d0bf0487bf1b..040ce0431fd2 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1204,7 +1204,7 @@ static struct platform_driver rz_ssi_driver = { .of_match_table = rz_ssi_of_match, }, .probe = rz_ssi_probe, - .remove_new = rz_ssi_remove, + .remove = rz_ssi_remove, }; module_platform_driver(rz_ssi_driver); diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index d0b5c543fd2f..7e771a164a80 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -788,7 +788,7 @@ static struct platform_driver siu_driver = { .name = "siu-pcm-audio", }, .probe = siu_probe, - .remove_new = siu_remove, + .remove = siu_remove, }; module_platform_driver(siu_driver); diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 9f24e3c283dd..f09ee0dea2cc 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -658,7 +658,7 @@ MODULE_DEVICE_TABLE(of, sof_of_imx8_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_imx8_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .driver = { .name = "sof-audio-of-imx8", .pm = &sof_of_pm, diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index cdd1e79ef9f6..01d3ad3314f3 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -505,7 +505,7 @@ MODULE_DEVICE_TABLE(of, sof_of_imx8m_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_imx8m_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .driver = { .name = "sof-audio-of-imx8m", .pm = &sof_of_pm, diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index 2585b1beef23..e5eee1c9f6da 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -507,7 +507,7 @@ MODULE_DEVICE_TABLE(of, sof_of_imx8ulp_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_imx8ulp_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .driver = { .name = "sof-audio-of-imx8ulp", .pm = &sof_of_pm, diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 7f18080e4e19..322ff118f0f6 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -684,7 +684,7 @@ static int sof_broadwell_probe(struct platform_device *pdev) /* acpi_driver definition */ static struct platform_driver snd_sof_acpi_intel_bdw_driver = { .probe = sof_broadwell_probe, - .remove_new = sof_acpi_remove, + .remove = sof_acpi_remove, .driver = { .name = "sof-audio-acpi-intel-bdw", .pm = &sof_acpi_pm, diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 7a57e162fb1c..f531710cf94e 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -465,7 +465,7 @@ static int sof_baytrail_probe(struct platform_device *pdev) /* acpi_driver definition */ static struct platform_driver snd_sof_acpi_intel_byt_driver = { .probe = sof_baytrail_probe, - .remove_new = sof_acpi_remove, + .remove = sof_acpi_remove, .driver = { .name = "sof-audio-acpi-intel-byt", .pm = &sof_acpi_pm, diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index 74522400207e..9466f7d2e535 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -656,7 +656,7 @@ MODULE_DEVICE_TABLE(of, sof_of_mt8186_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_mt8186_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .shutdown = sof_of_shutdown, .driver = { .name = "sof-audio-of-mt8186", diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 82d221f53a46..5b4423ed8023 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -612,7 +612,7 @@ MODULE_DEVICE_TABLE(of, sof_of_mt8195_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_mt8195_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .shutdown = sof_of_shutdown, .driver = { .name = "sof-audio-of-mt8195", diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index 688419c6b092..814a1cde1d35 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -993,7 +993,7 @@ MODULE_DEVICE_TABLE(of, sprd_mcdt_of_match); static struct platform_driver sprd_mcdt_driver = { .probe = sprd_mcdt_probe, - .remove_new = sprd_mcdt_remove, + .remove = sprd_mcdt_remove, .driver = { .name = "sprd-mcdt", .of_match_table = sprd_mcdt_of_match, diff --git a/sound/soc/starfive/jh7110_pwmdac.c b/sound/soc/starfive/jh7110_pwmdac.c index 4a4dd431b82b..a603dd17931c 100644 --- a/sound/soc/starfive/jh7110_pwmdac.c +++ b/sound/soc/starfive/jh7110_pwmdac.c @@ -516,7 +516,7 @@ static struct platform_driver jh7110_pwmdac_driver = { .pm = pm_ptr(&jh7110_pwmdac_pm_ops), }, .probe = jh7110_pwmdac_probe, - .remove_new = jh7110_pwmdac_remove, + .remove = jh7110_pwmdac_remove, }; module_platform_driver(jh7110_pwmdac_driver); diff --git a/sound/soc/starfive/jh7110_tdm.c b/sound/soc/starfive/jh7110_tdm.c index 1e0ff6720747..d38090e68df5 100644 --- a/sound/soc/starfive/jh7110_tdm.c +++ b/sound/soc/starfive/jh7110_tdm.c @@ -660,7 +660,7 @@ static struct platform_driver jh7110_tdm_driver = { .pm = pm_ptr(&jh7110_tdm_pm_ops), }, .probe = jh7110_tdm_probe, - .remove_new = jh7110_tdm_dev_remove, + .remove = jh7110_tdm_dev_remove, }; module_platform_driver(jh7110_tdm_driver); diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index fb5dd9a68bea..9351727dce1a 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -398,7 +398,7 @@ static struct platform_driver stm32_adfsdm_driver = { .of_match_table = stm32_adfsdm_of_match, }, .probe = stm32_adfsdm_probe, - .remove_new = stm32_adfsdm_remove, + .remove = stm32_adfsdm_remove, }; module_platform_driver(stm32_adfsdm_driver); diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index a96aa308681a..faa00103ee7f 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -1216,7 +1216,7 @@ static struct platform_driver stm32_i2s_driver = { .pm = &stm32_i2s_pm_ops, }, .probe = stm32_i2s_probe, - .remove_new = stm32_i2s_remove, + .remove = stm32_i2s_remove, }; module_platform_driver(stm32_i2s_driver); diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index ad2492efb1cd..7bc4a96b7503 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1619,7 +1619,7 @@ static struct platform_driver stm32_sai_sub_driver = { .pm = &stm32_sai_sub_pm_ops, }, .probe = stm32_sai_sub_probe, - .remove_new = stm32_sai_sub_remove, + .remove = stm32_sai_sub_remove, }; module_platform_driver(stm32_sai_sub_driver); diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index 9eed3c57e3f1..d1b32ba1e1a2 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -1072,7 +1072,7 @@ static struct platform_driver stm32_spdifrx_driver = { .pm = &stm32_spdifrx_pm_ops, }, .probe = stm32_spdifrx_probe, - .remove_new = stm32_spdifrx_remove, + .remove = stm32_spdifrx_remove, }; module_platform_driver(stm32_spdifrx_driver); diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 25af47b63bdd..330bc0c09f56 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1828,7 +1828,7 @@ static struct platform_driver sun4i_codec_driver = { .of_match_table = sun4i_codec_of_match, }, .probe = sun4i_codec_probe, - .remove_new = sun4i_codec_remove, + .remove = sun4i_codec_remove, }; module_platform_driver(sun4i_codec_driver); diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 3af0b2aab291..40de99a34bc3 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1684,7 +1684,7 @@ static const struct dev_pm_ops sun4i_i2s_pm_ops = { static struct platform_driver sun4i_i2s_driver = { .probe = sun4i_i2s_probe, - .remove_new = sun4i_i2s_remove, + .remove = sun4i_i2s_remove, .driver = { .name = "sun4i-i2s", .of_match_table = sun4i_i2s_match, diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index f41c30955857..0aa416423246 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -726,7 +726,7 @@ static struct platform_driver sun4i_spdif_driver = { .pm = &sun4i_spdif_pm, }, .probe = sun4i_spdif_probe, - .remove_new = sun4i_spdif_remove, + .remove = sun4i_spdif_remove, }; module_platform_driver(sun4i_spdif_driver); diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c index 884394ddaf86..3e751b5694fe 100644 --- a/sound/soc/sunxi/sun50i-dmic.c +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -426,7 +426,7 @@ static struct platform_driver sun50i_dmic_driver = { .pm = &sun50i_dmic_pm, }, .probe = sun50i_dmic_probe, - .remove_new = sun50i_dmic_remove, + .remove = sun50i_dmic_remove, }; module_platform_driver(sun50i_dmic_driver); diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index b5dafb749c3f..8c645e04d571 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -1713,7 +1713,7 @@ static struct platform_driver sun8i_codec_driver = { .pm = &sun8i_codec_pm_ops, }, .probe = sun8i_codec_probe, - .remove_new = sun8i_codec_remove, + .remove = sun8i_codec_remove, }; module_platform_driver(sun8i_codec_driver); diff --git a/sound/soc/tegra/tegra186_asrc.c b/sound/soc/tegra/tegra186_asrc.c index 22af5135d77a..d914dba56013 100644 --- a/sound/soc/tegra/tegra186_asrc.c +++ b/sound/soc/tegra/tegra186_asrc.c @@ -1034,7 +1034,7 @@ static struct platform_driver tegra186_asrc_driver = { .pm = &tegra186_asrc_pm_ops, }, .probe = tegra186_asrc_platform_probe, - .remove_new = tegra186_asrc_platform_remove, + .remove = tegra186_asrc_platform_remove, }; module_platform_driver(tegra186_asrc_driver) diff --git a/sound/soc/tegra/tegra186_dspk.c b/sound/soc/tegra/tegra186_dspk.c index 21cd41fec7a9..508128b7783e 100644 --- a/sound/soc/tegra/tegra186_dspk.c +++ b/sound/soc/tegra/tegra186_dspk.c @@ -542,7 +542,7 @@ static struct platform_driver tegra186_dspk_driver = { .pm = &tegra186_dspk_pm_ops, }, .probe = tegra186_dspk_platform_probe, - .remove_new = tegra186_dspk_platform_remove, + .remove = tegra186_dspk_platform_remove, }; module_platform_driver(tegra186_dspk_driver); diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 8011afe93c96..08c58e8f3c22 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -448,7 +448,7 @@ static struct platform_driver tegra20_ac97_driver = { .of_match_table = tegra20_ac97_of_match, }, .probe = tegra20_ac97_platform_probe, - .remove_new = tegra20_ac97_platform_remove, + .remove = tegra20_ac97_platform_remove, }; module_platform_driver(tegra20_ac97_driver); diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index f11618e8f13e..3b9823d1a87a 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -500,7 +500,7 @@ static struct platform_driver tegra20_i2s_driver = { .pm = &tegra20_i2s_pm_ops, }, .probe = tegra20_i2s_platform_probe, - .remove_new = tegra20_i2s_platform_remove, + .remove = tegra20_i2s_platform_remove, }; module_platform_driver(tegra20_i2s_driver); diff --git a/sound/soc/tegra/tegra210_admaif.c b/sound/soc/tegra/tegra210_admaif.c index 9f9334e48049..a866aeb2719d 100644 --- a/sound/soc/tegra/tegra210_admaif.c +++ b/sound/soc/tegra/tegra210_admaif.c @@ -856,7 +856,7 @@ static const struct dev_pm_ops tegra_admaif_pm_ops = { static struct platform_driver tegra_admaif_driver = { .probe = tegra_admaif_probe, - .remove_new = tegra_admaif_remove, + .remove = tegra_admaif_remove, .driver = { .name = "tegra210-admaif", .of_match_table = tegra_admaif_of_match, diff --git a/sound/soc/tegra/tegra210_adx.c b/sound/soc/tegra/tegra210_adx.c index d2530443a221..109f763fe211 100644 --- a/sound/soc/tegra/tegra210_adx.c +++ b/sound/soc/tegra/tegra210_adx.c @@ -532,7 +532,7 @@ static struct platform_driver tegra210_adx_driver = { .pm = &tegra210_adx_pm_ops, }, .probe = tegra210_adx_platform_probe, - .remove_new = tegra210_adx_platform_remove, + .remove = tegra210_adx_platform_remove, }; module_platform_driver(tegra210_adx_driver); diff --git a/sound/soc/tegra/tegra210_ahub.c b/sound/soc/tegra/tegra210_ahub.c index ab3c6b2544d2..1920b996e9aa 100644 --- a/sound/soc/tegra/tegra210_ahub.c +++ b/sound/soc/tegra/tegra210_ahub.c @@ -1416,7 +1416,7 @@ static const struct dev_pm_ops tegra_ahub_pm_ops = { static struct platform_driver tegra_ahub_driver = { .probe = tegra_ahub_probe, - .remove_new = tegra_ahub_remove, + .remove = tegra_ahub_remove, .driver = { .name = "tegra210-ahub", .of_match_table = tegra_ahub_of_match, diff --git a/sound/soc/tegra/tegra210_amx.c b/sound/soc/tegra/tegra210_amx.c index 91e405909e0f..38a2d6ec033b 100644 --- a/sound/soc/tegra/tegra210_amx.c +++ b/sound/soc/tegra/tegra210_amx.c @@ -589,7 +589,7 @@ static struct platform_driver tegra210_amx_driver = { .pm = &tegra210_amx_pm_ops, }, .probe = tegra210_amx_platform_probe, - .remove_new = tegra210_amx_platform_remove, + .remove = tegra210_amx_platform_remove, }; module_platform_driver(tegra210_amx_driver); diff --git a/sound/soc/tegra/tegra210_dmic.c b/sound/soc/tegra/tegra210_dmic.c index e53c0278ae9a..d9b577f146dc 100644 --- a/sound/soc/tegra/tegra210_dmic.c +++ b/sound/soc/tegra/tegra210_dmic.c @@ -559,7 +559,7 @@ static struct platform_driver tegra210_dmic_driver = { .pm = &tegra210_dmic_pm_ops, }, .probe = tegra210_dmic_probe, - .remove_new = tegra210_dmic_remove, + .remove = tegra210_dmic_remove, }; module_platform_driver(tegra210_dmic_driver) diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index e93ceb7afb4c..a3908b15dfdc 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -1019,7 +1019,7 @@ static struct platform_driver tegra210_i2s_driver = { .pm = &tegra210_i2s_pm_ops, }, .probe = tegra210_i2s_probe, - .remove_new = tegra210_i2s_remove, + .remove = tegra210_i2s_remove, }; module_platform_driver(tegra210_i2s_driver) diff --git a/sound/soc/tegra/tegra210_mixer.c b/sound/soc/tegra/tegra210_mixer.c index 024614f6ec0b..e07e2f1d2f70 100644 --- a/sound/soc/tegra/tegra210_mixer.c +++ b/sound/soc/tegra/tegra210_mixer.c @@ -674,7 +674,7 @@ static struct platform_driver tegra210_mixer_driver = { .pm = &tegra210_mixer_pm_ops, }, .probe = tegra210_mixer_platform_probe, - .remove_new = tegra210_mixer_platform_remove, + .remove = tegra210_mixer_platform_remove, }; module_platform_driver(tegra210_mixer_driver); diff --git a/sound/soc/tegra/tegra210_mvc.c b/sound/soc/tegra/tegra210_mvc.c index b89f5edafa03..4ead52564ab6 100644 --- a/sound/soc/tegra/tegra210_mvc.c +++ b/sound/soc/tegra/tegra210_mvc.c @@ -766,7 +766,7 @@ static struct platform_driver tegra210_mvc_driver = { .pm = &tegra210_mvc_pm_ops, }, .probe = tegra210_mvc_platform_probe, - .remove_new = tegra210_mvc_platform_remove, + .remove = tegra210_mvc_platform_remove, }; module_platform_driver(tegra210_mvc_driver) diff --git a/sound/soc/tegra/tegra210_ope.c b/sound/soc/tegra/tegra210_ope.c index 136ed17f3650..e2bc604e8b79 100644 --- a/sound/soc/tegra/tegra210_ope.c +++ b/sound/soc/tegra/tegra210_ope.c @@ -407,7 +407,7 @@ static struct platform_driver tegra210_ope_driver = { .pm = &tegra210_ope_pm_ops, }, .probe = tegra210_ope_probe, - .remove_new = tegra210_ope_remove, + .remove = tegra210_ope_remove, }; module_platform_driver(tegra210_ope_driver) diff --git a/sound/soc/tegra/tegra210_sfc.c b/sound/soc/tegra/tegra210_sfc.c index 028747c44f37..e16bbb44cc77 100644 --- a/sound/soc/tegra/tegra210_sfc.c +++ b/sound/soc/tegra/tegra210_sfc.c @@ -3631,7 +3631,7 @@ static struct platform_driver tegra210_sfc_driver = { .pm = &tegra210_sfc_pm_ops, }, .probe = tegra210_sfc_platform_probe, - .remove_new = tegra210_sfc_platform_remove, + .remove = tegra210_sfc_platform_remove, }; module_platform_driver(tegra210_sfc_driver) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d2e8078e444a..c9b52f54cea8 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -608,7 +608,7 @@ static const struct dev_pm_ops tegra30_ahub_pm_ops = { static struct platform_driver tegra30_ahub_driver = { .probe = tegra30_ahub_probe, - .remove_new = tegra30_ahub_remove, + .remove = tegra30_ahub_remove, .driver = { .name = DRV_NAME, .of_match_table = tegra30_ahub_of_match, diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index a8ff51d12edb..0d3badfbe143 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -560,7 +560,7 @@ static struct platform_driver tegra30_i2s_driver = { .pm = &tegra30_i2s_pm_ops, }, .probe = tegra30_i2s_platform_probe, - .remove_new = tegra30_i2s_platform_remove, + .remove = tegra30_i2s_platform_remove, }; module_platform_driver(tegra30_i2s_driver); diff --git a/sound/soc/tegra/tegra_audio_graph_card.c b/sound/soc/tegra/tegra_audio_graph_card.c index feba9d42bbc5..8b48813c2c59 100644 --- a/sound/soc/tegra/tegra_audio_graph_card.c +++ b/sound/soc/tegra/tegra_audio_graph_card.c @@ -248,7 +248,7 @@ static struct platform_driver tegra_audio_graph_card = { .of_match_table = graph_of_tegra_match, }, .probe = tegra_audio_graph_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(tegra_audio_graph_card); diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index 76bda188e992..94645f275495 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -595,7 +595,7 @@ static struct platform_driver ams_delta_driver = { .name = DRV_NAME, }, .probe = ams_delta_probe, - .remove_new = ams_delta_remove, + .remove = ams_delta_remove, }; module_platform_driver(ams_delta_driver); diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index 0f15a743c798..d682b98d6848 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -920,7 +920,7 @@ MODULE_DEVICE_TABLE(of, davinci_i2s_match); static struct platform_driver davinci_mcbsp_driver = { .probe = davinci_i2s_probe, - .remove_new = davinci_i2s_remove, + .remove = davinci_i2s_remove, .driver = { .name = "davinci-mcbsp", .of_match_table = of_match_ptr(davinci_i2s_match), diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 2b1ed91a736c..a0b8cca31cba 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2535,7 +2535,7 @@ static const struct dev_pm_ops davinci_mcasp_pm_ops = { static struct platform_driver davinci_mcasp_driver = { .probe = davinci_mcasp_probe, - .remove_new = davinci_mcasp_remove, + .remove = davinci_mcasp_remove, .driver = { .name = "davinci-mcasp", .pm = &davinci_mcasp_pm_ops, diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 2110ffe5281c..411970399271 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -1430,7 +1430,7 @@ static struct platform_driver asoc_mcbsp_driver = { }, .probe = asoc_mcbsp_probe, - .remove_new = asoc_mcbsp_remove, + .remove = asoc_mcbsp_remove, }; module_platform_driver(asoc_mcbsp_driver); diff --git a/sound/soc/uniphier/aio-ld11.c b/sound/soc/uniphier/aio-ld11.c index 01cc3b961999..a041ce8e23a4 100644 --- a/sound/soc/uniphier/aio-ld11.c +++ b/sound/soc/uniphier/aio-ld11.c @@ -347,7 +347,7 @@ static struct platform_driver uniphier_aio_driver = { .of_match_table = of_match_ptr(uniphier_aio_of_match), }, .probe = uniphier_aio_probe, - .remove_new = uniphier_aio_remove, + .remove = uniphier_aio_remove, }; module_platform_driver(uniphier_aio_driver); diff --git a/sound/soc/uniphier/aio-pxs2.c b/sound/soc/uniphier/aio-pxs2.c index fba13a212bdb..889f64b2c01f 100644 --- a/sound/soc/uniphier/aio-pxs2.c +++ b/sound/soc/uniphier/aio-pxs2.c @@ -256,7 +256,7 @@ static struct platform_driver uniphier_aio_driver = { .of_match_table = of_match_ptr(uniphier_aio_of_match), }, .probe = uniphier_aio_probe, - .remove_new = uniphier_aio_remove, + .remove = uniphier_aio_remove, }; module_platform_driver(uniphier_aio_driver); diff --git a/sound/soc/uniphier/evea.c b/sound/soc/uniphier/evea.c index d90b3e4b0104..662e45882c90 100644 --- a/sound/soc/uniphier/evea.c +++ b/sound/soc/uniphier/evea.c @@ -560,7 +560,7 @@ static struct platform_driver evea_codec_driver = { .of_match_table = of_match_ptr(evea_of_match), }, .probe = evea_probe, - .remove_new = evea_remove, + .remove = evea_remove, }; module_platform_driver(evea_codec_driver); diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index e0ab4534fe3e..ae6d326167d1 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -157,7 +157,7 @@ static struct platform_driver snd_soc_mop500_driver = { .of_match_table = snd_soc_mop500_match, }, .probe = mop500_probe, - .remove_new = mop500_remove, + .remove = mop500_remove, }; module_platform_driver(snd_soc_mop500_driver); diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 3fd13e8dd110..a2dd739fdf2d 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -816,7 +816,7 @@ static struct platform_driver msp_i2s_driver = { .of_match_table = ux500_msp_i2s_match, }, .probe = ux500_msp_drv_probe, - .remove_new = ux500_msp_drv_remove, + .remove = ux500_msp_drv_remove, }; module_platform_driver(msp_i2s_driver); diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 158fc21a86c1..17ef05309469 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -713,7 +713,7 @@ MODULE_DEVICE_TABLE(of, xlnx_formatter_pcm_of_match); static struct platform_driver xlnx_formatter_pcm_driver = { .probe = xlnx_formatter_pcm_probe, - .remove_new = xlnx_formatter_pcm_remove, + .remove = xlnx_formatter_pcm_remove, .driver = { .name = DRV_NAME, .of_match_table = xlnx_formatter_pcm_of_match, diff --git a/sound/soc/xilinx/xlnx_spdif.c b/sound/soc/xilinx/xlnx_spdif.c index d52d5fc7b5b8..7febb3830dc2 100644 --- a/sound/soc/xilinx/xlnx_spdif.c +++ b/sound/soc/xilinx/xlnx_spdif.c @@ -325,7 +325,7 @@ static struct platform_driver xlnx_spdif_driver = { .of_match_table = xlnx_spdif_of_match, }, .probe = xlnx_spdif_probe, - .remove_new = xlnx_spdif_remove, + .remove = xlnx_spdif_remove, }; module_platform_driver(xlnx_spdif_driver); diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c index 6e2b72d7a65d..4eaa9011405f 100644 --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -635,7 +635,7 @@ static const struct dev_pm_ops xtfpga_i2s_pm_ops = { static struct platform_driver xtfpga_i2s_driver = { .probe = xtfpga_i2s_probe, - .remove_new = xtfpga_i2s_remove, + .remove = xtfpga_i2s_remove, .driver = { .name = "xtfpga-i2s", .of_match_table = of_match_ptr(xtfpga_i2s_of_match), base-commit: 100cc857359b5d731407d1038f7e76cd0e871d94 -- 2.45.2

1 0

[PATCH] ASoC: tlv320aic31xx: Add support for loading filter coefficients
by Romain Gantois 09 Sep '24

09 Sep '24

The TLV320DAC3100 Audio DAC has 25 built-in digital audio processing blocks (PRBs). Each of these PRBs has a static filter structure with programmable coefficients. Once a PRB is selected for use by the DAC, its filter coefficients can be configured via a dedicated set of registers. Define a new optional firmware which can be loaded by the TLV320DAC driver. This firmware describes a full set of filter coefficients for all blocks used by the various PRBs. The firmware's binary format is heavily inspired by the one used in the peb2466 driver. It includes a version marker to allow for potential evolutions of the format. Note that adaptive filtering is not supported i.e. filter coefficients are loaded once before power-on and then cannot be changed while the DAC is powered. This is why only page A coefficients are modified. Page B coefficients are only used for adaptive filtering. Signed-off-by: Romain Gantois <romain.gantois(a)bootlin.com> --- sound/soc/codecs/tlv320aic31xx.c | 100 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 100 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 2f94cfda0e33..7e624c4b77b6 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -12,6 +12,7 @@ * and mono/stereo Class-D speaker driver. */ +#include <asm/unaligned.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -22,6 +23,7 @@ #include <linux/gpio/consumer.h> #include <linux/regulator/consumer.h> #include <linux/acpi.h> +#include <linux/firmware.h> #include <linux/of.h> #include <linux/slab.h> #include <sound/core.h> @@ -1638,6 +1640,98 @@ static const struct i2c_device_id aic31xx_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); +static int tlv320dac3100_fw_load(struct aic31xx_priv *aic31xx, + const u8 *data, size_t size) +{ + int ret, reg; + u16 val16; + + /* + * Coefficients firmware binary structure. Multi-byte values are big-endian. + * + * @0, 16bits: Magic (0xB30C) + * @2, 16bits: Version (0x0100 for version 1.0) + * @4, 8bits: DAC Processing Block Selection + * @5, 62 16-bit values: Page 8 buffer A DAC programmable filter coefficients + * @129, 12 16-bit values: Page 9 Buffer A DAC programmable filter coefficients + * + * Filter coefficients are interpreted as two's complement values + * ranging from -32 768 to 32 767. For more details on filter coefficients, + * please refer to the TLV320DAC3100 datasheet, tables 6-120 and 6-123. + */ + + if (size != 153) { + dev_err(aic31xx->dev, "firmware size is %zu, expected 153 bytes\n", size); + return -EINVAL; + } + + /* Check magic */ + val16 = get_unaligned_be16(data); + if (val16 != 0xb30c) { + dev_err(aic31xx->dev, "fw magic is 0x%04x expected 0xb30c\n", val16); + return -EINVAL; + } + data += 2; + + /* Check version */ + val16 = get_unaligned_be16(data); + if (val16 != 0x0100) { + dev_err(aic31xx->dev, "invalid firmware version 0x%04x! expected 1", val16); + return -EINVAL; + } + data += 2; + + ret = regmap_write(aic31xx->regmap, AIC31XX_DACPRB, *data); + if (ret) { + dev_err(aic31xx->dev, "failed to write PRB index: err %d\n", ret); + return ret; + } + data += 1; + + /* Page 8 Buffer A coefficients */ + for (reg = 2; reg < 126; reg++) { + ret = regmap_write(aic31xx->regmap, AIC31XX_REG(8, reg), *data); + if (ret) { + dev_err(aic31xx->dev, + "failed to write page 8 filter coefficient %d: err %d\n", reg, ret); + return ret; + } + data++; + } + + /* Page 9 Buffer A coefficients */ + for (reg = 2; reg < 26; reg++) { + ret = regmap_write(aic31xx->regmap, AIC31XX_REG(9, reg), *data); + if (ret) { + dev_err(aic31xx->dev, + "failed to write page 9 filter coefficient %d: err %d\n", reg, ret); + return ret; + } + data++; + } + + dev_info(aic31xx->dev, "done loading DAC filter coefficients\n"); + + return ret; +} + +static int tlv320dac3100_load_coeffs(struct aic31xx_priv *aic31xx, + const char *fw_name) +{ + const struct firmware *fw; + int ret; + + ret = request_firmware(&fw, fw_name, aic31xx->dev); + if (ret) + return ret; + + ret = tlv320dac3100_fw_load(aic31xx, fw->data, fw->size); + + release_firmware(fw); + + return ret; +} + static int aic31xx_i2c_probe(struct i2c_client *i2c) { struct aic31xx_priv *aic31xx; @@ -1727,6 +1821,12 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c) } } + if (aic31xx->codec_type == DAC3100) { + ret = tlv320dac3100_load_coeffs(aic31xx, "tlv320dac3100-coeffs.bin"); + if (ret) + dev_warn(aic31xx->dev, "Did not load any filter coefficients\n"); + } + if (aic31xx->codec_type & DAC31XX_BIT) return devm_snd_soc_register_component(&i2c->dev, &soc_codec_driver_aic31xx, --- base-commit: 8400291e289ee6b2bf9779ff1c83a291501f017b change-id: 20240806-tlv320-filter-189edd66496c Best regards, -- Romain Gantois <romain.gantois(a)bootlin.com>

2 1

[PATCH] soundwire: stream: Revert "soundwire: stream: fix programming slave ports for non-continous port maps"
by Krzysztof Kozlowski 09 Sep '24

09 Sep '24

This reverts commit ab8d66d132bc8f1992d3eb6cab8d32dda6733c84 because it breaks codecs using non-continuous masks in source and sink ports. The commit missed the point that port numbers are not used as indices for iterating over prop.sink_ports or prop.source_ports. Soundwire core and existing codecs expect that the array passed as prop.sink_ports and prop.source_ports is continuous. The port mask still might be non-continuous, but that's unrelated. Reported-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Closes: https://lore.kernel.org/all/b6c75eee-761d-44c8-8413-2a5b34ee2f98@linux.inte… Fixes: ab8d66d132bc ("soundwire: stream: fix programming slave ports for non-continous port maps") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- I will need to fix my codec drivers, but that's independent. --- drivers/soundwire/stream.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index f275143d7b18..7aa4900dcf31 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1291,18 +1291,18 @@ struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, unsigned int port_num) { struct sdw_dpn_prop *dpn_prop; - unsigned long mask; + u8 num_ports; int i; if (direction == SDW_DATA_DIR_TX) { - mask = slave->prop.source_ports; + num_ports = hweight32(slave->prop.source_ports); dpn_prop = slave->prop.src_dpn_prop; } else { - mask = slave->prop.sink_ports; + num_ports = hweight32(slave->prop.sink_ports); dpn_prop = slave->prop.sink_dpn_prop; } - for_each_set_bit(i, &mask, 32) { + for (i = 0; i < num_ports; i++) { if (dpn_prop[i].num == port_num) return &dpn_prop[i]; } -- 2.43.0

5 6

[PATCH] ASoC: tas2781: fix to save the dsp bin file name into the correct array in case name_prefix is not NULL
by Shenghao Ding 09 Sep '24

09 Sep '24

fix to save the dsp bin file name into the correct array, coef_binaryname, instead of rca_binaryname, in case name_prefix is not NULL. Fixes: 55f0a1fa6ea4 ("ASoC: tas2781: Add name_prefix as the prefix name of DSP firmwares and calibrated data files") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 59fcce7fef7b..3bc10d615fd3 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -498,7 +498,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw, */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; if (tas_priv->name_prefix) - scnprintf(tas_priv->rca_binaryname, 64, "%s-%s_coef.bin", + scnprintf(tas_priv->coef_binaryname, 64, "%s-%s_coef.bin", tas_priv->name_prefix, tas_priv->dev_name); else scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", -- 2.34.1

3 3

[PATCH] soundwire: stream: fix programming slave ports for non-continous port maps
by Krzysztof Kozlowski 09 Sep '24

09 Sep '24

Two bitmasks in 'struct sdw_slave_prop' - 'source_ports' and 'sink_ports' - define which ports to program in sdw_program_slave_port_params(). The masks are used to get the appropriate data port properties ('struct sdw_get_slave_dpn_prop') from an array. Bitmasks can be non-continuous or can start from index different than 0, thus when looking for matching port property for given port, we must iterate over mask bits, not from 0 up to number of ports. This fixes allocation and programming slave ports, when a source or sink masks start from further index. Fixes: f8101c74aa54 ("soundwire: Add Master and Slave port programming") Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/stream.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 7aa4900dcf31..f275143d7b18 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1291,18 +1291,18 @@ struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, unsigned int port_num) { struct sdw_dpn_prop *dpn_prop; - u8 num_ports; + unsigned long mask; int i; if (direction == SDW_DATA_DIR_TX) { - num_ports = hweight32(slave->prop.source_ports); + mask = slave->prop.source_ports; dpn_prop = slave->prop.src_dpn_prop; } else { - num_ports = hweight32(slave->prop.sink_ports); + mask = slave->prop.sink_ports; dpn_prop = slave->prop.sink_dpn_prop; } - for (i = 0; i < num_ports; i++) { + for_each_set_bit(i, &mask, 32) { if (dpn_prop[i].num == port_num) return &dpn_prop[i]; } -- 2.43.0

6 15

[PATCH v2] ASoC: dt-bindings: renesas,rsnd: add post-init-providers property
by Kuninori Morimoto 09 Sep '24

09 Sep '24

At least if rsnd is using DPCM connection with Audio-Graph-Card2, fw_devlink might doesn't have enough information to break the cycle (Same problem might occur with Multi-CPU/Codec or Codec2Codec). In such case, rsnd driver will not be probed. Add post-init-providers support to break the link cycle. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - remove basic description/type - add detail description - update git-log Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 07ec6247d9def..458877ce4aa42 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -112,6 +112,13 @@ properties: description: List of necessary clock names. # details are defined below + post-init-providers: + description: | + At least if rsnd is using DPCM connection with Audio-Graph-Card2, fw_devlink might doesn't + have enough information to break the cycle. rsnd driver will not be probed in such case. + Add post-init-providers property to indicate which link in the cycle to break. + Same problem might occur with Multi-CPU/Codec or Codec2Codec. + # ports is below port: $ref: audio-graph-port.yaml#/definitions/port-base -- 2.43.0

2 1

How to handle complex Codec2Codec
by Kuninori Morimoto 09 Sep '24

09 Sep '24

Hi ALSA ML ALSA SoC is supporting Codec2Codec connection, and Audio Graph Card2 is supporting it. I think its image is... +------+ +--- |Codec1| <-- | +------+ | +------+ +--> |Codec2| --> +------+ This is OK, but how to handle below ? +----------------------+ |CPU | +------+ | +--[rate convert]<--| <-- |Codec1| <-- | | | +------+ | | | +------+ | +------------------>| --> |Codec2| --> +----------------------+ +------+ I think there are 2 issues. (1) How to handle [CPU] part on Codec2Codec ? (2) How to handle [rate convert] on Codec2Codec ? I guess (1) can be handled if [CPU] was worked as platform, but do you have any other idea/solution for it ? I think (2) [rate convert] needs DPCM connection, but how to use it with Codec2Codec ? Or, can we handle Codec2Codec via DPCM somehow ?? FE BE +------+ +------+ --> |Codec1| -> |Codec2| -> +------+ +------+ If we can do this, both (1) and (2) can be solved, like below ? In such case, [Codec1] need to work as [capture] when playback. But I'm not sure how to implement it... FE BE +------+ +-----------------+ +-----+ +------+ --> |Codec1|-|CPU[rate convert]| -> |dummy|-|Codec2| -> +------+ +-----------------+ +-----+ +------+ Thank you for your help !! Best regards --- Kuninori Morimoto

2 8

Re: [PATCH] ALSA: Add logitech Audio profile quirk
by Joshua Pius 06 Sep '24

06 Sep '24

1 3

[PATCH] ASoC: codecs: Use devm_clk_get_enabled() helpers
by ying zuxin 06 Sep '24

06 Sep '24

From: ying zuxin <11154159(a)vivo.com> The devm_clk_get_enabled() helpers: - call devm_clk_get() - call clk_prepare_enable() and register what is needed in order to call clk_disable_unprepare() when needed, as a managed resource. This simplifies the code and avoids the calls to clk_disable_unprepare(). Signed-off-by: ying zuxin <11154159(a)vivo.com> --- sound/soc/codecs/peb2466.c | 14 +------------- 1 file changed, 1 insertion(+), 13 deletions(-) diff --git a/sound/soc/codecs/peb2466.c b/sound/soc/codecs/peb2466.c index 76ee7e3f4d9b..b67cfad4fc32 100644 --- a/sound/soc/codecs/peb2466.c +++ b/sound/soc/codecs/peb2466.c @@ -1975,12 +1975,9 @@ static int peb2466_spi_probe(struct spi_device *spi) if (IS_ERR(peb2466->reset_gpio)) return PTR_ERR(peb2466->reset_gpio); - peb2466->mclk = devm_clk_get(&peb2466->spi->dev, "mclk"); + peb2466->mclk = devm_clk_get_enabled(&peb2466->spi->dev, "mclk"); if (IS_ERR(peb2466->mclk)) return PTR_ERR(peb2466->mclk); - ret = clk_prepare_enable(peb2466->mclk); - if (ret) - return ret; if (peb2466->reset_gpio) { gpiod_set_value_cansleep(peb2466->reset_gpio, 1); @@ -2031,17 +2028,9 @@ static int peb2466_spi_probe(struct spi_device *spi) return 0; failed: - clk_disable_unprepare(peb2466->mclk); return ret; } -static void peb2466_spi_remove(struct spi_device *spi) -{ - struct peb2466 *peb2466 = spi_get_drvdata(spi); - - clk_disable_unprepare(peb2466->mclk); -} - static const struct of_device_id peb2466_of_match[] = { { .compatible = "infineon,peb2466", }, { } @@ -2061,7 +2050,6 @@ static struct spi_driver peb2466_spi_driver = { }, .id_table = peb2466_id_table, .probe = peb2466_spi_probe, - .remove = peb2466_spi_remove, }; module_spi_driver(peb2466_spi_driver); -- 2.39.0

3 2

[PATCH] ASoC: makes rtd->initialized bit field
by Kuninori Morimoto 06 Sep '24

06 Sep '24

rtd->initialized is used to know whether soc_init_pcm_runtime() was correctly fined, and used to call snd_soc_link_exit(). We don't need to have it as bool, let's make it bit-field same as other flags. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/soc.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 81ef380b2e06..e6e359c1a2ac 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1207,8 +1207,7 @@ struct snd_soc_pcm_runtime { /* bit field */ unsigned int pop_wait:1; unsigned int fe_compr:1; /* for Dynamic PCM */ - - bool initialized; + unsigned int initialized:1; /* CPU/Codec/Platform */ int num_components; -- 2.43.0

2 1

[PATCH 1/2] pcm: dmix: Fix resume with multiple instances
by Takashi Iwai 06 Sep '24

06 Sep '24

The fix for PCM dmix suspend/resume checks spcm->info bit of SND_PCM_INFO_RESUME for applying a workaround for drivers with the full resume support. This assumed that scpm->info is exposed from the underlying slave PCM device. The above is true for the first opened instance, but for the second opened instance, it's a copy from the saved data in shmem. And, we dropped SND_PCM_INFO_RESUME bit there to assure not to expose the full resume capability to applications. This resulted in the inconsistencies, and when the second instance is resumed at first, it misses the snd_pcm_resume() call, hence the driver doesn't react properly any longer. For addressing it, we keep SND_PCM_INFO_RESUME bit in shmptr->s.info bits as is, while dropping the bit exposed to apps in snd_pcm_direct_hw_refine() and *_hw_params() callbacks. Fixes: 6d1d620eadf3 ("pcm: dmix: resume workaround for buggy driver") Reported-and-tested-by: Chancel Liu <chancel.liu(a)nxp.com> Closes: https://lore.kernel.org/DB9PR04MB94988752ED7C43B399E0BC00E3942@DB9PR04MB949… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index e53e59238119..17e677f60b19 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -1018,6 +1018,7 @@ int snd_pcm_direct_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) } dshare->timer_ticks = hw_param_interval(params, SND_PCM_HW_PARAM_PERIOD_SIZE)->max / dshare->slave_period_size; params->info = dshare->shmptr->s.info; + params->info &= ~SND_PCM_INFO_RESUME; #ifdef REFINE_DEBUG snd_output_puts(log, "DMIX REFINE (end):\n"); snd_pcm_hw_params_dump(params, log); @@ -1031,6 +1032,7 @@ int snd_pcm_direct_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) snd_pcm_direct_t *dmix = pcm->private_data; params->info = dmix->shmptr->s.info; + params->info &= ~SND_PCM_INFO_RESUME; params->rate_num = dmix->shmptr->s.rate; params->rate_den = 1; params->fifo_size = 0; @@ -1183,8 +1185,6 @@ static void save_slave_setting(snd_pcm_direct_t *dmix, snd_pcm_t *spcm) COPY_SLAVE(buffer_time); COPY_SLAVE(sample_bits); COPY_SLAVE(frame_bits); - - dmix->shmptr->s.info &= ~SND_PCM_INFO_RESUME; } #undef COPY_SLAVE -- 2.43.0

1 1

Suspend/resume Issue on pcm_dmix.c in alsa-lib
by Chancel Liu 06 Sep '24

06 Sep '24

Hi Takashi Iwai, Jaroslav Kysela We found an issue on dmix in alsa-lib when do suspend and resume. It can be easily reproduced by following steps: 1. Run two dmix clients in parallel. (Only one client doesn't has such issue) ~# aplay xxx1.wav & ~# aplay xxx2.wav & Here I attach the asound.conf we're using. ~# cat /etc/asound.conf defaults.pcm.rate_converter "linear" pcm.dmix_44100{ type dmix ipc_key 5678293 ipc_key_add_uid yes slave{ pcm "hw:0,0" period_time 40000 format S16_LE rate 44100 } } pcm.asymed{ type asym playback.pcm "dmix_44100" capture.pcm "dsnoop_44100" } pcm.!default{ type plug route_policy "average" slave.pcm "asymed" } 2. Let linux enter into suspend and then resume(Repeat this step if not reproduced) 3. After resume, aplay will get stuck in snd_pcm_wait(). The GDB shows: (gdb) bt #0 0x0000fffff7da9264 in __GI___poll (fds=fds@entry=0xfffffffff480, nfds=nfds@entry=1, timeout=timeout@entry=240) at /usr/src/debug/glibc/2.39+git/sysdeps/unix/sysv/linux/poll.c:41 #1 0x0000fffff7edf468 in poll (__timeout=240, __nfds=1, __fds=0xfffffffff480) #2 snd1_pcm_wait_nocheck (pcm=pcm@entry=0xaaaaaaad2cb0, timeout=240, timeout@entry=-10001) at pcm.c:2993 #3 0x0000fffff7ee54a0 in snd1_pcm_write_areas (pcm=pcm@entry=0xaaaaaaad2cb0, areas=areas@entry=0xfffffffff560, offset=<optimized out>, offset@entry=0, size=<optimized out>, size@entry=1768, func=func@entry=0xfffff7ef5190 <snd_pcm_plugin_write_areas>) at pcm.c:7699 #4 0x0000fffff7ef5020 in snd_pcm_plugin_writei (pcm=0xaaaaaaad2cb0, buffer=<optimized out>, size=1768) at pcm_plugin.c:354 It seems that sometimes after suspend and resume there's no available space for data written into buffer. Then aplay keeps stuck in snd_pcm_wait(). I checked the hw_ptr of dmix and found that hw_ptr is always 0 after resume. I don't have a solution now so I turn to you for help. The version of alsa-lib is v1.2.11. Could you please help check it? Regards, Chancel Liu

3 12

[GIT PULL] ASoC fixes for v6.11-rc6
by Mark Brown 06 Sep '24

06 Sep '24

The following changes since commit bb4485562f5907708f1c218b5d70dce04165d1e1: ASoC: cs-amp-lib: Ignore empty UEFI calibration entries (2024-08-22 17:52:26 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.11-rc6 for you to fetch changes up to 77212f300bfd6fb3edaabd1daf863cabb521854a: ASoC: codecs: lpass-va-macro: set the default codec version for sm8250 (2024-08-30 13:54:24 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.11 A larger set of fixes than I'd like at this point, but mainly due to people working on fixing module autoloading by adding missing exports of ID tables rather than anything particularly concerning. There are some other runtime fixes and quirks, and a tweak to the ABI definition for SOF which ensures that a struct layout doesn't vary depending on the architecture of the host. ---------------------------------------------------------------- Chen-Yu Tsai (1): ASoc: SOF: topology: Clear SOF link platform name upon unload Hans de Goede (2): ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards harder ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict Laurentiu Mihalcea (1): ASoC: SOF: ipc: replace "enum sof_comp_type" field with "uint32_t" Liao Chen (4): ASoC: intel: fix module autoloading ASoC: google: fix module autoloading ASoC: tda7419: fix module autoloading ASoC: fix module autoloading Mark Brown (1): ASoC: fix module autoloading Markuss Broks (1): ASoC: amd: yc: Add a quirk for MSI Bravo 17 (D7VEK) Matteo Martelli (1): ASoC: sunxi: sun4i-i2s: fix LRCLK polarity in i2s mode Mohan Kumar (1): ASoC: tegra: Fix CBB error during probe() Srinivas Kandagatla (1): ASoC: codecs: lpass-va-macro: set the default codec version for sm8250 Zhang Yi (1): ASoC: mediatek: mt8188-mt6359: Modify key robelin (1): ASoC: dapm: Fix UAF for snd_soc_pcm_runtime object include/sound/sof/topology.h | 2 +- include/uapi/sound/sof/abi.h | 2 +- sound/soc/amd/yc/acp6x-mach.c | 7 ++ sound/soc/codecs/chv3-codec.c | 1 + sound/soc/codecs/lpass-va-macro.c | 11 +- sound/soc/codecs/tda7419.c | 1 + sound/soc/google/chv3-i2s.c | 1 + sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 1 - sound/soc/intel/keembay/kmb_platform.c | 1 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 17 ++- sound/soc/soc-dapm.c | 1 + sound/soc/sof/topology.c | 2 + sound/soc/sunxi/sun4i-i2s.c | 143 +++++++++++----------- sound/soc/tegra/tegra210_ahub.c | 10 +- 23 files changed, 127 insertions(+), 91 deletions(-)

2 1

[PATCH] ASoC: Intel: skl_hda_dsp_generic: convert comma to semicolon
by Chen Ni 05 Sep '24

05 Sep '24

Replace comma between expressions with semicolons. Using a ',' in place of a ';' can have unintended side effects. Although that is not the case here, it is seems best to use ';' unless ',' is intended. Found by inspection. No functional change intended. Compile tested only. Signed-off-by: Chen Ni <nichen(a)iscas.ac.cn> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 225867bb3310..8e104874d58c 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -217,14 +217,14 @@ static int skl_hda_audio_probe(struct platform_device *pdev) return -ENOMEM; card = &ctx->card; - card->name = "hda-dsp", - card->owner = THIS_MODULE, - card->dai_link = skl_hda_be_dai_links, - card->dapm_widgets = skl_hda_widgets, - card->dapm_routes = skl_hda_map, - card->add_dai_link = skl_hda_add_dai_link, - card->fully_routed = true, - card->late_probe = skl_hda_card_late_probe, + card->name = "hda-dsp"; + card->owner = THIS_MODULE; + card->dai_link = skl_hda_be_dai_links; + card->dapm_widgets = skl_hda_widgets; + card->dapm_routes = skl_hda_map; + card->add_dai_link = skl_hda_add_dai_link; + card->fully_routed = true; + card->late_probe = skl_hda_card_late_probe; snd_soc_card_set_drvdata(card, ctx); -- 2.25.1

1 0

[PATCH 1/2 4.19.y] ALSA: usb-audio: Sanity checks for each pipe and EP types
by Takashi Iwai 05 Sep '24

05 Sep '24

[ Upstream commit 801ebf1043ae7b182588554cc9b9ad3c14bc2ab5 ] The recent USB core code performs sanity checks for the given pipe and EP types, and it can be hit by manipulated USB descriptors by syzbot. For making syzbot happier, this patch introduces a local helper for a sanity check in the driver side and calls it at each place before the message handling, so that we can avoid the WARNING splats. Reported-by: syzbot+d952e5e28f5fb7718d23(a)syzkaller.appspotmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/usb/helper.c | 17 +++++++++++++++++ sound/usb/helper.h | 1 + sound/usb/quirks.c | 14 +++++++++++--- 3 files changed, 29 insertions(+), 3 deletions(-) diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 7712e2b84183..b1cc9499c57e 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -76,6 +76,20 @@ void *snd_usb_find_csint_desc(void *buffer, int buflen, void *after, u8 dsubtype return NULL; } +/* check the validity of pipe and EP types */ +int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe) +{ + static const int pipetypes[4] = { + PIPE_CONTROL, PIPE_ISOCHRONOUS, PIPE_BULK, PIPE_INTERRUPT + }; + struct usb_host_endpoint *ep; + + ep = usb_pipe_endpoint(dev, pipe); + if (usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)]) + return -EINVAL; + return 0; +} + /* * Wrapper for usb_control_msg(). * Allocates a temp buffer to prevent dmaing from/to the stack. @@ -88,6 +102,9 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, void *buf = NULL; int timeout; + if (snd_usb_pipe_sanity_check(dev, pipe)) + return -EINVAL; + if (size > 0) { buf = kmemdup(data, size, GFP_KERNEL); if (!buf) diff --git a/sound/usb/helper.h b/sound/usb/helper.h index f5b4c6647e4d..5e8a18b4e7b9 100644 --- a/sound/usb/helper.h +++ b/sound/usb/helper.h @@ -7,6 +7,7 @@ unsigned int snd_usb_combine_bytes(unsigned char *bytes, int size); void *snd_usb_find_desc(void *descstart, int desclen, void *after, u8 dtype); void *snd_usb_find_csint_desc(void *descstart, int desclen, void *after, u8 dsubtype); +int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe); int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 43cbaaff163f..087eef5e249d 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -743,11 +743,13 @@ static int snd_usb_novation_boot_quirk(struct usb_device *dev) static int snd_usb_accessmusic_boot_quirk(struct usb_device *dev) { int err, actual_length; - /* "midi send" enable */ static const u8 seq[] = { 0x4e, 0x73, 0x52, 0x01 }; + void *buf; - void *buf = kmemdup(seq, ARRAY_SIZE(seq), GFP_KERNEL); + if (snd_usb_pipe_sanity_check(dev, usb_sndintpipe(dev, 0x05))) + return -EINVAL; + buf = kmemdup(seq, ARRAY_SIZE(seq), GFP_KERNEL); if (!buf) return -ENOMEM; err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x05), buf, @@ -772,7 +774,11 @@ static int snd_usb_accessmusic_boot_quirk(struct usb_device *dev) static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev) { - int ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + int ret; + + if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), 0xaf, USB_TYPE_VENDOR | USB_RECIP_DEVICE, 1, 0, NULL, 0, 1000); @@ -879,6 +885,8 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) dev_dbg(&dev->dev, "Waiting for Axe-Fx III to boot up...\n"); + if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; /* If the Axe-Fx III has not fully booted, it will timeout when trying * to enable the audio streaming interface. A more generous timeout is * used here to detect when the Axe-Fx III has finished booting as the -- 2.45.2

4 4

[PATCH] ASoC: atmel: mchp-i2s-mcc: Improve maxburst calculation for better performance
by Andrei Simion 05 Sep '24

05 Sep '24

From: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> The period size represents the size of the DMA descriptor. To ensure all DMA descriptors start from a well-aligned address, the period size must be divided by (sample size * maxburst), not just by maxburst. This adjustment allows for computing a higher maxburst value, thereby increasing the performance of the DMA transfer. Previously, snd_pcm_lib_period_bytes() returned 0 because the runtime HW parameters are computed after the hw_params() callbacks are used. To address this, we now use params_*() functions to compute the period size accurately. This change optimizes the DMA transfer performance by ensuring proper alignment and efficient use of maxburst values. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> [andrei.simion(a)microchip.com: Reword commit message and commit title. Add macros with values for maximum DMA chunk size allowed. Add DMA_BURST_ALIGNED preprocessor function to check the alignment of the DMA burst] Signed-off-by: Andrei Simion <andrei.simion(a)microchip.com> --- sound/soc/atmel/mchp-i2s-mcc.c | 38 +++++++++++++++++++++++++++++----- 1 file changed, 33 insertions(+), 5 deletions(-) diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 193dd7acceb0..35a56b266b8d 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -221,6 +221,15 @@ #define MCHP_I2SMCC_MAX_CHANNELS 8 #define MCHP_I2MCC_TDM_SLOT_WIDTH 32 +/* + * ---- DMA chunk size allowed ---- + */ +#define MCHP_I2SMCC_DMA_8_WORD_CHUNK 8 +#define MCHP_I2SMCC_DMA_4_WORD_CHUNK 4 +#define MCHP_I2SMCC_DMA_2_WORD_CHUNK 2 +#define MCHP_I2SMCC_DMA_1_WORD_CHUNK 1 +#define DMA_BURST_ALIGNED(_p, _s, _w) !(_p % (_s * _w)) + static const struct regmap_config mchp_i2s_mcc_regmap_config = { .reg_bits = 32, .reg_stride = 4, @@ -504,12 +513,30 @@ static int mchp_i2s_mcc_is_running(struct mchp_i2s_mcc_dev *dev) return !!(sr & (MCHP_I2SMCC_SR_TXEN | MCHP_I2SMCC_SR_RXEN)); } +static inline int mchp_i2s_mcc_period_to_maxburst(int period_size, int sample_size) +{ + int p_size = period_size; + int s_size = sample_size; + + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_I2SMCC_DMA_8_WORD_CHUNK)) + return MCHP_I2SMCC_DMA_8_WORD_CHUNK; + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_I2SMCC_DMA_4_WORD_CHUNK)) + return MCHP_I2SMCC_DMA_4_WORD_CHUNK; + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_I2SMCC_DMA_2_WORD_CHUNK)) + return MCHP_I2SMCC_DMA_2_WORD_CHUNK; + return MCHP_I2SMCC_DMA_1_WORD_CHUNK; +} + static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { unsigned long rate = 0; struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); + int sample_bytes = params_physical_width(params) / 8; + int period_bytes = params_period_size(params) * + params_channels(params) * sample_bytes; + int maxburst; u32 mra = 0; u32 mrb = 0; unsigned int channels = params_channels(params); @@ -519,9 +546,9 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, int ret; bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - dev_dbg(dev->dev, "%s() rate=%u format=%#x width=%u channels=%u\n", + dev_dbg(dev->dev, "%s() rate=%u format=%#x width=%u channels=%u period_bytes=%d\n", __func__, params_rate(params), params_format(params), - params_width(params), params_channels(params)); + params_width(params), params_channels(params), period_bytes); switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -630,11 +657,12 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, * We must have the same burst size configured * in the DMA transfer and in out IP */ - mrb |= MCHP_I2SMCC_MRB_DMACHUNK(channels); + maxburst = mchp_i2s_mcc_period_to_maxburst(period_bytes, sample_bytes); + mrb |= MCHP_I2SMCC_MRB_DMACHUNK(maxburst); if (is_playback) - dev->playback.maxburst = 1 << (fls(channels) - 1); + dev->playback.maxburst = maxburst; else - dev->capture.maxburst = 1 << (fls(channels) - 1); + dev->capture.maxburst = maxburst; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: base-commit: fdadd93817f124fd0ea6ef251d4a1068b7feceba -- 2.34.1

1 0

[PATCH 2/2 4.19.y] ALSA: usb-audio: Fix gpf in snd_usb_pipe_sanity_check
by Hillf Danton 05 Sep '24

05 Sep '24

[ Upstream commit 5d78e1c2b7f4be00bbe62141603a631dc7812f35 ] syzbot found the following crash on: general protection fault: 0000 [#1] SMP KASAN RIP: 0010:snd_usb_pipe_sanity_check+0x80/0x130 sound/usb/helper.c:75 Call Trace: snd_usb_motu_microbookii_communicate.constprop.0+0xa0/0x2fb sound/usb/quirks.c:1007 snd_usb_motu_microbookii_boot_quirk sound/usb/quirks.c:1051 [inline] snd_usb_apply_boot_quirk.cold+0x163/0x370 sound/usb/quirks.c:1280 usb_audio_probe+0x2ec/0x2010 sound/usb/card.c:576 usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361 really_probe+0x281/0x650 drivers/base/dd.c:548 .... It was introduced in commit 801ebf1043ae for checking pipe and endpoint types. It is fixed by adding a check of the ep pointer in question. BugLink: https://syzkaller.appspot.com/bug?extid=d59c4387bfb6eced94e2 Reported-by: syzbot <syzbot+d59c4387bfb6eced94e2(a)syzkaller.appspotmail.com> Fixes: 801ebf1043ae ("ALSA: usb-audio: Sanity checks for each pipe and EP types") Cc: Andrey Konovalov <andreyknvl(a)google.com> Signed-off-by: Hillf Danton <hdanton(a)sina.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/usb/helper.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/helper.c b/sound/usb/helper.c index b1cc9499c57e..029489b490ca 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -85,7 +85,7 @@ int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe) struct usb_host_endpoint *ep; ep = usb_pipe_endpoint(dev, pipe); - if (usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)]) + if (!ep || usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)]) return -EINVAL; return 0; } -- 2.45.2

1 0

[PATCH 0/2 4.19.y] ALSA: usb-audio: sanity checks for pipes
by Dan Carpenter 05 Sep '24

05 Sep '24

Backporting the sanity checks on pipes seems like a good idea. These are basically the same as Takashi Iwai's patches upstream. The difference is that upstream added two sanity checks to code that doesn't exist in 4.19. I had talked about backporting fcc2cc1f3561 ("USB: move snd_usb_pipe_sanity_check into the USB core") but that's just a refactor and not a bug fix. Hillf Danton (1): ALSA: usb-audio: Fix gpf in snd_usb_pipe_sanity_check Takashi Iwai (1): ALSA: usb-audio: Sanity checks for each pipe and EP types sound/usb/helper.c | 17 +++++++++++++++++ sound/usb/helper.h | 1 + sound/usb/quirks.c | 14 +++++++++++--- 3 files changed, 29 insertions(+), 3 deletions(-) -- 2.45.2

1 0

[PATCH] ASoC/HDA cs35l56: Remove redundant IRQ handling
by Simon Trimmer 05 Sep '24

05 Sep '24

The IRQ handling in the cs35l56 driver was purely informational. It was not necessary to support the HDA or ASoC driver functionality and added unnecessary complexity to the drivers. As the IRQ signal GPIO line could be connected and shared with other components the handling is replaced with a regmap patch to ensure the cs35l56 IRQ sources are masked and will not generate interrupts that go unhandled. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- v1 - As this changes both the ASoC and HDA cs35l56 device drivers we considered a series of patches that stepped through the removal of the common code but thought that the co-ordination would be unnecessarily complicated so we initially offer a combined patch that that applies to both Mark and Takashi's for-next branches so that either route can be taken. include/sound/cs35l56.h | 4 -- sound/pci/hda/cs35l56_hda.c | 53 +-------------- sound/pci/hda/cs35l56_hda_i2c.c | 9 +-- sound/pci/hda/cs35l56_hda_spi.c | 9 +-- sound/soc/codecs/cs35l56-i2c.c | 2 - sound/soc/codecs/cs35l56-sdw.c | 104 +----------------------------- sound/soc/codecs/cs35l56-shared.c | 101 +---------------------------- sound/soc/codecs/cs35l56-spi.c | 2 - sound/soc/codecs/cs35l56.c | 92 ++------------------------ sound/soc/codecs/cs35l56.h | 10 --- 10 files changed, 14 insertions(+), 372 deletions(-) diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index 94e8185c4795..acc43d85f5df 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -257,8 +257,6 @@ struct cs35l56_base { struct device *dev; struct regmap *regmap; - int irq; - struct mutex irq_lock; u8 type; u8 rev; bool init_done; @@ -298,8 +296,6 @@ int cs35l56_wait_for_firmware_boot(struct cs35l56_base *cs35l56_base); void cs35l56_wait_control_port_ready(void); void cs35l56_wait_min_reset_pulse(void); void cs35l56_system_reset(struct cs35l56_base *cs35l56_base, bool is_soundwire); -int cs35l56_irq_request(struct cs35l56_base *cs35l56_base, int irq); -irqreturn_t cs35l56_irq(int irq, void *data); int cs35l56_is_fw_reload_needed(struct cs35l56_base *cs35l56_base); int cs35l56_runtime_suspend_common(struct cs35l56_base *cs35l56_base); int cs35l56_runtime_resume_common(struct cs35l56_base *cs35l56_base, bool is_soundwire); diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 31cc92bac89a..8ec06a3f4765 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -626,8 +626,6 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) goto err_fw_release; } - mutex_lock(&cs35l56->base.irq_lock); - /* * If the firmware hasn't been patched it must be shutdown before * doing a full patch and reset afterwards. If it is already @@ -637,14 +635,14 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) if (firmware_missing && (wmfw_firmware || coeff_firmware)) { ret = cs35l56_firmware_shutdown(&cs35l56->base); if (ret) - goto err; + goto err_fw_release; } ret = cs_dsp_power_up(&cs35l56->cs_dsp, wmfw_firmware, wmfw_filename, coeff_firmware, coeff_filename, "misc"); if (ret) { dev_dbg(cs35l56->base.dev, "%s: cs_dsp_power_up ret %d\n", __func__, ret); - goto err; + goto err_fw_release; } if (wmfw_filename) @@ -687,8 +685,6 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) err_powered_up: if (!cs35l56->base.fw_patched) cs_dsp_power_down(&cs35l56->cs_dsp); -err: - mutex_unlock(&cs35l56->base.irq_lock); err_fw_release: cs35l56_hda_release_firmware_files(wmfw_firmware, wmfw_filename, coeff_firmware, coeff_filename); @@ -778,15 +774,6 @@ static int cs35l56_hda_system_suspend(struct device *dev) cs35l56->suspended = true; - /* - * The interrupt line is normally shared, but after we start suspending - * we can't check if our device is the source of an interrupt, and can't - * clear it. Prevent this race by temporarily disabling the parent irq - * until we reach _no_irq. - */ - if (cs35l56->base.irq) - disable_irq(cs35l56->base.irq); - return pm_runtime_force_suspend(dev); } @@ -806,34 +793,6 @@ static int cs35l56_hda_system_suspend_late(struct device *dev) return 0; } -static int cs35l56_hda_system_suspend_no_irq(struct device *dev) -{ - struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); - - /* Handlers are now disabled so the parent IRQ can safely be re-enabled. */ - if (cs35l56->base.irq) - enable_irq(cs35l56->base.irq); - - return 0; -} - -static int cs35l56_hda_system_resume_no_irq(struct device *dev) -{ - struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); - - /* - * WAKE interrupts unmask if the CS35L56 hibernates, which can cause - * spurious interrupts, and the interrupt line is normally shared. - * We can't check if our device is the source of an interrupt, and can't - * clear it, until it has fully resumed. Prevent this race by temporarily - * disabling the parent irq until we complete resume(). - */ - if (cs35l56->base.irq) - disable_irq(cs35l56->base.irq); - - return 0; -} - static int cs35l56_hda_system_resume_early(struct device *dev) { struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); @@ -856,11 +815,8 @@ static int cs35l56_hda_system_resume(struct device *dev) struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); int ret; - /* Undo pm_runtime_force_suspend() before re-enabling the irq */ + /* Undo pm_runtime_force_suspend() */ ret = pm_runtime_force_resume(dev); - if (cs35l56->base.irq) - enable_irq(cs35l56->base.irq); - if (ret) return ret; @@ -987,7 +943,6 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) { int ret; - mutex_init(&cs35l56->base.irq_lock); dev_set_drvdata(cs35l56->base.dev, cs35l56); INIT_WORK(&cs35l56->dsp_work, cs35l56_hda_dsp_work); @@ -1120,8 +1075,6 @@ const struct dev_pm_ops cs35l56_hda_pm_ops = { SYSTEM_SLEEP_PM_OPS(cs35l56_hda_system_suspend, cs35l56_hda_system_resume) LATE_SYSTEM_SLEEP_PM_OPS(cs35l56_hda_system_suspend_late, cs35l56_hda_system_resume_early) - NOIRQ_SYSTEM_SLEEP_PM_OPS(cs35l56_hda_system_suspend_no_irq, - cs35l56_hda_system_resume_no_irq) }; EXPORT_SYMBOL_NS_GPL(cs35l56_hda_pm_ops, SND_HDA_SCODEC_CS35L56); diff --git a/sound/pci/hda/cs35l56_hda_i2c.c b/sound/pci/hda/cs35l56_hda_i2c.c index 40f2f97944d5..822c1251f703 100644 --- a/sound/pci/hda/cs35l56_hda_i2c.c +++ b/sound/pci/hda/cs35l56_hda_i2c.c @@ -34,14 +34,7 @@ static int cs35l56_hda_i2c_probe(struct i2c_client *clt) return ret; } - ret = cs35l56_hda_common_probe(cs35l56, id->driver_data, clt->addr); - if (ret) - return ret; - ret = cs35l56_irq_request(&cs35l56->base, clt->irq); - if (ret < 0) - cs35l56_hda_remove(cs35l56->base.dev); - - return ret; + return cs35l56_hda_common_probe(cs35l56, id->driver_data, clt->addr); } static void cs35l56_hda_i2c_remove(struct i2c_client *clt) diff --git a/sound/pci/hda/cs35l56_hda_spi.c b/sound/pci/hda/cs35l56_hda_spi.c index 7f02155fe61e..37a159546e59 100644 --- a/sound/pci/hda/cs35l56_hda_spi.c +++ b/sound/pci/hda/cs35l56_hda_spi.c @@ -34,14 +34,7 @@ static int cs35l56_hda_spi_probe(struct spi_device *spi) return ret; } - ret = cs35l56_hda_common_probe(cs35l56, id->driver_data, spi_get_chipselect(spi, 0)); - if (ret) - return ret; - ret = cs35l56_irq_request(&cs35l56->base, spi->irq); - if (ret < 0) - cs35l56_hda_remove(cs35l56->base.dev); - - return ret; + return cs35l56_hda_common_probe(cs35l56, id->driver_data, spi_get_chipselect(spi, 0)); } static void cs35l56_hda_spi_remove(struct spi_device *spi) diff --git a/sound/soc/codecs/cs35l56-i2c.c b/sound/soc/codecs/cs35l56-i2c.c index 2bd2ff75cd50..4419a94c5e73 100644 --- a/sound/soc/codecs/cs35l56-i2c.c +++ b/sound/soc/codecs/cs35l56-i2c.c @@ -41,8 +41,6 @@ static int cs35l56_i2c_probe(struct i2c_client *client) return ret; ret = cs35l56_init(cs35l56); - if (ret == 0) - ret = cs35l56_irq_request(&cs35l56->base, client->irq); if (ret < 0) cs35l56_remove(cs35l56); diff --git a/sound/soc/codecs/cs35l56-sdw.c b/sound/soc/codecs/cs35l56-sdw.c index 7c9a17fe2195..134631eff861 100644 --- a/sound/soc/codecs/cs35l56-sdw.c +++ b/sound/soc/codecs/cs35l56-sdw.c @@ -266,75 +266,14 @@ static void cs35l56_sdw_init(struct sdw_slave *peripheral) } ret = cs35l56_init(cs35l56); - if (ret < 0) { + if (ret < 0) regcache_cache_only(cs35l56->base.regmap, true); - goto out; - } - - /* - * cs35l56_init can return with !init_done if it triggered - * a soft reset. - */ - if (cs35l56->base.init_done) { - /* Enable SoundWire interrupts */ - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_MASK_1, - CS35L56_SDW_INT_MASK_CODEC_IRQ); - } out: pm_runtime_mark_last_busy(cs35l56->base.dev); pm_runtime_put_autosuspend(cs35l56->base.dev); } -static int cs35l56_sdw_interrupt(struct sdw_slave *peripheral, - struct sdw_slave_intr_status *status) -{ - struct cs35l56_private *cs35l56 = dev_get_drvdata(&peripheral->dev); - - /* SoundWire core holds our pm_runtime when calling this function. */ - - dev_dbg(cs35l56->base.dev, "int control_port=%#x\n", status->control_port); - - if ((status->control_port & SDW_SCP_INT1_IMPL_DEF) == 0) - return 0; - - /* - * Prevent bus manager suspending and possibly issuing a - * bus-reset before the queued work has run. - */ - pm_runtime_get_noresume(cs35l56->base.dev); - - /* - * Mask and clear until it has been handled. The read of GEN_INT_STAT_1 - * is required as per the SoundWire spec for interrupt status bits - * to clear. GEN_INT_MASK_1 masks the _inputs_ to GEN_INT_STAT1. - * None of the interrupts are time-critical so use the - * power-efficient queue. - */ - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0); - sdw_read_no_pm(peripheral, CS35L56_SDW_GEN_INT_STAT_1); - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF); - queue_work(system_power_efficient_wq, &cs35l56->sdw_irq_work); - - return 0; -} - -static void cs35l56_sdw_irq_work(struct work_struct *work) -{ - struct cs35l56_private *cs35l56 = container_of(work, - struct cs35l56_private, - sdw_irq_work); - - cs35l56_irq(-1, &cs35l56->base); - - /* unmask interrupts */ - if (!cs35l56->sdw_irq_no_unmask) - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, - CS35L56_SDW_INT_MASK_CODEC_IRQ); - - pm_runtime_put_autosuspend(cs35l56->base.dev); -} - static int cs35l56_sdw_read_prop(struct sdw_slave *peripheral) { struct cs35l56_private *cs35l56 = dev_get_drvdata(&peripheral->dev); @@ -349,7 +288,7 @@ static int cs35l56_sdw_read_prop(struct sdw_slave *peripheral) prop->sink_ports = BIT(CS35L56_SDW1_PLAYBACK_PORT); prop->paging_support = true; prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; - prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY | SDW_SCP_INT1_IMPL_DEF; + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; /* DP1 - playback */ ports[0].num = CS35L56_SDW1_PLAYBACK_PORT; @@ -406,7 +345,6 @@ static int __maybe_unused cs35l56_sdw_clk_stop(struct sdw_slave *peripheral, static const struct sdw_slave_ops cs35l56_sdw_ops = { .read_prop = cs35l56_sdw_read_prop, - .interrupt_callback = cs35l56_sdw_interrupt, .update_status = cs35l56_sdw_update_status, #ifdef DEBUG .clk_stop = cs35l56_sdw_clk_stop, @@ -465,10 +403,6 @@ static int __maybe_unused cs35l56_sdw_runtime_resume(struct device *dev) if (ret) return ret; - /* Re-enable SoundWire interrupts */ - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, - CS35L56_SDW_INT_MASK_CODEC_IRQ); - return 0; } @@ -479,32 +413,9 @@ static int __maybe_unused cs35l56_sdw_system_suspend(struct device *dev) if (!cs35l56->base.init_done) return 0; - /* - * Disable SoundWire interrupts. - * Flush - don't cancel because that could leave an unbalanced pm_runtime_get. - */ - cs35l56->sdw_irq_no_unmask = true; - flush_work(&cs35l56->sdw_irq_work); - - /* Mask interrupts and flush in case sdw_irq_work was queued again */ - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0); - sdw_read_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1); - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF); - flush_work(&cs35l56->sdw_irq_work); - return cs35l56_system_suspend(dev); } -static int __maybe_unused cs35l56_sdw_system_resume(struct device *dev) -{ - struct cs35l56_private *cs35l56 = dev_get_drvdata(dev); - - cs35l56->sdw_irq_no_unmask = false; - /* runtime_resume re-enables the interrupt */ - - return cs35l56_system_resume(dev); -} - static int cs35l56_sdw_probe(struct sdw_slave *peripheral, const struct sdw_device_id *id) { struct device *dev = &peripheral->dev; @@ -517,7 +428,6 @@ static int cs35l56_sdw_probe(struct sdw_slave *peripheral, const struct sdw_devi cs35l56->base.dev = dev; cs35l56->sdw_peripheral = peripheral; - INIT_WORK(&cs35l56->sdw_irq_work, cs35l56_sdw_irq_work); dev_set_drvdata(dev, cs35l56); @@ -542,13 +452,6 @@ static int cs35l56_sdw_remove(struct sdw_slave *peripheral) { struct cs35l56_private *cs35l56 = dev_get_drvdata(&peripheral->dev); - /* Disable SoundWire interrupts */ - cs35l56->sdw_irq_no_unmask = true; - cancel_work_sync(&cs35l56->sdw_irq_work); - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0); - sdw_read_no_pm(peripheral, CS35L56_SDW_GEN_INT_STAT_1); - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF); - cs35l56_remove(cs35l56); return 0; @@ -556,9 +459,8 @@ static int cs35l56_sdw_remove(struct sdw_slave *peripheral) static const struct dev_pm_ops cs35l56_sdw_pm = { SET_RUNTIME_PM_OPS(cs35l56_sdw_runtime_suspend, cs35l56_sdw_runtime_resume, NULL) - SYSTEM_SLEEP_PM_OPS(cs35l56_sdw_system_suspend, cs35l56_sdw_system_resume) + SYSTEM_SLEEP_PM_OPS(cs35l56_sdw_system_suspend, cs35l56_system_resume) LATE_SYSTEM_SLEEP_PM_OPS(cs35l56_system_suspend_late, cs35l56_system_resume_early) - /* NOIRQ stage not needed, SoundWire doesn't use a hard IRQ */ }; static const struct sdw_device_id cs35l56_sdw_id[] = { diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index e45e9ae01bc6..55adbc3d26ae 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -37,6 +37,7 @@ static const struct reg_sequence cs35l56_patch[] = { { CS35L56_SWIRE_DP3_CH3_INPUT, 0x00000029 }, { CS35L56_SWIRE_DP3_CH4_INPUT, 0x00000028 }, { CS35L56_IRQ1_MASK_18, 0x1f7df0ff }, + { CS35L56_IRQ1_MASK_20, 0x15fff19c }, /* These are not reset by a soft-reset, so patch to defaults. */ { CS35L56_MAIN_RENDER_USER_MUTE, 0x00000000 }, @@ -70,12 +71,6 @@ static const struct reg_default cs35l56_reg_defaults[] = { { CS35L56_SWIRE_DP3_CH2_INPUT, 0x00000019 }, { CS35L56_SWIRE_DP3_CH3_INPUT, 0x00000029 }, { CS35L56_SWIRE_DP3_CH4_INPUT, 0x00000028 }, - { CS35L56_IRQ1_MASK_1, 0x83ffffff }, - { CS35L56_IRQ1_MASK_2, 0xffff7fff }, - { CS35L56_IRQ1_MASK_4, 0xe0ffffff }, - { CS35L56_IRQ1_MASK_8, 0xfc000fff }, - { CS35L56_IRQ1_MASK_18, 0x1f7df0ff }, - { CS35L56_IRQ1_MASK_20, 0x15c00000 }, { CS35L56_MAIN_RENDER_USER_MUTE, 0x00000000 }, { CS35L56_MAIN_RENDER_USER_VOLUME, 0x00000000 }, { CS35L56_MAIN_POSTURE_NUMBER, 0x00000000 }, @@ -329,91 +324,6 @@ void cs35l56_system_reset(struct cs35l56_base *cs35l56_base, bool is_soundwire) } EXPORT_SYMBOL_NS_GPL(cs35l56_system_reset, SND_SOC_CS35L56_SHARED); -int cs35l56_irq_request(struct cs35l56_base *cs35l56_base, int irq) -{ - int ret; - - if (irq < 1) - return 0; - - ret = devm_request_threaded_irq(cs35l56_base->dev, irq, NULL, cs35l56_irq, - IRQF_ONESHOT | IRQF_SHARED | IRQF_TRIGGER_LOW, - "cs35l56", cs35l56_base); - if (!ret) - cs35l56_base->irq = irq; - else - dev_err(cs35l56_base->dev, "Failed to get IRQ: %d\n", ret); - - return ret; -} -EXPORT_SYMBOL_NS_GPL(cs35l56_irq_request, SND_SOC_CS35L56_SHARED); - -irqreturn_t cs35l56_irq(int irq, void *data) -{ - struct cs35l56_base *cs35l56_base = data; - unsigned int status1 = 0, status8 = 0, status20 = 0; - unsigned int mask1, mask8, mask20; - unsigned int val; - int rv; - - irqreturn_t ret = IRQ_NONE; - - if (!cs35l56_base->init_done) - return IRQ_NONE; - - mutex_lock(&cs35l56_base->irq_lock); - - rv = pm_runtime_resume_and_get(cs35l56_base->dev); - if (rv < 0) { - dev_err(cs35l56_base->dev, "irq: failed to get pm_runtime: %d\n", rv); - goto err_unlock; - } - - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_STATUS, &val); - if ((val & CS35L56_IRQ1_STS_MASK) == 0) { - dev_dbg(cs35l56_base->dev, "Spurious IRQ: no pending interrupt\n"); - goto err; - } - - /* Ack interrupts */ - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_EINT_1, &status1); - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_MASK_1, &mask1); - status1 &= ~mask1; - regmap_write(cs35l56_base->regmap, CS35L56_IRQ1_EINT_1, status1); - - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_EINT_8, &status8); - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_MASK_8, &mask8); - status8 &= ~mask8; - regmap_write(cs35l56_base->regmap, CS35L56_IRQ1_EINT_8, status8); - - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_EINT_20, &status20); - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_MASK_20, &mask20); - status20 &= ~mask20; - /* We don't want EINT20 but they default to unmasked: force mask */ - regmap_write(cs35l56_base->regmap, CS35L56_IRQ1_MASK_20, 0xffffffff); - - dev_dbg(cs35l56_base->dev, "%s: %#x %#x\n", __func__, status1, status8); - - /* Check to see if unmasked bits are active */ - if (!status1 && !status8 && !status20) - goto err; - - if (status1 & CS35L56_AMP_SHORT_ERR_EINT1_MASK) - dev_crit(cs35l56_base->dev, "Amp short error\n"); - - if (status8 & CS35L56_TEMP_ERR_EINT1_MASK) - dev_crit(cs35l56_base->dev, "Overtemp error\n"); - - ret = IRQ_HANDLED; - -err: - pm_runtime_put(cs35l56_base->dev); -err_unlock: - mutex_unlock(&cs35l56_base->irq_lock); - - return ret; -} -EXPORT_SYMBOL_NS_GPL(cs35l56_irq, SND_SOC_CS35L56_SHARED); int cs35l56_is_fw_reload_needed(struct cs35l56_base *cs35l56_base) { @@ -763,15 +673,6 @@ int cs35l56_hw_init(struct cs35l56_base *cs35l56_base) cs35l56_base->type, cs35l56_base->secured ? "s" : "", cs35l56_base->rev, otpid, fw_ver >> 16, (fw_ver >> 8) & 0xff, fw_ver & 0xff, !fw_missing); - /* Wake source and *_BLOCKED interrupts default to unmasked, so mask them */ - regmap_write(cs35l56_base->regmap, CS35L56_IRQ1_MASK_20, 0xffffffff); - regmap_update_bits(cs35l56_base->regmap, CS35L56_IRQ1_MASK_1, - CS35L56_AMP_SHORT_ERR_EINT1_MASK, - 0); - regmap_update_bits(cs35l56_base->regmap, CS35L56_IRQ1_MASK_8, - CS35L56_TEMP_ERR_EINT1_MASK, - 0); - return 0; } EXPORT_SYMBOL_NS_GPL(cs35l56_hw_init, SND_SOC_CS35L56_SHARED); diff --git a/sound/soc/codecs/cs35l56-spi.c b/sound/soc/codecs/cs35l56-spi.c index b07b798b0b45..1382b8903ea7 100644 --- a/sound/soc/codecs/cs35l56-spi.c +++ b/sound/soc/codecs/cs35l56-spi.c @@ -39,8 +39,6 @@ static int cs35l56_spi_probe(struct spi_device *spi) return ret; ret = cs35l56_init(cs35l56); - if (ret == 0) - ret = cs35l56_irq_request(&cs35l56->base, spi->irq); if (ret < 0) cs35l56_remove(cs35l56); diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 757ade6373ed..cf2aee6cda6c 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -12,7 +12,6 @@ #include <linux/delay.h> #include <linux/err.h> #include <linux/gpio/consumer.h> -#include <linux/interrupt.h> #include <linux/math.h> #include <linux/module.h> #include <linux/pm.h> @@ -702,23 +701,9 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56, bool firmware_missing { int ret; - /* - * Disable SoundWire interrupts to prevent race with IRQ work. - * Setting sdw_irq_no_unmask prevents the handler re-enabling - * the SoundWire interrupt. - */ - if (cs35l56->sdw_peripheral) { - cs35l56->sdw_irq_no_unmask = true; - flush_work(&cs35l56->sdw_irq_work); - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0); - sdw_read_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1); - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF); - flush_work(&cs35l56->sdw_irq_work); - } - ret = cs35l56_firmware_shutdown(&cs35l56->base); if (ret) - goto err; + return; /* * Use wm_adsp to load and apply the firmware patch and coefficient files, @@ -728,11 +713,9 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56, bool firmware_missing ret = wm_adsp_power_up(&cs35l56->dsp, !!firmware_missing); if (ret) { dev_dbg(cs35l56->base.dev, "%s: wm_adsp_power_up ret %d\n", __func__, ret); - goto err; + return; } - mutex_lock(&cs35l56->base.irq_lock); - reinit_completion(&cs35l56->init_completion); cs35l56->soft_resetting = true; @@ -748,10 +731,10 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56, bool firmware_missing msecs_to_jiffies(5000))) { dev_err(cs35l56->base.dev, "%s: init_completion timed out (SDW)\n", __func__); - goto err_unlock; + return; } } else if (cs35l56_init(cs35l56)) { - goto err_unlock; + return; } regmap_clear_bits(cs35l56->base.regmap, CS35L56_PROTECTION_STATUS, @@ -760,16 +743,6 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56, bool firmware_missing if (cs35l56_write_cal(cs35l56) == 0) cs35l56_mbox_send(&cs35l56->base, CS35L56_MBOX_CMD_AUDIO_REINIT); - -err_unlock: - mutex_unlock(&cs35l56->base.irq_lock); -err: - /* Re-enable SoundWire interrupts */ - if (cs35l56->sdw_peripheral) { - cs35l56->sdw_irq_no_unmask = false; - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, - CS35L56_SDW_INT_MASK_CODEC_IRQ); - } } static void cs35l56_dsp_work(struct work_struct *work) @@ -956,15 +929,6 @@ int cs35l56_system_suspend(struct device *dev) if (cs35l56->component) flush_work(&cs35l56->dsp_work); - /* - * The interrupt line is normally shared, but after we start suspending - * we can't check if our device is the source of an interrupt, and can't - * clear it. Prevent this race by temporarily disabling the parent irq - * until we reach _no_irq. - */ - if (cs35l56->base.irq) - disable_irq(cs35l56->base.irq); - return pm_runtime_force_suspend(dev); } EXPORT_SYMBOL_GPL(cs35l56_system_suspend); @@ -991,40 +955,6 @@ int cs35l56_system_suspend_late(struct device *dev) } EXPORT_SYMBOL_GPL(cs35l56_system_suspend_late); -int cs35l56_system_suspend_no_irq(struct device *dev) -{ - struct cs35l56_private *cs35l56 = dev_get_drvdata(dev); - - dev_dbg(dev, "system_suspend_no_irq\n"); - - /* Handlers are now disabled so the parent IRQ can safely be re-enabled. */ - if (cs35l56->base.irq) - enable_irq(cs35l56->base.irq); - - return 0; -} -EXPORT_SYMBOL_GPL(cs35l56_system_suspend_no_irq); - -int cs35l56_system_resume_no_irq(struct device *dev) -{ - struct cs35l56_private *cs35l56 = dev_get_drvdata(dev); - - dev_dbg(dev, "system_resume_no_irq\n"); - - /* - * WAKE interrupts unmask if the CS35L56 hibernates, which can cause - * spurious interrupts, and the interrupt line is normally shared. - * We can't check if our device is the source of an interrupt, and can't - * clear it, until it has fully resumed. Prevent this race by temporarily - * disabling the parent irq until we complete resume(). - */ - if (cs35l56->base.irq) - disable_irq(cs35l56->base.irq); - - return 0; -} -EXPORT_SYMBOL_GPL(cs35l56_system_resume_no_irq); - int cs35l56_system_resume_early(struct device *dev) { struct cs35l56_private *cs35l56 = dev_get_drvdata(dev); @@ -1065,11 +995,8 @@ int cs35l56_system_resume(struct device *dev) */ cs35l56_wait_control_port_ready(); - /* Undo pm_runtime_force_suspend() before re-enabling the irq */ + /* Undo pm_runtime_force_suspend() */ ret = pm_runtime_force_resume(dev); - if (cs35l56->base.irq) - enable_irq(cs35l56->base.irq); - if (ret) return ret; @@ -1267,7 +1194,6 @@ int cs35l56_common_probe(struct cs35l56_private *cs35l56) int ret; init_completion(&cs35l56->init_completion); - mutex_init(&cs35l56->base.irq_lock); cs35l56->base.cal_index = -1; cs35l56->speaker_id = -ENOENT; @@ -1428,13 +1354,6 @@ void cs35l56_remove(struct cs35l56_private *cs35l56) { cs35l56->base.init_done = false; - /* - * WAKE IRQs unmask if CS35L56 hibernates so free the handler to - * prevent it racing with remove(). - */ - if (cs35l56->base.irq) - devm_free_irq(cs35l56->base.dev, cs35l56->base.irq, &cs35l56->base); - flush_workqueue(cs35l56->dsp_wq); destroy_workqueue(cs35l56->dsp_wq); @@ -1454,7 +1373,6 @@ EXPORT_NS_GPL_DEV_PM_OPS(cs35l56_pm_ops_i2c_spi, SND_SOC_CS35L56_CORE) = { SET_RUNTIME_PM_OPS(cs35l56_runtime_suspend_i2c_spi, cs35l56_runtime_resume_i2c_spi, NULL) SYSTEM_SLEEP_PM_OPS(cs35l56_system_suspend, cs35l56_system_resume) LATE_SYSTEM_SLEEP_PM_OPS(cs35l56_system_suspend_late, cs35l56_system_resume_early) - NOIRQ_SYSTEM_SLEEP_PM_OPS(cs35l56_system_suspend_no_irq, cs35l56_system_resume_no_irq) }; #endif diff --git a/sound/soc/codecs/cs35l56.h b/sound/soc/codecs/cs35l56.h index 8a987ec01507..0ebae11c1cd0 100644 --- a/sound/soc/codecs/cs35l56.h +++ b/sound/soc/codecs/cs35l56.h @@ -16,10 +16,6 @@ #include <sound/cs35l56.h> #include "wm_adsp.h" -#define CS35L56_SDW_GEN_INT_STAT_1 0xc0 -#define CS35L56_SDW_GEN_INT_MASK_1 0xc1 -#define CS35L56_SDW_INT_MASK_CODEC_IRQ BIT(0) - #define CS35L56_SDW_INVALID_BUS_SCALE 0xf #define CS35L56_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) @@ -38,8 +34,6 @@ struct cs35l56_private { struct snd_soc_component *component; struct regulator_bulk_data supplies[CS35L56_NUM_BULK_SUPPLIES]; struct sdw_slave *sdw_peripheral; - struct work_struct sdw_irq_work; - bool sdw_irq_no_unmask; bool soft_resetting; bool sdw_attached; struct completion init_completion; @@ -57,12 +51,8 @@ extern const struct dev_pm_ops cs35l56_pm_ops_i2c_spi; int cs35l56_system_suspend(struct device *dev); int cs35l56_system_suspend_late(struct device *dev); -int cs35l56_system_suspend_no_irq(struct device *dev); -int cs35l56_system_resume_no_irq(struct device *dev); int cs35l56_system_resume_early(struct device *dev); int cs35l56_system_resume(struct device *dev); -irqreturn_t cs35l56_irq(int irq, void *data); -int cs35l56_irq_request(struct cs35l56_base *cs35l56_base, int irq); int cs35l56_common_probe(struct cs35l56_private *cs35l56); int cs35l56_init(struct cs35l56_private *cs35l56); void cs35l56_remove(struct cs35l56_private *cs35l56); -- 2.43.0

2 5

[PATCH v26 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 04 Sep '24

04 Sep '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userspace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platform card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v26: - Cleaned up drivers based on errors from checkpatch - Fixed several typos using codespell - Removed any vendor specific notation from USB SND offload mixer patch Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addressed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnecessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userspsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (31): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: Allow for secondary interrupter to set IMOD usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: qcom: Use card and PCM index from QMI request ALSA: usb-audio: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 429 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 34 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 54 +- drivers/usb/host/xhci-sideband.c | 425 ++++ drivers/usb/host/xhci.c | 45 +- drivers/usb/host/xhci.h | 19 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 71 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 135 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 194 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 391 ++++ sound/soc/qcom/sm8250.c | 24 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 30 + sound/soc/soc-usb.c | 338 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 4 +- sound/usb/card.c | 110 + sound/usb/card.h | 17 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1968 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 863 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 164 ++ 53 files changed, 5820 insertions(+), 56 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

2 64

[PATCH] ASoC: dt-bindings: renesas,rsnd: add post-init-providers property
by Kuninori Morimoto 04 Sep '24

04 Sep '24

sometimes, fwnode dependency cycles can be broken, and in such case, rsnd driver will never be probed. To avoid such case, we would like to use post-init-providers to avoid it. Enable post-init-providers property on rsnd. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 07ec6247d9def..cb44478d9e420 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -112,6 +112,10 @@ properties: description: List of necessary clock names. # details are defined below + post-init-providers: + description: ignore fw_devlink dependency cycles and avoid driver probe dead-lock + $ref: /schemas/types.yaml#/definitions/phandle-array + # ports is below port: $ref: audio-graph-port.yaml#/definitions/port-base -- 2.43.0

3 2

[PATCH 00/12] Add i2s/tdm support for acp7.0 and acp7.1 platforms
by Venkata Prasad Potturu 04 Sep '24

04 Sep '24

1. Refactor acp generic driver to support all platforms. 2. Add i2s/tdm and support for acp7.0 and acp7.1 platforms. Venkata Prasad Potturu (12): ASoC: amd: acp: Refactor TDM slots selction based on acp revision id ASoC: amd: acp: Refactor I2S dai driver ASoC: amd: acp: Update pcm hardware capabilities for acp6.3 platform ASoC: amd: acp: Add I2S TDM support for acp6.3 platform ASoC: amd: acp: Update pcm hardware capabilities for acp7.0 platform ASoC: amd: acp: Add I2S master clock generation support for acp7.0 platform ASoC: amd: acp: Set i2s clock for acp7.0 platform ASoC: amd: acp: Modify max channels and sample rate support for acp70 dai driver ASoC: amd: acp: Add I2S TDM support for acp7.0 platform ASoC: amd: acp: Add pte configuration for ACP7.0 platform ASoC: amd: acp: Add i2s master clock generation support for acp7.1 platform ASoC: amd: acp: Add I2S TDM support for acp7.1 platform sound/soc/amd/acp/acp-i2s.c | 188 ++++++++++++++++++++------ sound/soc/amd/acp/acp-legacy-common.c | 32 ++--- sound/soc/amd/acp/acp-pdm.c | 7 +- sound/soc/amd/acp/acp-platform.c | 117 ++++++++++++++-- sound/soc/amd/acp/acp70.c | 73 +++++++--- sound/soc/amd/acp/amd.h | 24 ++-- sound/soc/amd/acp/chip_offset_byte.h | 84 +++++++----- 7 files changed, 387 insertions(+), 138 deletions(-) -- 2.39.2

2 13

[PATCH] ALSA: hda/realtek: Support mute LED on HP Laptop 14-dq2xxx
by Maximilien Perreault 04 Sep '24

04 Sep '24

The mute LED on this HP laptop uses ALC236 and requires a quirk to function. This patch enables the existing quirk for the device. Signed-off-by: Maximilien Perreault <maximilienperreault(a)gmail.com> --- 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 f67190a8c..6f96c0c67 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10183,6 +10183,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f6, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), + SND_PCI_QUIRK(0x103c, 0x87fd, "HP Laptop 14-dq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x87fe, "HP Laptop 15s-fq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), -- 2.46.0

1 0

[PATCH] soc: qcom: apr: Fix some parenthesis alignment coding style issues
by Riyan Dhiman 04 Sep '24

04 Sep '24

Adhere to Linux kernel coding style. Issue reported by checkpatch: - CHECK: Alignment should match open parenthesis It has no functional changes. Signed-off-by: Riyan Dhiman <riyandhiman14(a)gmail.com> --- drivers/soc/qcom/apr.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/soc/qcom/apr.c b/drivers/soc/qcom/apr.c index 4fbff3a890e2..db1326bdb170 100644 --- a/drivers/soc/qcom/apr.c +++ b/drivers/soc/qcom/apr.c @@ -88,7 +88,7 @@ void gpr_free_port(gpr_port_t *port) EXPORT_SYMBOL_GPL(gpr_free_port); gpr_port_t *gpr_alloc_port(struct apr_device *gdev, struct device *dev, - gpr_port_cb cb, void *priv) + gpr_port_cb cb, void *priv) { struct packet_router *pr = dev_get_drvdata(gdev->dev.parent); gpr_port_t *port; @@ -159,7 +159,7 @@ static void apr_dev_release(struct device *dev) } static int apr_callback(struct rpmsg_device *rpdev, void *buf, - int len, void *priv, u32 addr) + int len, void *priv, u32 addr) { struct packet_router *apr = dev_get_drvdata(&rpdev->dev); struct apr_rx_buf *abuf; @@ -222,9 +222,9 @@ static int apr_do_rx_callback(struct packet_router *apr, struct apr_rx_buf *abuf } if (hdr->src_domain >= APR_DOMAIN_MAX || - hdr->dest_domain >= APR_DOMAIN_MAX || - hdr->src_svc >= APR_SVC_MAX || - hdr->dest_svc >= APR_SVC_MAX) { + hdr->dest_domain >= APR_DOMAIN_MAX || + hdr->src_svc >= APR_SVC_MAX || + hdr->dest_svc >= APR_SVC_MAX) { dev_err(apr->dev, "APR: Wrong APR header\n"); return -EINVAL; } -- 2.46.0

1 0

Qestion: of property has dead-lock to call .probe()
by Kuninori Morimoto 04 Sep '24

04 Sep '24

Hi Saravana, Rob Cc Mark/ALSA-ML, Geert, Laurent I got dead-lock issue on drivers/of/property.c to call .probe() I'm now using v6.11-rc5. I'm using 3 devices, and these are connected via OF-Graph. These are connected like this so far. Let's call it as [Style-A] [Style-A] +---+ +---+ |(A)+--+ | +-+-+ | | | |(B)| +-+-+ | | |(C)+--+ | +---+ +---+ These are using port/endpoint to connect. It works well for now, no issue I have so far. (B) is connector for (A)-(C) data, thus, (A) and (C) should be probed first, and (B) needs probe later (For ALSA member, this (B) is Audio Graph Card2). Here, (A)-(C) can connect directly for some data. Now it is using both data connection (= (A)-(B)-(C) and (A)-(C)) Now, I want to disconnect (A)-(C) connection, like below Let's call it as [Style-B] [Style-B] +---+ +---+ |(A)+--+ | +---+ | | |(B)| +---+ | | |(C)+--+ | +---+ +---+ Then, it seems dead-lock happen. In my debug, it seems... - (B) is handled as supplier for (A). - (B) probe() is called, but it needs (A) info which is not yet probed. So it returns -EPROBE_DEFER. - Because (B) is not probed, (A) probe() never called In [Style-A], it seems __fwnode_link_cycle() is called for (A)-(B) connection, so the dead-lock will be solved. But it is not called for [Style-B]. Because of that (B) is always handled as supplier for (A). If I used below patch, and use "non-supplier" property on (B), this dead-lock issue was solved. But I know this is not a good solution. I think it is very normal connection, not super special. How can I solve this issue on correct way ? Or how can I indicate you my issue more detail ? I can add debug patch and test it if you can indicate it to me. ----------------------------------------------- diff --git a/drivers/of/property.c b/drivers/of/property.c index def152c61049..2f08210c2ea4 100644 --- a/drivers/of/property.c +++ b/drivers/of/property.c @@ -1502,11 +1502,22 @@ static struct device_node *parse_remote_endpoint(struct device_node *np, const char *prop_name, int index) { + struct device_node *node; + /* Return NULL for index > 0 to signify end of remote-endpoints. */ if (index > 0 || strcmp(prop_name, "remote-endpoint")) return NULL; - return of_graph_get_remote_port_parent(np); + node = of_graph_get_remote_port_parent(np); + + /* + * There is clearly non-supplier node which is connected via "remote-endpoint". + * Ignore it, otherwise dead-lock might occur + */ + if (of_property_present(node, "non-supplier")) + return NULL; + + return node; } ----------------------------------------------- Thank you for your help !! Best regards --- Kuninori Morimoto

2 2

[PATCH] ASoC: tlv320aic32x4: Add multi endpoint support
by Marek Vasut 04 Sep '24

04 Sep '24

Support multiple endpoints on TLV320AIC32x4 codec port when used in of_graph context. This patch allows to share the codec port between two CPU DAIs. Example: Custom STM32MP157C board uses TLV320AIC32x4 audio codec. This codec is connected to two serial audio interfaces, which are configured either as rx or tx. From AsoC point of view the topolgy is the following: // 2 CPU DAIs (SAI2A/B), 1 Codec (TLV320AIC32x4) Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0 Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0 In the DT two endpoints have to be associated to the codec port: tlv320aic32x4_port: port { tlv320aic32x4_tx_endpoint: endpoint@0 { remote-endpoint = <&sai2a_endpoint>; }; tlv320aic32x4_rx_endpoint: endpoint@1 { remote-endpoint = <&sai2b_endpoint>; }; }; However, when the audio graph card parses the codec nodes, it expects to find DAI interface indexes matching the endpoints indexes. The current patch forces the use of DAI id 0 for both endpoints, which allows to share the codec DAI between the two CPU DAIs for playback and capture streams respectively. Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Baojun Xu <baojun.xu(a)ti.com> Cc: Charles Keepax <ckeepax(a)opensource.cirrus.com> Cc: Christoph Niedermaier <cniedermaier(a)dh-electronics.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Kevin Lu <kevin-lu(a)ti.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Philipp Zabel <p.zabel(a)pengutronix.de> Cc: Shenghao Ding <shenghao-ding(a)ti.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: kernel(a)dh-electronics.com Cc: linux-sound(a)vger.kernel.org --- sound/soc/codecs/tlv320aic32x4.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 5c0c81da06dba..54ea4bc58c276 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1073,6 +1073,13 @@ static int aic32x4_component_probe(struct snd_soc_component *component) return 0; } +static int aic32x4_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + /* return dai id 0, whatever the endpoint index */ + return 0; +} + static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .probe = aic32x4_component_probe, .set_bias_level = aic32x4_set_bias_level, @@ -1082,6 +1089,7 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets), .dapm_routes = aic32x4_dapm_routes, .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), + .of_xlate_dai_id = aic32x4_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, @@ -1203,6 +1211,7 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4_tas2505 = .num_dapm_widgets = ARRAY_SIZE(aic32x4_tas2505_dapm_widgets), .dapm_routes = aic32x4_tas2505_dapm_routes, .num_dapm_routes = ARRAY_SIZE(aic32x4_tas2505_dapm_routes), + .of_xlate_dai_id = aic32x4_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, -- 2.45.2

2 1

[PATCH] ASOC: Intel: fix parenthesis alignment and space after cast coding style issue
by Riyan Dhiman 03 Sep '24

03 Sep '24

Adhere to Linux kernel coding style. Reported by checkpatch: - CHECK: Alignment should match open parenthesis - CHECK: No space is necessary after a cast It has no functional changes. Signed-off-by: Riyan Dhiman <riyandhiman14(a)gmail.com> --- sound/soc/intel/atom/sst/sst.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index e0357d257c6c..38d29d545b52 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -50,7 +50,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) union sst_imr_reg_mrfld imr; struct ipc_post *msg = NULL; unsigned int size; - struct intel_sst_drv *drv = (struct intel_sst_drv *) context; + struct intel_sst_drv *drv = (struct intel_sst_drv *)context; irqreturn_t retval = IRQ_HANDLED; /* Interrupt arrived, check src */ @@ -60,7 +60,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) /* Clear done bit */ spin_lock(&drv->ipc_spin_lock); header.full = sst_shim_read64(drv->shim, - drv->ipc_reg.ipcx); + drv->ipc_reg.ipcx); header.p.header_high.part.done = 0; sst_shim_write64(drv->shim, drv->ipc_reg.ipcx, header.full); @@ -92,7 +92,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) size = header.p.header_low_payload; if (sst_validate_mailbox_size(size)) { memcpy_fromio(msg->mailbox_data, - drv->mailbox + drv->mailbox_recv_offset, size); + drv->mailbox + drv->mailbox_recv_offset, size); } else { dev_err(drv->dev, "Mailbox not copied, payload size is: %u\n", size); @@ -114,7 +114,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) static irqreturn_t intel_sst_irq_thread_mrfld(int irq, void *context) { - struct intel_sst_drv *drv = (struct intel_sst_drv *) context; + struct intel_sst_drv *drv = (struct intel_sst_drv *)context; struct ipc_post *__msg, *msg; unsigned long irq_flags; @@ -146,8 +146,8 @@ static int sst_save_dsp_context_v2(struct intel_sst_drv *sst) int ret = 0; ret = sst_prepare_and_post_msg(sst, SST_TASK_ID_MEDIA, IPC_CMD, - IPC_PREP_D3, PIPE_RSVD, 0, NULL, NULL, - true, true, false, true); + IPC_PREP_D3, PIPE_RSVD, 0, NULL, NULL, + true, true, false, true); if (ret < 0) { dev_err(sst->dev, "not suspending FW!!, Err: %d\n", ret); @@ -228,7 +228,7 @@ static void sst_init_locks(struct intel_sst_drv *ctx) * we are fine with using unsigned short as dev_id type. */ int sst_alloc_drv_context(struct intel_sst_drv **ctx, - struct device *dev, unsigned short dev_id) + struct device *dev, unsigned short dev_id) { *ctx = devm_kzalloc(dev, sizeof(struct intel_sst_drv), GFP_KERNEL); if (!(*ctx)) @@ -242,7 +242,7 @@ int sst_alloc_drv_context(struct intel_sst_drv **ctx, EXPORT_SYMBOL_GPL(sst_alloc_drv_context); static ssize_t firmware_version_show(struct device *dev, - struct device_attribute *attr, char *buf) + struct device_attribute *attr, char *buf) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); @@ -302,7 +302,7 @@ int sst_context_init(struct intel_sst_drv *ctx) ctx->ipc_reg.ipcd = SST_IPCD + ctx->pdata->ipc_info->ipc_offset; dev_info(ctx->dev, "Got drv data max stream %d\n", - ctx->info.max_streams); + ctx->info.max_streams); for (i = 1; i <= ctx->info.max_streams; i++) { struct stream_info *stream = &ctx->streams[i]; @@ -325,7 +325,7 @@ int sst_context_init(struct intel_sst_drv *ctx) sst_shim_write64(ctx->shim, SST_IMRX, 0xFFFF0038); ctx->qos = devm_kzalloc(ctx->dev, - sizeof(struct pm_qos_request), GFP_KERNEL); + sizeof(struct pm_qos_request), GFP_KERNEL); if (!ctx->qos) { ret = -ENOMEM; goto do_free_mem; -- 2.46.0

1 0

ucm2: sof-soundwire: add RT1320 amplifier
by GitHub pull_request - opened 03 Sep '24

03 Sep '24

alsa-project/alsa-ucm-conf pull request #439 was opened from shumingfan: Add support for rt1320 amplifier Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/439 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/439.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: tas2781: Add Calibration Kcontrols for Chromebook
by Shenghao Ding 03 Sep '24

03 Sep '24

Add calibration related kcontrol for speaker impedance calibration and speaker leakage check for Chromebook Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Add channel no into the log for error debug - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/202407131… - renamae tasdevice_chn_switch to put it in a single line - With inverted check this entire function becomes neater in tasdev_chn_switch. - Add registers for TAS2563 and TAS2781 calibration. - Add cali_data_backup for regsiter restore after calibration. - Add is_user_space_calidata to store the flag where the calibrated from, user space or bin file from driver parsing. - tasdevice_chn_switch for chip switch among multiple chips(tas2563 or tas2781) - Add loading the calibrated values from user space in tasdev_load_calibrated_data - Correct no dsp no work, it can still work in bypass mode. - Add calibrated register setting for tas2563&tas2781. - Define tas2781_cali_control and tas2563_cali_control for calibration --- include/sound/tas2781.h | 68 +++ sound/soc/codecs/tas2781-comlib.c | 26 + sound/soc/codecs/tas2781-fmwlib.c | 60 +- sound/soc/codecs/tas2781-i2c.c | 961 +++++++++++++++++++++++++++++- 4 files changed, 1104 insertions(+), 11 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index dbda552398b5..a9ccbbd1945c 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -49,12 +49,59 @@ /*I2C Checksum */ #define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) +/* XM_340 */ +#define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x02, 0x4c) +/* XM_341 */ +#define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x02, 0x64) + /* Volume control */ #define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) #define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) #define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) +#define TAS2563_IDLE TASDEVICE_REG(0x00, 0x00, 0x3e) +#define TAS2563_PRM_R0_REG TASDEVICE_REG(0x00, 0x0f, 0x34) + +#define TAS2563_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x02, 0x70) +#define TAS2563_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x02, 0x48) + +#define TAS2563_PRM_ENFF_REG TASDEVICE_REG(0x00, 0x0d, 0x54) +#define TAS2563_PRM_DISTCK_REG TASDEVICE_REG(0x00, 0x0d, 0x58) +#define TAS2563_PRM_TE_SCTHR_REG TASDEVICE_REG(0x00, 0x0f, 0x60) +#define TAS2563_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x0d, 0x74) +#define TAS2563_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x0d, 0x7c) +/* prm_Int_B0 */ +#define TAS2563_TE_TA1_REG TASDEVICE_REG(0x00, 0x10, 0x0c) +/* prm_Int_A1 */ +#define TAS2563_TE_TA1_AT_REG TASDEVICE_REG(0x00, 0x10, 0x10) +/* prm_TE_Beta */ +#define TAS2563_TE_TA2_REG TASDEVICE_REG(0x00, 0x0f, 0x64) +/* prm_TE_Beta1 */ +#define TAS2563_TE_AT_REG TASDEVICE_REG(0x00, 0x0f, 0x68) +/* prm_TE_1_Beta1 */ +#define TAS2563_TE_DT_REG TASDEVICE_REG(0x00, 0x0f, 0x70) + +#define TAS2781_PRM_INT_MASK_REG TASDEVICE_REG(0x00, 0x00, 0x3b) +#define TAS2781_PRM_CLK_CFG_REG TASDEVICE_REG(0x00, 0x00, 0x5c) +#define TAS2781_PRM_RSVD_REG TASDEVICE_REG(0x00, 0x01, 0x19) +#define TAS2781_PRM_TEST_57_REG TASDEVICE_REG(0x00, 0xfd, 0x39) +#define TAS2781_PRM_TEST_62_REG TASDEVICE_REG(0x00, 0xfd, 0x3e) +#define TAS2781_PRM_PVDD_UVLO_REG TASDEVICE_REG(0x00, 0x00, 0x71) +#define TAS2781_PRM_CHNL_0_REG TASDEVICE_REG(0x00, 0x00, 0x03) +#define TAS2781_PRM_NG_CFG0_REG TASDEVICE_REG(0x00, 0x00, 0x35) +#define TAS2781_PRM_IDLE_CH_DET_REG TASDEVICE_REG(0x00, 0x00, 0x66) +#define TAS2781_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x14, 0x38) +#define TAS2781_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x14, 0x40) +#define TAS2781_PRM_SINEGAIN2_REG TASDEVICE_REG(0x00, 0x14, 0x44) + +#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xFD, 0x0D) +#define TAS2781_TEST_PAGE_UNLOCK 0x0D + +#define TAS2781_RUNTIME_LATCH_RE_REG TASDEVICE_REG(0x00, 0x00, 0x49) +#define TAS2781_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x62, 0x48) +#define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) + #define TASDEVICE_CMD_SING_W 0x1 #define TASDEVICE_CMD_BURST 0x2 #define TASDEVICE_CMD_DELAY 0x3 @@ -70,7 +117,15 @@ enum device_catlog_id { OTHERS }; +struct bulk_reg_val { + int reg; + unsigned char val[4]; + unsigned char val_len; + bool is_locked; +}; + struct tasdevice { + struct bulk_reg_val *cali_data_backup; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -81,9 +136,19 @@ struct tasdevice { bool is_loaderr; }; +struct cali_reg { + unsigned int r0_reg; + unsigned int r0_low_reg; + unsigned int invr0_reg; + unsigned int pow_reg; + unsigned int tlimit_reg; +}; + struct calidata { unsigned char *data; unsigned long total_sz; + struct cali_reg cali_reg_array; + unsigned int cali_dat_sz_per_dev; }; struct tasdevice_priv { @@ -119,6 +184,7 @@ struct tasdevice_priv { bool force_fwload_status; bool playback_started; bool isacpi; + bool is_user_space_calidata; unsigned int global_addr; int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, @@ -145,6 +211,8 @@ int tasdevice_init(struct tasdevice_priv *tas_priv); void tasdevice_remove(struct tasdevice_priv *tas_priv); int tasdevice_save_calibration(struct tasdevice_priv *tas_priv); void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv); +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn); int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *value); int tasdevice_dev_write(struct tasdevice_priv *tas_priv, diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 664c371796d6..1e0b3aa95749 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -88,6 +88,32 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, return ret; } +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + struct regmap *map = tas_priv->regmap; + int ret; + + if (client->addr != tasdev->dev_addr) { + client->addr = tasdev->dev_addr; + /* All devices share the same regmap, clear the page + * inside regmap once switching to another device. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + return ret; + } + return 1; + } + return 0; +} +EXPORT_SYMBOL_GPL(tasdev_chn_switch); + int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *val) { diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index f3a7605f0710..3de0132c345d 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2151,20 +2151,61 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) { + struct tasdevice_fw *cal_fmw = priv->tasdevice[i].cali_data_fmw; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *data = cali_data->data; struct tasdevice_calibration *cal; - struct tasdevice_fw *cal_fmw; + int k = i * (cali_data->cali_dat_sz_per_dev + 1); + int rc; - cal_fmw = priv->tasdevice[i].cali_data_fmw; + /* Load the calibrated data from cal bin file */ + if (!priv->is_user_space_calidata && cal_fmw) { + cal = cal_fmw->calibrations; - /* No calibrated data for current devices, playback will go ahead. */ - if (!cal_fmw) + if (cal) + load_calib_data(priv, &cal->dev_data); return; - - cal = cal_fmw->calibrations; - if (!cal) + } + if (!priv->is_user_space_calidata) + return; + /* load calibrated data from user space */ + if (data[k] != i) { + dev_err(priv->dev, "%s: no cal-data for dev %d from usr-spc\n", + __func__, i); return; + } + k++; - load_calib_data(priv, &cal->dev_data); + rc = tasdevice_dev_bulk_write(priv, i, p->r0_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_reg bulk_wr err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->r0_low_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_low_reg err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->invr0_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d invr0_reg err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->pow_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d pow_reg bulk_wr err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->tlimit_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d tlimit_reg err = %d\n", i, rc); + return; + } } int tasdevice_select_tuningprm_cfg(void *context, int prm_no, @@ -2259,9 +2300,10 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, tas_priv->tasdevice[i].cur_conf = cfg_no; } } - } else + } else { dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", __func__, cfg_no); + } status |= cfg_info[rca_conf_no]->active_dev; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 59fcce7fef7b..ebd71c7eaa59 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -33,6 +33,143 @@ #include <sound/tas2781-tlv.h> #include <asm/unaligned.h> +static const struct bulk_reg_val tas2563_cali_start_reg[] = { + { + .reg = TAS2563_IDLE, + .val_len = 1, + .val = { 0x00 }, + }, + { + .reg = TAS2563_PRM_ENFF_REG, + .val_len = 4, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_DISTCK_REG, + .val_len = 4, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_TE_SCTHR_REG, + .val_len = 4, + .val = { 0x7f, 0xff, 0xff, 0xff }, + }, + { + .reg = TAS2563_PRM_PLT_FLAG_REG, + .val_len = 4, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_SINEGAIN_REG, + .val_len = 4, + .val = { 0x0a, 0x3d, 0x70, 0xa4 }, + }, + { + .reg = TAS2563_TE_TA1_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_TA1_AT_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_TA2_REG, + .val_len = 4, + .val = { 0x00, 0x06, 0xd3, 0x72 }, + }, + { + .reg = TAS2563_TE_AT_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_DT_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, +}; + +static const struct bulk_reg_val tas2781_cali_start_reg[] = { + { + .reg = TAS2781_PRM_INT_MASK_REG, + .val = { 0xfe }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_CLK_CFG_REG, + .val = { 0xdd }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_RSVD_REG, + .val = { 0x20 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_TEST_57_REG, + .val = { 0x14 }, + .val_len = 1, + .is_locked = true + }, + { + .reg = TAS2781_PRM_TEST_62_REG, + .val = { 0x45 }, + .val_len = 1, + .is_locked = true + }, + { + .reg = TAS2781_PRM_PVDD_UVLO_REG, + .val = { 0x03 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_CHNL_0_REG, + .val = { 0xA8 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_NG_CFG0_REG, + .val = { 0xb9 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_IDLE_CH_DET_REG, + .val = { 0x92 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = 0, + .val = { 0x00, 0x00, 0x00, 0x56 }, // 1mA + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_PLT_FLAG_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_SINEGAIN_REG, + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_SINEGAIN2_REG, + .val_len = 4, + .is_locked = false + }, +}; + static const struct i2c_device_id tasdevice_id[] = { { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, @@ -141,6 +278,556 @@ static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static int tasdev_cali_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned char *data = cali_data->data; + unsigned int i = 0; + unsigned int j, k; + int rc; + + guard(mutex)(&priv->codec_lock); + if (!priv->is_user_space_calidata) + return -1; + + if (!p->r0_reg) + return -1; + + dst[i++] = bytes_ext->max; + dst[i++] = 'r'; + + dst[i++] = TASDEVICE_BOOK_ID(p->r0_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->r0_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->r0_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->r0_low_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->r0_low_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->r0_low_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->invr0_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->invr0_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->invr0_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->pow_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->pow_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->pow_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->tlimit_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->tlimit_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->tlimit_reg); + + for (j = 0, k = 0; j < priv->ndev; j++) { + if (j == data[k]) { + dst[i++] = j; + k++; + } else { + dev_err(priv->dev, "chn %d device %u not match\n", + j, data[k]); + k += 21; + continue; + } + rc = tasdevice_dev_bulk_read(priv, j, p->r0_reg, &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_reg bulk_rd err = %d\n", + j, rc); + i += 20; + k += 20; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d r0_data is not same\n", j); + k += 4; + i += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->r0_low_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_low bulk_rd err = %d\n", + j, rc); + i += 16; + k += 16; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d r0_low is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->invr0_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d invr0 bulk_rd err = %d\n", + j, rc); + i += 12; + k += 12; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d invr0 is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->pow_reg, &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d pow_reg bulk_rd err = %d\n", + j, rc); + i += 8; + k += 8; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d pow_reg is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->tlimit_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d tlimit bulk_rd err = %d\n", + j, rc); + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d tlimit is not same\n", j); + i += 4; + k += 4; + } + return 0; +} + +static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, + unsigned char *dst) +{ + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int rc = -1; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + rc = tasdevice_dev_bulk_read(tas_priv, i, reg, &dst[1], + 4); + break; + } + } + + return rc; +} + +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; + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int j; + + if (p == NULL) + return; + + /* Store the current setting from the chip */ + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_read(tas_priv, i, p[j].reg, + (int *)&p[j].val[0]); + } else { + switch (p[j].reg) { + case 0: + if (!reg[0]) + continue; + p[j].reg = reg[0]; + 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); + } + } + + /* Update the setting for calibration */ + for (j = 0; j < sum - 2; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + 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); +} + +static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dat = ucontrol->value.bytes.data; + int i, reg[4]; + int j = 0; + + guard(mutex)(&priv->codec_lock); + if (priv->chip_id != TAS2781 || bytes_ext->max != dat[0] || + dat[1] != 'r') { + dev_err(priv->dev, "%s: package fmt or chipid incorrect\n", + __func__); + return 0; + } + j += 2; + /* refresh pilot tone and SineGain register */ + for (i = 0; i < ARRAY_SIZE(reg); i++) { + reg[i] = TASDEVICE_REG(dat[j], dat[j + 1], dat[j + 2]); + j += 3; + } + + for (i = 0; i < priv->ndev; i++) { + int k = i * 9 + j; + + if (dat[k] != i) { + dev_err(priv->dev, "%s:no cal-setting for dev %d\n", + __func__, i); + continue; + } + sngl_calib_start(priv, i, reg, dat + k); + } + return 1; +} + +static void tas2781_calib_stop_put(struct tasdevice_priv *tas_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; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_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, + p[j].val, 4); + } + } + } +} + +static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct bulk_reg_val *q = (struct bulk_reg_val *)tas2563_cali_start_reg; + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int rc = 1; + int i, j; + + guard(mutex)(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2563) { + rc = -1; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_read(tas_priv, + i, p[j].reg, + (unsigned int *)&p[j].val[0]); + else + tasdevice_dev_bulk_read(tas_priv, + i, p[j].reg, p[j].val, 4); + } + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_write(tas_priv, i, p[j].reg, + q[j].val[0]); + else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + q[j].val, 4); + } + } +out: + return rc; +} + +static void tas2563_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tasdev_calib_stop_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + + guard(mutex)(&priv->codec_lock); + if (priv->chip_id == TAS2563) + tas2563_calib_stop_put(priv); + else + tas2781_calib_stop_put(priv); + + return 1; +} + +static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *src = ucontrol->value.bytes.data; + unsigned char *dst = cali_data->data; + int rc = 1, i = 0; + int j; + + guard(mutex)(&priv->codec_lock); + if (src[0] != bytes_ext->max || src[1] != 'r') { + dev_err(priv->dev, "%s: pkg fmt invalid\n", __func__); + return 0; + } + for (j = 0; j < priv->ndev; j++) { + if (src[17 + j * 21] != j) { + dev_err(priv->dev, "%s: pkg fmt invalid\n", __func__); + return 0; + } + } + 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; + + memcpy(dst, &src[i], cali_data->total_sz); + return rc; +} + +static int tas2781_latch_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct tasdevice *tasdev = tas_priv->tasdevice; + unsigned char *dst = ucontrol->value.bytes.data; + int i, val, rc = -1; + + dst[0] = bytes_ext->max; + guard(mutex)(&tas_priv->codec_lock); + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[1] = i; + rc = tasdevice_dev_read(tas_priv, i, + TAS2781_RUNTIME_LATCH_RE_REG, &val); + if (rc < 0) + dev_err(tas_priv->dev, "%s, get value error\n", + __func__); + else + dst[2] = val; + + break; + } + } + + return rc; +} + +static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG_TF; + else + reg = TAS2563_RUNTIME_RE_REG_TF; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG; + else + reg = TAS2563_RUNTIME_RE_REG; + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct calidata *cali_data = &tas_priv->cali_data; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + + if (tas_priv->chip_id == TAS2563) + reg = TAS2563_PRM_R0_REG; + else if (cali_data->cali_reg_array.r0_reg) + reg = cali_data->cali_reg_array.r0_reg; + else + return -1; + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + 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; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + 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; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_nop_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + static int tas2563_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -241,6 +928,16 @@ static const struct snd_kcontrol_new tasdevice_snd_controls[] = { tasdev_force_fwload_get, tasdev_force_fwload_put), }; +static const struct snd_kcontrol_new tasdevice_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tasdev_calib_stop_put), + SND_SOC_BYTES_EXT("Amp TF Data", 6, tasdev_tf_data_get, NULL), + SND_SOC_BYTES_EXT("Amp RE Data", 6, tasdev_re_data_get, NULL), + SND_SOC_BYTES_EXT("Amp R0 Data", 6, tasdev_r0_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA1 Data", 6, tasdev_XMA1_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA2 Data", 6, tasdev_XMA2_data_get, NULL), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -250,6 +947,10 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, dvc_tlv), }; +static const struct snd_kcontrol_new tas2781_cali_controls[] = { + SND_SOC_BYTES_EXT("Amp Latch Data", 3, tas2781_latch_reg_get, NULL), +}; + static const struct snd_kcontrol_new tas2563_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2563_DVC_LVL, 0, 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, @@ -257,6 +958,11 @@ static const struct snd_kcontrol_new tas2563_snd_controls[] = { tas2563_dvc_tlv), }; +static const struct snd_kcontrol_new tas2563_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Start", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tas2563_calib_start_put), +}; + static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -274,6 +980,31 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, return ret; } +static int tasdevice_info_active_num(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->ndev - 1; + + return 0; +} + +static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = TAS2563; + uinfo->value.integer.max = TAS2781; + + return 0; +} + static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -330,6 +1061,17 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, return 0; } +static int tasdevice_get_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->chip_id; + + return 0; +} + static int tasdevice_create_control(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *prof_ctrls; @@ -421,11 +1163,47 @@ static int tasdevice_configuration_put( return ret; } +static int tasdevice_active_num_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + ucontrol->value.integer.value[0] = i; + return 0; + } + } + + return -1; +} + +static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int dev_id = ucontrol->value.integer.value[0]; + int max = tas_priv->ndev - 1, rc; + + dev_id = clamp(dev_id, 0, max); + + guard(mutex)(&tas_priv->codec_lock); + rc = tasdev_chn_switch(tas_priv, dev_id); + + return rc; +} + static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *prog_name, *conf_name; - int nr_controls = 2; + char *active_dev_num, *chip_id; + char *conf_name, *prog_name; + int nr_controls = 4; int mix_index = 0; int ret; @@ -466,6 +1244,30 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].put = tasdevice_configuration_put; mix_index++; + active_dev_num = devm_kstrdup(tas_priv->dev, "Activate Tasdevice Num", + GFP_KERNEL); + if (!active_dev_num) { + ret = -ENOMEM; + goto out; + } + dsp_ctrls[mix_index].name = active_dev_num; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_active_num; + dsp_ctrls[mix_index].get = tasdevice_active_num_get; + dsp_ctrls[mix_index].put = tasdevice_active_num_put; + mix_index++; + + chip_id = devm_kstrdup(tas_priv->dev, "Tasdevice Chip Id", GFP_KERNEL); + if (!chip_id) { + ret = -ENOMEM; + goto out; + } + dsp_ctrls[mix_index].name = chip_id; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_chip_id; + dsp_ctrls[mix_index].get = tasdevice_get_chip_id; + mix_index++; + ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); @@ -473,6 +1275,150 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) return ret; } +static int tasdevice_create_cali_ctrls(struct tasdevice_priv *tas_priv) +{ + struct calidata *cali_data = &tas_priv->cali_data; + struct tasdevice *tasdev = tas_priv->tasdevice; + struct soc_bytes_ext *ext_cali_data; + struct snd_kcontrol_new *cali_ctrls; + unsigned int num_controls; + char *cali_name; + int rc, i; + + rc = snd_soc_add_component_controls(tas_priv->codec, + tasdevice_cali_controls, ARRAY_SIZE(tasdevice_cali_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add cali controls err rc = %d", + __func__, rc); + return rc; + } + + if (tas_priv->chip_id == TAS2781) { + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; + num_controls = ARRAY_SIZE(tas2781_cali_controls); + for (i = 0; i < tas_priv->ndev; i++) { + 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; + } + } else { + cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; + num_controls = ARRAY_SIZE(tas2563_cali_controls); + for (i = 0; i < tas_priv->ndev; i++) { + tasdev[i].cali_data_backup = + kmemdup(tas2563_cali_start_reg, + sizeof(tas2563_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_backup) + return -ENOMEM; + } + } + + rc = snd_soc_add_component_controls(tas_priv->codec, cali_ctrls, + num_controls); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add chip cali ctrls err rc = %d", + __func__, rc); + return rc; + } + + /* index for cali_ctrls */ + i = 0; + if (tas_priv->chip_id == TAS2781) + num_controls = 2; + else + num_controls = 1; + + /* + * Alloc kcontrol via devm_kzalloc, which don't manually + * free the kcontrol + */ + cali_ctrls = devm_kcalloc(tas_priv->dev, num_controls, + sizeof(cali_ctrls[0]), GFP_KERNEL); + if (!cali_ctrls) + return -ENOMEM; + + ext_cali_data = devm_kzalloc(tas_priv->dev, sizeof(*ext_cali_data), + GFP_KERNEL); + if (!ext_cali_data) + return -ENOMEM; + + cali_name = devm_kstrdup(tas_priv->dev, "Speaker Calibrated Data", + GFP_KERNEL); + if (!cali_name) + return -ENOMEM; + /* the number of calibrated data per tas2563/tas2781 */ + cali_data->cali_dat_sz_per_dev = 20; + /* + * Data structure for tas2563/tas2781 calibrated data: + * Pkg len (1 byte) + * Reg id (1 byte, constant 'r') + * book, page, register array for calibrated data (15 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info (1 byte) + * Calibrated data for Device #i (20 bytes) + * } + */ + ext_cali_data->max = tas_priv->ndev * + (cali_data->cali_dat_sz_per_dev + 1) + 1 + 15 + 1; + tas_priv->cali_data.total_sz = tas_priv->ndev * + (cali_data->cali_dat_sz_per_dev + 1); + tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + ext_cali_data->max, GFP_KERNEL); + cali_ctrls[i].name = cali_name; + cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cali_ctrls[i].info = snd_soc_bytes_info_ext; + cali_ctrls[i].get = tasdev_cali_data_get; + cali_ctrls[i].put = tasdev_cali_data_put; + cali_ctrls[i].private_value = (unsigned long)ext_cali_data; + i++; + + cali_data->data = devm_kzalloc(tas_priv->dev, cali_data->total_sz, + GFP_KERNEL); + if (!cali_data->data) + return -ENOMEM; + + if (tas_priv->chip_id == TAS2781) { + struct soc_bytes_ext *ext_cali_start; + char *cali_start_name; + + ext_cali_start = devm_kzalloc(tas_priv->dev, + sizeof(*ext_cali_start), GFP_KERNEL); + if (!ext_cali_start) + return -ENOMEM; + + cali_start_name = devm_kstrdup(tas_priv->dev, + "Calibration Start", GFP_KERNEL); + if (!cali_start_name) + return -ENOMEM; + /* + * package structure for tas2781 ftc start: + * Pkg len (1 byte) + * Reg id (1 byte, constant 'r') + * book, page, register for pilot threshold, pilot tone + * and sine gain (12 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info (1 byte) + * Sine gain for Device #i (8 bytes) + * } + */ + ext_cali_data->max = 14 + tas_priv->ndev * 9; + cali_ctrls[i].name = cali_start_name; + cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cali_ctrls[i].info = snd_soc_bytes_info_ext; + cali_ctrls[i].put = tas2781_calib_start_put; + cali_ctrls[i].get = tasdev_nop_get; + cali_ctrls[i].private_value = (unsigned long)ext_cali_start; + i++; + } + + rc = snd_soc_add_component_controls(tas_priv->codec, cali_ctrls, + num_controls < i ? num_controls : i); + + return rc; +} + static void tasdevice_fw_ready(const struct firmware *fmw, void *context) { @@ -519,6 +1465,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, goto out; } + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; /* If calibrated data occurs error, dsp will still works with default @@ -720,6 +1672,11 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) static void tasdevice_deinit(void *context) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) + kfree(tasdev[i].cali_data_backup); tasdevice_config_info_remove(tas_priv); tasdevice_dsp_remove(tas_priv); -- 2.35.7

3 2

[PATCH v2 0/3] ASoC: dt-bindings: amlogic-sound-cards: document clocks property
by Neil Armstrong 02 Sep '24

02 Sep '24

Following an off-list discution with Jerome about fixing the following DTBs check errors: sound: Unevaluated properties are not allowed ('assigned-clock-parents', 'assigned-clock-rates', 'assigned-clocks' were unexpected) from schema $id: http://devicetree.org/schemas/sound/amlogic,axg-sound-card.yaml# sound: Unevaluated properties are not allowed ('assigned-clock-parents', 'assigned-clock-rates', 'assigned-clocks' were unexpected) from schema $id: http://devicetree.org/schemas/sound/amlogic,gx-sound-card.yaml# sound: 'anyOf' conditional failed, one must be fixed: 'clocks' is a required property '#clock-cells' is a required property from schema $id: http://devicetree.org/schemas/clock/clock.yaml# It has been agreed documenting the clock in the sound card is a better solution than moving them to a random clock controller or consumer node which is not related to the actual meaning of those root frequencies. The patchset adds the clocks proprty to the bindings and finally adds the properties to the DT files. Signed-off-by: Neil Armstrong <neil.armstrong(a)linaro.org> --- Changes in v2: - drop clock-names in bindings & commit messages - fix new boards - drop RFS - Link to v1: https://lore.kernel.org/r/20240614-topic-amlogic-upstream-bindings-fixes-au… --- Neil Armstrong (3): ASoC: dt-bindings: amlogic,axg-sound-card: document clocks property ASoC: dt-bindings: amlogic,gx-sound-card: document clocks property arm64: dts: amlogic: add clock and clock-names to sound cards .../devicetree/bindings/sound/amlogic,axg-sound-card.yaml | 7 +++++++ Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml | 7 +++++++ arch/arm64/boot/dts/amlogic/meson-axg-s400.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-fbx8am.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-radxa-zero.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-sei510.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-u200.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-x96-max.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-bananapi-cm4-cm4io.dts | 4 ++++ .../arm64/boot/dts/amlogic/meson-g12b-bananapi-cm4-mnt-reform2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-bananapi.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-dreambox.dtsi | 6 ++++++ arch/arm64/boot/dts/amlogic/meson-g12b-gsking-x.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-gtking-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-gtking.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-odroid-go-ultra.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-odroid-n2.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-odroid-n2l.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-radxa-zero2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-ugoos-am6.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gx-libretech-pc.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gx-p23x-q20x.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-kii-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-nanopi-k2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-nexbox-a95x.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-odroidc2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-p200.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-p201.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-vega-s95.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-wetek-hub.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-wetek-play2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s805x-libretech-ac.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s805x-p241.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-khadas-vim.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-libretech-cc-v2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-libretech-cc.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-p212.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-vero4k.dts | 5 +++++ arch/arm64/boot/dts/amlogic/meson-gxm-khadas-vim2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxm-nexbox-a1.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxm-rbox-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-khadas-vim3.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-libretech-cottonwood.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-a95xf3-air-gbit.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-a95xf3-air.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-bananapi-m2-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-bananapi-m5.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-h96-max.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-odroid.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-sei610.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-x96-air-gbit.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-x96-air.dts | 4 ++++ 52 files changed, 217 insertions(+) --- base-commit: cb2361c78068ec47c3a6222d71f664b67665550e change-id: 20240614-topic-amlogic-upstream-bindings-fixes-audio-snd-card-2c642ccd2c6e Best regards, -- Neil Armstrong <neil.armstrong(a)linaro.org>

3 6

[tiwai-sound:for-next 12/12] sound/core/pcm_memory.c:218:29: warning: format specifies type 'size_t' (aka 'unsigned int') but the argument has type 'unsigned long'
by kernel test robot 02 Sep '24

02 Sep '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 43b42ed438bfff6bb5a51cc27a1658c03cd223fd commit: 43b42ed438bfff6bb5a51cc27a1658c03cd223fd [12/12] ALSA: pcm: Fix the previous conversion to kstrtoul() config: i386-buildonly-randconfig-002-20240902 (https://download.01.org/0day-ci/archive/20240902/202409020722.T2ymGxmq-lkp@…) compiler: clang version 18.1.5 (https://github.com/llvm/llvm-project 617a15a9eac96088ae5e9134248d8236e34b91b1) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240902/202409020722.T2ymGxmq-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202409020722.T2ymGxmq-lkp@intel.com/ All warnings (new ones prefixed by >>): >> sound/core/pcm_memory.c:218:29: warning: format specifies type 'size_t' (aka 'unsigned int') but the argument has type 'unsigned long' [-Wformat] 215 | pr_debug("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", | ~~~ | %lu 216 | substream->pcm->card->number, substream->pcm->device, 217 | substream->stream ? 'c' : 'p', substream->number, 218 | substream->pcm->name, size); | ^~~~ include/linux/printk.h:595:38: note: expanded from macro 'pr_debug' 595 | no_printk(KERN_DEBUG pr_fmt(fmt), ##__VA_ARGS__) | ~~~ ^~~~~~~~~~~ include/linux/printk.h:133:18: note: expanded from macro 'no_printk' 133 | _printk(fmt, ##__VA_ARGS__); \ | ~~~ ^~~~~~~~~~~ 1 warning generated. vim +218 sound/core/pcm_memory.c c7132aeb72ad11 Jaroslav Kysela 2006-10-06 174 ^1da177e4c3f41 Linus Torvalds 2005-04-16 175 /* ^1da177e4c3f41 Linus Torvalds 2005-04-16 176 * write callback for prealloc proc file ^1da177e4c3f41 Linus Torvalds 2005-04-16 177 * ^1da177e4c3f41 Linus Torvalds 2005-04-16 178 * accepts the preallocation size in kB. ^1da177e4c3f41 Linus Torvalds 2005-04-16 179 */ 877211f5e1b119 Takashi Iwai 2005-11-17 180 static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, 877211f5e1b119 Takashi Iwai 2005-11-17 181 struct snd_info_buffer *buffer) ^1da177e4c3f41 Linus Torvalds 2005-04-16 182 { 877211f5e1b119 Takashi Iwai 2005-11-17 183 struct snd_pcm_substream *substream = entry->private_data; d4cfb30fce0309 Takashi Iwai 2020-01-20 184 struct snd_card *card = substream->pcm->card; ^1da177e4c3f41 Linus Torvalds 2005-04-16 185 char line[64], str[64]; 43b42ed438bfff Takashi Iwai 2024-09-01 186 unsigned long size; ^1da177e4c3f41 Linus Torvalds 2005-04-16 187 struct snd_dma_buffer new_dmab; ^1da177e4c3f41 Linus Torvalds 2005-04-16 188 dd0da75b9a2768 Takashi Iwai 2024-02-27 189 guard(mutex)(&substream->pcm->open_mutex); ^1da177e4c3f41 Linus Torvalds 2005-04-16 190 if (substream->runtime) { ^1da177e4c3f41 Linus Torvalds 2005-04-16 191 buffer->error = -EBUSY; dd0da75b9a2768 Takashi Iwai 2024-02-27 192 return; ^1da177e4c3f41 Linus Torvalds 2005-04-16 193 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 194 if (!snd_info_get_line(buffer, line, sizeof(line))) { ^1da177e4c3f41 Linus Torvalds 2005-04-16 195 snd_info_get_str(str, line, sizeof(str)); 61bc4deff03318 Hongbo Li 2024-08-31 196 buffer->error = kstrtoul(str, 10, &size); 61bc4deff03318 Hongbo Li 2024-08-31 197 if (buffer->error != 0) 61bc4deff03318 Hongbo Li 2024-08-31 198 return; 61bc4deff03318 Hongbo Li 2024-08-31 199 size *= 1024; ^1da177e4c3f41 Linus Torvalds 2005-04-16 200 if ((size != 0 && size < 8192) || size > substream->dma_max) { ^1da177e4c3f41 Linus Torvalds 2005-04-16 201 buffer->error = -EINVAL; dd0da75b9a2768 Takashi Iwai 2024-02-27 202 return; ^1da177e4c3f41 Linus Torvalds 2005-04-16 203 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 204 if (substream->dma_buffer.bytes == size) dd0da75b9a2768 Takashi Iwai 2024-02-27 205 return; ^1da177e4c3f41 Linus Torvalds 2005-04-16 206 memset(&new_dmab, 0, sizeof(new_dmab)); ^1da177e4c3f41 Linus Torvalds 2005-04-16 207 new_dmab.dev = substream->dma_buffer.dev; ^1da177e4c3f41 Linus Torvalds 2005-04-16 208 if (size > 0) { d4cfb30fce0309 Takashi Iwai 2020-01-20 209 if (do_alloc_pages(card, d4cfb30fce0309 Takashi Iwai 2020-01-20 210 substream->dma_buffer.dev.type, ^1da177e4c3f41 Linus Torvalds 2005-04-16 211 substream->dma_buffer.dev.dev, a25684a956468e Takashi Iwai 2021-10-17 212 substream->stream, ^1da177e4c3f41 Linus Torvalds 2005-04-16 213 size, &new_dmab) < 0) { ^1da177e4c3f41 Linus Torvalds 2005-04-16 214 buffer->error = -ENOMEM; dc85fc9d05d235 Amadeusz Sławiński 2021-03-18 215 pr_debug("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", dc85fc9d05d235 Amadeusz Sławiński 2021-03-18 216 substream->pcm->card->number, substream->pcm->device, dc85fc9d05d235 Amadeusz Sławiński 2021-03-18 217 substream->stream ? 'c' : 'p', substream->number, dc85fc9d05d235 Amadeusz Sławiński 2021-03-18 @218 substream->pcm->name, size); dd0da75b9a2768 Takashi Iwai 2024-02-27 219 return; ^1da177e4c3f41 Linus Torvalds 2005-04-16 220 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 221 substream->buffer_bytes_max = size; ^1da177e4c3f41 Linus Torvalds 2005-04-16 222 } else { ^1da177e4c3f41 Linus Torvalds 2005-04-16 223 substream->buffer_bytes_max = UINT_MAX; ^1da177e4c3f41 Linus Torvalds 2005-04-16 224 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 225 if (substream->dma_buffer.area) d4cfb30fce0309 Takashi Iwai 2020-01-20 226 do_free_pages(card, &substream->dma_buffer); ^1da177e4c3f41 Linus Torvalds 2005-04-16 227 substream->dma_buffer = new_dmab; ^1da177e4c3f41 Linus Torvalds 2005-04-16 228 } else { ^1da177e4c3f41 Linus Torvalds 2005-04-16 229 buffer->error = -EINVAL; ^1da177e4c3f41 Linus Torvalds 2005-04-16 230 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 231 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 232 :::::: The code at line 218 was first introduced by commit :::::: dc85fc9d05d23591ddfde400c817413765611ec7 ALSA: pcm: Add debug print on memory allocation failure :::::: TO: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[RESEND PATCH v4] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 01 Sep '24

01 Sep '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. So far, the Software maintainer has not been confirmed. Once the maintainer was confimred, the guy will update his mail into audio converter section. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v4: - Add Touch Screen support v3: - Add Audio converter section - Set the section of LOW/MIDDLE-POWER AUDIO AMPLIFIER as Supported v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 34 +++++++++++++++++++++++++--------- 1 file changed, 25 insertions(+), 9 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index fe83ba7194ea..a2448dab3601 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22596,26 +22596,43 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml +F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt -F: Documentation/devicetree/bindings/sound/tpa6130a2.txt -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -22625,9 +22642,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

Re: [PATCH v7 0/2] ASoC: fsl-asoc-card: add S/PDIF controller support
by Shawn Guo 01 Sep '24

01 Sep '24

On Tue, Aug 20, 2024 at 01:28:25PM +0200, Elinor Montmasson wrote: > Elinor Montmasson (2): > arm64: dts: imx8m: update spdif sound card node properties > ARM: dts: imx6: update spdif sound card node properties Applied both, thanks!

1 0

[tiwai-sound:for-next 11/11] sound/core/pcm_memory.c:196:37: error: incompatible pointer types passing 'size_t *' (aka 'unsigned int *') to parameter of type 'unsigned long *'
by kernel test robot 31 Aug '24

31 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 61bc4deff033181992408f973b48fca08757d3ff commit: 61bc4deff033181992408f973b48fca08757d3ff [11/11] ALSA: pcm: replace simple_strtoul to kstrtoul config: hexagon-randconfig-002-20240901 (https://download.01.org/0day-ci/archive/20240901/202409010509.nOL6xTD7-lkp@…) compiler: clang version 20.0.0git (https://github.com/llvm/llvm-project 46fe36a4295f05d5d3731762e31fc4e6e99863e9) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240901/202409010509.nOL6xTD7-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202409010509.nOL6xTD7-lkp@intel.com/ All errors (new ones prefixed by >>): In file included from sound/core/pcm_memory.c:7: In file included from include/linux/io.h:14: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:548:31: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 548 | val = __raw_readb(PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:561:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 561 | val = __le16_to_cpu((__le16 __force)__raw_readw(PCI_IOBASE + addr)); | ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:37:51: note: expanded from macro '__le16_to_cpu' 37 | #define __le16_to_cpu(x) ((__force __u16)(__le16)(x)) | ^ In file included from sound/core/pcm_memory.c:7: In file included from include/linux/io.h:14: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:574:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 574 | val = __le32_to_cpu((__le32 __force)__raw_readl(PCI_IOBASE + addr)); | ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:35:51: note: expanded from macro '__le32_to_cpu' 35 | #define __le32_to_cpu(x) ((__force __u32)(__le32)(x)) | ^ In file included from sound/core/pcm_memory.c:7: In file included from include/linux/io.h:14: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:585:33: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 585 | __raw_writeb(value, PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:595:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 595 | __raw_writew((u16 __force)cpu_to_le16(value), PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:605:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 605 | __raw_writel((u32 __force)cpu_to_le32(value), PCI_IOBASE + addr); | ~~~~~~~~~~ ^ In file included from sound/core/pcm_memory.c:14: In file included from include/sound/pcm.h:15: In file included from include/linux/mm.h:2228: include/linux/vmstat.h:517:36: warning: arithmetic between different enumeration types ('enum node_stat_item' and 'enum lru_list') [-Wenum-enum-conversion] 517 | return node_stat_name(NR_LRU_BASE + lru) + 3; // skip "nr_" | ~~~~~~~~~~~ ^ ~~~ >> sound/core/pcm_memory.c:196:37: error: incompatible pointer types passing 'size_t *' (aka 'unsigned int *') to parameter of type 'unsigned long *' [-Werror,-Wincompatible-pointer-types] 196 | buffer->error = kstrtoul(str, 10, &size); | ^~~~~ include/linux/kstrtox.h:30:90: note: passing argument to parameter 'res' here 30 | static inline int __must_check kstrtoul(const char *s, unsigned int base, unsigned long *res) | ^ 7 warnings and 1 error generated. vim +196 sound/core/pcm_memory.c 174 175 /* 176 * write callback for prealloc proc file 177 * 178 * accepts the preallocation size in kB. 179 */ 180 static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, 181 struct snd_info_buffer *buffer) 182 { 183 struct snd_pcm_substream *substream = entry->private_data; 184 struct snd_card *card = substream->pcm->card; 185 char line[64], str[64]; 186 size_t size; 187 struct snd_dma_buffer new_dmab; 188 189 guard(mutex)(&substream->pcm->open_mutex); 190 if (substream->runtime) { 191 buffer->error = -EBUSY; 192 return; 193 } 194 if (!snd_info_get_line(buffer, line, sizeof(line))) { 195 snd_info_get_str(str, line, sizeof(str)); > 196 buffer->error = kstrtoul(str, 10, &size); 197 if (buffer->error != 0) 198 return; 199 size *= 1024; 200 if ((size != 0 && size < 8192) || size > substream->dma_max) { 201 buffer->error = -EINVAL; 202 return; 203 } 204 if (substream->dma_buffer.bytes == size) 205 return; 206 memset(&new_dmab, 0, sizeof(new_dmab)); 207 new_dmab.dev = substream->dma_buffer.dev; 208 if (size > 0) { 209 if (do_alloc_pages(card, 210 substream->dma_buffer.dev.type, 211 substream->dma_buffer.dev.dev, 212 substream->stream, 213 size, &new_dmab) < 0) { 214 buffer->error = -ENOMEM; 215 pr_debug("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", 216 substream->pcm->card->number, substream->pcm->device, 217 substream->stream ? 'c' : 'p', substream->number, 218 substream->pcm->name, size); 219 return; 220 } 221 substream->buffer_bytes_max = size; 222 } else { 223 substream->buffer_bytes_max = UINT_MAX; 224 } 225 if (substream->dma_buffer.area) 226 do_free_pages(card, &substream->dma_buffer); 227 substream->dma_buffer = new_dmab; 228 } else { 229 buffer->error = -EINVAL; 230 } 231 } 232 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 11/11] sound/core/pcm_memory.c:196:51: error: passing argument 3 of 'kstrtoul' from incompatible pointer type
by kernel test robot 31 Aug '24

31 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 61bc4deff033181992408f973b48fca08757d3ff commit: 61bc4deff033181992408f973b48fca08757d3ff [11/11] ALSA: pcm: replace simple_strtoul to kstrtoul config: sh-allmodconfig (https://download.01.org/0day-ci/archive/20240901/202409010425.YPS7cWeJ-lkp@…) compiler: sh4-linux-gcc (GCC) 14.1.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240901/202409010425.YPS7cWeJ-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202409010425.YPS7cWeJ-lkp@intel.com/ All errors (new ones prefixed by >>): sound/core/pcm_memory.c: In function 'snd_pcm_lib_preallocate_proc_write': >> sound/core/pcm_memory.c:196:51: error: passing argument 3 of 'kstrtoul' from incompatible pointer type [-Wincompatible-pointer-types] 196 | buffer->error = kstrtoul(str, 10, &size); | ^~~~~ | | | size_t * {aka unsigned int *} In file included from include/linux/kernel.h:25, from arch/sh/include/asm/fixmap.h:13, from arch/sh/include/asm/pgtable.h:22, from include/linux/pgtable.h:6, from arch/sh/include/asm/io.h:21, from include/linux/io.h:14, from sound/core/pcm_memory.c:7: include/linux/kstrtox.h:30:90: note: expected 'long unsigned int *' but argument is of type 'size_t *' {aka 'unsigned int *'} 30 | static inline int __must_check kstrtoul(const char *s, unsigned int base, unsigned long *res) | ~~~~~~~~~~~~~~~^~~ vim +/kstrtoul +196 sound/core/pcm_memory.c 174 175 /* 176 * write callback for prealloc proc file 177 * 178 * accepts the preallocation size in kB. 179 */ 180 static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, 181 struct snd_info_buffer *buffer) 182 { 183 struct snd_pcm_substream *substream = entry->private_data; 184 struct snd_card *card = substream->pcm->card; 185 char line[64], str[64]; 186 size_t size; 187 struct snd_dma_buffer new_dmab; 188 189 guard(mutex)(&substream->pcm->open_mutex); 190 if (substream->runtime) { 191 buffer->error = -EBUSY; 192 return; 193 } 194 if (!snd_info_get_line(buffer, line, sizeof(line))) { 195 snd_info_get_str(str, line, sizeof(str)); > 196 buffer->error = kstrtoul(str, 10, &size); 197 if (buffer->error != 0) 198 return; 199 size *= 1024; 200 if ((size != 0 && size < 8192) || size > substream->dma_max) { 201 buffer->error = -EINVAL; 202 return; 203 } 204 if (substream->dma_buffer.bytes == size) 205 return; 206 memset(&new_dmab, 0, sizeof(new_dmab)); 207 new_dmab.dev = substream->dma_buffer.dev; 208 if (size > 0) { 209 if (do_alloc_pages(card, 210 substream->dma_buffer.dev.type, 211 substream->dma_buffer.dev.dev, 212 substream->stream, 213 size, &new_dmab) < 0) { 214 buffer->error = -ENOMEM; 215 pr_debug("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", 216 substream->pcm->card->number, substream->pcm->device, 217 substream->stream ? 'c' : 'p', substream->number, 218 substream->pcm->name, size); 219 return; 220 } 221 substream->buffer_bytes_max = size; 222 } else { 223 substream->buffer_bytes_max = UINT_MAX; 224 } 225 if (substream->dma_buffer.area) 226 do_free_pages(card, &substream->dma_buffer); 227 substream->dma_buffer = new_dmab; 228 } else { 229 buffer->error = -EINVAL; 230 } 231 } 232 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

Fix TLV dB parser in case of used container
by GitHub pull_request - edited 31 Aug '24

31 Aug '24

alsa-project/alsa-lib pull request #409 was edited from arkq: In case when dB information does not appear as the only TLV type in the stream (it might be wrapped in a container, but the container can not have any other type), the TLV parser fails to get the dB TLV pointer. This commit fixes it by distinguishing between TLV parse error and dB information not being found in a container, so the parser can iterate over all elements in the container. Also, it fixes out-of-bounds read in case of malicious TLV record. EDIT: Other possibility to fix is to return `-ENOENT` in case where dB information was not found and handle that accordingly when iterating over items of a container. Request URL : https://github.com/alsa-project/alsa-lib/pull/409 Patch URL : https://github.com/alsa-project/alsa-lib/pull/409.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Fix TLV dB parser in case of used container
by GitHub pull_request - opened 31 Aug '24

31 Aug '24

alsa-project/alsa-lib pull request #409 was opened from arkq: In case when dB information does not appear as the only TLV type in the stream (it might be wrapped in a container, but the container can not have any other type), the TLV parser fails to get the dB TLV pointer. This commit fixes it by distinguishing between TLV parse error and dB information not being found in a container, so the parser can iterate over all elements in the container. Also, it fixes out-of-bounds read in case of malicious TLV record. Request URL : https://github.com/alsa-project/alsa-lib/pull/409 Patch URL : https://github.com/alsa-project/alsa-lib/pull/409.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] ASoC: codecs: Switch to use dev_err_probe()
by Yang Ruibin 30 Aug '24

30 Aug '24

Using dev_err_probe() instead of dev_err() in probe() simplifies the error path and standardizes the format of the error code. Signed-off-by: Yang Ruibin <11162571(a)vivo.com> --- sound/soc/codecs/ad1980.c | 8 +++----- sound/soc/codecs/adau1701.c | 19 +++++++------------ sound/soc/codecs/ssm2602.c | 6 ++---- 3 files changed, 12 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 3c1ae13c1..d2f679d9d 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -237,11 +237,9 @@ static int ad1980_soc_probe(struct snd_soc_component *component) u16 ext_status; ac97 = snd_soc_new_ac97_component(component, 0, 0); - if (IS_ERR(ac97)) { - ret = PTR_ERR(ac97); - dev_err(component->dev, "Failed to register AC97 component: %d\n", ret); - return ret; - } + if (IS_ERR(ac97)) + return dev_err_probe(component->dev, PTR_ERR(ac97), + "Failed to register AC97 component\n"); regmap = regmap_init_ac97(ac97, &ad1980_regmap_config); if (IS_ERR(regmap)) { diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 8bd6067df..f4c574447 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -677,10 +677,9 @@ static int adau1701_probe(struct snd_soc_component *component) ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); - if (ret < 0) { - dev_err(component->dev, "Failed to enable regulators: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(component->dev, ret, + "Failed to enable regulators\n"); /* * Let the pll_clkdiv variable default to something that won't happen @@ -798,17 +797,13 @@ static int adau1701_i2c_probe(struct i2c_client *client) ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(adau1701->supplies), adau1701->supplies); - if (ret < 0) { - dev_err(dev, "Failed to get regulators: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(dev, ret, "Failed to get regulators\n"); ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); - if (ret < 0) { - dev_err(dev, "Failed to enable regulators: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(dev, ret, "Failed to enable regulators\n"); adau1701->client = client; adau1701->regmap = devm_regmap_init(dev, NULL, client, diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index c29324403..153eb55a3 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -605,10 +605,8 @@ static int ssm260x_component_probe(struct snd_soc_component *component) int ret; ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0); - if (ret < 0) { - dev_err(component->dev, "Failed to issue reset: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(component->dev, ret, "Failed to issue reset\n"); regmap_register_patch(ssm2602->regmap, ssm2602_patch, ARRAY_SIZE(ssm2602_patch)); -- 2.34.1

2 1

[PATCH v2] ASoC: codecs: lpass-va-macro: set the default codec version for sm8250
by srinivas.kandagatla＠linaro.org 30 Aug '24

30 Aug '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> sm8250 and sc7280 have lpass codec version 1.0, as these are very old platforms, they do not have a reliable way to get the codec version from core_id registers. On codec versions below 2.0, even though the core_id registers are available to read, the values of these registers are not unique to be able to determine the version of the codec dynamically. Add the version info into of_data, so that driver does not need to use core_id registers to get version number for such situations. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: - updated commit text to add more details sound/soc/codecs/lpass-va-macro.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 8454193ed22a..e95d1f29ef18 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -228,11 +228,13 @@ struct va_macro { struct va_macro_data { bool has_swr_master; bool has_npl_clk; + int version; }; static const struct va_macro_data sm8250_va_data = { .has_swr_master = false, .has_npl_clk = false, + .version = LPASS_CODEC_VERSION_1_0, }; static const struct va_macro_data sm8450_va_data = { @@ -1587,7 +1589,14 @@ static int va_macro_probe(struct platform_device *pdev) goto err_npl; } - va_macro_set_lpass_codec_version(va); + /** + * old version of codecs do not have a reliable way to determine the + * version from registers, get them from soc specific data + */ + if (data->version) + lpass_macro_set_codec_version(data->version); + else /* read version from register */ + va_macro_set_lpass_codec_version(va); if (va->has_swr_master) { /* Set default CLK div to 1 */ -- 2.25.1

5 4

[PATCH v1] ASoC: fsl_ssi: Simplify with dev_err_probe()
by Shen Lichuan 30 Aug '24

30 Aug '24

Use dev_err_probe() to simplify the error path and unify a message template. Using this helper is totally fine even if err is known to never be -EPROBE_DEFER. The benefit compared to a normal dev_err() is the standardized format of the error code, it being emitted symbolically and the fact that the error code is returned which allows more compact error paths. Signed-off-by: Shen Lichuan <shenlichuan(a)vivo.com> --- sound/soc/fsl/fsl_ssi.c | 33 +++++++++++++++------------------ 1 file changed, 15 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c4c1d9c44056..51927f89241f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1354,19 +1354,16 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, ssi->clk = devm_clk_get(dev, "ipg"); else ssi->clk = devm_clk_get(dev, NULL); - if (IS_ERR(ssi->clk)) { - ret = PTR_ERR(ssi->clk); - dev_err(dev, "failed to get clock: %d\n", ret); - return ret; - } + if (IS_ERR(ssi->clk)) + return dev_err_probe(dev, PTR_ERR(ssi->clk), + "failed to get clock\n"); /* Enable the clock since regmap will not handle it in this case */ if (!ssi->has_ipg_clk_name) { ret = clk_prepare_enable(ssi->clk); - if (ret) { - dev_err(dev, "clk_prepare_enable failed: %d\n", ret); - return ret; - } + if (ret) + return dev_err_probe(dev, ret, + "clk_prepare_enable failed\n"); } /* Do not error out for consumer cases that live without a baud clock */ @@ -1552,10 +1549,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) &regconfig); else ssi->regs = devm_regmap_init_mmio(dev, iomem, &regconfig); - if (IS_ERR(ssi->regs)) { - dev_err(dev, "failed to init register map\n"); - return PTR_ERR(ssi->regs); - } + if (IS_ERR(ssi->regs)) + return dev_err_probe(dev, PTR_ERR(ssi->regs), + "failed to init register map\n"); ssi->irq = platform_get_irq(pdev, 0); if (ssi->irq < 0) @@ -1607,7 +1603,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) mutex_init(&ssi->ac97_reg_lock); ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); if (ret) { - dev_err(dev, "failed to set AC'97 ops\n"); + dev_err_probe(dev, ret, "failed to set AC'97 ops\n"); goto error_ac97_ops; } } @@ -1615,7 +1611,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_ssi_component, &ssi->cpu_dai_drv, 1); if (ret) { - dev_err(dev, "failed to register DAI: %d\n", ret); + dev_err_probe(dev, ret, "failed to register DAI\n"); goto error_asoc_register; } @@ -1623,7 +1619,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = devm_request_irq(dev, ssi->irq, fsl_ssi_isr, 0, dev_name(dev), ssi); if (ret < 0) { - dev_err(dev, "failed to claim irq %u\n", ssi->irq); + dev_err_probe(dev, ret, "failed to claim irq %u\n", + ssi->irq); goto error_asoc_register; } } @@ -1649,8 +1646,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi->card_name, ssi->card_idx, NULL, 0); if (IS_ERR(ssi->card_pdev)) { ret = PTR_ERR(ssi->card_pdev); - dev_err(dev, "failed to register %s: %d\n", - ssi->card_name, ret); + dev_err_probe(dev, ret, "failed to register %s\n", + ssi->card_name); goto error_sound_card; } } -- 2.17.1

1 0

ALSA question
by Hai Nguyen 30 Aug '24

30 Aug '24

Hi, I have a C++ audio application that plays back audio data on the sound card, Digigram VX882e. https://www.digigram.com/support/vx882e-support/ The application runs on HPE ProLiant DL380 Gen9 Server. HPE ProLiant DL380 Gen9 Server It has worked well under RHEL 7 (kernel 3.10) and RHEL 8 (kernel 4.x). Asthe OS is upgraded to RHEL 9 (kernel 5.14), there is an buffer underrun. When investigating the issue, I run “speaker-test” utility. It fails with the error below. I'm just wonder if this has something to do with the changes in ALSA driver from kernel 4.x to kernel 5.14? If not, what could cause the buffer underrun issue? Your advice/help is highly appreciated, Thanks, Kevin Nguyen sudo speaker-test -D plughw:VX882e0 -c 2 speaker-test 1.2.9 Playback device is plughw:VX882e0 Stream parameters are 48000Hz, S16_LE, 2 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 128 to 8192 Period size range from 64 to 4096 Using max buffer size 8192 Periods = 4 was set period_size = 2048 was set buffer_size = 8192 0 - Front Left Write error: -5,Input/outputerror xrun_recovery failed:-5,Input/output error Transfer failed: Input/outputerror The Digigram card is detected with “aplay -l” command. sudo aplay -l **** List of PLAYBACK Hardware Devices **** card 0: VX882e0 [Digigram [PCM #0]], device 0: pcxhr 0[pcxhr 0] Subdevices: 4/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 card 1: VX882e1 [Digigram [PCM #1]], device 0: pcxhr 1[pcxhr 1] Subdevices: 4/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 card 2: VX882e2 [Digigram [PCM #2]], device 0: pcxhr 2 [pcxhr2] Subdevices: 4/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 card 3: VX882e3 [Digigram [PCM #3]], device 0: pcxhr 3[pcxhr 3] Subdevices: 4/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3

2 1

[PATCH v3] ASoC: codecs: wsa884x: Implement temperature reading and hwmon
by Krzysztof Kozlowski 29 Aug '24

29 Aug '24

Read temperature of the speaker and expose it via hwmon interface, which will be later used during calibration of speaker protection algorithms. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v3: 1. Replace Kconfig HWMON dependency in favor of IS_REACHABLE(CONFIG_HWMON) which solves Kernel Test Robot warning for !HWMON and HWMON=y && WSA884x=m Changes in v2: 1. Add missing dependency on HWMON for y!=m builds (Kernel Test Robot report: undefined reference to `devm_hwmon_device_register_with_info'). --- sound/soc/codecs/wsa884x.c | 201 +++++++++++++++++++++++++++++++++++++ 1 file changed, 201 insertions(+) diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c index 8db1380d1f10..86df5152c547 100644 --- a/sound/soc/codecs/wsa884x.c +++ b/sound/soc/codecs/wsa884x.c @@ -5,11 +5,14 @@ */ #include <linux/bitfield.h> +#include <linux/cleanup.h> #include <linux/device.h> #include <linux/gpio/consumer.h> +#include <linux/hwmon.h> #include <linux/init.h> #include <linux/kernel.h> #include <linux/module.h> +#include <linux/mutex.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> @@ -301,8 +304,28 @@ #define WSA884X_PA_FSM_MSK1 (WSA884X_DIG_CTRL0_BASE + 0x3b) #define WSA884X_PA_FSM_BYP_CTL (WSA884X_DIG_CTRL0_BASE + 0x3c) #define WSA884X_PA_FSM_BYP0 (WSA884X_DIG_CTRL0_BASE + 0x3d) +#define WSA884X_PA_FSM_BYP0_DC_CAL_EN_MASK 0x01 +#define WSA884X_PA_FSM_BYP0_DC_CAL_EN_SHIFT 0 +#define WSA884X_PA_FSM_BYP0_CLK_WD_EN_MASK 0x02 +#define WSA884X_PA_FSM_BYP0_CLK_WD_EN_SHIFT 1 +#define WSA884X_PA_FSM_BYP0_BG_EN_MASK 0x04 +#define WSA884X_PA_FSM_BYP0_BG_EN_SHIFT 2 +#define WSA884X_PA_FSM_BYP0_BOOST_EN_MASK 0x08 +#define WSA884X_PA_FSM_BYP0_BOOST_EN_SHIFT 3 +#define WSA884X_PA_FSM_BYP0_PA_EN_MASK 0x10 +#define WSA884X_PA_FSM_BYP0_PA_EN_SHIFT 4 +#define WSA884X_PA_FSM_BYP0_D_UNMUTE_MASK 0x20 +#define WSA884X_PA_FSM_BYP0_D_UNMUTE_SHIFT 5 +#define WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_MASK 0x40 +#define WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_SHIFT 6 +#define WSA884X_PA_FSM_BYP0_TSADC_EN_MASK 0x80 +#define WSA884X_PA_FSM_BYP0_TSADC_EN_SHIFT 7 #define WSA884X_PA_FSM_BYP1 (WSA884X_DIG_CTRL0_BASE + 0x3e) #define WSA884X_TADC_VALUE_CTL (WSA884X_DIG_CTRL0_BASE + 0x50) +#define WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK 0x01 +#define WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_SHIFT 0 +#define WSA884X_TADC_VALUE_CTL_VBAT_VALUE_RD_EN_MASK 0x02 +#define WSA884X_TADC_VALUE_CTL_VBAT_VALUE_RD_EN_SHIFT 1 #define WSA884X_TEMP_DETECT_CTL (WSA884X_DIG_CTRL0_BASE + 0x51) #define WSA884X_TEMP_DIN_MSB (WSA884X_DIG_CTRL0_BASE + 0x52) #define WSA884X_TEMP_DIN_LSB (WSA884X_DIG_CTRL0_BASE + 0x53) @@ -691,6 +714,17 @@ SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) +/* Two-point trimming for temperature calibration */ +#define WSA884X_T1_TEMP -10L +#define WSA884X_T2_TEMP 150L + +/* + * Device will report senseless data in many cases, so discard any measurements + * outside of valid range. + */ +#define WSA884X_LOW_TEMP_THRESHOLD 5 +#define WSA884X_HIGH_TEMP_THRESHOLD 45 + struct wsa884x_priv { struct regmap *regmap; struct device *dev; @@ -706,6 +740,13 @@ struct wsa884x_priv { int active_ports; int dev_mode; bool hw_init; + /* + * Protects temperature reading code (related to speaker protection) and + * fields: temperature and pa_on. + */ + struct mutex sp_lock; + unsigned int temperature; + bool pa_on; }; enum { @@ -1660,6 +1701,10 @@ static int wsa884x_spkr_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: + mutex_lock(&wsa884x->sp_lock); + wsa884x->pa_on = true; + mutex_unlock(&wsa884x->sp_lock); + wsa884x_spkr_post_pmu(component, wsa884x); snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, @@ -1671,6 +1716,10 @@ static int wsa884x_spkr_event(struct snd_soc_dapm_widget *w, snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, WSA884X_PDM_WD_CTL_PDM_WD_EN_MASK, 0x0); + + mutex_lock(&wsa884x->sp_lock); + wsa884x->pa_on = false; + mutex_unlock(&wsa884x->sp_lock); break; } @@ -1810,6 +1859,144 @@ static struct snd_soc_dai_driver wsa884x_dais[] = { }, }; +static int wsa884x_get_temp(struct wsa884x_priv *wsa884x, long *temp) +{ + unsigned int d1_msb = 0, d1_lsb = 0, d2_msb = 0, d2_lsb = 0; + unsigned int dmeas_msb = 0, dmeas_lsb = 0; + int d1, d2, dmeas; + unsigned int mask; + long val; + int ret; + + guard(mutex)(&wsa884x->sp_lock); + + if (wsa884x->pa_on) { + /* + * Reading temperature is possible only when Power Amplifier is + * off. Report last cached data. + */ + *temp = wsa884x->temperature; + return 0; + } + + ret = pm_runtime_resume_and_get(wsa884x->dev); + if (ret < 0) + return ret; + + mask = WSA884X_PA_FSM_BYP0_DC_CAL_EN_MASK | + WSA884X_PA_FSM_BYP0_CLK_WD_EN_MASK | + WSA884X_PA_FSM_BYP0_BG_EN_MASK | + WSA884X_PA_FSM_BYP0_D_UNMUTE_MASK | + WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_MASK | + WSA884X_PA_FSM_BYP0_TSADC_EN_MASK; + /* + * Here and further do not care about read or update failures. + * For example, before turning on Power Amplifier for the first + * time, reading WSA884X_TEMP_DIN_MSB will always return 0. + * Instead, check if returned value is within reasonable + * thresholds. + */ + regmap_update_bits(wsa884x->regmap, WSA884X_PA_FSM_BYP0, mask, mask); + + regmap_update_bits(wsa884x->regmap, WSA884X_TADC_VALUE_CTL, + WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, + FIELD_PREP(WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, 0x0)); + + regmap_read(wsa884x->regmap, WSA884X_TEMP_DIN_MSB, &dmeas_msb); + regmap_read(wsa884x->regmap, WSA884X_TEMP_DIN_LSB, &dmeas_lsb); + + regmap_update_bits(wsa884x->regmap, WSA884X_TADC_VALUE_CTL, + WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, + FIELD_PREP(WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, 0x1)); + + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_1, &d1_msb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_2, &d1_lsb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_3, &d2_msb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_4, &d2_lsb); + + regmap_update_bits(wsa884x->regmap, WSA884X_PA_FSM_BYP0, mask, 0x0); + + dmeas = (((dmeas_msb & 0xff) << 0x8) | (dmeas_lsb & 0xff)) >> 0x6; + d1 = (((d1_msb & 0xff) << 0x8) | (d1_lsb & 0xff)) >> 0x6; + d2 = (((d2_msb & 0xff) << 0x8) | (d2_lsb & 0xff)) >> 0x6; + + if (d1 == d2) { + /* Incorrect data in OTP? */ + ret = -EINVAL; + goto out; + } + + val = WSA884X_T1_TEMP + (((dmeas - d1) * (WSA884X_T2_TEMP - WSA884X_T1_TEMP))/(d2 - d1)); + + dev_dbg(wsa884x->dev, "Measured temp %ld (dmeas=%d, d1=%d, d2=%d)\n", + val, dmeas, d1, d2); + + if ((val > WSA884X_LOW_TEMP_THRESHOLD) && + (val < WSA884X_HIGH_TEMP_THRESHOLD)) { + wsa884x->temperature = val; + *temp = val; + ret = 0; + } else { + ret = -EAGAIN; + } + +out: + pm_runtime_mark_last_busy(wsa884x->dev); + pm_runtime_put_autosuspend(wsa884x->dev); + + return ret; +} + +static umode_t wsa884x_hwmon_is_visible(const void *data, + enum hwmon_sensor_types type, u32 attr, + int channel) +{ + if (type != hwmon_temp) + return 0; + + switch (attr) { + case hwmon_temp_input: + return 0444; + default: + break; + } + + return 0; +} + +static int wsa884x_hwmon_read(struct device *dev, + enum hwmon_sensor_types type, + u32 attr, int channel, long *temp) +{ + int ret; + + switch (attr) { + case hwmon_temp_input: + ret = wsa884x_get_temp(dev_get_drvdata(dev), temp); + break; + default: + ret = -EOPNOTSUPP; + break; + } + + return ret; +} + +static const struct hwmon_channel_info *const wsa884x_hwmon_info[] = { + HWMON_CHANNEL_INFO(temp, HWMON_T_INPUT), + NULL +}; + +static const struct hwmon_ops wsa884x_hwmon_ops = { + .is_visible = wsa884x_hwmon_is_visible, + .read = wsa884x_hwmon_read, +}; + +static const struct hwmon_chip_info wsa884x_hwmon_chip_info = { + .ops = &wsa884x_hwmon_ops, + .info = wsa884x_hwmon_info, +}; + static void wsa884x_reset_powerdown(void *data) { struct wsa884x_priv *wsa884x = data; @@ -1866,6 +2053,8 @@ static int wsa884x_probe(struct sdw_slave *pdev, if (!wsa884x) return -ENOMEM; + mutex_init(&wsa884x->sp_lock); + for (i = 0; i < WSA884X_SUPPLIES_NUM; i++) wsa884x->supplies[i].supply = wsa884x_supply_name[i]; @@ -1923,6 +2112,18 @@ static int wsa884x_probe(struct sdw_slave *pdev, regcache_cache_only(wsa884x->regmap, true); wsa884x->hw_init = true; + if (IS_REACHABLE(CONFIG_HWMON)) { + struct device *hwmon; + + hwmon = devm_hwmon_device_register_with_info(dev, "wsa884x", + wsa884x, + &wsa884x_hwmon_chip_info, + NULL); + if (IS_ERR(hwmon)) + return dev_err_probe(dev, PTR_ERR(hwmon), + "Failed to register hwmon sensor\n"); + } + pm_runtime_set_autosuspend_delay(dev, 3000); pm_runtime_use_autosuspend(dev); pm_runtime_mark_last_busy(dev); -- 2.43.0

2 1

Re: [PATCH] ALSA: hda/realtek: Autodetect Cirrus Logic companion amplifier bindings
by Takashi Iwai 29 Aug '24

29 Aug '24

On Thu, 29 Aug 2024 18:11:14 +0200, Simon Trimmer wrote: > > We do not need model specific HDA quirks to construct the component > binding details of Cirrus Logic companion amplifiers as this information > is already present in the ACPI. > > Quirks are then only required for special workarounds not described in > the ACPI such as internal configuration of the Realtek codecs. > > The codec pointer is now initialized in hda_component_manager_init() so > that we can detect when companion amplifiers are found in the ACPI but > the SSID invokes a quirk that also attempts to create the component > binding. > > Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> > --- > v1 - This patch applies to for-next with this additional patch: > b1faff28a2bd ("ALSA: hda: hda_component: Fix mutex crash if nothing ever binds") > which is currently only in the for-linus. OK, I now merged for-linus to for-next, and applied yours. > We're looking to do the same for the CS35L41 devices, but that will > come later because there are many more quirks to work through. That's understandable. Thanks! Takashi

1 0

[PATCH] ASoC: mediatek: mt8188-mt6359: Modify key
by Zhang Yi 29 Aug '24

29 Aug '24

In order to get the correct keys when using the ES8326.We will associate SND_JACK_BTN_1 to KEY_VOLUMEUP and SND_JACK_BTN_2 to KEY_VOLUMEDOWN when the ES8326 flag is recognized. Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index eba6f4c445ff..08ae962afeb9 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -734,6 +734,7 @@ static int mt8188_headset_codec_init(struct snd_soc_pcm_runtime *rtd) struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8188_JACK_HEADSET]; struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; + struct mtk_platform_card_data *card_data = soc_card_data->card_data; int ret; ret = snd_soc_dapm_new_controls(&card->dapm, mt8188_nau8825_widgets, @@ -762,10 +763,18 @@ static int mt8188_headset_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + if (card_data->flags & ES8326_HS_PRESENT) { + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + } else { + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + } + ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { -- 2.17.1

2 1

[PATCH v1] ASoC: codecs: Simplify with dev_err_probe()
by Yang Ruibin 29 Aug '24

29 Aug '24

Using dev_err_probe() instead of dev_err() in probe() simplifies the error path and standardizes the format of the error code. Signed-off-by: Yang Ruibin <11162571(a)vivo.com> --- sound/soc/codecs/ad1980.c | 4 ++-- sound/soc/codecs/adau1701.c | 12 ++++++------ sound/soc/codecs/ssm2602.c | 4 ++-- 3 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 3c1ae13c1..25fd8bc4f 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -239,8 +239,8 @@ static int ad1980_soc_probe(struct snd_soc_component *component) ac97 = snd_soc_new_ac97_component(component, 0, 0); if (IS_ERR(ac97)) { ret = PTR_ERR(ac97); - dev_err(component->dev, "Failed to register AC97 component: %d\n", ret); - return ret; + return dev_err_probe(component->dev, ret, + "Failed to register AC97 component: %d\n", ret); } regmap = regmap_init_ac97(ac97, &ad1980_regmap_config); diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 8bd6067df..9fe18b22b 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -678,8 +678,8 @@ static int adau1701_probe(struct snd_soc_component *component) ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); if (ret < 0) { - dev_err(component->dev, "Failed to enable regulators: %d\n", ret); - return ret; + return dev_err_probe(component->dev, ret, + "Failed to enable regulators: %d\n", ret); } /* @@ -799,15 +799,15 @@ static int adau1701_i2c_probe(struct i2c_client *client) ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(adau1701->supplies), adau1701->supplies); if (ret < 0) { - dev_err(dev, "Failed to get regulators: %d\n", ret); - return ret; + return dev_err_probe(dev, ret, + "Failed to get regulators: %d\n", ret); } ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); if (ret < 0) { - dev_err(dev, "Failed to enable regulators: %d\n", ret); - return ret; + return dev_err_probe(dev, ret, + "Failed to enable regulators: %d\n", ret); } adau1701->client = client; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index c29324403..727bcccad 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -606,8 +606,8 @@ static int ssm260x_component_probe(struct snd_soc_component *component) ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0); if (ret < 0) { - dev_err(component->dev, "Failed to issue reset: %d\n", ret); - return ret; + return dev_err_probe(component->dev, ret, + "Failed to issue reset: %d\n", ret); } regmap_register_patch(ssm2602->regmap, ssm2602_patch, -- 2.34.1

1 1

[PATCH -next] aoa: core: Using helper function for_each_child_of_node()
by Lin Ruifeng 29 Aug '24

29 Aug '24

Helper function for_each_child_of_node() can iterate through the DT node, so we don't need to use while loop. Signed-off-by: Lin Ruifeng <linruifeng4(a)huawei.com> --- sound/aoa/core/gpio-feature.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c index 39bb409b27f6..01060f7c3862 100644 --- a/sound/aoa/core/gpio-feature.c +++ b/sound/aoa/core/gpio-feature.c @@ -72,7 +72,7 @@ static struct device_node *get_gpio(char *name, gpio = of_find_node_by_name(NULL, "gpio"); if (!gpio) return NULL; - while ((np = of_get_next_child(gpio, np))) { + for_each_child_of_node(gpio, np) { audio_gpio = of_get_property(np, "audio-gpio", NULL); if (!audio_gpio) continue; -- 2.17.1

1 0

[PATCH 0/2] Add a driver for the Iron Device SMA1307 Amp
by Kiseok Jo 29 Aug '24

29 Aug '24

This adds basic audio support for the Iron Device SMA1307 amplifier Kiseok Jo (2): ASoC: sma1307: Add driver for Iron Device SMA1307 ASoC: sma1307: Add bindings for Iron Device SMA1307 amplifier .../bindings/sound/irondevice,sma1307.yaml | 67 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2630 +++++++++++++++++ sound/soc/codecs/sma1307.h | 456 +++ 5 files changed, 3163 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h -- 2.39.2

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[PATCH v25 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 28 Aug '24

28 Aug '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addresed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (31): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: Allow for secondary interrupter to set IMOD usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: qcom: Use card and PCM index from QMI request ALSA: usb-audio: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 429 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 34 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 50 +- drivers/usb/host/xhci-sideband.c | 424 ++++ drivers/usb/host/xhci.c | 43 +- drivers/usb/host/xhci.h | 18 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 68 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 134 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 193 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 389 ++++ sound/soc/qcom/sm8250.c | 23 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 29 + sound/soc/soc-usb.c | 337 +++ sound/usb/Kconfig | 26 + sound/usb/Makefile | 4 +- sound/usb/card.c | 109 + sound/usb/card.h | 16 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1964 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 53 files changed, 5827 insertions(+), 53 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

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[PATCH v2-resend 0/2] ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local function
by Kuninori Morimoto 28 Aug '24

28 Aug '24

Hi Mark This is resend patch-set No driver is calling snd_soc_dpcm_can_be_xxx() functions. We don't need to have EXPORT_SYMBOL_GPL() for them. This patch-set makes it static function. v1 -> v2 - rebased to latest branch (for-6.12) Kuninori Morimoto (2): ASoC: soc-pcm: move snd_soc_dpcm_can_be_xxx() to top ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local function include/sound/soc-dpcm.h | 18 ---- sound/soc/soc-pcm.c | 201 +++++++++++++++++++-------------------- 2 files changed, 99 insertions(+), 120 deletions(-) -- 2.43.0

2 3

ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local function
by Kuninori Morimoto 28 Aug '24

28 Aug '24

Hi Mark No driver is calling snd_soc_dpcm_can_be_xxx() functions. We don't need to have EXPORT_SYMBOL_GPL() for them. This patch-set makes it static function. Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH 00/18] ASoC: Intel: boards: updates for 6.11
by Pierre-Louis Bossart 28 Aug '24

28 Aug '24

More simplifications from Brent Lu for Chromebooks, Bard Liao for SoundWire, and support for new Dell SKUs w/ Cirrus Logic codecs added by Charles Keepax. Bard Liao (7): ASoC: Intel: sof_sdw_cs42l42: use dai parameter ASoC: Intel: sof_sdw_rt711: use dai parameter ASoC: Intel: sof_sdw_rt5682: use dai parameter ASoC: Intel: sof_sdw_rt700: use dai parameter ASoC: Intel: sof_sdw_rt_dmic: use from dai parameter ASoC: Intel: sof_sdw_rt_sdca_jack_common: use dai parameter ASoC: Intel: sof_sdw: remove get_codec_dai_by_name Brent Lu (7): ASoC: SOF: sof-audio: rename dai clock setting query function ASoC: SOF: sof-audio: add sof_dai_get_tdm_slots function ASoC: SOF: ipc3-topology: support tdm slot number query ASoC: SOF: ipc4-topology: support tdm slot number query ASoC: Intel: maxim-common: rewrite max_98373_hw_params function ASoC: Intel: sof_da7219: remove local max98373 ops ASoC: Intel: sof_da7219: disable max98373 speaker pins in late_probe Charles Keepax (4): ASoC: Intel: sof_sdw: Add missing controls for cs42l43/cs35l56 ASoC: Intel: soc-acpi: Add match entries for some cs42l43 laptops ASoC: Intel: sof_sdw: Add quirks for some new Dell laptops ASoC: Intel: sof_sdw: Add quirks from some new Dell laptops include/sound/sof.h | 1 + sound/soc/intel/boards/sof_board_helpers.h | 2 - sound/soc/intel/boards/sof_da7219.c | 57 ++------ sound/soc/intel/boards/sof_maxim_common.c | 84 ++++++++++-- sound/soc/intel/boards/sof_sdw.c | 128 +++++++++++++++--- sound/soc/intel/boards/sof_sdw_common.h | 4 - sound/soc/intel/boards/sof_sdw_cs42l42.c | 11 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 11 +- sound/soc/intel/boards/sof_sdw_rt700.c | 11 +- sound/soc/intel/boards/sof_sdw_rt711.c | 11 +- sound/soc/intel/boards/sof_sdw_rt_dmic.c | 14 +- .../boards/sof_sdw_rt_sdca_jack_common.c | 11 +- .../intel/common/soc-acpi-intel-arl-match.c | 50 +++++++ .../intel/common/soc-acpi-intel-rpl-match.c | 50 +++++++ sound/soc/sof/ipc3-topology.c | 14 +- sound/soc/sof/ipc4-topology.c | 14 +- sound/soc/sof/sof-audio.c | 20 ++- sound/soc/sof/sof-audio.h | 9 +- 18 files changed, 338 insertions(+), 164 deletions(-) -- 2.43.0

4 21

[PATCH v7 0/4] ASoC: grace time for DPCM cleanup
by Kuninori Morimoto 28 Aug '24

28 Aug '24

Hi Mark, Pierre-Louis, Jerome Cc each ASoC driver maintainer This is v7 of DPCM cleanup As we discussed in [1], we don't need to use dpcm_playback/capture flag, so we remove it. But we have been using it for 10 years, some driver might get damage. The most likely case is that the device/driver can use both playback/capture, but have only one flag, and not using xxx_only flag. [1/3] patch indicates warning in such case. These adds grace time for DPCM cleanup. I'm not sure when dpcm_xxx will be removed, and Codec check bypass will be error, but maybe v6.12 or v6.13 ? Please check each driver by that time. Previous patch-set try to check both CPU and Codec in DPCM, but we noticed that there are some special DAI which we can't handle today [2]. So I will escape it in this patch-set. [1] https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com [2] https://lore.kernel.org/all/3e67d62d-fe08-4f55-ab5b-ece8a57154f9@linux.inte… Link: https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87wmo6dyxg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87msole5wc.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871q5tnuok.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87bk4oqerx.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/8734pctmte.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87r0ctwzr4.wl-kuninori.morimoto.gx@renesas.com v6 -> v7 - based on latest ASoC branch (for-6.12) v5 -> v6 - based on latest ASoC branch v4 -> v5 - typo fix: limition -> limitation - typo fix: catpure -> capture - include Intel patch v3 -> v4 - don't check Codec on DPCM - include Jerome's dpcm_xxx update patch v2 -> v3 - tidyup typo (reuqsts -> requests) - add Tested-by on git-log v1 -> v2 - tidyup Codec check warning output condition Jerome Brunet (1): ASoC: amlogic: do not use dpcm_playback/capture flags Kuninori Morimoto (2): ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: remove snd_soc_dai_link_set_capabilities() Pierre-Louis Bossart (1): ASoC: Intel: sof_sdw: use playback/capture_only flags include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/fsl/imx-card.c | 3 -- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/intel/boards/sof_sdw.c | 4 +- sound/soc/meson/axg-card.c | 11 +++-- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/soc-dai.c | 38 ---------------- sound/soc/soc-pcm.c | 65 +++++++++++++++------------ 13 files changed, 47 insertions(+), 88 deletions(-) -- 2.43.0

2 7

[PATCH v6 0/4] ASoC: grace time for DPCM cleanup
by Kuninori Morimoto 28 Aug '24

28 Aug '24

Hi Mark, Pierre-Louis, Jerome Cc each ASoC driver maintainer This is resend of v6 of DPCM cleanup As we discussed in [1], we don't need to use dpcm_playback/capture flag, so we remove it. But we have been using it for 10 years, some driver might get damage. The most likely case is that the device/driver can use both playback/capture, but have only one flag, and not using xxx_only flag. [1/3] patch indicates warning in such case. These adds grace time for DPCM cleanup. I'm not sure when dpcm_xxx will be removed, and Codec check bypass will be error, but maybe v6.12 or v6.13 ? Please check each driver by that time. Previous patch-set try to check both CPU and Codec in DPCM, but we noticed that there are some special DAI which we can't handle today [2]. So I will escape it in this patch-set. [1] https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com [2] https://lore.kernel.org/all/3e67d62d-fe08-4f55-ab5b-ece8a57154f9@linux.inte… Link: https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87wmo6dyxg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87msole5wc.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871q5tnuok.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87bk4oqerx.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/8734pctmte.wl-kuninori.morimoto.gx@renesas.com v5 -> v6 - based on latest ASoC branch v4 -> v5 - typo fix: limition -> limitation - typo fix: catpure -> capture - include Intel patch v3 -> v4 - don't check Codec on DPCM - include Jerome's dpcm_xxx update patch v2 -> v3 - tidyup typo (reuqsts -> requests) - add Tested-by on git-log v1 -> v2 - tidyup Codec check warning output condition Jerome Brunet (1): ASoC: amlogic: do not use dpcm_playback/capture flags Kuninori Morimoto (2): ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: remove snd_soc_dai_link_set_capabilities() Pierre-Louis Bossart (1): ASoC: Intel: sof_sdw: use playback/capture_only flags include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/fsl/imx-card.c | 3 -- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/intel/boards/sof_sdw.c | 4 +- sound/soc/meson/axg-card.c | 11 +++-- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/soc-dai.c | 38 ---------------- sound/soc/soc-pcm.c | 65 +++++++++++++++------------ 13 files changed, 47 insertions(+), 88 deletions(-) -- 2.43.0

2 5

[PATCH v5 0/4] ASoC: grace time for DPCM cleanup
by Kuninori Morimoto 28 Aug '24

28 Aug '24

Hi Mark, Pierre-Louis, Jerome Cc each ASoC driver maintainer This is v5 of DPCM cleanup As we discussed in [1], we don't need to use dpcm_playback/capture flag, so we remove it. But we have been using it for 10 years, some driver might get damage. The most likely case is that the device/driver can use both playback/capture, but have only one flag, and not using xxx_only flag. [1/3] patch indicates warning in such case. These adds grace time for DPCM cleanup. I'm not sure when dpcm_xxx will be removed, and Codec check bypass will be error, but maybe v6.11 or v6.12 ? Please check each driver by that time. Previous patch-set try to check both CPU and Codec in DPCM, but we noticed that there are some special DAI which we can't handle today [2]. So I will escape it in this patch-set. [1] https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com [2] https://lore.kernel.org/all/3e67d62d-fe08-4f55-ab5b-ece8a57154f9@linux.inte… Link: https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87wmo6dyxg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87msole5wc.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871q5tnuok.wl-kuninori.morimoto.gx@renesas.com v4 -> v5 - typo fix: limition -> limitation - typo fix: catpure -> capture - include Intel patch v3 -> v4 - don't check Codec on DPCM - include Jerome's dpcm_xxx update patch v2 -> v3 - tidyup typo (reuqsts -> requests) - add Tested-by on git-log v1 -> v2 - tidyup Codec check warning output condition Jerome Brunet (1): ASoC: amlogic: do not use dpcm_playback/capture flags Kuninori Morimoto (2): ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: remove snd_soc_dai_link_set_capabilities() Pierre-Louis Bossart (1): ASoC: Intel: sof_sdw: use playback/capture_only flags include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/fsl/imx-card.c | 3 -- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/intel/boards/sof_sdw.c | 4 +- sound/soc/meson/axg-card.c | 11 +++-- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/soc-dai.c | 38 ---------------- sound/soc/soc-pcm.c | 65 +++++++++++++++------------ 13 files changed, 47 insertions(+), 88 deletions(-) -- 2.43.0

2 9

Specifying supported hw params (channels, format, etc.) by the plug plugin in alsa config?
by Pavel Hofman 27 Aug '24

27 Aug '24

Hi, Please is it possible to restrict the hw params space for the plug plugin by some alsa configuration, so that it reports/accepts only a specific number of channels/specific sample format? I was not able to figure it out, I apologize if it's trivial :-) Thanks a lot. With regards, Pavel.

2 4

[PATCH -next 0/4] ASoC: fix module autoloading
by Liao Chen 27 Aug '24

27 Aug '24

Hi all, This patchset aims to enable autoloading of some use modules. By registering MDT, the kernel is allowed to automatically bind modules to devices that match the specified compatible strings. Liao Chen (4): ASoC: intel: fix module autoloading ASoC: google: fix module autoloading ASoC: tda7419: fix module autoloading ASoC: fix module autoloading sound/soc/codecs/chv3-codec.c | 1 + sound/soc/codecs/tda7419.c | 1 + sound/soc/google/chv3-i2s.c | 1 + sound/soc/intel/keembay/kmb_platform.c | 1 + 4 files changed, 4 insertions(+) -- 2.34.1

1 0

[PATCH v2] ASoC: tas2781: replace devm_kzalloc and scnprintf with devm_kstrdup
by Shenghao Ding 26 Aug '24

26 Aug '24

Replace devm_kzalloc and scnprintf with devm_kstrdup. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - use devm_kstrdup instead of devm_kasprintf --- sound/soc/codecs/tas2781-i2c.c | 23 ++++++++++------------- 1 file changed, 10 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index eb8732b1bb99..59fcce7fef7b 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -346,13 +346,11 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) } /* Create a mixer item for selecting the active profile */ - name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - GFP_KERNEL); + name = devm_kstrdup(tas_priv->dev, "Speaker Profile Id", GFP_KERNEL); if (!name) { ret = -ENOMEM; goto out; } - scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "Speaker Profile Id"); prof_ctrls[mix_index].name = name; prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; prof_ctrls[mix_index].info = tasdevice_info_profile; @@ -441,18 +439,13 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) goto out; } - /* Create a mixer item for selecting the active profile */ - prog_name = devm_kzalloc(tas_priv->dev, - SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + /* Create mixer items for selecting the active Program and Config */ + prog_name = devm_kstrdup(tas_priv->dev, "Speaker Program Id", GFP_KERNEL); - if (!prog_name || !conf_name) { + if (!prog_name) { ret = -ENOMEM; goto out; } - - scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Program Id"); dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -460,8 +453,12 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].put = tasdevice_program_put; mix_index++; - scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Config Id"); + conf_name = devm_kstrdup(tas_priv->dev, "Speaker Config Id", + GFP_KERNEL); + if (!conf_name) { + ret = -ENOMEM; + goto out; + } dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; -- 2.34.1

3 2

[PATCH] ASoC: MAINTAINERS: Drop incorrect tlv320aic31xx.txt path
by Krzysztof Kozlowski 26 Aug '24

26 Aug '24

tlv320aic31xx.txt was converted to DT schema (YAML) and new file is already matched by wildcard. This fixes get_maintainers.pl self-test warning: ./MAINTAINERS:22739: warning: no file matches F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt Fixes: e486feb7b8ec ("ASoC: dt-bindings: convert tlv320aic31xx.txt to yaml") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 6574b4910f8a..028186bb4e8d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22736,7 +22736,6 @@ F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt F: include/sound/tas2*.h F: include/sound/tlv320*.h F: include/sound/tpa6130a2-plat.h -- 2.43.0

2 1

[PATCH] MAINTAINERS: adjust file entries in TEXAS INSTRUMENTS AUDIO DRIVERS
by Lukas Bulwahn 26 Aug '24

26 Aug '24

From: Lukas Bulwahn <lukas.bulwahn(a)redhat.com> Commit e486feb7b8ec ("ASoC: dt-bindings: convert tlv320aic31xx.txt to yaml") converts tlv320aic31xx.txt to ti,tlv320dac3100.yaml, but misses to adjust the file entry in TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS. Hence, ./scripts/get_maintainer.pl --self-test=patterns complains about a broken reference. The pattern ti,tlv320*.yaml in the section already covers ti,tlv320dac3100.yaml and ti,tlv320adcx140.yaml. So, further file entries in this section are redundant. Adjust the file entries in TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS accordingly. Signed-off-by: Lukas Bulwahn <lukas.bulwahn(a)redhat.com> --- MAINTAINERS | 2 -- 1 file changed, 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 5dd88c28c0f7..6aa85a43d8c1 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22753,8 +22753,6 @@ F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml -F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt F: include/sound/tas2*.h F: include/sound/tlv320*.h F: include/sound/tpa6130a2-plat.h -- 2.45.2

1 0

[PATCH v1] ASoC: tas2781: mark const variables tas2563_dvc_table as __maybe_unused
by Shenghao Ding 26 Aug '24

26 Aug '24

In case of tas2781, tas2563_dvc_table will be unused, so mark it as __maybe_unused. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2563-tlv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h index faa3e194f73b..bb269b21f460 100644 --- a/include/sound/tas2563-tlv.h +++ b/include/sound/tas2563-tlv.h @@ -18,7 +18,7 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); /* pow(10, db/20) * pow(2,30) */ -static const unsigned char tas2563_dvc_table[][4] = { +static const __maybe_unused unsigned char tas2563_dvc_table[][4] = { { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ -- 2.34.1

4 6

[DISCUSSION] Why doesn't ALSA autogenerate/autoconfigure the mono profile by default?
by Xerox Slayer 25 Aug '24

25 Aug '24

Sorry if this is the wrong place to discuss this, but from what I managed to find out, this is primarily the responsibility of ALSA. This was pointed out by a reddit user, and to be frank, I was kinda surprised that this is not done by default. The user in question is hearing impaired (can listen on only one ear), so the user has no benefit of actually listening to stereo or any other multichannel audio, thus wanted to just set everything to mono (dual mono I presume) and was puzzled when there was no such option in pulseaudio (which pulls the profiles from ALSA). I did some more investigation and as it turns out, this profile is autogenerated, but only if the hardware has such capability (as is the case with any other profile). You can enable it manually by editing config files (in which case I presume the channel downconversion is done in software), but it seemed odd that one has to go through such trouble just to listen to mono sound. Thus, I think the mono profile should be taken as a special case scenario and be autogenerated/configured by default, regardless if it's supported in hardware or not. Once again, I'm sorry if this is not the place to ask/discuss this, but would appreciate if someone pointed me to where I can actually do this.

2 2

[PATCH v4] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 25 Aug '24

25 Aug '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. So far, the Software maintainer has not been confirmed. Once the maintainer was confimred, the guy will update his mail into audio converter section. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v4: - Add Touch Screen support v3: - Add Audio converter section - Set the section of LOW/MIDDLE-POWER AUDIO AMPLIFIER as Supported v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 34 +++++++++++++++++++++++++--------- 1 file changed, 25 insertions(+), 9 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 878dcd23b331..ff363eac55b1 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22591,26 +22591,43 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml +F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt -F: Documentation/devicetree/bindings/sound/tpa6130a2.txt -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -22620,9 +22637,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

[PATCH 0/2] Add SOF support for ACP7.0 based platform
by Vijendar Mukunda 24 Aug '24

24 Aug '24

This patch series adds SOF support for ACP7.0 based platform. Link: https://github.com/thesofproject/linux/pull/5139 Vijendar Mukunda (2): ASoC: amd: Add acpi machine id for acp7.0 version based platform ASoC: SOF: amd: add support for acp7.0 based platform sound/soc/amd/acp-config.c | 12 +++ sound/soc/amd/mach-config.h | 1 + sound/soc/sof/amd/Kconfig | 10 ++ sound/soc/sof/amd/Makefile | 4 +- sound/soc/sof/amd/acp-dsp-offset.h | 24 ++++- sound/soc/sof/amd/acp.c | 65 ++++++++++--- sound/soc/sof/amd/acp.h | 9 ++ sound/soc/sof/amd/acp70.c | 142 +++++++++++++++++++++++++++++ sound/soc/sof/amd/pci-acp70.c | 112 +++++++++++++++++++++++ 9 files changed, 365 insertions(+), 14 deletions(-) create mode 100644 sound/soc/sof/amd/acp70.c create mode 100644 sound/soc/sof/amd/pci-acp70.c -- 2.34.1

2 3

[PATCH] ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict
by Hans de Goede 24 Aug '24

24 Aug '24

There are 2G and 4G RAM versions of the Lenovo Yoga Tab 3 X90F and it turns out that the 2G version has a DMI product name of "CHERRYVIEW D1 PLATFORM" where as the 4G version has "CHERRYVIEW C0 PLATFORM". The sys-vendor + product-version check are unique enough that the product-name check is not necessary. Drop the product-name check so that the existing DMI match for the 4G RAM version also matches the 2G RAM version. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 5e2ec60e2954..e4c3492a0c28 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -84,7 +84,6 @@ static const struct dmi_system_id lenovo_yoga_tab3_x90[] = { /* Lenovo Yoga Tab 3 Pro YT3-X90, codec missing from DSDT */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), - DMI_MATCH(DMI_PRODUCT_NAME, "CHERRYVIEW D1 PLATFORM"), DMI_MATCH(DMI_PRODUCT_VERSION, "Blade3-10A-001"), }, }, -- 2.46.0

3 2

[PATCH 6.11 regression fix] ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards harder
by Hans de Goede 24 Aug '24

24 Aug '24

Since commit 13f58267cda3 ("ASoC: soc.h: don't create dummy Component via COMP_DUMMY()") dummy codecs declared like this: SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); expand to: static struct snd_soc_dai_link_component dummy[] = { }; Which means that dummy is a zero sized array and thus dais[i].codecs should not be dereferenced *at all* since it points to the address of the next variable stored in the data section as the "dummy" variable has an address but no size, so even dereferencing dais[0] is already an out of bounds array reference. Which means that the if (dais[i].codecs->name) check added in commit 7d99a70b6595 ("ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards") relies on that the part of the next variable which the name member maps to just happens to be NULL. Which apparently so far it usually is, except when it isn't and then it results in crashes like this one: [ 28.795659] BUG: unable to handle page fault for address: 0000000000030011 ... [ 28.795780] Call Trace: [ 28.795787] <TASK> ... [ 28.795862] ? strcmp+0x18/0x40 [ 28.795872] 0xffffffffc150c605 [ 28.795887] platform_probe+0x40/0xa0 ... [ 28.795979] ? __pfx_init_module+0x10/0x10 [snd_soc_sst_bytcr_wm5102] Really fix things this time around by checking dais.num_codecs != 0. Fixes: 7d99a70b6595 ("ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards") Cc: stable(a)vger.kernel.org Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- 9 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index dce6a2086f2a..6da1517c53c6 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -605,7 +605,7 @@ static int broxton_audio_probe(struct platform_device *pdev) int i; for (i = 0; i < ARRAY_SIZE(broxton_rt298_dais); i++) { - if (card->dai_link[i].codecs->name && + if (card->dai_link[i].num_codecs && !strncmp(card->dai_link[i].codecs->name, "i2c-INT343A:00", I2C_NAME_SIZE)) { if (!strncmp(card->name, "broxton-rt298", diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index c014d85a08b2..df3c2a7b64d2 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -241,7 +241,7 @@ static int snd_byt_cht_cx2072x_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_cht_cx2072x_dais); i++) { - if (byt_cht_cx2072x_dais[i].codecs->name && + if (byt_cht_cx2072x_dais[i].num_codecs && !strcmp(byt_cht_cx2072x_dais[i].codecs->name, "i2c-14F10720:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index f4ac3ddd148b..08c598b7e1ee 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -245,7 +245,7 @@ static int bytcht_da7213_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(dailink); i++) { - if (dailink[i].codecs->name && + if (dailink[i].num_codecs && !strcmp(dailink[i].codecs->name, "i2c-DLGS7213:00")) { dai_index = i; break; diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 2fcec2e02bb5..77b91ea4dc32 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -546,7 +546,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_cht_es8316_dais); i++) { - if (byt_cht_es8316_dais[i].codecs->name && + if (byt_cht_es8316_dais[i].num_codecs && !strcmp(byt_cht_es8316_dais[i].codecs->name, "i2c-ESSX8316:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index a64d1989e28a..db4a33680d94 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1677,7 +1677,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_rt5640_dais); i++) { - if (byt_rt5640_dais[i].codecs->name && + if (byt_rt5640_dais[i].num_codecs && !strcmp(byt_rt5640_dais[i].codecs->name, "i2c-10EC5640:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 80c841b000a3..8514b79f389b 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -910,7 +910,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_rt5651_dais); i++) { - if (byt_rt5651_dais[i].codecs->name && + if (byt_rt5651_dais[i].num_codecs && !strcmp(byt_rt5651_dais[i].codecs->name, "i2c-10EC5651:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index cccb5e90c0fe..e5a7cc606aa9 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -605,7 +605,7 @@ static int snd_byt_wm5102_mc_probe(struct platform_device *pdev) /* find index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_wm5102_dais); i++) { - if (byt_wm5102_dais[i].codecs->name && + if (byt_wm5102_dais[i].num_codecs && !strcmp(byt_wm5102_dais[i].codecs->name, "wm5102-codec")) { dai_index = i; diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index eb41b7115d01..1da9ceee4d59 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -569,7 +569,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* set correct codec name */ for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) - if (cht_dailink[i].codecs->name && + if (cht_dailink[i].num_codecs && !strcmp(cht_dailink[i].codecs->name, "i2c-10EC5645:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index be2d1a8dbca8..d68e5bc755de 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -466,7 +466,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* find index of codec dai */ for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) { - if (cht_dailink[i].codecs->name && + if (cht_dailink[i].num_codecs && !strcmp(cht_dailink[i].codecs->name, RT5672_I2C_DEFAULT)) { dai_index = i; break; -- 2.46.0

3 3

[GIT PULL] ASoC fixes for v6.11-rc3
by Mark Brown 23 Aug '24

23 Aug '24

The following changes since commit 72776774b55bb59b7b1b09117e915a5030110304: ASoC: cs35l56: Patch CS35L56_IRQ1_MASK_18 to the default value (2024-08-08 20:34:56 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.11-rc3 for you to fetch changes up to bb4485562f5907708f1c218b5d70dce04165d1e1: ASoC: cs-amp-lib: Ignore empty UEFI calibration entries (2024-08-22 17:52:26 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.11 A relatively large collection of fixes here, all driver specific and none of them particularly major, plus one MAINTAINERS update. There's been a bunch of work on module autoloading from several people. ---------------------------------------------------------------- Albert Jakieła (1): ASoC: SOF: mediatek: Add missing board compatible Dmitry Baryshkov (2): ASoC: codecs: lpass-macro: fix version strings returned for 1.x codecs ASoC: codecs: lpass-va-macro: warn on unknown version Hongbo Li (2): ASoC: allow module autoloading for table db1200_pids ASoC: allow module autoloading for table board_ids Krzysztof Kozlowski (2): ASoC: MAINTAINERS: Drop Banajit Goswami from Qualcomm sound drivers ASoC: codecs: wcd937x: Fix missing de-assert of reset GPIO Mark Brown (1): ASoC: codecs: warn on unknown codec version Richard Fitzgerald (2): ASoC: cs-amp-lib-test: Force test calibration blob entries to be valid ASoC: cs-amp-lib: Ignore empty UEFI calibration entries Vijendar Mukunda (3): ASoC: SOF: amd: move iram-dram fence register programming sequence ASoC: SOF: amd: Fix for incorrect acp error register offsets ASoC: SOF: amd: Fix for acp init sequence YR Yang (1): ASoC: mediatek: mt8188: Mark AFE_DAC_CON0 register as volatile Yuntao Liu (1): ASoC: amd: acp: fix module autoloading MAINTAINERS | 1 - sound/soc/amd/acp/acp-legacy-mach.c | 2 ++ sound/soc/amd/acp/acp-sof-mach.c | 2 ++ sound/soc/au1x/db1200.c | 1 + sound/soc/codecs/cs-amp-lib-test.c | 9 ++++++ sound/soc/codecs/cs-amp-lib.c | 7 +++- sound/soc/codecs/lpass-macro-common.h | 6 ++++ sound/soc/codecs/lpass-va-macro.c | 4 +++ sound/soc/codecs/wcd937x.c | 5 ++- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 1 + sound/soc/sof/amd/acp-dsp-offset.h | 6 ++-- sound/soc/sof/amd/acp.c | 52 ++++++++++++++++++++---------- sound/soc/sof/amd/acp.h | 9 ++++-- sound/soc/sof/amd/pci-acp63.c | 2 ++ sound/soc/sof/amd/pci-rmb.c | 2 ++ sound/soc/sof/amd/pci-rn.c | 2 ++ sound/soc/sof/mediatek/mt8195/mt8195.c | 3 ++ 17 files changed, 88 insertions(+), 26 deletions(-)

2 1

[PATCH v8 0/4] ASoC: grace time for DPCM cleanup
by Kuninori Morimoto 23 Aug '24

23 Aug '24

Hi Mark, Pierre-Louis, Jerome Cc each ASoC driver maintainer This is v8 of DPCM cleanup As we discussed in [1], we don't need to use dpcm_playback/capture flag, so we remove it. But we have been using it for 10 years, some driver might get damage. The most likely case is that the device/driver can use both playback/capture, but have only one flag, and not using xxx_only flag. [1/3] patch indicates warning in such case. These adds grace time for DPCM cleanup. I'm not sure when dpcm_xxx will be removed, and Codec check bypass will be error, but maybe v6.12 or v6.13 ? Please check each driver by that time. Previous patch-set try to check both CPU and Codec in DPCM, but we noticed that there are some special DAI which we can't handle today [2]. So I will escape it in this patch-set. [1] https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com [2] https://lore.kernel.org/all/3e67d62d-fe08-4f55-ab5b-ece8a57154f9@linux.inte… Link: https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87wmo6dyxg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87msole5wc.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871q5tnuok.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87bk4oqerx.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/8734pctmte.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87r0ctwzr4.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87cymvlmki.wl-kuninori.morimoto.gx@renesas.com v8 -> v7 - rebase to latest ASoC branch (for-6.12) v6 -> v7 - based on latest ASoC branch (for-6.12) v5 -> v6 - based on latest ASoC branch v4 -> v5 - typo fix: limition -> limitation - typo fix: catpure -> capture - include Intel patch v3 -> v4 - don't check Codec on DPCM - include Jerome's dpcm_xxx update patch v2 -> v3 - tidyup typo (reuqsts -> requests) - add Tested-by on git-log v1 -> v2 - tidyup Codec check warning output condition Jerome Brunet (1): ASoC: amlogic: do not use dpcm_playback/capture flags Kuninori Morimoto (3): ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: remove snd_soc_dai_link_set_capabilities() ASoC: Intel: sof_sdw: use playback/capture_only flags include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/fsl/imx-card.c | 3 -- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/meson/axg-card.c | 11 +++-- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/sdw_utils/soc_sdw_utils.c | 4 +- sound/soc/soc-dai.c | 38 ---------------- sound/soc/soc-pcm.c | 65 +++++++++++++++------------ 13 files changed, 47 insertions(+), 88 deletions(-) -- 2.43.0

2 5

ALC285 headphone no sound
by Kailang 22 Aug '24

22 Aug '24

Hi Takashi, I also changed with ALC215 ALC285 ALC289 ALC225 ALC295 ALC299 codec. It maybe has similar issue as ALC256. Let it has same procedure. Many Thanks. BR, Kailang > -----Original Message----- > From: Kailang > Sent: Thursday, August 22, 2024 11:14 AM > To: Takashi Iwai (tiwai(a)suse.de) <tiwai(a)suse.de> > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > Subject: ALC256 headphone no sound > > Hi Takashi, > > Attach patch will solve headphone issue for alc256. > Many Thanks. > > BR, > Kailang

2 1

Re: [PATCH] [RFC] ASoC: Conditional PCM support
by Jaroslav Kysela 22 Aug '24

22 Aug '24

On 21. 08. 24 12:18, Cezary Rojewski wrote: > Conditional PCM (condpcm) helps facilitate modern audio usecases such as > Echo Cancellations and Noise Reduction. These are not invoked by the > means of userspace application opening an endpoint (FrontEnd) but are a > "side effect" of selected PCMs running simultaneously e.g.: if both > Speaker (source) and Microphone Array (sink) are running, reference > data from the Speaker and take it into account when processing capture > for better voice command detection ratio. > > Which PCMs are needed for given conditional PCM to be spawned is > determinated by the driver when registering the condpcm. Perhaps the naming may be changed to "sidepcm" or "monitorpcm"/"monpcm". It looks better in my eyes. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

2 1

ALC256 headphone no sound
by Kailang 22 Aug '24

22 Aug '24

Hi Takashi, Attach patch will solve headphone issue for alc256. Many Thanks. BR, Kailang

2 2

[PATCH v2 0/2] ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local function
by Kuninori Morimoto 22 Aug '24

22 Aug '24

Hi Mark No driver is calling snd_soc_dpcm_can_be_xxx() functions. We don't need to have EXPORT_SYMBOL_GPL() for them. This patch-set makes it static function. v1 -> v2 - rebased to latest branch (for-6.12) Kuninori Morimoto (2): ASoC: soc-pcm: move snd_soc_dpcm_can_be_xxx() to top ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local function include/sound/soc-dpcm.h | 18 ---- sound/soc/soc-pcm.c | 201 +++++++++++++++++++-------------------- 2 files changed, 99 insertions(+), 120 deletions(-) -- 2.43.0

2 6

USB-Audio: Add support for Focusrite 4th Gen devices
by GitHub pull_request - opened 21 Aug '24

21 Aug '24

alsa-project/alsa-ucm-conf pull request #438 was opened from ChrisLane: Extending the existing support for Focusrite Scarlett Solo and 2i2 interfaces with support for the 4th Gen devices. I have a Focusrite Scarlett 2i2 which is what I have tested with and used to get the device ID. For the Solo device, I grabbed the device ID from [linux-hardware.org](https://linux-hardware.org/?id=usb:1235-8218). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/438 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/438.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v24 00/34] Introduce QC USB SND audio offloading support
by Wesley Cheng 21 Aug '24

21 Aug '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addresed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (32): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: Allow for secondary interrupter to set IMOD usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: qcom: Use card and PCM index from QMI request ALSA: usb-audio: qcom: Populate PCM and USB chip information ALSA: usb-audio: qcom: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 435 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 36 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 50 +- drivers/usb/host/xhci-sideband.c | 419 ++++ drivers/usb/host/xhci.c | 43 +- drivers/usb/host/xhci.h | 18 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-sideband.h | 68 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 135 ++ sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 193 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 372 ++++ sound/soc/qcom/sm8250.c | 23 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 29 + sound/soc/soc-usb.c | 331 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 2 +- sound/usb/card.c | 109 + sound/usb/card.h | 15 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 6 + sound/usb/qcom/mixer_usb_offload.c | 101 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1937 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 49 files changed, 5768 insertions(+), 50 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/mixer_usb_offload.c create mode 100644 sound/usb/qcom/mixer_usb_offload.h create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

5 86

No sound from XonarDX
by GitHub issues - opened 21 Aug '24

21 Aug '24

alsa-project/alsa-lib issue #408 was opened from 3bitbrain: No matter what wiki or advice I try, no sound at all from a known good Xonar DX card. Distro : Gentoo, fresh install. Always un-muted everything in the console alsamixer. alsa-lib version 1.2.11 [alsa.txt](https://github.com/user-attachments/files/16697602/alsa.txt) Issue URL : https://github.com/alsa-project/alsa-lib/issues/408 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: soc-pcm: remove snd_soc_dpcm_stream_lock_irqsave_nested()
by Kuninori Morimoto 21 Aug '24

21 Aug '24

soc-pcm.c has snd_soc_dpcm_stream_lock_irqsave_nested() / snd_soc_dpcm_stream_unlock_irqrestore() helper function, but it is almost nothing help. It just makes a code complex. Let's remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - rebased to latest branch (for-6.12) sound/soc/soc-pcm.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index f79abcf36105..baf8b81e7110 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -49,12 +49,6 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, return ret; } -#define snd_soc_dpcm_stream_lock_irqsave_nested(rtd, stream, flags) \ - snd_pcm_stream_lock_irqsave_nested(snd_soc_dpcm_get_substream(rtd, stream), flags) - -#define snd_soc_dpcm_stream_unlock_irqrestore(rtd, stream, flags) \ - snd_pcm_stream_unlock_irqrestore(snd_soc_dpcm_get_substream(rtd, stream), flags) - #define DPCM_MAX_BE_USERS 8 static inline const char *soc_cpu_dai_name(struct snd_soc_pcm_runtime *rtd) @@ -2144,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be = dpcm->be; be_substream = snd_soc_dpcm_get_substream(be, stream); - snd_soc_dpcm_stream_lock_irqsave_nested(be, stream, flags); + snd_pcm_stream_lock_irqsave_nested(be_substream, flags); /* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) @@ -2291,7 +2285,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, break; } next: - snd_soc_dpcm_stream_unlock_irqrestore(be, stream, flags); + snd_pcm_stream_unlock_irqrestore(be_substream, flags); if (ret) break; } -- 2.43.0

2 1

[PATCH] ASoC: SOF: amd: remove unused variable from sof_amd_acp_desc structure
by Vijendar Mukunda 21 Aug '24

21 Aug '24

Remove unused structure member 'rev' from sof_amd_acp_desc structure. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/amd/acp.h | 1 - sound/soc/sof/amd/pci-acp63.c | 1 - sound/soc/sof/amd/pci-rmb.c | 1 - sound/soc/sof/amd/pci-rn.c | 1 - sound/soc/sof/amd/pci-vangogh.c | 1 - 5 files changed, 5 deletions(-) diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 321c40cc6d50..11def07efc0f 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -190,7 +190,6 @@ struct acp_dsp_stream { }; struct sof_amd_acp_desc { - unsigned int rev; const char *name; unsigned int host_bridge_id; u32 pgfsm_base; diff --git a/sound/soc/sof/amd/pci-acp63.c b/sound/soc/sof/amd/pci-acp63.c index e90658ba2bd7..b54ed61b79ed 100644 --- a/sound/soc/sof/amd/pci-acp63.c +++ b/sound/soc/sof/amd/pci-acp63.c @@ -28,7 +28,6 @@ #define ACP6x_REG_END 0x125C000 static const struct sof_amd_acp_desc acp63_chip_info = { - .rev = 6, .host_bridge_id = HOST_BRIDGE_ACP63, .pgfsm_base = ACP6X_PGFSM_BASE, .ext_intr_enb = ACP6X_EXTERNAL_INTR_ENB, diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index a366f904e6f3..c45256bf4fda 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -28,7 +28,6 @@ #define ACP6X_FUTURE_REG_ACLK_0 0x1854 static const struct sof_amd_acp_desc rembrandt_chip_info = { - .rev = 6, .host_bridge_id = HOST_BRIDGE_RMB, .pgfsm_base = ACP6X_PGFSM_BASE, .ext_intr_stat = ACP6X_EXT_INTR_STAT, diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index 2b7c53470ce8..386a0f1e7ee0 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -28,7 +28,6 @@ #define ACP3X_FUTURE_REG_ACLK_0 0x1860 static const struct sof_amd_acp_desc renoir_chip_info = { - .rev = 3, .host_bridge_id = HOST_BRIDGE_CZN, .pgfsm_base = ACP3X_PGFSM_BASE, .ext_intr_stat = ACP3X_EXT_INTR_STAT, diff --git a/sound/soc/sof/amd/pci-vangogh.c b/sound/soc/sof/amd/pci-vangogh.c index eba580840100..cb845f81795e 100644 --- a/sound/soc/sof/amd/pci-vangogh.c +++ b/sound/soc/sof/amd/pci-vangogh.c @@ -26,7 +26,6 @@ #define ACP5X_FUTURE_REG_ACLK_0 0x1864 static const struct sof_amd_acp_desc vangogh_chip_info = { - .rev = 5, .name = "vangogh", .host_bridge_id = HOST_BRIDGE_VGH, .pgfsm_base = ACP5X_PGFSM_BASE, -- 2.34.1

2 1

[PATCH] ASoC: codecs: wcd934x: make read-only array minCode_param static const
by Colin Ian King 21 Aug '24

21 Aug '24

Don't populate the read-only array minCode_param on the stack at run time, instead make it static const. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/codecs/wcd934x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 291d0c80a6fc..910852eb9698 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -2643,8 +2643,8 @@ static void wcd934x_mbhc_get_result_params(struct wcd934x_codec *wcd934x, s16 c1; s32 x1, d1; int32_t denom; - int minCode_param[] = { - 3277, 1639, 820, 410, 205, 103, 52, 26 + static const int minCode_param[] = { + 3277, 1639, 820, 410, 205, 103, 52, 26 }; regmap_update_bits(wcd934x->regmap, WCD934X_ANA_MBHC_ZDET, 0x20, 0x20); -- 2.39.2

2 1

[PATCH v1] ASoC: tas2781: Remove unnecessary line feed for tasdevice_codec_remove
by Baojun Xu 21 Aug '24

21 Aug '24

Remove unnecessary line feed for tasdevice_codec_remove. Add comma at the end the last member of the array. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v1: --- sound/soc/codecs/tas2781-i2c.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9a32e0504857..eb8732b1bb99 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -582,13 +582,13 @@ static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), SND_SOC_DAPM_OUTPUT("OUT"), - SND_SOC_DAPM_INPUT("DMIC") + SND_SOC_DAPM_INPUT("DMIC"), }; static const struct snd_soc_dapm_route tasdevice_audio_map[] = { {"SPK", NULL, "ASI"}, {"OUT", NULL, "SPK"}, - {"ASI OUT", NULL, "DMIC"} + {"ASI OUT", NULL, "DMIC"}, }; static int tasdevice_startup(struct snd_pcm_substream *substream, @@ -730,8 +730,7 @@ static void tasdevice_deinit(void *context) tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; } -static void tasdevice_codec_remove( - struct snd_soc_component *codec) +static void tasdevice_codec_remove(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); -- 2.43.0

1 0

[PATCH] ASoC: soc-pcm: remove snd_soc_dpcm_stream_lock_irqsave_nested()
by Kuninori Morimoto 21 Aug '24

21 Aug '24

soc-pcm.c has snd_soc_dpcm_stream_lock_irqsave_nested() / snd_soc_dpcm_stream_unlock_irqrestore() helper function, but it is almost nothing help. It just makes a code complex. Let's remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 58381801e41d4..4571785736f76 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -49,12 +49,6 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, return ret; } -#define snd_soc_dpcm_stream_lock_irqsave_nested(rtd, stream, flags) \ - snd_pcm_stream_lock_irqsave_nested(snd_soc_dpcm_get_substream(rtd, stream), flags) - -#define snd_soc_dpcm_stream_unlock_irqrestore(rtd, stream, flags) \ - snd_pcm_stream_unlock_irqrestore(snd_soc_dpcm_get_substream(rtd, stream), flags) - #define DPCM_MAX_BE_USERS 8 static inline const char *soc_cpu_dai_name(struct snd_soc_pcm_runtime *rtd) @@ -2144,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be = dpcm->be; be_substream = snd_soc_dpcm_get_substream(be, stream); - snd_soc_dpcm_stream_lock_irqsave_nested(be, stream, flags); + snd_pcm_stream_lock_irqsave_nested(be_substream, flags); /* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) @@ -2291,7 +2285,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, break; } next: - snd_soc_dpcm_stream_unlock_irqrestore(be, stream, flags); + snd_pcm_stream_unlock_irqrestore(be_substream, flags); if (ret) break; } -- 2.43.0

2 2

Re: [PATCH] ALSA: hda: cs35l56: Don't use the device index as a calibration index
by Takashi Iwai 21 Aug '24

21 Aug '24

On Wed, 21 Aug 2024 14:47:11 +0200, Simon Trimmer wrote: > > The HDA driver cannot assume that the order that the devices are > specified in the cirrus,dev-index matches the order of calibration > entries. > > Only a calibration entry with a matching silicon id will be used. > > Fixes: cfa43aaa7948 ("ALSA: hda: cs35l56: Apply amp calibration from EFI data") > Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Thanks, applied now. Takashi

1 0

Re: [RESEND] ASoC: mediatek: mt8188-mt6359: Modify key
by Mark Brown 21 Aug '24

21 Aug '24

On Wed, Aug 21, 2024 at 03:40:51PM +0800, Zhang Yi wrote: > In order to get the correct keys when using the ES8326.We will associate > SND_JACK_BTN_1 to KEY_VOLUMEUP and SND_JACK_BTN_2 to KEY_VOLUMEDOWN > when the ES8326 flag is recognized. Please please allow a reasonable time for review. People get busy, go on holiday, attend conferences and so on so unless there is some reason for urgency (like critical bug fixes) please allow at least a couple of weeks for review.

1 0

[PATCH v2 1/1] ASoC: dt-bindings: Convert tpa6130a2.txt to yaml
by Frank Li 20 Aug '24

20 Aug '24

Convert binding doc tpa6130a2.txt to yaml format. Additional change: - add ref to dai-common.yaml - add i2c node in example Fix below warning: arch/arm64/boot/dts/freescale/imx8mq-zii-ultra-rmb3.dtb: /soc@0/bus@30800000/i2c@30a20000/amp@60: failed to match any schema with compatible: ['ti,tpa6130a2'] Reviewed-by: Rob Herring (Arm) <robh(a)kernel.org> Signed-off-by: Frank Li <Frank.Li(a)nxp.com> --- Change from v1 to v2 - fix MAINTAINERS --- .../bindings/sound/ti,tpa6130a2.yaml | 55 +++++++++++++++++++ .../devicetree/bindings/sound/tpa6130a2.txt | 27 --------- MAINTAINERS | 2 +- 3 files changed, 56 insertions(+), 28 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tpa6130a2.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml new file mode 100644 index 0000000000000..a42bf9bde6940 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tpa6130a2.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments - tpa6130a2 Codec module + +maintainers: + - Sebastian Reichel <sre(a)kernel.org> + +description: + Stereo, analog input headphone amplifier + +properties: + compatible: + enum: + - ti,tpa6130a2 + - ti,tpa6140a2 + + reg: + maxItems: 1 + + Vdd-supply: + description: power supply regulator + + power-gpio: + description: gpio pin to power the device + +required: + - compatible + - reg + - Vdd-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@60 { + compatible = "ti,tpa6130a2"; + reg = <0x60>; + Vdd-supply = <&vmmc2>; + power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/tpa6130a2.txt b/Documentation/devicetree/bindings/sound/tpa6130a2.txt deleted file mode 100644 index 6dfa740e4b2d8..0000000000000 --- a/Documentation/devicetree/bindings/sound/tpa6130a2.txt +++ /dev/null @@ -1,27 +0,0 @@ -Texas Instruments - tpa6130a2 Codec module - -The tpa6130a2 serial control bus communicates through I2C protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tpa6130a2" - TPA6130A2 - "ti,tpa6140a2" - TPA6140A2 - - -- reg - <int> - I2C slave address - -- Vdd-supply - <phandle> - power supply regulator - -Optional properties: - -- power-gpio - gpio pin to power the device - -Example: - -tpa6130a2: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - reg = <0x60>; - Vdd-supply = <&vmmc2>; - power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; -}; diff --git a/MAINTAINERS b/MAINTAINERS index 118d9cf64e9cb..4ced49519930b 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22713,12 +22713,12 @@ F: Documentation/devicetree/bindings/sound/tas2552.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt -F: Documentation/devicetree/bindings/sound/tpa6130a2.txt F: include/sound/tas2*.h F: include/sound/tlv320*.h F: include/sound/tpa6130a2-plat.h -- 2.34.1

1 0

[PATCH] ASoC: amd: acp: replace desc->rev check with acp pci revision id
by Vijendar Mukunda 20 Aug '24

20 Aug '24

Replace acp descriptor structure member 'rev' check with acp pci revision id. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/sof/amd/acp.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 0f6115c8b005..e4d46fdda88b 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -236,7 +236,6 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, unsigned int image_length) { struct snd_sof_dev *sdev = adata->dev; - const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int tx_count, fw_qualifier, val; int ret; @@ -265,8 +264,9 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_DESTINATION_ADDR, dest_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_MSG_LENGTH, image_length); - /* psp_send_cmd only required for vangogh platform (rev - 5) */ - if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { + /* psp_send_cmd only required for vangogh platform */ + if (adata->pci_rev == ACP_VANGOGH_PCI_ID && + !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { /* Modify IRAM and DRAM size */ ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); if (ret) @@ -285,8 +285,8 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return ret; } - /* psp_send_cmd only required for renoir platform (rev - 3) */ - if (desc->rev == 3) { + /* psp_send_cmd only required for renoir platform*/ + if (adata->pci_rev == ACP_RN_PCI_ID) { ret = psp_send_cmd(adata, MBOX_ACP_SHA_DMA_COMMAND); if (ret) return ret; @@ -405,7 +405,7 @@ static irqreturn_t acp_irq_handler(int irq, void *dev_id) snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_stat, ACP_ERROR_IRQ_MASK); snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->acp_sw0_i2s_err_reason, 0); /* ACP_SW1_I2S_ERROR_REASON is newly added register from rmb platform onwards */ - if (desc->rev >= 6) + if (adata->pci_rev >= ACP_RMB_PCI_ID) snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SW1_I2S_ERROR_REASON, 0); snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->acp_error_stat, 0); irq_flag = 1; @@ -431,6 +431,7 @@ static irqreturn_t acp_irq_handler(int irq, void *dev_id) static int acp_power_on(struct snd_sof_dev *sdev) { const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); + struct acp_dev_data *adata = sdev->pdata->hw_pdata; unsigned int base = desc->pgfsm_base; unsigned int val; unsigned int acp_pgfsm_status_mask, acp_pgfsm_cntl_mask; @@ -441,13 +442,14 @@ static int acp_power_on(struct snd_sof_dev *sdev) if (val == ACP_POWERED_ON) return 0; - switch (desc->rev) { - case 3: - case 5: + switch (adata->pci_rev) { + case ACP_RN_PCI_ID: + case ACP_VANGOGH_PCI_ID: acp_pgfsm_status_mask = ACP3X_PGFSM_STATUS_MASK; acp_pgfsm_cntl_mask = ACP3X_PGFSM_CNTL_POWER_ON_MASK; break; - case 6: + case ACP_RMB_PCI_ID: + case ACP63_PCI_ID: acp_pgfsm_status_mask = ACP6X_PGFSM_STATUS_MASK; acp_pgfsm_cntl_mask = ACP6X_PGFSM_CNTL_POWER_ON_MASK; break; -- 2.34.1

2 1

[RFC PATCH 00/16] soundwire/ASoC: speed-up downloads with BTP/BRA protocol
by Pierre-Louis Bossart 20 Aug '24

20 Aug '24

This RFC patchset suggests a new API for ASoC codec drivers to use for firmware/table downloads. Problem statement: All existing transfers initiated by codec drivers rely on SoundWire read/write commands, which can only support ONE byte per frame. With the typical 48kHz frame rate, this means 384 kbits/s. In addition, the command/control is typically handled with a FIFO and interrupts which adds more software overhead. To give a practical reference, sending 32Kb takes 2.5s on Intel platforms, which means about 105kbit/s only. Additional non-audio activity is likely to adversely impact interrupt scheduling and further decrease the transfer speeds. New SDCA-based codecs have a need to download tables and DSP firmware which are typically between 20 and 256 Kb. The slow bus operation has a direct impact on boot/resume times, and clearly having to wait more than 300ms is a showstopper in terms of latency requirements and user-experience. Suggested solution: The MIPI specification and most of the new codecs support the Bulk Transfer Protocol (BTP) and specifically the Bulk Register Access (BRA) configuration. This mode reclaims the 'audio' data space of the SoundWire frame to send firmware/coefficients over the DataPort 0 (DP0). The API suggested is rather simple, with the following sequence expected: open(): reserve resources and prepare hardware send_async(): trigger DMAs and perform SoundWire bank switch wait(): wait for DMA completion and disable DMAs close(): release resources Benefits: Even after accounting for the protocol overhead, the data can be sent 8x or 16x faster on the link than with the regular commands. With the use of DMAs, the software overhead becomes limited to the initialization. Measured results show that transferring the same 32Kb takes about 100ms, a 25x improvement on the baseline write() commands with an actual bitrate of 2.6 Mbits/s. These results are a measure of bus/hardware performance mainly, and will typically not be too modified by the CPU activity and scheduling. The performance for reads is similar, with a 25x speedup measured. Limitations: Setting up the transfers over DP0 takes time, and the reliance on DMAs on the host side brings alignment restrictions. The BTP/BRA protocol is really only relevant for "large" transfers done during boot/resume *before* audio transfers take place. Mixing BTP/BRA and audio is a nightmare, this patchset suggests a mutual-exclusion between two usages. Scope: This patchset only exposes the API and a debugfs interface to initiate commands, validate results and measure performance. The actual use of the API is left as an exercise for codec driver developers. This patchset depends on a number of pre-requisite patches and will not build on top of any for-next branches. The main intent of this RFC is to gather comments on the usage, API, benefits and restrictions. The code and functionality was tested on an Intel LunarLake RVP platform connected to a Realtek RT711-SDCA device. Acknowledgements: Thanks to Zeek Tsai at Realtek for providing test sequences that helped reconcile the data formatted by the host driver with the expected results on the code side. Pierre-Louis Bossart (16): Documentation: driver: add SoundWire BRA description soundwire: cadence: add BTP support for DP0 soundwire: stream: extend sdw_alloc_stream() to take 'type' parameter soundwire: extend sdw_stream_type to BPT soundwire: stream: special-case the bus compute_params() routine soundwire: stream: reuse existing code for BPT stream soundwire: bus: add API for BPT protocol soundwire: bus: add bpt_stream pointer soundwire: crc8: add constant table soundwire: cadence: add BTP/BRA helpers to format data soundwire: intel_auxdevice: add indirection for BPT open/close/send_async/wait ASoC: SOF: Intel: hda-sdw-bpt: add helpers for SoundWire BPT DMA soundwire: intel: add BPT context definition soundwire: intel_ace2x: add BPT open/close/send_async/wait soundwire: debugfs: add interface for BPT/BRA transfers ASoC: rt711-sdca: add DP0 support Documentation/driver-api/soundwire/bra.rst | 478 +++++++++++++ Documentation/driver-api/soundwire/index.rst | 1 + Documentation/driver-api/soundwire/stream.rst | 2 +- .../driver-api/soundwire/summary.rst | 5 +- drivers/soundwire/Kconfig | 1 + drivers/soundwire/Makefile | 4 +- drivers/soundwire/amd_manager.c | 2 +- drivers/soundwire/bus.c | 77 +++ drivers/soundwire/bus.h | 18 + drivers/soundwire/cadence_master.c | 646 +++++++++++++++++- drivers/soundwire/cadence_master.h | 30 + drivers/soundwire/crc8.c | 277 ++++++++ drivers/soundwire/crc8.h | 11 + drivers/soundwire/debugfs.c | 122 +++- .../soundwire/generic_bandwidth_allocation.c | 84 ++- drivers/soundwire/intel.h | 12 + drivers/soundwire/intel_ace2x.c | 377 ++++++++++ drivers/soundwire/intel_auxdevice.c | 55 ++ drivers/soundwire/qcom.c | 2 +- drivers/soundwire/stream.c | 137 +++- include/linux/soundwire/sdw.h | 91 ++- include/linux/soundwire/sdw_intel.h | 16 + include/sound/hda-sdw-bpt.h | 76 +++ sound/soc/codecs/rt711-sdca-sdw.c | 8 + sound/soc/qcom/sdw.c | 2 +- sound/soc/sof/intel/Kconfig | 8 +- sound/soc/sof/intel/Makefile | 4 + sound/soc/sof/intel/hda-sdw-bpt.c | 328 +++++++++ 28 files changed, 2810 insertions(+), 64 deletions(-) create mode 100644 Documentation/driver-api/soundwire/bra.rst create mode 100644 drivers/soundwire/crc8.c create mode 100644 drivers/soundwire/crc8.h create mode 100644 include/sound/hda-sdw-bpt.h create mode 100644 sound/soc/sof/intel/hda-sdw-bpt.c -- 2.39.2

4 60

[PATCH] ASoC: dt-bindings: samsung,odroid: drop stale clocks
by Krzysztof Kozlowski 20 Aug '24

20 Aug '24

Clocks property was present only to allow usage of assigned-clocks in the sound card node, however in upstream DTS the assigned-clocks were moved in commit 4afb06afd768 ("ARM: dts: exynos: move assigned-clock* properties to i2s0 node in Odroid XU4") to respective I2S nodes. Linux drivers never parsed "clocks" so it can be safely dropped. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Documentation/devicetree/bindings/sound/samsung,odroid.yaml | 5 ----- 1 file changed, 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml index b77284e3e26a..c3dea852cc8d 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -27,11 +27,6 @@ properties: - const: samsung,odroid-xu4-audio deprecated: true - assigned-clock-parents: true - assigned-clock-rates: true - assigned-clocks: true - clocks: true - cpu: type: object additionalProperties: false -- 2.43.0

3 2

[PATCH alsa-lib] pcm: implement snd_pcm_hw_params_get_sync() and obsolete snd_pcm_info_get_sync()
by Jaroslav Kysela 20 Aug '24

20 Aug '24

Use the new clock source mechanism to get information about similar PCM clock sources for PCM streams. Link: https://lore.kernel.org/linux-sound/20240625172836.589380-1-perex@perex.cz/ Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- include/pcm.h | 4 ++++ include/sound/uapi/asound.h | 3 ++- src/pcm/pcm.c | 30 ++++++++++++++++++++++++++---- 3 files changed, 32 insertions(+), 5 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 102ff812..b5c67889 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -508,6 +508,9 @@ typedef union _snd_pcm_sync_id { unsigned int id32[4]; } snd_pcm_sync_id_t; +/** synchronization ID size (see snd_pcm_hw_params_get_sync) */ +#define SND_PCM_HW_PARAMS_SYNC_SIZE 16 + /** Infinite wait for snd_pcm_wait() */ #define SND_PCM_WAIT_INFINITE (-1) /** Wait for next i/o in snd_pcm_wait() */ @@ -747,6 +750,7 @@ int snd_pcm_hw_params_get_rate_numden(const snd_pcm_hw_params_t *params, unsigned int *rate_den); int snd_pcm_hw_params_get_sbits(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_get_fifo_size(const snd_pcm_hw_params_t *params); +const unsigned char * snd_pcm_hw_params_get_sync(const snd_pcm_hw_params_t *params); #if 0 typedef struct _snd_pcm_hw_strategy snd_pcm_hw_strategy_t; diff --git a/include/sound/uapi/asound.h b/include/sound/uapi/asound.h index f3b2b94d..012fd890 100644 --- a/include/sound/uapi/asound.h +++ b/include/sound/uapi/asound.h @@ -426,7 +426,8 @@ struct snd_pcm_hw_params { unsigned int rate_num; /* R: rate numerator */ unsigned int rate_den; /* R: rate denominator */ snd_pcm_uframes_t fifo_size; /* R: chip FIFO size in frames */ - unsigned char reserved[64]; /* reserved for future */ + unsigned char sync[16]; /* R: synchronization ID (perfect sync - one clock source) */ + unsigned char reserved[48]; /* reserved for future */ }; enum { diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 62e76e7e..f8581191 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -494,8 +494,8 @@ software parameter. There are two functions allowing link multiple streams together. In the case, the linking means that all operations are synchronized. Because the drivers cannot guarantee the synchronization (sample resolution) on hardware -lacking this feature, the #snd_pcm_info_get_sync() function -returns synchronization ID - #snd_pcm_sync_id_t, which is equal +lacking this feature, the #snd_pcm_hw_params_get_sync() function +returns 16-byte synchronization ID, which is equal for hardware synchronized streams. When the #snd_pcm_link() function is called, all operations managing the stream state for these two streams are joined. The opposite function is #snd_pcm_unlink(). @@ -3948,6 +3948,25 @@ int snd_pcm_hw_params_get_fifo_size(const snd_pcm_hw_params_t *params) return params->fifo_size; } +/** + * \brief Get hardware synchronization ID from a PCM info container + * \param params Configuration space + * \return 16-byte synchronization ID (use #SND_PCM_HW_PARAMS_SYNC_SIZE) + * + * This synchronization ID determines the similar clocks for the + * PCM stream between multiple devices (including different cards). + * "All zeros" means "not set". The contents of the ID can be used + * only for a comparison with the contents of another ID returned + * from this function. Applications should not do a comparison with + * hard-coded values, because the implementation generating such + * synchronization IDs may be changed in future. + */ +const unsigned char *snd_pcm_hw_params_get_sync(const snd_pcm_hw_params_t *params) +{ + assert(params); + return params->sync; +} + /** * \brief Fill params with a full configuration space for a PCM * \param pcm PCM handle @@ -7332,7 +7351,7 @@ unsigned int snd_pcm_info_get_subdevices_avail(const snd_pcm_info_t *obj) } /** - * \brief Get hardware synchronization ID from a PCM info container + * \brief (DEPRECATED) Get hardware synchronization ID from a PCM info container * \param obj PCM info container * \return hardware synchronization ID */ @@ -7340,9 +7359,12 @@ snd_pcm_sync_id_t snd_pcm_info_get_sync(const snd_pcm_info_t *obj) { snd_pcm_sync_id_t res; assert(obj); - memcpy(&res, &obj->sync, sizeof(res)); + bzero(&res, sizeof(res)); return res; } +#ifndef DOC_HIDDEN +link_warning(snd_pcm_info_get_sync, "Warning: snd_pcm_info_get_sync is deprecated, consider to use snd_pcm_hw_params_get_sync"); +#endif /** * \brief Set wanted device inside a PCM info container (see #snd_ctl_pcm_info) -- 2.45.2

1 0

[PATCH v2 00/14] ASoC: Intel: Remove skylake driver
by Cezary Rojewski 20 Aug '24

20 Aug '24

A spiritual successor to haswell/baytrail removal series [1]. The avs-driver found in sound/soc/intel/avs is a direct replacement to the existing skylake-driver. It covers all features supported by it and more and aligns with the recommended flows and requirements based on Windows driver equivalent. The skylake-driver related UAPI has been removed with "ASoC: Drop soc-topology ABI v4 support" [2]. For the official kernel tree the deprecation begun with v6.0. Most skylake-drivers users moved to avs- or SOF-driver when AudioDSP capabilities are available on the platform or to snd-hda-intel (sound/pci/hda) when such capabilities are not. For the supported trees the deprecation begun with v5.4 with v5.15 being the first where the skylake-driver is disabled entirely. All machine board drivers that consume this DSP driver have their replacements present within sound/soc/intel/avs/boards/ directory. Changes in v2: - added two new patches (leading the series) - the first moves all device entries previously under the skylake-driver dependency under the avs-driver - the second patch drops struct skl_machine_pdata usage to fix compilation errors as pointed out by IKP - no further changes, retained Acked-by tags for these [1]: https://lore.kernel.org/alsa-devel/20201006064907.16277-1-cezary.rojewski@i… [2]: https://lore.kernel.org/alsa-devel/20240403091629.647267-1-cezary.rojewski@… Cezary Rojewski (14): ALSA: hda: Move SST device entries to AVS ASoC: Intel: Drop skl_machine_pdata usage ASoC: Intel: Remove bxt_rt298 board driver ASoC: Intel: Remove bxt_da7219_max98357a board driver ASoC: Intel: Remove kbl_rt5663_rt5514_max98927 board driver ASoC: Intel: Remove kbl_rt5663_max98927 board driver ASoC: Intel: Remove kbl_rt5660 board driver ASoC: Intel: Remove kbl_da7219_max98927 board driver ASoC: Intel: Remove kbl_da7219_max98357a board driver ASoC: Intel: Remove skl_rt286 board driver ASoC: Intel: Remove skl_nau88l25_ssm4567 board driver ASoC: Intel: Remove skl_nau88l25_max98357a board driver ASoC: Intel: Remove skylake driver ASoC: Intel: avs: Enable by default for all SST configurations sound/hda/intel-dsp-config.c | 111 +- sound/soc/intel/Kconfig | 120 - sound/soc/intel/Makefile | 1 - sound/soc/intel/avs/core.c | 8 +- sound/soc/intel/boards/Kconfig | 152 +- sound/soc/intel/boards/Makefile | 20 - sound/soc/intel/boards/bxt_da7219_max98357a.c | 720 ---- sound/soc/intel/boards/bxt_rt298.c | 670 --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 688 ---- sound/soc/intel/boards/kbl_da7219_max98927.c | 1175 ------ sound/soc/intel/boards/kbl_rt5660.c | 567 --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 1073 ----- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 869 ---- .../soc/intel/boards/skl_nau88l25_max98357a.c | 704 ---- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 751 ---- sound/soc/intel/boards/skl_rt286.c | 568 --- .../intel/common/soc-acpi-intel-cnl-match.c | 6 - .../intel/common/soc-acpi-intel-ehl-match.c | 1 - .../intel/common/soc-acpi-intel-hda-match.c | 6 - .../intel/common/soc-acpi-intel-icl-match.c | 6 - .../intel/common/soc-acpi-intel-kbl-match.c | 11 - .../intel/common/soc-acpi-intel-skl-match.c | 5 - sound/soc/intel/skylake/Makefile | 15 - sound/soc/intel/skylake/bxt-sst.c | 629 --- sound/soc/intel/skylake/cnl-sst-dsp.c | 266 -- sound/soc/intel/skylake/cnl-sst-dsp.h | 103 - sound/soc/intel/skylake/cnl-sst.c | 508 --- sound/soc/intel/skylake/skl-debug.c | 248 -- sound/soc/intel/skylake/skl-i2s.h | 87 - sound/soc/intel/skylake/skl-messages.c | 1419 ------- sound/soc/intel/skylake/skl-nhlt.c | 269 -- sound/soc/intel/skylake/skl-pcm.c | 1507 ------- sound/soc/intel/skylake/skl-ssp-clk.c | 428 -- sound/soc/intel/skylake/skl-ssp-clk.h | 108 - sound/soc/intel/skylake/skl-sst-cldma.c | 373 -- sound/soc/intel/skylake/skl-sst-cldma.h | 243 -- sound/soc/intel/skylake/skl-sst-dsp.c | 462 --- sound/soc/intel/skylake/skl-sst-dsp.h | 256 -- sound/soc/intel/skylake/skl-sst-ipc.c | 1071 ----- sound/soc/intel/skylake/skl-sst-ipc.h | 169 - sound/soc/intel/skylake/skl-sst-utils.c | 425 -- sound/soc/intel/skylake/skl-sst.c | 599 --- sound/soc/intel/skylake/skl-topology.c | 3605 ----------------- sound/soc/intel/skylake/skl-topology.h | 524 --- sound/soc/intel/skylake/skl.c | 1177 ------ sound/soc/intel/skylake/skl.h | 207 - 46 files changed, 59 insertions(+), 22871 deletions(-) delete mode 100644 sound/soc/intel/boards/bxt_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/bxt_rt298.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5660.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_max98357a.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_ssm4567.c delete mode 100644 sound/soc/intel/boards/skl_rt286.c delete mode 100644 sound/soc/intel/skylake/Makefile delete mode 100644 sound/soc/intel/skylake/bxt-sst.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/cnl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-debug.c delete mode 100644 sound/soc/intel/skylake/skl-i2s.h delete mode 100644 sound/soc/intel/skylake/skl-messages.c delete mode 100644 sound/soc/intel/skylake/skl-nhlt.c delete mode 100644 sound/soc/intel/skylake/skl-pcm.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.c delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.h delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h delete mode 100644 sound/soc/intel/skylake/skl-sst-utils.c delete mode 100644 sound/soc/intel/skylake/skl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-topology.c delete mode 100644 sound/soc/intel/skylake/skl-topology.h delete mode 100644 sound/soc/intel/skylake/skl.c delete mode 100644 sound/soc/intel/skylake/skl.h -- 2.25.1

4 19

[PATCH 00/12] ASoC: Intel: Remove skylake driver
by Cezary Rojewski 20 Aug '24

20 Aug '24

A spiritual successor to haswell/baytrail removal series [1]. The avs-driver found in sound/soc/intel/avs is a direct replacement to the existing skylake-driver. It covers all features supported by it and more and aligns with the recommended flows and requirements based on Windows driver equivalent. The skylake-driver related UAPI has been removed with "ASoC: Drop soc-topology ABI v4 support" [2]. For the official kernel tree the deprecation begun with v6.0. Most skylake-drivers users moved to avs- or SOF-driver when AudioDSP capabilities are available on the platform or to snd-hda-intel (sound/pci/hda) when such capabilities are not. For the supported trees the deprecation begun with v5.4 with v5.15 being the first where the skylake-driver is disabled entirely. All machine board drivers that consume this DSP driver have their replacements present within sound/soc/intel/avs/boards/ directory. [1]: https://lore.kernel.org/alsa-devel/20201006064907.16277-1-cezary.rojewski@i… [2]: https://lore.kernel.org/alsa-devel/20240403091629.647267-1-cezary.rojewski@… Cezary Rojewski (12): ASoC: Intel: Remove bxt_rt298 board driver ASoC: Intel: Remove bxt_da7219_max98357a board driver ASoC: Intel: Remove kbl_rt5663_rt5514_max98927 board driver ASoC: Intel: Remove kbl_rt5663_max98927 board driver ASoC: Intel: Remove kbl_rt5660 board driver ASoC: Intel: Remove kbl_da7219_max98927 board driver ASoC: Intel: Remove kbl_da7219_max98357a board driver ASoC: Intel: Remove skl_rt286 board driver ASoC: Intel: Remove skl_nau88l25_ssm4567 board driver ASoC: Intel: Remove skl_nau88l25_max98357a board driver ASoC: Intel: Remove skylake driver ASoC: Intel: avs: Enable by default for all SST configurations sound/soc/intel/Kconfig | 120 - sound/soc/intel/Makefile | 1 - sound/soc/intel/avs/core.c | 8 +- sound/soc/intel/boards/Kconfig | 152 +- sound/soc/intel/boards/Makefile | 20 - sound/soc/intel/boards/bxt_da7219_max98357a.c | 720 ---- sound/soc/intel/boards/bxt_rt298.c | 670 --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 688 ---- sound/soc/intel/boards/kbl_da7219_max98927.c | 1175 ------ sound/soc/intel/boards/kbl_rt5660.c | 567 --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 1073 ----- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 869 ---- .../soc/intel/boards/skl_nau88l25_max98357a.c | 704 ---- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 751 ---- sound/soc/intel/boards/skl_rt286.c | 568 --- sound/soc/intel/skylake/Makefile | 15 - sound/soc/intel/skylake/bxt-sst.c | 629 --- sound/soc/intel/skylake/cnl-sst-dsp.c | 266 -- sound/soc/intel/skylake/cnl-sst-dsp.h | 103 - sound/soc/intel/skylake/cnl-sst.c | 508 --- sound/soc/intel/skylake/skl-debug.c | 248 -- sound/soc/intel/skylake/skl-i2s.h | 87 - sound/soc/intel/skylake/skl-messages.c | 1419 ------- sound/soc/intel/skylake/skl-nhlt.c | 269 -- sound/soc/intel/skylake/skl-pcm.c | 1507 ------- sound/soc/intel/skylake/skl-ssp-clk.c | 428 -- sound/soc/intel/skylake/skl-ssp-clk.h | 108 - sound/soc/intel/skylake/skl-sst-cldma.c | 373 -- sound/soc/intel/skylake/skl-sst-cldma.h | 243 -- sound/soc/intel/skylake/skl-sst-dsp.c | 462 --- sound/soc/intel/skylake/skl-sst-dsp.h | 256 -- sound/soc/intel/skylake/skl-sst-ipc.c | 1071 ----- sound/soc/intel/skylake/skl-sst-ipc.h | 169 - sound/soc/intel/skylake/skl-sst-utils.c | 425 -- sound/soc/intel/skylake/skl-sst.c | 599 --- sound/soc/intel/skylake/skl-topology.c | 3605 ----------------- sound/soc/intel/skylake/skl-topology.h | 524 --- sound/soc/intel/skylake/skl.c | 1177 ------ sound/soc/intel/skylake/skl.h | 207 - 39 files changed, 9 insertions(+), 22775 deletions(-) delete mode 100644 sound/soc/intel/boards/bxt_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/bxt_rt298.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5660.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_max98357a.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_ssm4567.c delete mode 100644 sound/soc/intel/boards/skl_rt286.c delete mode 100644 sound/soc/intel/skylake/Makefile delete mode 100644 sound/soc/intel/skylake/bxt-sst.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/cnl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-debug.c delete mode 100644 sound/soc/intel/skylake/skl-i2s.h delete mode 100644 sound/soc/intel/skylake/skl-messages.c delete mode 100644 sound/soc/intel/skylake/skl-nhlt.c delete mode 100644 sound/soc/intel/skylake/skl-pcm.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.c delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.h delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h delete mode 100644 sound/soc/intel/skylake/skl-sst-utils.c delete mode 100644 sound/soc/intel/skylake/skl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-topology.c delete mode 100644 sound/soc/intel/skylake/skl-topology.h delete mode 100644 sound/soc/intel/skylake/skl.c delete mode 100644 sound/soc/intel/skylake/skl.h -- 2.25.1

3 14

[REGRESSION] ALSA: firewire-lib: heavy digital distortion with Fireface 800
by edmund.raile 20 Aug '24

20 Aug '24

Between the kernel releases 6.9.9 and 6.10.0, something changed that causes heavy digital distortion on playback from at least ALSA and PipeWire. The rhythm of the music is discernable in the output, so it isn't just random noise. It sounds a bit like bitcrushing, but not quite. The generated overtones appear to be dependet on the sample rate. I am sorry I can not be more specific, there are no kernel messages this time. As Mr. Sakamoto recently committed all these changes to firewire he might be able to identify the issue easily, which is why I'd like to ask him to look into it. Kind regards, Edmund Raile. #regzbot introduced: v6.9.9..v6.10

2 1

[PATCH v1] ASoc: tas2781: fixed the issue that the chip do not shutdown immediatly after aplay stopped
by Shenghao Ding 20 Aug '24

20 Aug '24

Issue reported from customer that the chip do not shutdown after aplay stopped until 6 mins later. Drop tasdevice_dapm_event and implement .stream_mute in the tasdevice_dai_ops. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Drop tasdevice_dapm_event and implement .stream_mute in the tasdevice_dai_ops - Remove unnecessary line feed for tasdevice_codec_remove --- sound/soc/codecs/tas2781-i2c.c | 23 ++++++++--------------- 1 file changed, 8 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9a32e0504857..145c45349399 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -557,30 +557,22 @@ static void tasdevice_fw_ready(const struct firmware *fmw, release_firmware(fmw); } -static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, int stream) { - struct snd_soc_component *codec = snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *codec = dai->component; struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); - int state = 0; - /* Codec Lock Hold */ - mutex_lock(&tas_priv->codec_lock); - if (event == SND_SOC_DAPM_PRE_PMD) - state = 1; - tasdevice_tuning_switch(tas_priv, state); - /* Codec Lock Release*/ - mutex_unlock(&tas_priv->codec_lock); + /* Codec Lock/UnLock */ + guard(mutex)(&tas_priv->codec_lock); + tasdevice_tuning_switch(tas_priv, mute); return 0; } static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_AIF_OUT_E("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, - 0, 0, tasdevice_dapm_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SPK("SPK", NULL), SND_SOC_DAPM_OUTPUT("OUT"), SND_SOC_DAPM_INPUT("DMIC") }; @@ -667,6 +659,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { .startup = tasdevice_startup, .hw_params = tasdevice_hw_params, .set_sysclk = tasdevice_set_dai_sysclk, + .mute_stream = tasdevice_mute, }; static struct snd_soc_dai_driver tasdevice_dai_driver[] = { -- 2.34.1

3 3

[PATCH v7 2/2] ARM: dts: imx6: update spdif sound card node properties
by Elinor Montmasson 20 Aug '24

20 Aug '24

The merge of imx-spdif driver into fsl-asoc-card brought new DT properties that can be used with the "fsl,imx-audio-spdif" compatible: * The "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an S/PDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". While backward compatibility is kept to support properties "spdif-controller", "spdif-in" and "spdif-out", using new properties has several benefits: * "audio-cpu" and "audio-codec" are more generic names reflecting that the fsl-asoc-card driver supports multiple hardware. They are properties already used by devices using the fsl-asoc-card driver. They are also similar to properties of simple-card: "cpu" and "codec". * "spdif-in" and "spdif-out" imply the use of the dummy codec in the driver. However, there are already two codec drivers for the S/PDIF, spdif_transmitter and spdif_receiver. It is better to declare S/PDIF Tx and Rx devices in a DT, and then reference them with "audio-codec" than using the dummy codec. For those reasons, this commit updates in-tree DTs to use the new properties: * Rename "spdif-controller" property to "audio-cpu". * Declare S/PDIF transmitter and/or receiver devices, and use them with the "audio-codec" property instead of "spdif-out" and/or "spdif-in". These modifications were tested only on an imx8mn-evk board. Note that out-of-tree and old DTs are still supported. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi | 9 +++++++-- .../boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi | 9 +++++++-- 12 files changed, 104 insertions(+), 28 deletions(-) diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts b/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts index 95b49fc83f7b..299106fbe51c 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts @@ -127,12 +127,21 @@ simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts b/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts index a7d5693c5ab7..8d2b608e0b90 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts @@ -111,12 +111,21 @@ simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts b/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts index 7c298d9aa21e..5353a0c24420 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts @@ -90,11 +90,16 @@ sound-sgtl5000 { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "On-board SPDIF"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi index ea40623d12e5..edf55760a5c1 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi @@ -197,11 +197,20 @@ sound { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound_spdif: sound-spdif { compatible = "fsl,imx-audio-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; model = "imx-spdif"; status = "disabled"; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi index 3a46ade3b6bd..9e97ef5e43f2 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi @@ -121,11 +121,16 @@ sound { mux-ext-port = <3>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi index d3a7a6eeb8e0..b01670cdd52c 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi @@ -142,12 +142,21 @@ sound { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + /* Optional S/PDIF in on SODIMM 88 and out on SODIMM 90, 137 or 168 */ sound_spdif: sound-spdif { compatible = "fsl,imx-audio-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; model = "imx-spdif"; status = "disabled"; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi index 761566ae3cf5..bd66430c1d78 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi @@ -100,12 +100,17 @@ v_usb1: regulator-v-usb1 { vin-supply = <&v_5v0>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "Integrated SPDIF"; /* IMX6 doesn't implement this yet */ - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; gpio-keys { diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi index a955c77cd499..d1ad65ab6b72 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi @@ -140,12 +140,17 @@ sound_codec: simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "On-board SPDIF"; /* IMX6 doesn't implement this yet */ - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi index 0a3deaf92eea..55fb7b904220 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi @@ -143,12 +143,17 @@ sound-cs42888 { "AIN2R", "Line In Jack"; }; + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-sabreauto-spdif", "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_in>; }; backlight { diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi index 38abb6b50f6c..7130b9c3b3aa 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi @@ -26,11 +26,16 @@ sound { mux-ext-port = <3>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; reg_1p5v: regulator-1p5v { diff --git a/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts b/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts index b0c27b9b0244..dfbfb8119bf3 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts @@ -97,11 +97,16 @@ sound-cs42888 { "AIN2R", "Line In Jack"; }; + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi b/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi index 7d4170c27732..277a6e039045 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi @@ -183,12 +183,17 @@ panel_in: endpoint { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx6sx-sdb-spdif", "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; -- 2.34.1

1 0

[PATCH v7 1/2] arm64: dts: imx8m: update spdif sound card node properties
by Elinor Montmasson 20 Aug '24

20 Aug '24

The merge of imx-spdif driver into fsl-asoc-card brought new DT properties that can be used with the "fsl,imx-audio-spdif" compatible: * The "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an S/PDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". While backward compatibility is kept to support properties "spdif-controller", "spdif-in" and "spdif-out", using new properties has several benefits: * "audio-cpu" and "audio-codec" are more generic names reflecting that the fsl-asoc-card driver supports multiple hardware. They are properties already used by devices using the fsl-asoc-card driver. They are also similar to properties of simple-card: "cpu" and "codec". * "spdif-in" and "spdif-out" imply the use of the dummy codec in the driver. However, there are already two codec drivers for the S/PDIF, spdif_transmitter and spdif_receiver. It is better to declare S/PDIF Tx and Rx devices in a DT, and then reference them with "audio-codec" than using the dummy codec. For those reasons, this commit updates in-tree DTs to use the new properties: * Rename "spdif-controller" property to "audio-cpu". * Declare S/PDIF transmitter and/or receiver devices, and use them with the "audio-codec" property instead of "spdif-out" and/or "spdif-in". These modifications were tested only on an imx8mn-evk board. Note that out-of-tree and old DTs are still supported. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mq-evk.dts | 24 +++++++++++++++---- 3 files changed, 43 insertions(+), 11 deletions(-) diff --git a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi index 930e14fec423..5f8336217bb8 100644 --- a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi @@ -180,12 +180,21 @@ cpu { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi index 9e0259ddf4bc..33d73f3dc187 100644 --- a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi @@ -124,12 +124,21 @@ sound-wm8524 { "Line Out Jack", "LINEVOUTR"; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; sound-micfil { diff --git a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts index 7507548cdb16..a87d0692c3bb 100644 --- a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts +++ b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts @@ -125,19 +125,33 @@ link_codec: simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; + }; + + hdmi_arc_in: hdmi-arc-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; }; sound-hdmi-arc { compatible = "fsl,imx-audio-spdif"; model = "imx-hdmi-arc"; - spdif-controller = <&spdif2>; - spdif-in; + audio-cpu = <&spdif2>; + audio-codec = <&hdmi_arc_in>; }; }; -- 2.34.1

1 0

[PATCH] ASoc: tas2781: replace devm_kzalloc and scnprintf with devm_kasprintf
by Shenghao Ding 20 Aug '24

20 Aug '24

Replace devm_kzalloc and scnprintf with devm_kasprintf. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 23 ++++++++++------------- 1 file changed, 10 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9a32e0504857..1451ba0fa5b4 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -346,13 +346,11 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) } /* Create a mixer item for selecting the active profile */ - name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - GFP_KERNEL); + name = devm_kasprintf(tas_priv->dev, GFP_KERNEL, "Speaker Profile Id"); if (!name) { ret = -ENOMEM; goto out; } - scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "Speaker Profile Id"); prof_ctrls[mix_index].name = name; prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; prof_ctrls[mix_index].info = tasdevice_info_profile; @@ -441,18 +439,13 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) goto out; } - /* Create a mixer item for selecting the active profile */ - prog_name = devm_kzalloc(tas_priv->dev, - SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - GFP_KERNEL); - if (!prog_name || !conf_name) { + /* Create mixer items for selecting the active Program and Config */ + prog_name = devm_kasprintf(tas_priv->dev, GFP_KERNEL, + "Speaker Program Id"); + if (!prog_name) { ret = -ENOMEM; goto out; } - - scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Program Id"); dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -460,8 +453,12 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].put = tasdevice_program_put; mix_index++; - scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + conf_name = devm_kasprintf(tas_priv->dev, GFP_KERNEL, "Speaker Config Id"); + if (!conf_name) { + ret = -ENOMEM; + goto out; + } dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; -- 2.34.1

3 2

[PATCH] soundwire: qcom: Use devm_clk_get_enabled() helpers
by Huan Yang 20 Aug '24

20 Aug '24

The devm_clk_get_enabled() helpers: - call devm_clk_get() - call clk_prepare_enable() and register what is needed in order to call clk_disable_unprepare() when needed, as a managed resource. This simplifies the code and avoids the calls to clk_disable_unprepare(). Signed-off-by: Huan Yang <link(a)vivo.com> --- drivers/soundwire/qcom.c | 22 ++++++++-------------- 1 file changed, 8 insertions(+), 14 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index aed57002fd0e..2a57ed9464b4 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1527,24 +1527,22 @@ static int qcom_swrm_probe(struct platform_device *pdev) if (IS_ERR(ctrl->audio_cgcr)) { dev_err(dev, "Failed to get cgcr reset ctrl required for SW gating\n"); ret = PTR_ERR(ctrl->audio_cgcr); - goto err_init; + return ret; } } ctrl->irq = of_irq_get(dev->of_node, 0); if (ctrl->irq < 0) { ret = ctrl->irq; - goto err_init; + return ret; } - ctrl->hclk = devm_clk_get(dev, "iface"); + ctrl->hclk = devm_clk_get_enabled(dev, "iface"); if (IS_ERR(ctrl->hclk)) { ret = dev_err_probe(dev, PTR_ERR(ctrl->hclk), "unable to get iface clock\n"); - goto err_init; + return ret; } - clk_prepare_enable(ctrl->hclk); - ctrl->dev = dev; dev_set_drvdata(&pdev->dev, ctrl); mutex_init(&ctrl->port_lock); @@ -1558,7 +1556,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) ret = qcom_swrm_get_port_config(ctrl); if (ret) - goto err_clk; + return ret; params = &ctrl->bus.params; params->max_dr_freq = DEFAULT_CLK_FREQ; @@ -1586,7 +1584,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) "soundwire", ctrl); if (ret) { dev_err(dev, "Failed to request soundwire irq\n"); - goto err_clk; + return ret; } ctrl->wake_irq = of_irq_get(dev->of_node, 1); @@ -1597,7 +1595,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) "swr_wake_irq", ctrl); if (ret) { dev_err(dev, "Failed to request soundwire wake irq\n"); - goto err_init; + return ret; } } @@ -1612,7 +1610,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) if (ret) { dev_err(dev, "Failed to register Soundwire controller (%d)\n", ret); - goto err_clk; + return ret; } qcom_swrm_init(ctrl); @@ -1642,9 +1640,6 @@ static int qcom_swrm_probe(struct platform_device *pdev) err_master_add: sdw_bus_master_delete(&ctrl->bus); -err_clk: - clk_disable_unprepare(ctrl->hclk); -err_init: return ret; } @@ -1653,7 +1648,6 @@ static void qcom_swrm_remove(struct platform_device *pdev) struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(&pdev->dev); sdw_bus_master_delete(&ctrl->bus); - clk_disable_unprepare(ctrl->hclk); } static int __maybe_unused swrm_runtime_resume(struct device *dev) -- 2.45.2

1 0

[PATCH v3 0/8] MSM8953/MSM8976 ASoC support
by Adam Skladowski 20 Aug '24

20 Aug '24

Introduce support for basic sound card setup on MSM8953/MSM8976 platforms, document new compatibles and introduce support for more dais. Most of code is sourced from msm8953-mainline fork over github with some changes implemented by me,some basic changes are mentioned in each patch. Changes since v2 =============== 1. Guarded ioremap on quinary mux with checking if resource is not NULL 2. Introduced compatible based reg overriding in wcd-analog driver 3. Separated PM8950/PM8953 codec patches 4. Moved reg/reg-names back into top-level and rewrote as in comments 5. Separated MSM8953/MSM8976 QDSP6 patches 6. Added Lineout patch [1] - https://lore.kernel.org/lkml/20240723083300.35605-1-krzysztof.kozlowski@lin… Changes since v1 ================ 1. Rebased dt-bindings documentation based on Krzysztof's split patch[1] 2. Resolved clang errors by guarding ret verification inside if 3. Switched quin-iomux to devm_ioremap_resource to not fail on msm8916 Signed-off-by: Adam Skladowski <a39.skl(a)gmail.com> --- Adam Skladowski (4): ASoC: dt-bindings: pm8916-wcd-analog-codec: Document pm8950/pm8953 ASoC: msm8916-wcd-analog: add pm8950 codec ASoC: dt-bindings: apq8016-sbc: Add msm8953/msm8976-qdsp6-sndcard ASoC: qcom: apq8016_sbc: Add support for msm8976 SoC Vladimir Lypak (4): ASoC: qcom: apq8016_sbc.c: Add Quinary support ASoC: msm8916-wcd-analog: add pm8953 codec ASoC: qcom: apq8016_sbc: Add support for msm8953 SoC ASoC: msm8916-wcd-analog: add lineout output .../bindings/sound/qcom,apq8016-sbc-sndcard.yaml | 27 ++++ .../sound/qcom,pm8916-wcd-analog-codec.yaml | 6 +- sound/soc/codecs/msm8916-wcd-analog.c | 145 ++++++++++++++++++++- sound/soc/qcom/apq8016_sbc.c | 88 ++++++++++++- 4 files changed, 257 insertions(+), 9 deletions(-) --- base-commit: 31c2a5c4bf3673b0bf098f358ea931b9ada1e7de change-id: 20240729-msm8953-msm8976-asoc-65f973a13165 Best regards, -- Adam Skladowski <a39.skl(a)gmail.com>

6 21

[DISCUSSION] Why doesn't ALSA autogenerate/autoconfigure the mono profile by default?
by xeroxslayer＠proton.me 20 Aug '24

20 Aug '24

Sorry if this is the wrong place to discuss this, but from what I managed to find out, this is primarily the responsibility of ALSA. This was pointed out by a reddit user, and to be frank, I was kinda surprised that this is not done by default. The user in question is hearing impaired (can listen on only one ear), so the user has no benefit of actually listening to stereo or any other multichannel audio, thus wanted to just set everything to mono (dual mono I presume) and was puzzled when there was no such option in pulseaudio (which pulls the profiles from ALSA). I did some more investigation and as it turns out, this profile is autogenerated, but only if the hardware has such capability (as is the case with any other profile). You can enable it manually by editing config files (in which case I presume the channel downconversion is done in software), but it seemed odd that one has to go through such trouble just to listen to mono sound. Thus, I think the mono profile should be taken as a special case scenario and be autogenerated/configured by default, regardless if it's supported in hardware or not. Once again, I'm sorry if this is not the place to ask/discuss this, but would appreciate if someone pointed me to where I can actually do this.

1 0

discussion: why is the mono profile in ALSA not autogenerated by default?
by xeroxslayer＠proton.me 20 Aug '24

20 Aug '24

Sorry if this is the wrong place to discuss this, but from what I managed to find out, this is primarily the responsibility of ALSA. This was pointed out by a reddit user, and to be frank, I was kinda surprised that this is not done by default. The user in question is hearing impaired (can listen on only one ear), so the user has no benefit of actually listening to stereo or any other multichannel audio, thus wanted to just set everything to mono (dual mono I presume) and was puzzled when there was no such option in pulseaudio (which pulls the profiles from ALSA). I did some more investigation and as it turns out, this profile is autogenerated, but only if the hardware has such capability (as is the case with any other profile). You can enable it manually by editing config files (in which case I presume the channel downconversion is done in software), but it seemed odd that one has to go through such trouble just to listen to mono sound. Thus, I think the mono profile should be taken as a special case scenario and be autogenerated/configured by default, regardless if it's supported in hardware or not. Once again, I'm sorry if this is not the place to ask/discuss this, but would appreciate if someone pointed me to where I can actually do this.

1 0

[RFC PATCH v2 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 20 Aug '24

20 Aug '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: reserve space in snd_compr_metadata.key for private usage ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_offload.h | 2 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 176 +++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 68 +++ sound/soc/fsl/fsl_asrc_m2m.c | 791 ++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 259 ++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1297 insertions(+), 8 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

4 13

[PATCH alsa-utils] aseqdump: Add missing dump of UMP Set Key Signature Message
by Takashi Iwai 20 Aug '24

20 Aug '24

The handling of Set Key Signature message of the Flex Data type was missing by some reason. Now the definition was added in alsa-lib ump_msg.h, so let's show the contents. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- seq/aseqdump/aseqdump.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/seq/aseqdump/aseqdump.c b/seq/aseqdump/aseqdump.c index 1d9c01e01dd3..e37c87b45fc1 100644 --- a/seq/aseqdump/aseqdump.c +++ b/seq/aseqdump/aseqdump.c @@ -964,6 +964,14 @@ static void dump_ump_flex_data_event(const unsigned int *ump) return; } + if (fh->meta.status_bank == SND_UMP_FLEX_DATA_MSG_BANK_SETUP && + fh->meta.status == SND_UMP_FLEX_DATA_MSG_STATUS_SET_KEY_SIGNATURE) { + printf("UMP Set Key Signature sharps/flats %d, tonic %d\n", + fh->set_key_sig.sharps_flats, + fh->set_key_sig.tonic_note); + return; + } + if (fh->meta.status_bank == SND_UMP_FLEX_DATA_MSG_BANK_SETUP && fh->meta.status == SND_UMP_FLEX_DATA_MSG_STATUS_SET_CHORD_NAME) { printf("UMP Set Chord Name tonic %d %d %d, alt1 %d/%d, alt2 %d/%d, alt3 %d/%d, alt4 %d/%d, bass %d %d %d, alt1 %d/%d alt2 %d/%d\n", -- 2.43.0

1 0

[PATCH alsa-lib] ump_msg: Add missing definition for Set Key Signature Message
by Takashi Iwai 20 Aug '24

20 Aug '24

The Set Key Signature message was missing at the time we defined for Flex Data types. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/ump_msg.h | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/include/ump_msg.h b/include/ump_msg.h index a64cb317889e..5f0a45acd44c 100644 --- a/include/ump_msg.h +++ b/include/ump_msg.h @@ -606,6 +606,39 @@ typedef struct _snd_ump_msg_set_metronome { #endif } __attribute((packed)) snd_ump_msg_set_metronome_t; +/** Set Key Signature Message (128bit) */ +typedef struct _snd_ump_msg_set_key_sig { +#ifdef SNDRV_BIG_ENDIAN_BITFIELD + uint8_t type:4; /**< UMP packet type */ + uint8_t group:4; /**< UMP Group */ + uint8_t format:2; /**< Format */ + uint8_t addrs:2; /**< Address */ + uint8_t channel:4; /**< Channel */ + uint8_t status_bank; /**< Status Bank */ + uint8_t status; /**< Status */ + + uint8_t sharps_flats:4; /**< Sharps/Flats */ + uint8_t tonic_note:4; /**< Tonic Note 1 */ + uint8_t reserved1[3]; /**< Unused */ + + uint32_t reserved2[2]; /**< Unused */ +#else + uint8_t status; /**< Status */ + uint8_t status_bank; /**< Status Bank */ + uint8_t channel:4; /**< Channel */ + uint8_t addrs:2; /**< Address */ + uint8_t format:2; /**< Format */ + uint8_t group:4; /**< UMP Group */ + uint8_t type:4; /**< UMP packet type */ + + uint8_t reserved1[3]; /**< Unused */ + uint8_t tonic_note:4; /**< Tonic Note */ + uint8_t sharps_flats:4; /**< Sharps/Flats */ + + uint32_t reserved2[2]; /**< Unused */ +#endif +} __attribute((packed)) snd_ump_msg_set_key_sig_t; + /** Set Chord Name Message (128bit) */ typedef struct _snd_ump_msg_set_chord_name { #ifdef SNDRV_BIG_ENDIAN_BITFIELD @@ -678,6 +711,7 @@ typedef union _snd_ump_msg_flex_data { snd_ump_msg_set_tempo_t set_tempo; /**< Set Tempo */ snd_ump_msg_set_time_sig_t set_time_sig; /**< Set Time Signature */ snd_ump_msg_set_metronome_t set_metronome; /**< Set Metronome */ + snd_ump_msg_set_key_sig_t set_key_sig; /**< Set Key Signature */ snd_ump_msg_set_chord_name_t set_chord_name; /**< Set Chord Name */ snd_ump_msg_hdr_t hdr; /**< UMP header */ uint32_t raw[4]; /**< raw UMP packet */ -- 2.43.0

1 0

[PATCH 1/1] ASoC: dt-bindings: convert tlv320aic31xx.txt to yaml
by Frank Li 19 Aug '24

19 Aug '24

Convert binding doc tlv320aic31xx.txt to yaml format. Additional change: - add i2c node in example. - replace MICBIAS_OFF with MICBIAS_2_0v in example because MICBIAS_OFF have been defined in header file. - add ref to dai-common.yaml. - add #sound-dai-cells. Fix below warning: arch/arm64/boot/dts/freescale/imx8mq-zii-ultra-rmb3.dtb: /soc@0/bus@30800000/i2c@30a30000/codec@18: failed to match any schema with compatible: ['ti,tlv320dac3100'] Signed-off-by: Frank Li <Frank.Li(a)nxp.com> --- .../bindings/sound/ti,tlv320dac3100.yaml | 127 ++++++++++++++++++ .../bindings/sound/tlv320aic31xx.txt | 77 ----------- 2 files changed, 127 insertions(+), 77 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml new file mode 100644 index 0000000000000..85e937e34962d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml @@ -0,0 +1,127 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320dac3100.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments - tlv320aic31xx Codec module + +maintainers: + - Shenghao Ding <shenghao-ding(a)ti.com> + +description: | + CODEC output pins: + * HPL + * HPR + * SPL, devices with stereo speaker amp + * SPR, devices with stereo speaker amp + * SPK, devices with mono speaker amp + * MICBIAS + + CODEC input pins: + * MIC1LP, devices with ADC + * MIC1RP, devices with ADC + * MIC1LM, devices with ADC + * AIN1, devices without ADC + * AIN2, devices without ADC + + The pins can be used in referring sound node's audio-routing property. + +properties: + compatible: + enum: + - ti,tlv320aic310x # - Generic TLV320AIC31xx with mono speaker amp + - ti,tlv320aic311x # - Generic TLV320AIC31xx with stereo speaker amp + - ti,tlv320aic3100 # - TLV320AIC3100 (mono speaker amp, no MiniDSP) + - ti,tlv320aic3110 # - TLV320AIC3110 (stereo speaker amp, no MiniDSP) + - ti,tlv320aic3120 # - TLV320AIC3120 (mono speaker amp, MiniDSP) + - ti,tlv320aic3111 # - TLV320AIC3111 (stereo speaker amp, MiniDSP) + - ti,tlv320dac3100 # - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP) + - ti,tlv320dac3101 # - TLV320DAC3101 (no ADC, stereo speaker amp, no MiniDSP) + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + HPVDD-supply: true + + SPRVDD-supply: true + + SPLVDD-supply: true + + AVDD-supply: true + + IOVDD-supply: true + + DVDD-supply: true + + reset-gpios: + description: GPIO specification for the active low RESET input. + + ai31xx-micbias-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 1 + enum: [1, 2, 3] + description: | + MicBias Voltage setting + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD + + ai31xx-ocmv: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2, 3] + description: | + output common-mode voltage setting + 0 - 1.35V, + 1 - 1.5V, + 2 - 1.65V, + 3 - 1.8V + + gpio-reset: + description: gpio pin number used for codec reset + deprecated: true + + +required: + - compatible + - reg + - HPVDD-supply + - SPRVDD-supply + - SPLVDD-supply + - AVDD-supply + - IOVDD-supply + - DVDD-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320aic31xx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + sound@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + ai31xx-micbias-vg = <MICBIAS_2_0V>; + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + HPVDD-supply = <&regulator>; + SPRVDD-supply = <&regulator>; + SPLVDD-supply = <&regulator>; + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt deleted file mode 100644 index bbad98d5b9862..0000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +++ /dev/null @@ -1,77 +0,0 @@ -Texas Instruments - tlv320aic31xx Codec module - -The tlv320aic31xx serial control bus communicates through I2C protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp - "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp - "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP) - "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP) - "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP) - "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP) - "ti,tlv320dac3100" - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP) - "ti,tlv320dac3101" - TLV320DAC3101 (no ADC, stereo speaker amp, no MiniDSP) - -- reg - <int> - I2C slave address -- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, - DVDD-supply : power supplies for the device as covered in - Documentation/devicetree/bindings/regulator/regulator.txt - - -Optional properties: - -- reset-gpios - GPIO specification for the active low RESET input. -- ai31xx-micbias-vg - MicBias Voltage setting - 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V - 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V - 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD - If this node is not mentioned or if the value is unknown, then - micbias is set to 2.0V. -- ai31xx-ocmv - output common-mode voltage setting - 0 - 1.35V, - 1 - 1.5V, - 2 - 1.65V, - 3 - 1.8V - -Deprecated properties: - -- gpio-reset - gpio pin number used for codec reset - -CODEC output pins: - * HPL - * HPR - * SPL, devices with stereo speaker amp - * SPR, devices with stereo speaker amp - * SPK, devices with mono speaker amp - * MICBIAS - -CODEC input pins: - * MIC1LP, devices with ADC - * MIC1RP, devices with ADC - * MIC1LM, devices with ADC - * AIN1, devices without ADC - * AIN2, devices without ADC - -The pins can be used in referring sound node's audio-routing property. - -Example: -#include <dt-bindings/gpio/gpio.h> -#include <dt-bindings/sound/tlv320aic31xx.h> - -tlv320aic31xx: tlv320aic31xx@18 { - compatible = "ti,tlv320aic311x"; - reg = <0x18>; - - ai31xx-micbias-vg = <MICBIAS_OFF>; - - reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; - - HPVDD-supply = <&regulator>; - SPRVDD-supply = <&regulator>; - SPLVDD-supply = <&regulator>; - AVDD-supply = <&regulator>; - IOVDD-supply = <&regulator>; - DVDD-supply = <&regulator>; -}; -- 2.34.1

3 2

Re: [PATCH] ALSA: hda/realtek: Convert existing CS35L56 products to use autodetect fixup function
by Takashi Iwai 19 Aug '24

19 Aug '24

On Mon, 19 Aug 2024 14:37:36 +0200, Simon Trimmer wrote: > > The existing CS35L56 products can make use of the fixup function that > works out the component binding details so we can remove the fixed > configuration fixup functions. > > Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Looks like a nice cleanup. Applied to for-next branch now. Thanks. Takashi

1 0

[PATCH v6 6/7] arm64: dts: imx8m: update spdif sound card node properties
by Elinor Montmasson 19 Aug '24

19 Aug '24

The merge of imx-spdif driver into fsl-asoc-card brought new DT properties that can be used with the "fsl,imx-audio-spdif" compatible: * The "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an S/PDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". While backward compatibility is kept to support properties "spdif-controller", "spdif-in" and "spdif-out", using new properties has several benefits: * "audio-cpu" and "audio-codec" are more generic names reflecting that the fsl-asoc-card driver supports multiple hardware. They are properties already used by devices using the fsl-asoc-card driver. They are also similar to properties of simple-card: "cpu" and "codec". * "spdif-in" and "spdif-out" imply the use of the dummy codec in the driver. However, there are already two codec drivers for the S/PDIF, spdif_transmitter and spdif_receiver. It is better to declare S/PDIF Tx and Rx devices in a DT, and then reference them with "audio-codec" than using the dummy codec. For those reasons, this commit updates in-tree DTs to use the new properties: * Rename "spdif-controller" property to "audio-cpu". * Declare S/PDIF transmitter and/or receiver devices, and use them with the "audio-codec" property instead of "spdif-out" and/or "spdif-in". These modifications were tested only on an imx8mn-evk board. Note that out-of-tree and old DTs are still supported. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mq-evk.dts | 24 +++++++++++++++---- 3 files changed, 43 insertions(+), 11 deletions(-) diff --git a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi index 90d1901df2b1..348855a41852 100644 --- a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi @@ -180,12 +180,21 @@ cpu { }; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi index 9e0259ddf4bc..6a47e09703a7 100644 --- a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi @@ -124,12 +124,21 @@ sound-wm8524 { "Line Out Jack", "LINEVOUTR"; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; sound-micfil { diff --git a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts index 7507548cdb16..b953865f0b46 100644 --- a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts +++ b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts @@ -125,19 +125,33 @@ link_codec: simple-audio-card,codec { }; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; + }; + + hdmi_arc_in: hdmi-arc-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; }; sound-hdmi-arc { compatible = "fsl,imx-audio-spdif"; model = "imx-hdmi-arc"; - spdif-controller = <&spdif2>; - spdif-in; + audio-cpu = <&spdif2>; + audio-codec = <&hdmi_arc_in>; }; }; -- 2.34.1

2 2

[PATCH v11 00/10] spi: Add support for stacked/parallel memories
by Amit Kumar Mahapatra 19 Aug '24

19 Aug '24

This patch series updated the spi-nor, spi core and the AMD-Xilinx GQSPI driver to add stacked and parallel memories support. The first nine patches https://lore.kernel.org/all/20230119185342.2093323-1-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-2-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-3-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-4-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-5-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-6-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-7-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-8-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-9-amit.kumar-mahapatra@a… of the previous series got applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/spi.git for-next But the rest of the patches in the series did not get applied as failure was reported for spi driver with GPIO CS, so send the remaining patches in the series after rebasing it on top of for-next branch and fixing the issue. CS GPIO is not tested on our hardware, but it has been tested by @Stefan https://lore.kernel.org/all/005001da1efc$619ad5a0$24d080e0$@opensource.cirr… --- BRANCH: for-next Changes in v11: - Rebased patch series on top of latest for-next branch. - Added a new patch(1/10) to replace spi->chip_select with spi_get_chipselect() call in tps6594-spi.c. - Added a new patch(2/10) to replace spi->chip_select with spi_get_chipseletc() call in cs35l56_hda_spi.c. - In spi.c initialized unused CS[] to 0xff and spi->cs_index_mask to 0x01 in all flows. - Updated spi_dev_check() to compare the CS of old spi device with the new spi device CS. - Updated cover letter description to add information regarding GPIO CS testing and added Stefen's Tested-by tag in 3/10 patch. Changes in v10: - Rebased patch series on top of latest for-next branch and fixed merge conflicts. Changes in v9: - Updated 1/8 patch description to add an high-level overview of parallel(multi-cs) & stacked design. - Initialized all unused CS to 0xFF. - Moved CS check from spi_add_device() to __spi_add_device(). - Updated __spi_add_device() to check to make sure that multiple logical CS don't map to the same physical CS and same physical CS doesn't map to different logical CS. - Updated 1/8, 5/8 & 7/8 to support arbitrary number of flash devices connected in parallel or stacked mode. - Updated documentation for chip_select. - Added a new spi-nor structure member nor->num_flash to keep track of the number of flashes connected. - Added a new patch in the series 4/8 to move write_enable call just before spi_mem ops call in SPI-NOR. - Added comments in SPI core & SPI-NOR. - Rebased the patch series on top of the latest for-next branch. Changes in v8: - Updated __spi_add_device() and spi_set_cs() to fix spi driver failure with GPIO CS. - Rebased the patch series on top of latest for-next branch and fixed merge conflicts. - Updated cover letter description to add information regarding GPIO CS testing and request Stefan to provide his Tested-by tag for 1/7 patch. - Updated 1/7 patch description. Changes in v7: - Updated spi_dev_check() to avoid failures for spi driver GPIO CS and moved the error message from __spi_add_device() to spi_dev_check(). - Resolved code indentation issue in spi_set_cs(). - In spi_set_cs() call spi_delay_exec( ) once if the controller supports multi cs with both the CS backed by GPIO. - Updated __spi_validate()to add checks for both the GPIO CS. - Replaced cs_index_mask bit mask with SPI_CS_CNT_MAX. - Updated struct spi_controller to represent multi CS capability of the spi controller through a flag bit SPI_CONTROLLER_MULTI_CS instead of a boolen structure member "multi_cs_cap". - Updated 1/7 patch description . Changes in v6: - Rebased on top of latest v6.3-rc1 and fixed merge conflicts in spi-mpc512x-psc.c, sfdp.c, spansion.c files and removed spi-omap-100k.c. - Updated spi_dev_check( ) to reject new devices if any one of the chipselect is used by another device. Changes in v5: - Rebased the patches on top of v6.3-rc1 and fixed the merge conflicts. - Fixed compilation warnings in spi-sh-msiof.c with shmobile_defconfig Changes in v4: - Fixed build error in spi-pl022.c file - reported by Mark. - Fixed build error in spi-sn-f-ospi.c file. - Added Reviewed-by: Serge Semin <fancer.lancer(a)gmail.com> tag. - Added two more patches to replace spi->chip_select with API calls in mpc832x_rdb.c & cs35l41_hda_spi.c files. Changes in v3: - Rebased the patches on top of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/spi.git for-next - Added a patch to convert spi_nor_otp_region_len(nor) & spi_nor_otp_n_regions(nor) macros into inline functions - Added Reviewed-by & Acked-by tags Changes in v2: - Rebased the patches on top of v6.2-rc1 - Created separate patch to add get & set APIs for spi->chip_select & spi->cs_gpiod, and replaced all spi->chip_select and spi->cs_gpiod references with the API calls. - Created separate patch to add get & set APIs for nor->params. --- Amit Kumar Mahapatra (10): mfd: tps6594: Use set/get APIs to access spi->chip_select ALSA: hda/cs35l56: Use set/get APIs to access spi->chip_select spi: Add multi-cs memories support in SPI core mtd: spi-nor: Convert macros with inline functions mtd: spi-nor: Add APIs to set/get nor->params mtd: spi-nor: Move write enable inside specific write & erase APIs mtd: spi-nor: Add stacked memories support in spi-nor spi: spi-zynqmp-gqspi: Add stacked memories support in GQSPI driver mtd: spi-nor: Add parallel memories support in spi-nor spi: spi-zynqmp-gqspi: Add parallel memories support in GQSPI driver drivers/mfd/tps6594-spi.c | 2 +- drivers/mtd/spi-nor/atmel.c | 17 +- drivers/mtd/spi-nor/core.c | 683 +++++++++++++++++++++++++------ drivers/mtd/spi-nor/core.h | 8 + drivers/mtd/spi-nor/debugfs.c | 4 +- drivers/mtd/spi-nor/gigadevice.c | 4 +- drivers/mtd/spi-nor/issi.c | 11 +- drivers/mtd/spi-nor/macronix.c | 10 +- drivers/mtd/spi-nor/micron-st.c | 35 +- drivers/mtd/spi-nor/otp.c | 48 ++- drivers/mtd/spi-nor/sfdp.c | 33 +- drivers/mtd/spi-nor/spansion.c | 57 +-- drivers/mtd/spi-nor/sst.c | 7 +- drivers/mtd/spi-nor/swp.c | 25 +- drivers/mtd/spi-nor/winbond.c | 2 +- drivers/mtd/spi-nor/xilinx.c | 18 +- drivers/spi/spi-zynqmp-gqspi.c | 58 ++- drivers/spi/spi.c | 259 ++++++++++-- include/linux/mtd/spi-nor.h | 21 +- include/linux/spi/spi.h | 51 ++- sound/pci/hda/cs35l56_hda_spi.c | 2 +- 21 files changed, 1085 insertions(+), 270 deletions(-) -- 2.17.1

10 61

[PATCH v3] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 19 Aug '24

19 Aug '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. So far, the Software maintainer has not been confirmed. Once the maintainer was confimred, the guy will update his mail into audio converter section. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - Add Audio converter section - Set the section of LOW/MIDDLE-POWER AUDIO AMPLIFIER as Supported v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 33 ++++++++++++++++++++++++--------- 1 file changed, 24 insertions(+), 9 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index f328373463b0..32954c5a63c3 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22569,26 +22569,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml +F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) DRIVERS +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt -F: Documentation/devicetree/bindings/sound/tpa6130a2.txt -F: include/sound/tas2*.h F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -22598,9 +22614,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

Port API coverage reporting to Python 3 and adapt to current doxygen output
by GitHub pull_request - opened 18 Aug '24

18 Aug '24

alsa-project/alsa-python pull request #15 was opened from tormodvolden: Tested by comparing the new output from alsa-python-coverage.py with the old COVERAGE file. For transparency and easier review there are multiple commits for the Python 3 fixups, but this can be squashed if desired. Request URL : https://github.com/alsa-project/alsa-python/pull/15 Patch URL : https://github.com/alsa-project/alsa-python/pull/15.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

Add persistent ID's for soundcards and midi devices (like coreaudio)
by GitHub issues - opened 18 Aug '24

18 Aug '24

alsa-project/alsa-lib issue #407 was opened from sjomae: "This is something where macOS' Coreaudio/MIDI shines. Unlike macOS Linux/ALSA has no persistent unique IDs for soundcards or MIDI devices. ALSA supports hotplug, and first come first server sequential numeric IDs. The best you^Wpipewire can do is keep track of cards by name. So this is not something pipewire can reliably address, until ALSA get support to identify cards by vendor and serial-number, and provide a UUID." https://lists.linuxaudio.org/hyperkitty/list/linux-audio-dev@lists.linuxaud… Issue URL : https://github.com/alsa-project/alsa-lib/issues/407 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/1] ASoC: dt-bindings: Convert tpa6130a2.txt to yaml
by Frank Li 16 Aug '24

16 Aug '24

Convert binding doc tpa6130a2.txt to yaml format. Additional change: - add ref to dai-common.yaml - add i2c node in example Fix below warning: arch/arm64/boot/dts/freescale/imx8mq-zii-ultra-rmb3.dtb: /soc@0/bus@30800000/i2c@30a20000/amp@60: failed to match any schema with compatible: ['ti,tpa6130a2'] Signed-off-by: Frank Li <Frank.Li(a)nxp.com> --- .../bindings/sound/ti,tpa6130a2.yaml | 55 +++++++++++++++++++ .../devicetree/bindings/sound/tpa6130a2.txt | 27 --------- 2 files changed, 55 insertions(+), 27 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tpa6130a2.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml new file mode 100644 index 0000000000000..a42bf9bde6940 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tpa6130a2.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments - tpa6130a2 Codec module + +maintainers: + - Sebastian Reichel <sre(a)kernel.org> + +description: + Stereo, analog input headphone amplifier + +properties: + compatible: + enum: + - ti,tpa6130a2 + - ti,tpa6140a2 + + reg: + maxItems: 1 + + Vdd-supply: + description: power supply regulator + + power-gpio: + description: gpio pin to power the device + +required: + - compatible + - reg + - Vdd-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@60 { + compatible = "ti,tpa6130a2"; + reg = <0x60>; + Vdd-supply = <&vmmc2>; + power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/tpa6130a2.txt b/Documentation/devicetree/bindings/sound/tpa6130a2.txt deleted file mode 100644 index 6dfa740e4b2d8..0000000000000 --- a/Documentation/devicetree/bindings/sound/tpa6130a2.txt +++ /dev/null @@ -1,27 +0,0 @@ -Texas Instruments - tpa6130a2 Codec module - -The tpa6130a2 serial control bus communicates through I2C protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tpa6130a2" - TPA6130A2 - "ti,tpa6140a2" - TPA6140A2 - - -- reg - <int> - I2C slave address - -- Vdd-supply - <phandle> - power supply regulator - -Optional properties: - -- power-gpio - gpio pin to power the device - -Example: - -tpa6130a2: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - reg = <0x60>; - Vdd-supply = <&vmmc2>; - power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; -}; -- 2.34.1

3 2

ASUS Maximus Z790 Hero (ALC4082) - USB Audio 5.1 speakers assigned incorrectly and subwoofer not working
by GitHub issues - opened 16 Aug '24

16 Aug '24

alsa-project/alsa-ucm-conf issue #437 was opened from Gess79: I hope I'm correct to post this here. Still new to things Linux. **Situation** OS: Kubuntu 24.04 Mainboard = ASUS ROG Maximus Z790 Hero USB Audio = 0b05:1a53 ASUSTek Computer, Inc. USB Audio Speakers = Logitech Z906 Surround Kernel = 6.8.0-40-generic Alsa version = 1.2.9 **Issue** The speakers seem to be assigned incorrectly and the subwoofer isn't working. I'm testing the speakers through pulseaudio as follows: ![image](https://github.com/user-attachments/assets/8bf65600-63ad-46cc-af68-d51ab6d4a7e6) I'm listing the results below in three categories: - The icon clicked within pulseaudio - The soundfile played - The speaker the sound is actually output to _Icon clicked - Soundfile played - Physical speaker output_ Front Left - Front Left - Front Left Front Center - Front Center - Rear Left Front Right - Front Right - Front Right Rear Left - Rear Left - Front Center Subwoofer - Rear Center - Rear Right Rear Right - (nothing) - (nothing) I've had a lengthy discussion on Reddit but I couldn't find a fix for it so far: https://www.reddit.com/r/linux4noobs/comments/1980uv1/comment/kieqsor/ I'm happy to provide all information necessary to pinpoint the issue and fix it. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/437 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Use PyBytes and not Unicode strings for IEC958 values
by GitHub pull_request - edited 16 Aug '24

16 Aug '24

alsa-project/alsa-python pull request #14 was edited from tormodvolden: Suggested fix for #13. - the first commit is not strictly related but also needed to avoid hctltest1.py crashing with my audio devices, where some have info.count = 0. Let me know if more information is needed. I don't know if this is expected or if a warning should be issued. - I went for PyBytes instead of PyByteArray - for setting values, if the provided PyBytes value is shorter than the value, it will be zero-padded. - if it is too long, it will be silently truncated - following the existing code style, no empty lines were inserted, but this could help readability - the third commit is not for fixing the crash, but for fixing set_tuple() to actually do anything for IEC958 - I haven't changed the reference counting, which seems wrong to me in the original code as well. This would be a different issue. Request URL : https://github.com/alsa-project/alsa-python/pull/14 Patch URL : https://github.com/alsa-project/alsa-python/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

Use PyBytes and not Unicode strings for IEC958 values
by GitHub pull_request - edited 16 Aug '24

16 Aug '24

alsa-project/alsa-python pull request #14 was edited from tormodvolden: Suggested fix for #13. - the first commit is not strictly related but also needed to avoid hctltest1.py crashing with my audio devices, where some have info.count = 0. Let me know if more information is needed. I don't know if this is expected or if a warning should be issued. - I went for PyBytes instead of PyByteArray - for setting values, if the provided PyBytes value is shorter than the value, it will be zero-padded. - if it is too long, it will be silently truncated - following the existing code style, no empty lines were inserted, but this could help readability - the third commit is not for fixing the crash, but for fixing set_tuple() to actually do anything for IEC958 Request URL : https://github.com/alsa-project/alsa-python/pull/14 Patch URL : https://github.com/alsa-project/alsa-python/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

[PATCH] ASoC: dt-bindings: qcom,lpass-wsa-macro: correct clocks on SM8250
by srinivas.kandagatla＠linaro.org 16 Aug '24

16 Aug '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> we seems to have ended up with duplicate clocks for frame-sync on sm8250, it has both va and fsgen which are exactly same things. Remove the redundant va clock and make it align with other SoCs. Codec driver does not even handle va clock, so remove this from the bindings and examples to avoid any confusion. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 22 ++----------------- 1 file changed, 2 insertions(+), 20 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml index 06b5f7be3608..6f5644a89feb 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml @@ -64,6 +64,7 @@ allOf: compatible: enum: - qcom,sc7280-lpass-wsa-macro + - qcom,sm8250-lpass-wsa-macro - qcom,sm8450-lpass-wsa-macro - qcom,sc8280xp-lpass-wsa-macro then: @@ -79,24 +80,6 @@ allOf: - const: dcodec - const: fsgen - - if: - properties: - compatible: - enum: - - qcom,sm8250-lpass-wsa-macro - then: - properties: - clocks: - minItems: 6 - clock-names: - items: - - const: mclk - - const: npl - - const: macro - - const: dcodec - - const: va - - const: fsgen - - if: properties: compatible: @@ -130,8 +113,7 @@ examples: <&audiocc 0>, <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, - <&aoncc LPASS_CDC_VA_MCLK>, <&vamacro>; - clock-names = "mclk", "npl", "macro", "dcodec", "va", "fsgen"; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; clock-output-names = "mclk"; }; -- 2.25.1

3 2

[PATCH v1] ASoc: tas2781: Rename dai_driver name to unify the name between TAS2563 and TAS2781
by Shenghao Ding 16 Aug '24

16 Aug '24

Rename dai_driver name to unify the name between TAS2563 and TAS2781, remove unnecessary line feed, strscpy replace scnprintf. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Use strscpy to replace scnprintf. - rename dai_driver name from "tas2781_codec" to "tasdev_codec", in case of misunderstanding whether tas2781 or tas2563. - Remove unnecessary line feed for tasdevice_dsp_create_ctrls --- sound/soc/codecs/tas2781-i2c.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8a57c045afdd..6631254d4985 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -353,7 +353,7 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) ret = -ENOMEM; goto out; } - scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "Speaker Profile Id"); + strscpy(name, "Speaker Profile Id", SNDRV_CTL_ELEM_ID_NAME_MAXLEN); prof_ctrls[mix_index].name = name; prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; prof_ctrls[mix_index].info = tasdevice_info_profile; @@ -424,8 +424,7 @@ static int tasdevice_configuration_put( return ret; } -static int tasdevice_dsp_create_ctrls( - struct tasdevice_priv *tas_priv) +static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; char *prog_name, *conf_name; @@ -453,8 +452,8 @@ static int tasdevice_dsp_create_ctrls( goto out; } - scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Program Id"); + strscpy(prog_name, "Speaker Program Id", + SNDRV_CTL_ELEM_ID_NAME_MAXLEN); dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -462,8 +461,7 @@ static int tasdevice_dsp_create_ctrls( dsp_ctrls[mix_index].put = tasdevice_program_put; mix_index++; - scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Config Id"); + strscpy(conf_name, "Speaker Config Id", SNDRV_CTL_ELEM_ID_NAME_MAXLEN); dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; @@ -673,7 +671,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", -- 2.34.1

4 6

[PATCH] ASoC: SOF: amd: Fix for acp init sequence
by Vijendar Mukunda 16 Aug '24

16 Aug '24

When ACP is not powered on by default, acp power on sequence explicitly invoked by programming pgfsm control mask. The existing implementation checks the same PGFSM status mask and programs the same PGFSM control mask in all ACP variants which breaks acp power on sequence for ACP6.0 and ACP6.3 variants. So to fix this issue, update ACP pgfsm control mask and status mask based on acp descriptor rev field, which will vary based on acp variant. Fixes: 846aef1d7cc0 ("ASoC: SOF: amd: Add Renoir ACP HW support") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/sof/amd/acp.c | 19 +++++++++++++++++-- sound/soc/sof/amd/acp.h | 7 +++++-- 2 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 35e56d22930f..0f6115c8b005 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -433,6 +433,7 @@ static int acp_power_on(struct snd_sof_dev *sdev) const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int base = desc->pgfsm_base; unsigned int val; + unsigned int acp_pgfsm_status_mask, acp_pgfsm_cntl_mask; int ret; val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, base + PGFSM_STATUS_OFFSET); @@ -440,9 +441,23 @@ static int acp_power_on(struct snd_sof_dev *sdev) if (val == ACP_POWERED_ON) return 0; - if (val & ACP_PGFSM_STATUS_MASK) + switch (desc->rev) { + case 3: + case 5: + acp_pgfsm_status_mask = ACP3X_PGFSM_STATUS_MASK; + acp_pgfsm_cntl_mask = ACP3X_PGFSM_CNTL_POWER_ON_MASK; + break; + case 6: + acp_pgfsm_status_mask = ACP6X_PGFSM_STATUS_MASK; + acp_pgfsm_cntl_mask = ACP6X_PGFSM_CNTL_POWER_ON_MASK; + break; + default: + return -EINVAL; + } + + if (val & acp_pgfsm_status_mask) snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + PGFSM_CONTROL_OFFSET, - ACP_PGFSM_CNTL_POWER_ON_MASK); + acp_pgfsm_cntl_mask); ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, base + PGFSM_STATUS_OFFSET, val, !val, ACP_REG_POLL_INTERVAL, ACP_REG_POLL_TIMEOUT_US); diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index f6f0fcfeb691..321c40cc6d50 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -25,8 +25,11 @@ #define ACP_REG_POLL_TIMEOUT_US 2000 #define ACP_DMA_COMPLETE_TIMEOUT_US 5000 -#define ACP_PGFSM_CNTL_POWER_ON_MASK 0x01 -#define ACP_PGFSM_STATUS_MASK 0x03 +#define ACP3X_PGFSM_CNTL_POWER_ON_MASK 0x01 +#define ACP3X_PGFSM_STATUS_MASK 0x03 +#define ACP6X_PGFSM_CNTL_POWER_ON_MASK 0x07 +#define ACP6X_PGFSM_STATUS_MASK 0x0F + #define ACP_POWERED_ON 0x00 #define ACP_ASSERT_RESET 0x01 #define ACP_RELEASE_RESET 0x00 -- 2.34.1

2 1

Use PyBytes and not Unicode strings for IEC958 values
by GitHub pull_request - edited 16 Aug '24

16 Aug '24

alsa-project/alsa-python pull request #14 was edited from tormodvolden: Suggested fix for #13. - the first commit is not strictly related but also needed to avoid hctltest1.py crashing with my audio devices, where some have info.count = 0. Let me know if more information is needed. I don't know if this is expected or if a warning should be issued. - I went for PyBytes instead of PyByteArray - for setting values, if the provided PyBytes value is shorter than the value, it will be zero-padded. - if it is too long, it will be silently truncated - following the existing code style, no empty lines were inserted, but this could help readability Request URL : https://github.com/alsa-project/alsa-python/pull/14 Patch URL : https://github.com/alsa-project/alsa-python/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

Use PyBytes and not Unicode strings for IEC958 values
by GitHub pull_request - opened 16 Aug '24

16 Aug '24

alsa-project/alsa-python pull request #14 was opened from tormodvolden: Suggested fix for #13. - the first commit is not strictly related but also needed to avoid hctltest1.py crashing with my audio devices, where some have info.count = 0. Let me know if more information is needed. - I went for PyBytes instead of PyByteArray - for setting values, if the provided PyBytes value is shorter than the value, it will be zero-padded. - if it is too long, it will be silently truncated - following the existing code style, no empty lines were inserted, but this could help readability Request URL : https://github.com/alsa-project/alsa-python/pull/14 Patch URL : https://github.com/alsa-project/alsa-python/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

[PATCH] ASoC: codecs: lpass-va-macro: set the default codec version for sm8250
by srinivas.kandagatla＠linaro.org 16 Aug '24

16 Aug '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> sm8250 and sc7280 have lpass codec version 1.0, as these are very old platforms, they do not have a reliable way to get the codec version from core_id registers. Add the version info into of_data, so that it does not need to use core_id registers to get version number. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Fixes: 727de4fbc546 ("ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/lpass-va-macro.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 8454193ed22a..e95d1f29ef18 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -228,11 +228,13 @@ struct va_macro { struct va_macro_data { bool has_swr_master; bool has_npl_clk; + int version; }; static const struct va_macro_data sm8250_va_data = { .has_swr_master = false, .has_npl_clk = false, + .version = LPASS_CODEC_VERSION_1_0, }; static const struct va_macro_data sm8450_va_data = { @@ -1587,7 +1589,14 @@ static int va_macro_probe(struct platform_device *pdev) goto err_npl; } - va_macro_set_lpass_codec_version(va); + /** + * old version of codecs do not have a reliable way to determine the + * version from registers, get them from soc specific data + */ + if (data->version) + lpass_macro_set_codec_version(data->version); + else /* read version from register */ + va_macro_set_lpass_codec_version(va); if (va->has_swr_master) { /* Set default CLK div to 1 */ -- 2.25.1

3 4

[PATCH v1] ASoC: codecs: ES8326: Adjust buttons
by zhangyi＠everest-semi.com 16 Aug '24

16 Aug '24

The content of this message was lost. It was probably cross-posted to multiple lists and previously handled on another list.

2 1

[PATCH] ALSA: aoa: Use helper function for_each_child_of_node()
by Zhang Zekun 16 Aug '24

16 Aug '24

for_each_child_of_node can help to iterate through the device_node, and we don't need to use while loop. No functional change with this conversion. Signed-off-by: Zhang Zekun <zhangzekun11(a)huawei.com> --- sound/aoa/soundbus/i2sbus/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 5431d2c49421..ce84288168e4 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -335,7 +335,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) { - struct device_node *np = NULL; + struct device_node *np; int got = 0, err; struct i2sbus_control *control = NULL; @@ -347,7 +347,7 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) return -ENODEV; } - while ((np = of_get_next_child(dev->ofdev.dev.of_node, np))) { + for_each_child_of_node(dev->ofdev.dev.of_node, np) { if (of_device_is_compatible(np, "i2sbus") || of_device_is_compatible(np, "i2s-modem")) { got += i2sbus_add_dev(dev, control, np); -- 2.17.1

1 0

[PATCH v2 0/6] ASoC: codecs: wsa88xx: add support for static port mapping.
by srinivas.kandagatla＠linaro.org 15 Aug '24

15 Aug '24

Existing way of allocating soundwire master ports on Qualcommm platforms is dynamic, and in linear order starting from 1 to MAX_PORTS. This will work as long as soundwire device ports are 1:1 mapped linearly. However on most Qcom SoCs like SM8550, SM8650, x1e80100, these are NOT mapped in that order. The result of this is that only one speaker among the pair of speakers is always silent, With recent changes for WSA codec to support codec versions and along with these patches we are able to get all speakers working on these SoCs. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes in v2: - used dev_dbg instead of dev_info - Link to v1: https://lore.kernel.org/r/20240626-port-map-v1-0-bd8987d2b332@linaro.org --- Manikantan R (1): ASoC: dt-bindings: wsa883x: Document port mapping property Srinivas Kandagatla (5): ASoC: codecs: wsa883x: parse port-mapping information ASoC: dt-bindings: wsa8840: Document port mapping property ASoC: codecs: wsa884x: parse port-mapping information arm64: dts: x1e80100-crd: fix wsa soundwire port mapping arm64: dts: x1e80100-qcp: fix wsa soundwire port mapping Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml | 8 ++++++++ Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml | 8 ++++++++ arch/arm64/boot/dts/qcom/x1e80100-crd.dts | 4 ++++ arch/arm64/boot/dts/qcom/x1e80100-qcp.dts | 2 ++ sound/soc/codecs/wsa883x.c | 8 ++++++++ sound/soc/codecs/wsa884x.c | 8 ++++++++ 6 files changed, 38 insertions(+) --- base-commit: 9935be184a55dd84fc3275094f2df095491f6ea1 change-id: 20240626-port-map-ef50c3304d4a Best regards, -- Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>

5 10

[PATCH 0/6] soc: qcom: use scoped device node handling to simplify error paths
by Krzysztof Kozlowski 15 Aug '24

15 Aug '24

Use cleanup.h/scope on device nodes to get rid of of_node_put(). Code is a bit simpler. Best regards, Krzysztof --- Krzysztof Kozlowski (6): soc: qcom: apr: simplify with scoped for each OF child loop soc: qcom: aoss: simplify with scoped for each OF child loop soc: qcom: ice: use scoped device node handling to simplify error paths soc: qcom: ocmem: use scoped device node handling to simplify error paths soc: qcom: pbs: use scoped device node handling to simplify error paths soc: qcom: smp2p: use scoped device node handling to simplify error paths drivers/soc/qcom/apr.c | 5 +---- drivers/soc/qcom/ice.c | 14 +++++--------- drivers/soc/qcom/ocmem.c | 7 ++----- drivers/soc/qcom/qcom-pbs.c | 16 ++++++---------- drivers/soc/qcom/qcom_aoss.c | 8 +++----- drivers/soc/qcom/smp2p.c | 16 ++++------------ 6 files changed, 21 insertions(+), 45 deletions(-) --- base-commit: d54f33dafd9e93367ffaf95318ed371138a6c8c0 change-id: 20240813-b4-cleanup-h-of-node-put-other-7e9c7128abe4 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

2 7

hctltest1.py crashes with utf-8 decoding error on IEC958 "status"
by GitHub issues - edited 15 Aug '24

15 Aug '24

alsa-project/alsa-python issue #13 was edited from tormodvolden: Version 1.2.12, on Ubuntu 24.04, Python 3.12.3. It seems like the "status" member returned by `snd_ctl_elem_value_get_iec958(self->value, iec958);` on pyalsa/alsahcontrol.c line 1053 starts with 0xff then a 0x00. I haven't been able to find a corresponding node under /proc/asound to check this. The relevant aplay -L entry is ``` iec958:CARD=Stereo,DEV=0 Logitech H570e Stereo, USB Audio IEC958 (S/PDIF) Digital Audio Output ``` Here is a dump of /proc/asound/card1 (card 1 is called "Stereo") <details> ``` $ more $(find /proc/asound/card1 -type f) :::::::::::::: /proc/asound/card1/id :::::::::::::: Stereo :::::::::::::: /proc/asound/card1/pcm0c/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: CAPTURE id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 :::::::::::::: /proc/asound/card1/pcm0c/sub0/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: CAPTURE id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 :::::::::::::: /proc/asound/card1/pcm0c/sub0/status :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0c/sub0/hw_params :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0c/sub0/sw_params :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0p/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 :::::::::::::: /proc/asound/card1/pcm0p/sub0/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 :::::::::::::: /proc/asound/card1/pcm0p/sub0/status :::::::::::::: state: RUNNING owner_pid : 1928 trigger_time: 270044.979966617 tstamp : 309371.219688371 delay : 1872 avail : 31520 avail_max : 32032 ----- hw_ptr : 1887652896 appl_ptr : 1887654144 :::::::::::::: /proc/asound/card1/pcm0p/sub0/hw_params :::::::::::::: access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 256 buffer_size: 32768 :::::::::::::: /proc/asound/card1/pcm0p/sub0/sw_params :::::::::::::: tstamp_mode: ENABLE period_step: 1 avail_min: 256 start_threshold: 9223372036854775807 stop_threshold: 32768 silence_threshold: 0 silence_size: 0 boundary: 4611686018427387904 :::::::::::::: /proc/asound/card1/usbid :::::::::::::: 046d:0a56 :::::::::::::: /proc/asound/card1/usbbus :::::::::::::: 003/006 :::::::::::::: /proc/asound/card1/stream0 :::::::::::::: Logitech Inc Logitech H570e Stereo at usb-0000:00:14.0-4.4.2, full speed : USB A udio Playback: Status: Running Interface = 2 Altset = 1 Packet Size = 192 Momentary freq = 48000 Hz (0x30.0000) Interface 2 Altset 1 Format: S16_LE Channels: 2 Endpoint: 0x01 (1 OUT) (ADAPTIVE) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Bits: 16 Channel map: FL FR Capture: Status: Stop Interface 1 Altset 1 Format: S16_LE Channels: 2 Endpoint: 0x81 (1 IN) (ASYNC) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Bits: 16 Channel map: FL FR :::::::::::::: /proc/asound/card1/usbmixer :::::::::::::: USB Mixer: usb_id=0x046d0a56, ctrlif=0, ctlerr=0 Card: Logitech Inc Logitech H570e Stereo at usb-0000:00:14.0-4.4.2, full speed Unit: 2 Control: name="Auto Gain Control", index=0 Info: id=2, control=7, cmask=0x0, channels=1, type="BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 2 Control: name="Headset Capture Volume", index=0 Info: id=2, control=2, cmask=0x0, channels=1, type="S16" Volume: min=-7680, max=1280, dBmin=-3000, dBmax=500 Unit: 2 Control: name="Headset Capture Switch", index=0 Info: id=2, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 5 Control: name="Headset Playback Volume", index=0 Info: id=5, control=2, cmask=0x3, channels=2, type="S16" Volume: min=-15360, max=1280, dBmin=-6000, dBmax=500 Unit: 5 Control: name="Headset Playback Switch", index=0 Info: id=5, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 11 Control: name="Sidetone Playback Volume", index=0 Info: id=11, control=2, cmask=0x0, channels=1, type="S16" Volume: min=0, max=4125, dBmin=0, dBmax=1611 Unit: 11 Control: name="Sidetone Playback Switch", index=0 Info: id=11, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 ``` </details> This is the last part of the output from running `python3 test/hctltest1.py ` ``` ***** count: 1 device: 0 dimensions: None element: <alsahcontrol.Element object at 0x7e418ebb9650> id: (19, 2, 0, 0, 'IEC958 Playback Con Mask', 0) index: 0 interface: 2 is_inactive: False is_locked: False is_owner: False is_readable: True is_tlv_Commandable: False is_tlv_Writable: False is_tlv_readable: False is_user: False is_volatile: False is_writable: False name: IEC958 Playback Con Mask numid: 19 owner: -1 subdevice: 0 type: 5 (IEC958) ----- device: 0 element: <alsahcontrol.Element object at 0x7e418ebb9650> get_array: <built-in method get_array of alsahcontrol.Value object at 0x7e418ebb9620> get_tuple: <built-in method get_tuple of alsahcontrol.Value object at 0x7e418ebb9620> id: (19, 2, 0, 0, 'IEC958 Playback Con Mask', 0) index: 0 interface: 2 name: IEC958 Playback Con Mask numid: 19 read: <built-in method read of alsahcontrol.Value object at 0x7e418ebb9620> set_array: <built-in method set_array of alsahcontrol.Value object at 0x7e418ebb9620> set_tuple: <built-in method set_tuple of alsahcontrol.Value object at 0x7e418ebb9620> subdevice: 0 write: <built-in method write of alsahcontrol.Value object at 0x7e418ebb9620> UnicodeDecodeError: 'utf-8' codec can't decode byte 0xff in position 1: invalid start byte The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/home/tormod/SOURCES/pyalsa-debian/python-pyalsa/test/hctltest1.py", line 69, in <module> value(element1) File "/home/tormod/SOURCES/pyalsa-debian/python-pyalsa/test/hctltest1.py", line 36, in value values = value.get_tuple(info.type, info.count) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ SystemError: <method 'get_tuple' of 'alsahcontrol.Value' objects> returned a result with an exception set ``` info.type is 5 info.count is 1 Issue URL : https://github.com/alsa-project/alsa-python/issues/13 Repository URL: https://github.com/alsa-project/alsa-python

1 0

hctltest1.py crashes with utf-8 decoding error on IEC958 "status"
by GitHub issues - edited 15 Aug '24

15 Aug '24

alsa-project/alsa-python issue #13 was edited from tormodvolden: Version 1.2.12, on Ubuntu 24.04, Python 3.12. It seems like the "status" member returned by `snd_ctl_elem_value_get_iec958(self->value, iec958);` on pyalsa/alsahcontrol.c line 1053 starts with 0xff then a 0x00. I haven't been able to find a corresponding node under /proc/asound to check this. The relevant aplay -L entry is ``` iec958:CARD=Stereo,DEV=0 Logitech H570e Stereo, USB Audio IEC958 (S/PDIF) Digital Audio Output ``` Here is a dump of /proc/asound/card1 (card 1 is called "Stereo") <details> ``` $ more $(find /proc/asound/card1 -type f) :::::::::::::: /proc/asound/card1/id :::::::::::::: Stereo :::::::::::::: /proc/asound/card1/pcm0c/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: CAPTURE id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 :::::::::::::: /proc/asound/card1/pcm0c/sub0/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: CAPTURE id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 :::::::::::::: /proc/asound/card1/pcm0c/sub0/status :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0c/sub0/hw_params :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0c/sub0/sw_params :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0p/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 :::::::::::::: /proc/asound/card1/pcm0p/sub0/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 :::::::::::::: /proc/asound/card1/pcm0p/sub0/status :::::::::::::: state: RUNNING owner_pid : 1928 trigger_time: 270044.979966617 tstamp : 309371.219688371 delay : 1872 avail : 31520 avail_max : 32032 ----- hw_ptr : 1887652896 appl_ptr : 1887654144 :::::::::::::: /proc/asound/card1/pcm0p/sub0/hw_params :::::::::::::: access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 256 buffer_size: 32768 :::::::::::::: /proc/asound/card1/pcm0p/sub0/sw_params :::::::::::::: tstamp_mode: ENABLE period_step: 1 avail_min: 256 start_threshold: 9223372036854775807 stop_threshold: 32768 silence_threshold: 0 silence_size: 0 boundary: 4611686018427387904 :::::::::::::: /proc/asound/card1/usbid :::::::::::::: 046d:0a56 :::::::::::::: /proc/asound/card1/usbbus :::::::::::::: 003/006 :::::::::::::: /proc/asound/card1/stream0 :::::::::::::: Logitech Inc Logitech H570e Stereo at usb-0000:00:14.0-4.4.2, full speed : USB A udio Playback: Status: Running Interface = 2 Altset = 1 Packet Size = 192 Momentary freq = 48000 Hz (0x30.0000) Interface 2 Altset 1 Format: S16_LE Channels: 2 Endpoint: 0x01 (1 OUT) (ADAPTIVE) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Bits: 16 Channel map: FL FR Capture: Status: Stop Interface 1 Altset 1 Format: S16_LE Channels: 2 Endpoint: 0x81 (1 IN) (ASYNC) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Bits: 16 Channel map: FL FR :::::::::::::: /proc/asound/card1/usbmixer :::::::::::::: USB Mixer: usb_id=0x046d0a56, ctrlif=0, ctlerr=0 Card: Logitech Inc Logitech H570e Stereo at usb-0000:00:14.0-4.4.2, full speed Unit: 2 Control: name="Auto Gain Control", index=0 Info: id=2, control=7, cmask=0x0, channels=1, type="BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 2 Control: name="Headset Capture Volume", index=0 Info: id=2, control=2, cmask=0x0, channels=1, type="S16" Volume: min=-7680, max=1280, dBmin=-3000, dBmax=500 Unit: 2 Control: name="Headset Capture Switch", index=0 Info: id=2, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 5 Control: name="Headset Playback Volume", index=0 Info: id=5, control=2, cmask=0x3, channels=2, type="S16" Volume: min=-15360, max=1280, dBmin=-6000, dBmax=500 Unit: 5 Control: name="Headset Playback Switch", index=0 Info: id=5, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 11 Control: name="Sidetone Playback Volume", index=0 Info: id=11, control=2, cmask=0x0, channels=1, type="S16" Volume: min=0, max=4125, dBmin=0, dBmax=1611 Unit: 11 Control: name="Sidetone Playback Switch", index=0 Info: id=11, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 ``` </details> This is the last part of the output from running `python3 test/hctltest1.py ` ``` ***** count: 1 device: 0 dimensions: None element: <alsahcontrol.Element object at 0x7e418ebb9650> id: (19, 2, 0, 0, 'IEC958 Playback Con Mask', 0) index: 0 interface: 2 is_inactive: False is_locked: False is_owner: False is_readable: True is_tlv_Commandable: False is_tlv_Writable: False is_tlv_readable: False is_user: False is_volatile: False is_writable: False name: IEC958 Playback Con Mask numid: 19 owner: -1 subdevice: 0 type: 5 (IEC958) ----- device: 0 element: <alsahcontrol.Element object at 0x7e418ebb9650> get_array: <built-in method get_array of alsahcontrol.Value object at 0x7e418ebb9620> get_tuple: <built-in method get_tuple of alsahcontrol.Value object at 0x7e418ebb9620> id: (19, 2, 0, 0, 'IEC958 Playback Con Mask', 0) index: 0 interface: 2 name: IEC958 Playback Con Mask numid: 19 read: <built-in method read of alsahcontrol.Value object at 0x7e418ebb9620> set_array: <built-in method set_array of alsahcontrol.Value object at 0x7e418ebb9620> set_tuple: <built-in method set_tuple of alsahcontrol.Value object at 0x7e418ebb9620> subdevice: 0 write: <built-in method write of alsahcontrol.Value object at 0x7e418ebb9620> UnicodeDecodeError: 'utf-8' codec can't decode byte 0xff in position 1: invalid start byte The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/home/tormod/SOURCES/pyalsa-debian/python-pyalsa/test/hctltest1.py", line 69, in <module> value(element1) File "/home/tormod/SOURCES/pyalsa-debian/python-pyalsa/test/hctltest1.py", line 36, in value values = value.get_tuple(info.type, info.count) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ SystemError: <method 'get_tuple' of 'alsahcontrol.Value' objects> returned a result with an exception set ``` info.type is 5 info.count is 1 Issue URL : https://github.com/alsa-project/alsa-python/issues/13 Repository URL: https://github.com/alsa-project/alsa-python

1 0

hctltest1.py crashes with utf-8 decoding error on IEC958 "status"
by GitHub issues - opened 15 Aug '24

15 Aug '24

alsa-project/alsa-python issue #13 was opened from tormodvolden: It seems like the "status" member returned by `snd_ctl_elem_value_get_iec958(self->value, iec958);` on pyalsa/alsahcontrol.c line 1053 starts with 0xff then a 0x00. I haven't been able to find a corresponding node under /proc/asound to check this. The relevant aplay -L entry is ``` iec958:CARD=Stereo,DEV=0 Logitech H570e Stereo, USB Audio IEC958 (S/PDIF) Digital Audio Output ``` Here is a dump of /proc/asound/card1 (card 1 is called "Stereo") <details> ``` $ more $(find /proc/asound/card1 -type f) :::::::::::::: /proc/asound/card1/id :::::::::::::: Stereo :::::::::::::: /proc/asound/card1/pcm0c/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: CAPTURE id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 :::::::::::::: /proc/asound/card1/pcm0c/sub0/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: CAPTURE id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 :::::::::::::: /proc/asound/card1/pcm0c/sub0/status :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0c/sub0/hw_params :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0c/sub0/sw_params :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0p/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 :::::::::::::: /proc/asound/card1/pcm0p/sub0/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 :::::::::::::: /proc/asound/card1/pcm0p/sub0/status :::::::::::::: state: RUNNING owner_pid : 1928 trigger_time: 270044.979966617 tstamp : 309371.219688371 delay : 1872 avail : 31520 avail_max : 32032 ----- hw_ptr : 1887652896 appl_ptr : 1887654144 :::::::::::::: /proc/asound/card1/pcm0p/sub0/hw_params :::::::::::::: access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 256 buffer_size: 32768 :::::::::::::: /proc/asound/card1/pcm0p/sub0/sw_params :::::::::::::: tstamp_mode: ENABLE period_step: 1 avail_min: 256 start_threshold: 9223372036854775807 stop_threshold: 32768 silence_threshold: 0 silence_size: 0 boundary: 4611686018427387904 :::::::::::::: /proc/asound/card1/usbid :::::::::::::: 046d:0a56 :::::::::::::: /proc/asound/card1/usbbus :::::::::::::: 003/006 :::::::::::::: /proc/asound/card1/stream0 :::::::::::::: Logitech Inc Logitech H570e Stereo at usb-0000:00:14.0-4.4.2, full speed : USB A udio Playback: Status: Running Interface = 2 Altset = 1 Packet Size = 192 Momentary freq = 48000 Hz (0x30.0000) Interface 2 Altset 1 Format: S16_LE Channels: 2 Endpoint: 0x01 (1 OUT) (ADAPTIVE) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Bits: 16 Channel map: FL FR Capture: Status: Stop Interface 1 Altset 1 Format: S16_LE Channels: 2 Endpoint: 0x81 (1 IN) (ASYNC) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Bits: 16 Channel map: FL FR :::::::::::::: /proc/asound/card1/usbmixer :::::::::::::: USB Mixer: usb_id=0x046d0a56, ctrlif=0, ctlerr=0 Card: Logitech Inc Logitech H570e Stereo at usb-0000:00:14.0-4.4.2, full speed Unit: 2 Control: name="Auto Gain Control", index=0 Info: id=2, control=7, cmask=0x0, channels=1, type="BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 2 Control: name="Headset Capture Volume", index=0 Info: id=2, control=2, cmask=0x0, channels=1, type="S16" Volume: min=-7680, max=1280, dBmin=-3000, dBmax=500 Unit: 2 Control: name="Headset Capture Switch", index=0 Info: id=2, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 5 Control: name="Headset Playback Volume", index=0 Info: id=5, control=2, cmask=0x3, channels=2, type="S16" Volume: min=-15360, max=1280, dBmin=-6000, dBmax=500 Unit: 5 Control: name="Headset Playback Switch", index=0 Info: id=5, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 11 Control: name="Sidetone Playback Volume", index=0 Info: id=11, control=2, cmask=0x0, channels=1, type="S16" Volume: min=0, max=4125, dBmin=0, dBmax=1611 Unit: 11 Control: name="Sidetone Playback Switch", index=0 Info: id=11, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 ``` </details> This is the last part of the output from running `python3 test/hctltest1.py ` ``` ***** count: 1 device: 0 dimensions: None element: <alsahcontrol.Element object at 0x7e418ebb9650> id: (19, 2, 0, 0, 'IEC958 Playback Con Mask', 0) index: 0 interface: 2 is_inactive: False is_locked: False is_owner: False is_readable: True is_tlv_Commandable: False is_tlv_Writable: False is_tlv_readable: False is_user: False is_volatile: False is_writable: False name: IEC958 Playback Con Mask numid: 19 owner: -1 subdevice: 0 type: 5 (IEC958) ----- device: 0 element: <alsahcontrol.Element object at 0x7e418ebb9650> get_array: <built-in method get_array of alsahcontrol.Value object at 0x7e418ebb9620> get_tuple: <built-in method get_tuple of alsahcontrol.Value object at 0x7e418ebb9620> id: (19, 2, 0, 0, 'IEC958 Playback Con Mask', 0) index: 0 interface: 2 name: IEC958 Playback Con Mask numid: 19 read: <built-in method read of alsahcontrol.Value object at 0x7e418ebb9620> set_array: <built-in method set_array of alsahcontrol.Value object at 0x7e418ebb9620> set_tuple: <built-in method set_tuple of alsahcontrol.Value object at 0x7e418ebb9620> subdevice: 0 write: <built-in method write of alsahcontrol.Value object at 0x7e418ebb9620> UnicodeDecodeError: 'utf-8' codec can't decode byte 0xff in position 1: invalid start byte The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/home/tormod/SOURCES/pyalsa-debian/python-pyalsa/test/hctltest1.py", line 69, in <module> value(element1) File "/home/tormod/SOURCES/pyalsa-debian/python-pyalsa/test/hctltest1.py", line 36, in value values = value.get_tuple(info.type, info.count) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ SystemError: <method 'get_tuple' of 'alsahcontrol.Value' objects> returned a result with an exception set ``` info.type is 5 info.count is 1 Issue URL : https://github.com/alsa-project/alsa-python/issues/13 Repository URL: https://github.com/alsa-project/alsa-python

1 0

[PATCH] ASoc: tas2781: Remove unnecessary line feed and space
by Shenghao Ding 15 Aug '24

15 Aug '24

Remove unnecessary line feed for tasdevice_dsp_create_ctrls, and remove two unnecessary spaces in tas2563_digital_gain_get and tas2563_digital_gain_put. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8c9efd9183ff..9a32e0504857 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -157,7 +157,7 @@ static int tas2563_digital_gain_get( mutex_lock(&tas_dev->codec_lock); /* Read the primary device */ - ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); if (ret) { dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); goto out; @@ -203,7 +203,7 @@ static int tas2563_digital_gain_put( vol = clamp(vol, 0, max); mutex_lock(&tas_dev->codec_lock); /* Read the primary device */ - ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); if (ret) { dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); rc = -1; @@ -423,8 +423,7 @@ static int tasdevice_configuration_put( return ret; } -static int tasdevice_dsp_create_ctrls( - struct tasdevice_priv *tas_priv) +static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; char *prog_name, *conf_name; -- 2.34.1

2 1

[PATCH alsa-lib] test: Add an example program to inquire UMP Endpoint and Block info
by Takashi Iwai 15 Aug '24

15 Aug '24

Provide an example program to inquire UMP Endpoint and Block info in various APIs. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- test/Makefile.am | 3 +- test/umpinfo.c | 147 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 149 insertions(+), 1 deletion(-) create mode 100644 test/umpinfo.c diff --git a/test/Makefile.am b/test/Makefile.am index 635fa39bb7e3..707063cc9fc0 100644 --- a/test/Makefile.am +++ b/test/Makefile.am @@ -1,7 +1,7 @@ SUBDIRS=. lsb check_PROGRAMS=control pcm pcm_min latency seq seq-ump-example \ - playmidi1 timer rawmidi midiloop \ + playmidi1 timer rawmidi midiloop umpinfo \ oldapi queue_timer namehint client_event_filter \ chmap audio_time user-ctl-element-set pcm-multi-thread @@ -17,6 +17,7 @@ playmidi1_LDADD=../src/libasound.la timer_LDADD=../src/libasound.la rawmidi_LDADD=../src/libasound.la midiloop_LDADD=../src/libasound.la +umpinfo_LDADD=../src/libasound.la oldapi_LDADD=../src/libasound.la queue_timer_LDADD=../src/libasound.la namehint_LDADD=../src/libasound.la diff --git a/test/umpinfo.c b/test/umpinfo.c new file mode 100644 index 000000000000..92a209361861 --- /dev/null +++ b/test/umpinfo.c @@ -0,0 +1,147 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// Inquire UMP Endpoint and Block info in various APIs + +#include <stdio.h> +#include <alsa/asoundlib.h> + +enum { UNKNOWN, SEQ, RAWMIDI, CTL }; +static int mode = UNKNOWN; + +static snd_seq_t *seq; +static int client; +static snd_ump_t *ump; +static snd_ctl_t *ctl; + +static int get_ump_endpoint_info(snd_ump_endpoint_info_t *ep) +{ + switch (mode) { + case SEQ: + return snd_seq_get_ump_endpoint_info(seq, client, ep); + case RAWMIDI: + return snd_ump_endpoint_info(ump, ep); + case CTL: + return snd_ctl_ump_endpoint_info(ctl, ep); + default: + return -EINVAL; + } +} + +static int get_ump_block_info(int blk, snd_ump_block_info_t *bp) +{ + switch (mode) { + case SEQ: + return snd_seq_get_ump_block_info(seq, client, blk, bp); + case RAWMIDI: + snd_ump_block_info_set_block_id(bp, blk); + return snd_ump_block_info(ump, bp); + case CTL: + snd_ump_block_info_set_block_id(bp, blk); + return snd_ctl_ump_block_info(ctl, bp); + default: + return -EINVAL; + } +} + +static void help(void) +{ + printf("umpinfo: inquire UMP Endpoint and Block info\n" + " umpinfo -c target inquiry via control API\n" + " umpinfo -m target inquiry via UMP rawmidi API\n" + " umpinfo -s client inquiry via sequencer API\n" + " target = hw:0, etc\n" + " client = sequencer client number\n"); + exit(1); +} + +int main(int argc, char **argv) +{ + const char *target = "hw:0"; + snd_ump_endpoint_info_t *ep; + snd_ump_block_info_t *blk; + const unsigned char *s; + int c, i, num_blks; + + while ((c = getopt(argc, argv, "s:m:c:h")) >= 0) { + switch (c) { + case 's': + mode = SEQ; + client = atoi(target); + break; + case 'm': + mode = RAWMIDI; + target = optarg; + break; + case 'c': + mode = CTL; + target = optarg; + break; + default: + help(); + break; + } + } + + switch (mode) { + case SEQ: + if (snd_seq_open(&seq, "default", SND_SEQ_OPEN_DUPLEX, 0)) { + fprintf(stderr, "failed to open sequencer\n"); + return 1; + } + break; + case RAWMIDI: + if (snd_ump_open(&ump, NULL, target, 0)) { + fprintf(stderr, "failed to open UMP rawmidi\n"); + return 1; + } + break; + case CTL: + if (snd_ctl_open(&ctl, target, 0)) { + fprintf(stderr, "failed to open control\n"); + return 1; + } + break; + default: + help(); + break; + } + + snd_ump_endpoint_info_alloca(&ep); + snd_ump_block_info_alloca(&blk); + + if (get_ump_endpoint_info(ep)) { + fprintf(stderr, "failed to get UMP EP info\n"); + return 1; + } + + printf("Name: %s\n", snd_ump_endpoint_info_get_name(ep)); + printf("Product ID: %s\n", snd_ump_endpoint_info_get_product_id(ep)); + printf("Flags: 0x%x\n", snd_ump_endpoint_info_get_flags(ep)); + printf("Protocol Caps: 0x%x\n", snd_ump_endpoint_info_get_protocol_caps(ep)); + printf("Protocol: 0x%x\n", snd_ump_endpoint_info_get_protocol(ep)); + printf("Num Blocks: %d\n", snd_ump_endpoint_info_get_num_blocks(ep)); + printf("Version: 0x%x\n", snd_ump_endpoint_info_get_version(ep)); + printf("Manufacturer ID: 0x%x\n", snd_ump_endpoint_info_get_manufacturer_id(ep)); + printf("Family ID: 0x%x\n", snd_ump_endpoint_info_get_family_id(ep)); + printf("Model ID: 0x%x\n", snd_ump_endpoint_info_get_model_id(ep)); + s = snd_ump_endpoint_info_get_sw_revision(ep); + printf("SW Revision: %02x:%02x:%02x:%02x\n", s[0], s[1], s[2], s[3]); + num_blks = snd_ump_endpoint_info_get_num_blocks(ep); + for (i = 0; i < num_blks; i++) { + if (get_ump_block_info(i, blk)) { + fprintf(stderr, "failed to get UMP Block info for %d\n", i); + continue; + } + printf("\n"); + printf("Block ID: %d\n", snd_ump_block_info_get_block_id(blk)); + printf("Name: %s\n", snd_ump_block_info_get_name(blk)); + printf("Active: %d\n", snd_ump_block_info_get_active(blk)); + printf("Flags: 0x%x\n", snd_ump_block_info_get_flags(blk)); + printf("Direction: %d\n", snd_ump_block_info_get_direction(blk)); + printf("First Group: %d\n", snd_ump_block_info_get_first_group(blk)); + printf("Num Groups: %d\n", snd_ump_block_info_get_num_groups(blk)); + printf("MIDI-CI Version: %d\n", snd_ump_block_info_get_midi_ci_version(blk)); + printf("Sysex8 Streams: %d\n", snd_ump_block_info_get_sysex8_streams(blk)); + } + + return 0; +} -- 2.43.0

1 0

[PATCH alsa-lib] control: Make ump_{endpoint|block}_info calls optional
by Takashi Iwai 15 Aug '24

15 Aug '24

Add the NULL check for ump_endpoint_info and ump_block_info calls. Those can be NULl depending on the target. Fixes: 81b0cf46d16a ("control: Add UMP Endpoint and Block info query support") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/control/control.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index d77ab24cc69c..e443d5436103 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -1293,7 +1293,10 @@ int snd_ctl_ump_next_device(snd_ctl_t *ctl, int *device) int snd_ctl_ump_endpoint_info(snd_ctl_t *ctl, snd_ump_endpoint_info_t *info) { assert(ctl && info); - return ctl->ops->ump_endpoint_info(ctl, info); + fprintf(stderr, "%s:%d\n", __func__, __LINE__); + if (ctl->ops->ump_endpoint_info) + return ctl->ops->ump_endpoint_info(ctl, info); + return -ENXIO; } /** @@ -1305,7 +1308,9 @@ int snd_ctl_ump_endpoint_info(snd_ctl_t *ctl, snd_ump_endpoint_info_t *info) int snd_ctl_ump_block_info(snd_ctl_t *ctl, snd_ump_block_info_t *info) { assert(ctl && info); - return ctl->ops->ump_block_info(ctl, info); + if (ctl->ops->ump_block_info) + return ctl->ops->ump_block_info(ctl, info); + return -ENXIO; } /** -- 2.43.0

1 0

[PATCH 5.10 031/352] ARM: pxa: spitz: use gpio descriptors for audio
by Greg Kroah-Hartman 15 Aug '24

15 Aug '24

5.10-stable review patch. If anyone has any objections, please let me know. ------------------ From: Arnd Bergmann <arnd(a)arndb.de> [ Upstream commit 726d8c965bae2d7468445d990849e281dca8cbf7 ] The audio driver should not use a hardwired gpio number from the header. Change it to use a lookup table. Acked-by: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Reviewed-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Stable-dep-of: 78ab3d352f29 ("ARM: spitz: fix GPIO assignment for backlight") Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- arch/arm/mach-pxa/spitz.c | 33 ++++++++++- arch/arm/mach-pxa/{include/mach => }/spitz.h | 2 +- arch/arm/mach-pxa/spitz_pm.c | 2 +- sound/soc/pxa/spitz.c | 58 ++++++++------------ 4 files changed, 57 insertions(+), 38 deletions(-) rename arch/arm/mach-pxa/{include/mach => }/spitz.h (99%) diff --git a/arch/arm/mach-pxa/spitz.c b/arch/arm/mach-pxa/spitz.c index 264de0bc97d68..8f880dbc05fdb 100644 --- a/arch/arm/mach-pxa/spitz.c +++ b/arch/arm/mach-pxa/spitz.c @@ -43,7 +43,7 @@ #include <linux/platform_data/mmc-pxamci.h> #include <linux/platform_data/usb-ohci-pxa27x.h> #include <linux/platform_data/video-pxafb.h> -#include <mach/spitz.h> +#include "spitz.h" #include "sharpsl_pm.h" #include <mach/smemc.h> @@ -954,11 +954,42 @@ static void __init spitz_i2c_init(void) static inline void spitz_i2c_init(void) {} #endif +static struct gpiod_lookup_table spitz_audio_gpio_table = { + .dev_id = "spitz-audio", + .table = { + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_L - SPITZ_SCP_GPIO_BASE, + "mute-l", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_R - SPITZ_SCP_GPIO_BASE, + "mute-r", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("sharp-scoop.1", SPITZ_GPIO_MIC_BIAS - SPITZ_SCP2_GPIO_BASE, + "mic", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct gpiod_lookup_table akita_audio_gpio_table = { + .dev_id = "spitz-audio", + .table = { + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_L - SPITZ_SCP_GPIO_BASE, + "mute-l", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_R - SPITZ_SCP_GPIO_BASE, + "mute-r", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("i2c-max7310", AKITA_GPIO_MIC_BIAS - AKITA_IOEXP_GPIO_BASE, + "mic", GPIO_ACTIVE_HIGH), + { }, + }, +}; + /****************************************************************************** * Audio devices ******************************************************************************/ static inline void spitz_audio_init(void) { + if (machine_is_akita()) + gpiod_add_lookup_table(&akita_audio_gpio_table); + else + gpiod_add_lookup_table(&spitz_audio_gpio_table); + platform_device_register_simple("spitz-audio", -1, NULL, 0); } diff --git a/arch/arm/mach-pxa/include/mach/spitz.h b/arch/arm/mach-pxa/spitz.h similarity index 99% rename from arch/arm/mach-pxa/include/mach/spitz.h rename to arch/arm/mach-pxa/spitz.h index 04828d8918aa3..f97e3ebd762d5 100644 --- a/arch/arm/mach-pxa/include/mach/spitz.h +++ b/arch/arm/mach-pxa/spitz.h @@ -11,7 +11,7 @@ #define __ASM_ARCH_SPITZ_H 1 #endif -#include "irqs.h" /* PXA_NR_BUILTIN_GPIO, PXA_GPIO_TO_IRQ */ +#include <mach/irqs.h> /* PXA_NR_BUILTIN_GPIO, PXA_GPIO_TO_IRQ */ #include <linux/fb.h> /* Spitz/Akita GPIOs */ diff --git a/arch/arm/mach-pxa/spitz_pm.c b/arch/arm/mach-pxa/spitz_pm.c index 25a1f8c5a7382..6167f96d7b41e 100644 --- a/arch/arm/mach-pxa/spitz_pm.c +++ b/arch/arm/mach-pxa/spitz_pm.c @@ -20,7 +20,7 @@ #include <asm/mach-types.h> #include <mach/hardware.h> -#include <mach/spitz.h> +#include "spitz.h" #include "pxa27x.h" #include "sharpsl_pm.h" diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 7c1384a869ca4..44303b6eb228b 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -14,13 +14,12 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <asm/mach-types.h> -#include <mach/spitz.h> #include "../codecs/wm8750.h" #include "pxa2xx-i2s.h" @@ -37,7 +36,7 @@ static int spitz_jack_func; static int spitz_spk_func; -static int spitz_mic_gpio; +static struct gpio_desc *gpiod_mic, *gpiod_mute_l, *gpiod_mute_r; static void spitz_ext_control(struct snd_soc_dapm_context *dapm) { @@ -56,8 +55,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 1); - gpio_set_value(SPITZ_GPIO_MUTE_R, 1); + gpiod_set_value(gpiod_mute_l, 1); + gpiod_set_value(gpiod_mute_r, 1); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ @@ -65,8 +64,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ @@ -74,8 +73,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ @@ -83,8 +82,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 1); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 1); break; case SPITZ_HP_OFF: @@ -93,8 +92,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; } @@ -199,7 +198,7 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol, static int spitz_mic_bias(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - gpio_set_value_cansleep(spitz_mic_gpio, SND_SOC_DAPM_EVENT_ON(event)); + gpiod_set_value_cansleep(gpiod_mic, SND_SOC_DAPM_EVENT_ON(event)); return 0; } @@ -287,39 +286,28 @@ static int spitz_probe(struct platform_device *pdev) struct snd_soc_card *card = &snd_soc_spitz; int ret; - if (machine_is_akita()) - spitz_mic_gpio = AKITA_GPIO_MIC_BIAS; - else - spitz_mic_gpio = SPITZ_GPIO_MIC_BIAS; - - ret = gpio_request(spitz_mic_gpio, "MIC GPIO"); - if (ret) - goto err1; - - ret = gpio_direction_output(spitz_mic_gpio, 0); - if (ret) - goto err2; + gpiod_mic = devm_gpiod_get(&pdev->dev, "mic", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mic)) + return PTR_ERR(gpiod_mic); + gpiod_mute_l = devm_gpiod_get(&pdev->dev, "mute-l", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mute_l)) + return PTR_ERR(gpiod_mute_l); + gpiod_mute_r = devm_gpiod_get(&pdev->dev, "mute-r", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mute_r)) + return PTR_ERR(gpiod_mute_r); card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { + if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); - goto err2; - } - - return 0; -err2: - gpio_free(spitz_mic_gpio); -err1: return ret; } static int spitz_remove(struct platform_device *pdev) { - gpio_free(spitz_mic_gpio); return 0; } -- 2.43.0

1 0

[PATCH 5.15 039/484] ARM: pxa: spitz: use gpio descriptors for audio
by Greg Kroah-Hartman 15 Aug '24

15 Aug '24

5.15-stable review patch. If anyone has any objections, please let me know. ------------------ From: Arnd Bergmann <arnd(a)arndb.de> [ Upstream commit 726d8c965bae2d7468445d990849e281dca8cbf7 ] The audio driver should not use a hardwired gpio number from the header. Change it to use a lookup table. Acked-by: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Reviewed-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Stable-dep-of: 78ab3d352f29 ("ARM: spitz: fix GPIO assignment for backlight") Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- arch/arm/mach-pxa/spitz.c | 33 ++++++++++- arch/arm/mach-pxa/{include/mach => }/spitz.h | 2 +- arch/arm/mach-pxa/spitz_pm.c | 2 +- sound/soc/pxa/spitz.c | 58 ++++++++------------ 4 files changed, 57 insertions(+), 38 deletions(-) rename arch/arm/mach-pxa/{include/mach => }/spitz.h (99%) diff --git a/arch/arm/mach-pxa/spitz.c b/arch/arm/mach-pxa/spitz.c index 264de0bc97d68..8f880dbc05fdb 100644 --- a/arch/arm/mach-pxa/spitz.c +++ b/arch/arm/mach-pxa/spitz.c @@ -43,7 +43,7 @@ #include <linux/platform_data/mmc-pxamci.h> #include <linux/platform_data/usb-ohci-pxa27x.h> #include <linux/platform_data/video-pxafb.h> -#include <mach/spitz.h> +#include "spitz.h" #include "sharpsl_pm.h" #include <mach/smemc.h> @@ -954,11 +954,42 @@ static void __init spitz_i2c_init(void) static inline void spitz_i2c_init(void) {} #endif +static struct gpiod_lookup_table spitz_audio_gpio_table = { + .dev_id = "spitz-audio", + .table = { + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_L - SPITZ_SCP_GPIO_BASE, + "mute-l", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_R - SPITZ_SCP_GPIO_BASE, + "mute-r", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("sharp-scoop.1", SPITZ_GPIO_MIC_BIAS - SPITZ_SCP2_GPIO_BASE, + "mic", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct gpiod_lookup_table akita_audio_gpio_table = { + .dev_id = "spitz-audio", + .table = { + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_L - SPITZ_SCP_GPIO_BASE, + "mute-l", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_R - SPITZ_SCP_GPIO_BASE, + "mute-r", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("i2c-max7310", AKITA_GPIO_MIC_BIAS - AKITA_IOEXP_GPIO_BASE, + "mic", GPIO_ACTIVE_HIGH), + { }, + }, +}; + /****************************************************************************** * Audio devices ******************************************************************************/ static inline void spitz_audio_init(void) { + if (machine_is_akita()) + gpiod_add_lookup_table(&akita_audio_gpio_table); + else + gpiod_add_lookup_table(&spitz_audio_gpio_table); + platform_device_register_simple("spitz-audio", -1, NULL, 0); } diff --git a/arch/arm/mach-pxa/include/mach/spitz.h b/arch/arm/mach-pxa/spitz.h similarity index 99% rename from arch/arm/mach-pxa/include/mach/spitz.h rename to arch/arm/mach-pxa/spitz.h index 04828d8918aa3..f97e3ebd762d5 100644 --- a/arch/arm/mach-pxa/include/mach/spitz.h +++ b/arch/arm/mach-pxa/spitz.h @@ -11,7 +11,7 @@ #define __ASM_ARCH_SPITZ_H 1 #endif -#include "irqs.h" /* PXA_NR_BUILTIN_GPIO, PXA_GPIO_TO_IRQ */ +#include <mach/irqs.h> /* PXA_NR_BUILTIN_GPIO, PXA_GPIO_TO_IRQ */ #include <linux/fb.h> /* Spitz/Akita GPIOs */ diff --git a/arch/arm/mach-pxa/spitz_pm.c b/arch/arm/mach-pxa/spitz_pm.c index 25a1f8c5a7382..6167f96d7b41e 100644 --- a/arch/arm/mach-pxa/spitz_pm.c +++ b/arch/arm/mach-pxa/spitz_pm.c @@ -20,7 +20,7 @@ #include <asm/mach-types.h> #include <mach/hardware.h> -#include <mach/spitz.h> +#include "spitz.h" #include "pxa27x.h" #include "sharpsl_pm.h" diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 7c1384a869ca4..44303b6eb228b 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -14,13 +14,12 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <asm/mach-types.h> -#include <mach/spitz.h> #include "../codecs/wm8750.h" #include "pxa2xx-i2s.h" @@ -37,7 +36,7 @@ static int spitz_jack_func; static int spitz_spk_func; -static int spitz_mic_gpio; +static struct gpio_desc *gpiod_mic, *gpiod_mute_l, *gpiod_mute_r; static void spitz_ext_control(struct snd_soc_dapm_context *dapm) { @@ -56,8 +55,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 1); - gpio_set_value(SPITZ_GPIO_MUTE_R, 1); + gpiod_set_value(gpiod_mute_l, 1); + gpiod_set_value(gpiod_mute_r, 1); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ @@ -65,8 +64,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ @@ -74,8 +73,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ @@ -83,8 +82,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 1); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 1); break; case SPITZ_HP_OFF: @@ -93,8 +92,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; } @@ -199,7 +198,7 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol, static int spitz_mic_bias(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - gpio_set_value_cansleep(spitz_mic_gpio, SND_SOC_DAPM_EVENT_ON(event)); + gpiod_set_value_cansleep(gpiod_mic, SND_SOC_DAPM_EVENT_ON(event)); return 0; } @@ -287,39 +286,28 @@ static int spitz_probe(struct platform_device *pdev) struct snd_soc_card *card = &snd_soc_spitz; int ret; - if (machine_is_akita()) - spitz_mic_gpio = AKITA_GPIO_MIC_BIAS; - else - spitz_mic_gpio = SPITZ_GPIO_MIC_BIAS; - - ret = gpio_request(spitz_mic_gpio, "MIC GPIO"); - if (ret) - goto err1; - - ret = gpio_direction_output(spitz_mic_gpio, 0); - if (ret) - goto err2; + gpiod_mic = devm_gpiod_get(&pdev->dev, "mic", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mic)) + return PTR_ERR(gpiod_mic); + gpiod_mute_l = devm_gpiod_get(&pdev->dev, "mute-l", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mute_l)) + return PTR_ERR(gpiod_mute_l); + gpiod_mute_r = devm_gpiod_get(&pdev->dev, "mute-r", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mute_r)) + return PTR_ERR(gpiod_mute_r); card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { + if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); - goto err2; - } - - return 0; -err2: - gpio_free(spitz_mic_gpio); -err1: return ret; } static int spitz_remove(struct platform_device *pdev) { - gpio_free(spitz_mic_gpio); return 0; } -- 2.43.0

1 0

[PATCH v11] ALSA: hda/tas2781: Add tas2781 hda SPI driver
by Baojun Xu 15 Aug '24

15 Aug '24

This patch was used to add TAS2781 devices on SPI support in sound/pci/hda. It use ACPI node descript about parameters of TAS2781 on SPI, it like: Scope (_SB.PC00.SPI0) { Device (GSPK) { Name (_HID, "TXNW2781") // _HID: Hardware ID Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { SpiSerialBusV2 (...) SpiSerialBusV2 (...) } } } } And in platform/x86/serial-multi-instantiate.c, those spi devices will be added into system as a single SPI device, so TAS2781 SPI driver will probe twice for every single SPI device. And driver will also parser mono DSP firmware binary and RCA binary for itself. The code support Realtek as the primary codec. In patch version-10, add multi devices firmware binary support, to compatble with windows driver, they can share same firmware binary. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v11: - Define config number for pre-reset and post-reset scripts. - Remove acpi_subsystem_id as it was not used. - Change variables order in structure, to avoid possible memory holes. - Remove pre-reset and post-reset scripts in reset function. - Change variables name to more common. - Add pre-reset and post-reset scripts in fw_ready. - Add empty line after end of function. - Remove reset function before fw ready. - Add result check for tasdevice_spi_prmg_load(). - Return real result of tasdevice_load_block_kernel(). - Change code for ppcver check, and update debug message. - Remove load_calib_data() as no calibration data in firmware now. - Remove tasdevice_spi_select_tuningprm_cfg() as config will be forced downloaded after program download. - Change comments for tasdevice_spi_tuning_switch(). - Remove unnecessary brace. v10: - Remove project id 103c8b93 in Patch_realtek.c, as it was used for test only, offcial project id is 103c8daa (Gemtree). - Add TAS2781_REG_CLK_CONFIG for check if AMP has been reset. - Add table for pre-reset and post-reset. - Perform pre-reset config before reset AMP. - Perform post-reset config after reset AMP. - Fix wrong offset in calibration data issue. - Remove tuning switch after firmware download. - Force stop play in runtime suspend. - Apply patch for component change in bind & unbind. - Remove program download and calibration data apply, switch tuning only in runtime resume. - Check if AMP has reseted in system resume, and do firmware download and calibration data apply. - Change flag 0xCx to 0x8x for coeff data force download. - Add device index for multi device firmware binary file compatible. - Check device index before registers write during firmware download. - Force config data download after program download. - Remove config download in tuning switch. v9: - Add parentheses for variables in micro define. - Add define for calibration registers address identify. - Change variable name from calibration_data to cali_data. - Add variable for calibration registers address in struct tasdevice_priv. - Change format for alignment to parentheses. - Remove tas2781_reg_defaults as it will be over write in firmware download. - Remove RESET for second device in tas2781_spi_reset(), as it will cause no sound on second device. - Add calibration registers address initial in tasdevice_spi_init(). - Remove unnecessary parentheses. - Remove exception in tas2781_hda_playback_hook(). - Remove hard code for calibration registers address. - Add support for calibration data format V3 for registers address identify, as new firmware tool will define calibration registers address dynamic, so can't hard code it, need read it from BIOS with calibration data. - Remove some debug print. - Change hard code value (3) to ARRAY_SIZE(tas2781_snd_controls)/2. - New patch apply for change position of component_del(). - Change label from out, out1 to config_err and block_err. - Remove unnecessary parentheses in tas2781_spi_fwlib.c. - Remove tasdevice_spi_prmg_calibdata_load(). v8: - Move *-y position in sound/pci/hda/Makefile. - Change + to | for bit operation in macro define. - Change TASDEVICE_Checksum to TASDEVICE_CHECKSUM. v7: - Add new project Gemtree (0x103c8daa) support for tas2781 spi driver. v6: - Remove client in tasdevice_priv struct as it was not used. - Change format for tasdevice_spi_dev_write() in tas2781_hda_spi.c. - Change update bits function to read & write mode, as tas2781 use last bit to mask read & write, it cause regmap_update_bits work not as our expected. - Add element in tas2781_snd_controls, tas2781_prof_ctrl, tas2781_dsp_conf_ctrl, to support second device. Previous version add sound card for first index only. - Change calibration registers address declaire to TASDEVICE_REG() mode. - Add snd_ctl_new1() for second device. - Change format for tas2781_hda_unbind() to fit 79 chars in one line. - Remove client for spi as it was not used. - Remove unused variables. - Remove cur_prog clean in runtime_suspend(). - Remove unused variables. - Add tasdevice_spi_tuning_switch() in runtime_resume. - Remove cur_prog clean in system_resume(). - Add tuning switch if playback started in system_resume(). - Change TASDEVICE_PAGE_REG to TASDEVICE_REG_ID in tas2781_spi_fwlib.c. - Remove calibration binary load functions as data will be loaded from EFI. v5: - Combined all modification into one patch for pass the compile. - Remove old hardware id "TIAS2781" as no production on market. - Change max page size to 256 in tas2781-spi.h. - Change book reg from 127 to GENMASK(7, 1), as one bit shift is needed. - Change register address define as one bit shift needed for offset. - Change block check sum comments to Firmware block from I2C. - Change define of (page + reg) to TASDEVICE_PAGE_REG from TASDEVICE_PGRG. - Change to lower case in defined value. - Add registers default value table in tas2781_hda_spi.c - Change rang_max to 256 * 256. - Add zero_flag_mask in regmap_config. - Change max_register to 256 * 256. - Add registers default table into structure regmap_config. - Change parameter from book number to whole register address. - Remove page 0 switch as SPI driver will do it. - Add bit(0) for read function as tas2781 will check last bit of address, to determine operation mode is read (1) or write (0). - Change switch case to if check in tas2781_hda_playback_hook(), as it may cause compile problem in clang-19 building. - Change variable declaration position. - Change order in if check. - Remove old hardware id ("TIAS2781") support. - Remove debug message in runtime suspend and resume. - Remove regmap.h include in tas2781_spi_fwlib.c. - Remove MODULE_ in fwlib as this file will work with tas2781_hda_spi.c only. v4: - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs v4: - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs - Move include of sound/tas2781-dsp.h to c source file tas2781_hda_spi.c, tas2781_spi_fwlib.c, as it's not needed in header file tas2781-spi.h - Use page and book number define in other micro define - Change 4000000 to (4 * HZ_PER_MHZ) - Add define for calibration variable name and size in UEFI. - Remove structure tasdevice & tasdevice_irqinfo, and move variables in to structure tasdevice_priv - Remove some unused variables in sttructure tasdevice_priv - Remove function declare of tascodec_init(), use tascodec_spi_init() - Remove function declare of tasdevice_remove() - Add array_size.h included in tas2781_hda_spi.c - Add cleanup.h included for change mutex to guard - Add units.h included - Include tas2781-dsp.h - Remove unused variables in tas2781_hda structure - Change 0xff to GENMASK(7, 0) - Change 128 to define of max page size - Change code as all variables was moved to tasdevice_priv structure - Change comments for error in book write, or page write - Remove initial of ret, and return 0 directly - Change comments for wrong typo, add space in front and end - Add check for bulk read, max size should be a page size - Change usleep_rang() to fsleep() - Change mutex_lock to guard(mutex), and remove mutex_unlock() - Change tasdevice_spi_init() to void from int as no value return required - Change second parameter to sizeof - Remove tasdevice_clamp(), use clamp() - Add compatible with old hardware id "TIAS2781" - Remove cs_gpio and spi device in tas2781_read_acpi() as use default CS - Change dev-index check, return -EINVAL if it's <= 0 - Change 0xFF to U8_MAX - Remove GPIO get for chip-select, use default from SPI device - Perform RESET in first AMP only as all AMPs share same RESET pin - return 0 directly in some simple routine to avoid variable initial - Change comments for function which was used by system - Change "ON" "OFF" to function str_on_off(...) - Change format of GUID - Add temp buffer for first efi.get_variable() - Free data if it was allocated - Change format of debug message of calibration date time print - Remove total_sz = 0, as it's not needed - Move save_calibration to after tuning_switch to avoid calibration data overwrite - Remove dev from structure tasdevice_hda, use dev in tasdevice_priv - Remove 0 from { "tas2781-hda", 0}, & {"TXNW2781", 0}, - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs v3: - Move HID up to above /* Non-conforming _HID ... */ in scan.c, for avoid misunderstanding. - Move HID up to above /* Non-conforming _HID ... */ in serial-multi-instantiate.c, for avoid misunderstanding. - Change objs to y for snd-hda-scodec-tas2781-spi- in Makefile. - Change format for some define in tas2781-spi.h. - Add define for TASDEVICE_MAX_BOOK_NUM, TASDEVICE_MAX_PAGE. - Move declare of "enum calib_data {" to header file from source file. - Remove "enum device_catlog_id {" as only one platform was supported now. - Remove "struct calidata", as we will save pure calibration data only. - Remove "struct calidata", "enum device_catlog_id" in tasdevice_priv. - Add calibration_data in tasdevice_priv to save pure calibration data. - Remove declare of tasdevice_save_calibration() and tasdevice_apply_calibration(), as it will be used within same c file. - Add below header files included in tas2781_hda_spi.c: bits.h, mutex.h, property.h, time.h, types.h. - Move two micro define to header file tas2781-spi.h. - Change format of some micro define. - Change format of some structure. - Remove irq in tas2781_hda, as have it already in tasdevice_priv. - Remove some local functions declare as not necessarily. - Return error if regmap_write() failed. - Change fixed value 2 to sizeof(data). - Remove all of EXPORT_SYMBOL_GPL, as all of function was called within same module. - Remove empty line after last return in some functions. - Remove some variable initialyzation. - Remove variable sub, store acpi_subsystem_id directly. - Remove wrong comments for device index, it's must, can't be NULL. - Remove some local variables, use elements in structure directly. - Append commas in last element of an array. - Change calibration data process, didn't save all of calibration data in EFI, get in local, and save pure calibration data in tasdevice_priv. - Call tasdevice_save_calibration() from function pointer in tasdevice_priv; - Remove subid as one platform was supported only, needn't check. - Add initialyzation for local variable. - Move regmap initialyzation to before acpi read. - Call tasdevice_apply_calibration() from function pointer in tasdevice_priv; - Remove MODULE_IMPORT_NS(SND_SOC_TAS2781_FMWLIB), as all functions was performed within same module. - Update format and variables declare order. - Change format of multi conditions for if. - Remove casting which is not needed. - Change variables type to avoid casting. - Remove Unneeded parentheses. - Change if check to avoid goto. - Add {} for keep same style. - Remove some local variables, use elements in structure directly. v2: - Change depends on SPI to SPI_MASTER in sound/pci/hda/Kconfig - Add select SND_HDA_SCODEC_COMPONENT in sound/pci/hda/Kconfig - Change comp_generic_fixup() 5th parameter from "-%s:00" to "-%s:00-tas2781-hda.%d" in sound/pci/hda/patch_realtek.c - Change chain_id from ALC285_FIXUP_THINKPAD_HEADSET_JACK to ALC285_FIXUP_HP_GPIO_LED in sound/pci/hda/patch_realtek.c - Remove project name "Gemtree" - Update information in file header of tas2781-spi.h. - Remove useless micro define. - Change TASDEVICE_I2CChecksum to TASDEVICE_Checksum - Remove enum control_bus as current code support SPI only. - Remove device define as current code support TAS2781 only. - Remove spi_device **spi_devs in structure tasdevice_priv. - Remove cal_binaryname in structure tasdevice_priv. - Remove ndev in structure tasdevice_priv. - Change isacpi and isspi, replace by control_bus(I2C or SPI). - Remove void tasdevice_spi_dsp_remove(void *context). - Add acpi_device and irq in structure tas2781_hda in tas2781_hda_spi.c. - Remove parameter chn in all of registers access functions. - Add tascodec_spi_init(). - Add chip select gpio set for SPI device. - Change tasdevice_tuning_switch() to tasdevice_spi_tuning_switch(). - Change device offset from tas_priv->ndev to tas_priv->index. - Change tasdevice_dsp_remove() to tasdevice_spi_dsp_remove(). - tasdevice_prmg_load() to tasdevice_spi_prmg_load(). - Change tasdevice_config_info_remove() to tasdevice_spi_config_info_remove(). - Add one dummy byte shift when read from page 2~127, or not book 0. - Change tasdevice_spi_change_chn_book() to tasdevice_spi_switch_book() - Perform put_device(physdev) before error return. - Change calibration data id to 2781 from 2783 in EFI - Change calibration data structure name in comments. - Remove file name in tas2781_spi_fwlib.c. - Remove #include <linux/i2c.h> as it not needed in SPI mode. - Change TAB to speace in micro define in tas2781_spi_fwlib.c. - Change to up-case for DSP. - Change all of multi-line comments format, empty first line. - Change all of sizeof(struct xx) to sizeof(*xx). - Remove Explicit casting for variables compare to avoid possible fault. - Return directly without goto. - Change 1 << xx to BIT(xx). - Remove deviceNumber[] as current SPI device will support one device only. - Add memory free before return error. - Change "buf[offset]; offset += 1" to buf[offset++]. - Remove some debug information print. - Change firmware binary to single device mode. - Change all bexx_to_cpup() to get_unaligned_bexx(). - Remove ndev process as current SPI device support single device only. - Remove chn and chnend for single device support only. - Change all of registers read/write function, remove parameter chn. - Create new function for single write, burst write, delay, field write in tasdevice_process_block(). --- drivers/acpi/scan.c | 1 + .../platform/x86/serial-multi-instantiate.c | 12 + sound/pci/hda/Kconfig | 14 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 13 + sound/pci/hda/tas2781-spi.h | 158 ++ sound/pci/hda/tas2781_hda_spi.c | 1273 +++++++++++ sound/pci/hda/tas2781_spi_fwlib.c | 2007 +++++++++++++++++ 8 files changed, 3480 insertions(+) create mode 100644 sound/pci/hda/tas2781-spi.h create mode 100644 sound/pci/hda/tas2781_hda_spi.c create mode 100644 sound/pci/hda/tas2781_spi_fwlib.c diff --git a/drivers/acpi/scan.c b/drivers/acpi/scan.c index 59771412686b..b1a8bf1043da 100644 --- a/drivers/acpi/scan.c +++ b/drivers/acpi/scan.c @@ -1769,6 +1769,7 @@ static bool acpi_device_enumeration_by_parent(struct acpi_device *device) {"CSC3557", }, {"INT33FE", }, {"INT3515", }, + {"TXNW2781", }, /* Non-conforming _HID for Cirrus Logic already released */ {"CLSA0100", }, {"CLSA0101", }, diff --git a/drivers/platform/x86/serial-multi-instantiate.c b/drivers/platform/x86/serial-multi-instantiate.c index 3be016cfe601..b15c819023d0 100644 --- a/drivers/platform/x86/serial-multi-instantiate.c +++ b/drivers/platform/x86/serial-multi-instantiate.c @@ -368,6 +368,17 @@ static const struct smi_node cs35l57_hda = { .bus_type = SMI_AUTO_DETECT, }; +static const struct smi_node tas2781_hda = { + .instances = { + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + {} + }, + .bus_type = SMI_AUTO_DETECT, +}; + /* * Note new device-ids must also be added to ignore_serial_bus_ids in * drivers/acpi/scan.c: acpi_device_enumeration_by_parent(). @@ -380,6 +391,7 @@ static const struct acpi_device_id smi_acpi_ids[] = { { "CSC3556", (unsigned long)&cs35l56_hda }, { "CSC3557", (unsigned long)&cs35l57_hda }, { "INT3515", (unsigned long)&int3515_data }, + { "TXNW2781", (unsigned long)&tas2781_hda }, /* Non-conforming _HID for Cirrus Logic already released */ { "CLSA0100", (unsigned long)&cs35l41_hda }, { "CLSA0101", (unsigned long)&cs35l41_hda }, diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bb15a0248250..62cd9ba408e2 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -206,6 +206,20 @@ config SND_HDA_SCODEC_TAS2781_I2C comment "Set to Y if you want auto-loading the side codec driver" depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_I2C=m +config SND_HDA_SCODEC_TAS2781_SPI + tristate "Build TAS2781 HD-audio side codec support for SPI Bus" + depends on SPI_MASTER + depends on ACPI + depends on EFI + depends on SND_SOC + select CRC32_SARWATE + help + Say Y or M here to include TAS2781 SPI HD-audio side codec support + in snd-hda-intel driver, such as ALC287. + +comment "Set to Y if you want auto-loading the side codec driver" + depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_SPI=m + config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 80210f845df2..210c406dfbc5 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -40,6 +40,7 @@ snd-hda-scodec-cs35l56-spi-y := cs35l56_hda_spi.o snd-hda-cs-dsp-ctls-y := hda_cs_dsp_ctl.o snd-hda-scodec-component-y := hda_component.o snd-hda-scodec-tas2781-i2c-y := tas2781_hda_i2c.o +snd-hda-scodec-tas2781-spi-y := tas2781_hda_spi.o tas2781_spi_fwlib.o # common driver obj-$(CONFIG_SND_HDA) := snd-hda-codec.o @@ -72,6 +73,7 @@ obj-$(CONFIG_SND_HDA_SCODEC_CS35L56_SPI) += snd-hda-scodec-cs35l56-spi.o obj-$(CONFIG_SND_HDA_CS_DSP_CONTROLS) += snd-hda-cs-dsp-ctls.o obj-$(CONFIG_SND_HDA_SCODEC_COMPONENT) += snd-hda-scodec-component.o obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_I2C) += snd-hda-scodec-tas2781-i2c.o +obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_SPI) += snd-hda-scodec-tas2781-spi.o # this must be the last entry after codec drivers; # otherwise the codec patches won't be hooked before the PCI probe diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ba0ce8750ca4..996e930b4cef 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6926,6 +6926,11 @@ static void tas2781_fixup_i2c(struct hda_codec *cdc, comp_generic_fixup(cdc, action, "i2c", "TIAS2781", "-%s:00", 1); } +static void tas2781_fixup_spi(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "spi", "TXNW2781", "-%s:00-tas2781-hda.%d", 2); +} + static void yoga7_14arb7_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { @@ -7504,6 +7509,7 @@ enum { ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, ALC287_FIXUP_TAS2781_I2C, + ALC245_FIXUP_TAS2781_SPI_2, ALC287_FIXUP_YOGA7_14ARB7_I2C, ALC245_FIXUP_HP_MUTE_LED_COEFBIT, ALC245_FIXUP_HP_X360_MUTE_LEDS, @@ -9733,6 +9739,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, }, + [ALC245_FIXUP_TAS2781_SPI_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_spi, + .chained = true, + .chain_id = ALC285_FIXUP_HP_GPIO_LED, + }, [ALC287_FIXUP_YOGA7_14ARB7_I2C] = { .type = HDA_FIXUP_FUNC, .v.func = yoga7_14arb7_fixup_i2c, @@ -10270,6 +10282,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8cdf, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ce0, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8daa, "HP Gemtree", ALC245_FIXUP_TAS2781_SPI_2), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), diff --git a/sound/pci/hda/tas2781-spi.h b/sound/pci/hda/tas2781-spi.h new file mode 100644 index 000000000000..ecfc3c8bb821 --- /dev/null +++ b/sound/pci/hda/tas2781-spi.h @@ -0,0 +1,158 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for TAS2781 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2781_SPI_H__ +#define __TAS2781_SPI_H__ + +#define TASDEVICE_RATES \ + (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_88200) + +#define TASDEVICE_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define TASDEVICE_MAX_BOOK_NUM 256 +#define TASDEVICE_MAX_PAGE 256 + +#define TASDEVICE_MAX_SIZE (TASDEVICE_MAX_BOOK_NUM * TASDEVICE_MAX_PAGE) + +/* PAGE Control Register (available in page0 of each book) */ +#define TASDEVICE_PAGE_SELECT 0x00 +#define TASDEVICE_BOOKCTL_PAGE 0x00 +#define TASDEVICE_BOOKCTL_REG GENMASK(7, 1) +#define TASDEVICE_BOOK_ID(reg) (((reg) & GENMASK(24, 16)) >> 16) +#define TASDEVICE_PAGE_ID(reg) (((reg) & GENMASK(15, 8)) >> 8) +#define TASDEVICE_REG_ID(reg) (((reg) & GENMASK(7, 1)) >> 1) +#define TASDEVICE_PAGE_REG(reg) ((reg) & GENMASK(15, 1)) +#define TASDEVICE_REG(book, page, reg) \ + (((book) << 16) | ((page) << 8) | ((reg) << 1)) + +/* Software Reset */ +#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) +#define TAS2781_REG_SWRESET_RESET BIT(0) + +/* System Reset Check Register */ +#define TAS2781_REG_CLK_CONFIG TASDEVICE_REG(0x0, 0x0, 0x5c) +#define TAS2781_REG_CLK_CONFIG_RESET (0x19) +#define TAS2781_PRE_POST_RESET_CFG 3 + +/* Block Checksum */ +#define TASDEVICE_CHECKSUM TASDEVICE_REG(0x0, 0x0, 0x7e) + +/* Volume control */ +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1a) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) + +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + +#define TAS2781_SPI_MAX_FREQ (4 * HZ_PER_MHZ) + +#define TASDEVICE_CRC8_POLYNOMIAL 0x4d +#define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 + +/* Flag of calibration registers address. */ +#define TASDEVICE_CALIBRATION_REG_ADDRESS BIT(7) + +#define TASDEVICE_CALIBRATION_DATA_NAME L"CALI_DATA" +#define TASDEVICE_CALIBRATION_DATA_SIZE 256 + +enum calib_data { + R0_VAL = 0, + INV_R0, + R0LOW, + POWER, + TLIM, + CALIB_MAX +}; + +struct tasdevice_priv { + struct tasdevice_fw *cali_data_fmw; + struct tasdevice_rca rcabin; + struct tasdevice_fw *fmw; + struct gpio_desc *reset; + struct mutex codec_lock; + struct regmap *regmap; + struct device *dev; + struct tm tm; + + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + unsigned char coef_binaryname[64]; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + + bool force_fwload_status; + bool playback_started; + bool is_loading; + bool is_loaderr; + unsigned int cali_reg_array[CALIB_MAX]; + unsigned int cali_data[CALIB_MAX]; + unsigned int err_code; + void *codec; + int cur_book; + int cur_prog; + int cur_conf; + int fw_state; + int index; + int irq; + + int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, + int offset); + int (*fw_parse_program_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*fw_parse_configuration_data)(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); + + int (*save_calibration)(struct tasdevice_priv *tas_priv); + void (*apply_calibration)(struct tasdevice_priv *tas_priv); +}; + +int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned int *value); +int tasdevice_spi_dev_write(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned int value); +int tasdevice_spi_dev_bulk_write(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned char *p_data, + unsigned int n_length); +int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned char *p_data, + unsigned int n_length); +int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tasdevice, + unsigned int reg, unsigned int mask, + unsigned int value); + +void tasdevice_spi_select_cfg_blk(void *context, int conf_no, + unsigned char block_type); +void tasdevice_spi_config_info_remove(void *context); +int tasdevice_spi_dsp_parser(void *context); +int tasdevice_spi_rca_parser(void *context, const struct firmware *fmw); +void tasdevice_spi_dsp_remove(void *context); +void tasdevice_spi_calbin_remove(void *context); +int tasdevice_spi_select_tuningprm_cfg(void *context, int prm, int cfg_no, + int rca_conf_no); +int tasdevice_spi_prmg_load(void *context, int prm_no); +int tasdevice_spi_prmg_calibdata_load(void *context, int prm_no); +void tasdevice_spi_tuning_switch(void *context, int state); +int tas2781_spi_load_calibration(void *context, char *file_name, + unsigned short i); +#endif /* __TAS2781_SPI_H__ */ diff --git a/sound/pci/hda/tas2781_hda_spi.c b/sound/pci/hda/tas2781_hda_spi.c new file mode 100644 index 000000000000..a426b5bee451 --- /dev/null +++ b/sound/pci/hda/tas2781_hda_spi.c @@ -0,0 +1,1272 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA SPI driver +// +// Copyright 2024 Texas Instruments, Inc. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> + +#include <linux/acpi.h> +#include <linux/array_size.h> +#include <linux/bits.h> +#include <linux/cleanup.h> +#include <linux/crc8.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/firmware.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/mutex.h> +#include <linux/pm_runtime.h> +#include <linux/property.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <linux/time.h> +#include <linux/types.h> +#include <linux/units.h> + +#include <sound/hda_codec.h> +#include <sound/soc.h> +#include <sound/tas2781-dsp.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2781-spi.h" + +#include "hda_local.h" +#include "hda_auto_parser.h" +#include "hda_component.h" +#include "hda_jack.h" +#include "hda_generic.h" + +/* + * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD + * Define two controls, one is Volume control callbacks, the other is + * flag setting control callbacks. + */ + +/* Volume control callbacks for tas2781 */ +#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) { \ + .reg = xreg, .rreg = xreg, \ + .shift = xshift, .rshift = xshift,\ + .min = xmin, .max = xmax, .invert = xinvert, \ + } \ +} + +/* Flag control callbacks for tas2781 */ +#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ + .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = xhandler_get, \ + .put = xhandler_put, \ + .private_value = xdata, \ +} + +struct tas2781_hda { + struct tasdevice_priv *priv; + struct acpi_device *dacpi; + struct snd_kcontrol *dsp_prog_ctl; + struct snd_kcontrol *dsp_conf_ctl; + struct snd_kcontrol *snd_ctls[3]; + struct snd_kcontrol *prof_ctl; +}; + +static const struct regmap_range_cfg tasdevice_ranges[] = { + { + .range_min = 0, + .range_max = TASDEVICE_MAX_SIZE, + .selector_reg = TASDEVICE_PAGE_SELECT, + .selector_mask = GENMASK(7, 0), + .selector_shift = 0, + .window_start = 0, + .window_len = TASDEVICE_MAX_PAGE, + }, +}; + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 8, + .val_bits = 8, + .zero_flag_mask = true, + .cache_type = REGCACHE_NONE, + .ranges = tasdevice_ranges, + .num_ranges = ARRAY_SIZE(tasdevice_ranges), + .max_register = TASDEVICE_MAX_SIZE, +}; + +static int tasdevice_spi_switch_book(struct tasdevice_priv *tas_priv, int reg) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { + ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) { + dev_err(tas_priv->dev, "Switch Book E=%d\n", ret); + return ret; + } + tas_priv->cur_book = TASDEVICE_BOOK_ID(reg); + } + return ret; +} + +int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned int *val) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + /* + * In our TAS2781 SPI mode, if read from other book (not book 0), + * or read from page number larger than 1 in book 0, one more byte + * read is needed, and first byte is a dummy byte, need to be ignored. + */ + if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { + unsigned char data[2]; + + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, + data, sizeof(data)); + *val = data[1]; + } else { + ret = regmap_read(map, TASDEVICE_PAGE_REG(reg) | 1, val); + } + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_write(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned int value) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + ret = regmap_write(map, TASDEVICE_PAGE_REG(reg), value); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_bulk_write(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned char *data, + unsigned int len) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + ret = regmap_bulk_write(map, TASDEVICE_PAGE_REG(reg), data, len); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned char *data, + unsigned int len) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + if (len > TASDEVICE_MAX_PAGE) + len = TASDEVICE_MAX_PAGE; + /* + * In our TAS2781 SPI mode, if read from other book (not book 0), + * or read from page number larger than 1 in book 0, one more byte + * read is needed, and first byte is a dummy byte, need to be ignored. + */ + if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { + unsigned char buf[TASDEVICE_MAX_PAGE+1]; + + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, buf, + len + 1); + memcpy(data, buf + 1, len); + } else { + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, data, + len); + } + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned int mask, + unsigned int value) +{ + struct regmap *map = tas_priv->regmap; + int ret, val; + + /* + * In our TAS2781 SPI mode, read/write was masked in last bit of + * address, it cause regmap_update_bits() not work as expected. + */ + ret = tasdevice_spi_dev_read(tas_priv, reg, &val); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + return ret; + } + ret = regmap_write(map, TASDEVICE_PAGE_REG(reg), + (val & ~mask) | (mask & value)); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +static void tas2781_spi_reset(struct tasdevice_priv *tas_dev) +{ + int ret; + + if (tas_dev->reset) { + gpiod_set_value_cansleep(tas_dev->reset, 0); + fsleep(800); + gpiod_set_value_cansleep(tas_dev->reset, 1); + } + ret = tasdevice_spi_dev_write(tas_dev, TAS2781_REG_SWRESET, + TAS2781_REG_SWRESET_RESET); + if (ret < 0) + dev_err(tas_dev->dev, "dev sw-reset fail, %d\n", ret); + fsleep(1000); +} + +static int tascodec_spi_init(struct tasdevice_priv *tas_priv, + void *codec, struct module *module, + void (*cont)(const struct firmware *fw, void *context)) +{ + int ret; + + /* + * Codec Lock Hold to ensure that codec_probe and firmware parsing and + * loading do not simultaneously execute. + */ + guard(mutex)(&tas_priv->codec_lock); + + ret = scnprintf(tas_priv->rca_binaryname, + sizeof(tas_priv->rca_binaryname), "%sRCA%d.bin", + tas_priv->dev_name, tas_priv->index); + if (ret <= 0) { + dev_err(tas_priv->dev, "rca name err:0x%08x\n", ret); + return ret; + } + crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); + tas_priv->codec = codec; + ret = request_firmware_nowait(module, FW_ACTION_UEVENT, + tas_priv->rca_binaryname, tas_priv->dev, GFP_KERNEL, tas_priv, + cont); + if (ret) + dev_err(tas_priv->dev, "request_firmware_nowait err:0x%08x\n", + ret); + + return ret; +} + +static void tasdevice_spi_init(struct tasdevice_priv *tas_priv) +{ + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + tas_priv->cur_book = -1; + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + /* Store default registers address for calibration data. */ + tas_priv->cali_reg_array[0] = TASDEVICE_REG(0, 0x17, 0x74); + tas_priv->cali_reg_array[1] = TASDEVICE_REG(0, 0x18, 0x0c); + tas_priv->cali_reg_array[2] = TASDEVICE_REG(0, 0x18, 0x14); + tas_priv->cali_reg_array[3] = TASDEVICE_REG(0, 0x13, 0x70); + tas_priv->cali_reg_array[4] = TASDEVICE_REG(0, 0x18, 0x7c); + + mutex_init(&tas_priv->codec_lock); +} + +static int tasdevice_spi_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask; + int max = mc->max; + int val, ret; + + mask = rounddown_pow_of_two(max); + mask <<= mc->shift; + val = clamp(invert ? max - ucontrol->value.integer.value[0] : + ucontrol->value.integer.value[0], 0, max); + ret = tasdevice_spi_dev_update_bits(tas_priv, + mc->reg, mask, (unsigned int)(val << mc->shift)); + if (ret) + dev_err(tas_priv->dev, "set AMP vol error in dev %d\n", + tas_priv->index); + + return ret; +} + +static int tasdevice_spi_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask = 0; + int max = mc->max; + int ret, val; + + /* Read the primary device */ + ret = tasdevice_spi_dev_read(tas_priv, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get AMP vol error\n", __func__); + return ret; + } + + mask = rounddown_pow_of_two(max); + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + val = clamp(invert ? max - val : val, 0, max); + ucontrol->value.integer.value[0] = val; + + return ret; +} + +static int tasdevice_spi_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int val, ret; + + val = clamp(invert ? max - ucontrol->value.integer.value[0] : + ucontrol->value.integer.value[0], 0, max); + ret = tasdevice_spi_dev_write(tas_priv, mc->reg, (unsigned int)val); + if (ret) + dev_err(tas_priv->dev, "set digital vol err in dev %d\n", + tas_priv->index); + + return ret; +} + +static int tasdevice_spi_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int ret, val; + + /* Read the primary device as the whole */ + ret = tasdevice_spi_dev_read(tas_priv, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get digital vol err\n", __func__); + return ret; + } + + val = clamp(invert ? max - val : val, 0, max); + ucontrol->value.integer.value[0] = val; + + return ret; +} + +static int tas2781_read_acpi(struct tas2781_hda *tas_hda, + const char *hid, + int id) +{ + struct tasdevice_priv *p = tas_hda->priv; + struct acpi_device *adev; + struct device *physdev; + u32 values[HDA_MAX_COMPONENTS]; + const char *property; + size_t nval; + int ret, i; + + adev = acpi_dev_get_first_match_dev(hid, NULL, -1); + if (!adev) { + dev_err(p->dev, "Failed to find ACPI device: %s\n", hid); + return -ENODEV; + } + + strscpy(p->dev_name, hid, sizeof(p->dev_name)); + tas_hda->dacpi = adev; + physdev = get_device(acpi_get_first_physical_node(adev)); + acpi_dev_put(adev); + + property = "ti,dev-index"; + ret = device_property_count_u32(physdev, property); + if (ret <= 0 || ret > ARRAY_SIZE(values)) { + ret = -EINVAL; + goto err; + } + nval = ret; + + ret = device_property_read_u32_array(physdev, property, values, nval); + if (ret) + goto err; + + p->index = U8_MAX; + for (i = 0; i < nval; i++) { + if (values[i] == id) { + p->index = i; + break; + } + } + if (p->index == U8_MAX) { + dev_dbg(p->dev, "No index found in %s\n", property); + ret = -ENODEV; + goto err; + } + + if (p->index == 0) { + /* All of amps share same RESET pin. */ + p->reset = devm_gpiod_get_index_optional(physdev, "reset", + p->index, GPIOD_OUT_LOW); + if (IS_ERR(p->reset)) { + dev_err_probe(p->dev, ret, "Failed on reset GPIO\n"); + goto err; + } + } + put_device(physdev); + + return 0; +err: + dev_err(p->dev, "read acpi error, ret: %d\n", ret); + put_device(physdev); + acpi_dev_put(adev); + + return ret; +} + +static void tas2781_hda_playback_hook(struct device *dev, int action) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + if (action == HDA_GEN_PCM_ACT_OPEN) { + pm_runtime_get_sync(dev); + guard(mutex)(&tas_hda->priv->codec_lock); + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + } else if (action == HDA_GEN_PCM_ACT_CLOSE) { + guard(mutex)(&tas_hda->priv->codec_lock); + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + } +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->rcabin.ncfgs - 1; + + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + + return 0; +} + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int max = tas_priv->rcabin.ncfgs - 1; + int val; + + val = clamp(ucontrol->value.integer.value[0], 0, max); + if (tas_priv->rcabin.profile_cfg_id != val) { + tas_priv->rcabin.profile_cfg_id = val; + return 1; + } + + return 0; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->fmw->nr_programs - 1; + + return 0; +} + +static int tasdevice_info_config(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->fmw->nr_configurations - 1; + + return 0; +} + +static int tasdevice_program_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int nr_program = ucontrol->value.integer.value[0]; + int max = tas_priv->fmw->nr_programs - 1; + int val; + + val = clamp(nr_program, 0, max); + + if (tas_priv->cur_prog != val) { + tas_priv->cur_prog = val; + return 1; + } + + return 0; +} + +static int tasdevice_config_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + + return 0; +} + +static int tasdevice_config_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int max = tas_priv->fmw->nr_configurations - 1; + int val; + + val = clamp(ucontrol->value.integer.value[0], 0, max); + + if (tas_priv->cur_conf != val) { + tas_priv->cur_conf = val; + return 1; + } + + return 0; +} + +/* + * tas2781_digital_getvol - get the volum control + * @kcontrol: control pointer + * @ucontrol: User data + * + * Customer Kcontrol for tas2781 is primarily for regmap booking, paging + * depends on internal regmap mechanism. + * tas2781 contains book and page two-level register map, especially + * book switching will set the register BXXP00R7F, after switching to the + * correct book, then leverage the mechanism for paging to access the + * register. + * + * Return 0 if succeeded. + */ +static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_spi_digital_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_spi_amp_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_digital_putvol. */ + return tasdevice_spi_digital_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_amp_putvol. */ + return tasdevice_spi_amp_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = (int)tas_priv->force_fwload_status; + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + str_on_off(tas_priv->force_fwload_status)); + + return 0; +} + +static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + bool change, val = (bool)ucontrol->value.integer.value[0]; + + if (tas_priv->force_fwload_status == val) { + change = false; + } else { + change = true; + tas_priv->force_fwload_status = val; + } + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + str_on_off(tas_priv->force_fwload_status)); + + return change; +} + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain 0", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain 0", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load 0", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain 1", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain 1", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load 1", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static const struct snd_kcontrol_new tas2781_prof_ctrl[] = { +{ + .name = "Speaker Profile Id - 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, +}, +{ + .name = "Speaker Profile Id - 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, +}, +}; +static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl[] = { +{ + .name = "Speaker Program Id 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, +}, +{ + .name = "Speaker Program Id 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, +}, +}; + +static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl[] = { +{ + .name = "Speaker Config Id 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_config, + .get = tasdevice_config_get, + .put = tasdevice_config_put, +}, +{ + .name = "Speaker Config Id 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_config, + .get = tasdevice_config_get, + .put = tasdevice_config_put, +}, +}; + +static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) +{ + int i, rc; + + /* + * If no calibration data exist in tasdevice_priv *tas_priv, + * calibration apply will be ignored, and use default values + * in firmware binary, which was loaded during firmware download. + */ + if (tas_priv->cali_data[0] == 0) + return; + /* + * Calibration data was saved in tasdevice_priv *tas_priv as: + * unsigned int cali_data[CALIB_MAX]; + * and every data (in 4 bytes) will be saved in register which in + * book 0, and page number in page_array[], offset was saved in + * rgno_array[]. + */ + for (i = 0; i < CALIB_MAX; i++) { + rc = tasdevice_spi_dev_bulk_write(tas_priv, + tas_priv->cali_reg_array[i], + (unsigned char *)&tas_priv->cali_data[i], 4); + if (rc < 0) + dev_err(tas_priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + tas_priv->index, i, rc); + } +} + +/* + * Update the calibration data, including speaker impedance, f0, etc, + * into algo. Calibrate data is done by manufacturer in the factory. + * These data are used by Algo for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * Calibration data format in EFI is V2, since 2024. + */ +static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) +{ + /* + * GUID was used for data access in BIOS, it was provided by board + * manufactory, like HP: "{02f9af02-7734-4233-b43d-93fe5aa35db3}" + */ + efi_guid_t efi_guid = + EFI_GUID(0x02f9af02, 0x7734, 0x4233, + 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + static efi_char16_t efi_name[] = TASDEVICE_CALIBRATION_DATA_NAME; + unsigned char data[TASDEVICE_CALIBRATION_DATA_SIZE], *buf; + unsigned int attr, crc, offset, *tmp_val; + struct tm *tm = &tas_priv->tm; + unsigned long total_sz = 0; + efi_status_t status; + + tas_priv->cali_data[0] = 0; + status = efi.get_variable(efi_name, &efi_guid, &attr, &total_sz, data); + if (status == EFI_BUFFER_TOO_SMALL) { + if (total_sz > TASDEVICE_CALIBRATION_DATA_SIZE) + return -ENOMEM; + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &total_sz, data); + } + if (status != EFI_SUCCESS) + return -EINVAL; + + tmp_val = (unsigned int *)data; + if (tmp_val[0] == 2781) { + /* + * New features were added in calibrated Data V3: + * 1. Added calibration registers address define in + * a node, marked as Device id == 0x80. + * New features were added in calibrated Data V2: + * 1. Added some the fields to store the link_id and + * uniqie_id for multi-link solutions + * 2. Support flexible number of devices instead of + * fixed one in V1. + * Layout of calibrated data V2 in UEFI(total 256 bytes): + * ChipID (2781, 4 bytes) + * Device-Sum (4 bytes) + * TimeStamp of Calibration (4 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info () { + * SDW link id (2bytes) + * SDW unique_id (2bytes) + * } // if Device number is 0x80, mean it's + * calibration registers address. + * Calibrated Data of Device #i (20 bytes) + * } + * CRC (4 bytes) + * Reserved (the rest) + */ + crc = crc32(~0, data, (3 + tmp_val[1] * 6) * 4) ^ ~0; + + if (crc != tmp_val[3 + tmp_val[1] * 6]) + return 0; + + time64_to_tm(tmp_val[2], 0, tm); + for (int j = 0; j < tmp_val[1]; j++) { + offset = j * 6 + 3; + if (tmp_val[offset] == tas_priv->index) { + for (int i = 0; i < CALIB_MAX; i++) + tas_priv->cali_data[i] = + tmp_val[offset + i + 1]; + } else if (tmp_val[offset] == + TASDEVICE_CALIBRATION_REG_ADDRESS) { + for (int i = 0; i < CALIB_MAX; i++) { + buf = &data[(offset + i + 1) * 4]; + tas_priv->cali_reg_array[i] = + TASDEVICE_REG(buf[1], buf[2], + buf[3]); + } + } + tas_priv->apply_calibration(tas_priv); + } + } else { + /* + * Calibration data is in V1 format. + * struct cali_data { + * char cali_data[20]; + * } + * + * struct { + * struct cali_data cali_data[4]; + * int TimeStamp of Calibration (4 bytes) + * int CRC (4 bytes) + * } ueft; + */ + crc = crc32(~0, data, 84) ^ ~0; + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + for (int i = 0; i < CALIB_MAX; i++) + tas_priv->cali_data[i] = + tmp_val[tas_priv->index * 5 + i]; + tas_priv->apply_calibration(tas_priv); + } + } + + return 0; +} + +static void tas2781_hda_remove_controls(struct tas2781_hda *tas_hda) +{ + struct hda_codec *codec = tas_hda->priv->codec; + + snd_ctl_remove(codec->card, tas_hda->dsp_prog_ctl); + + snd_ctl_remove(codec->card, tas_hda->dsp_conf_ctl); + + for (int i = ARRAY_SIZE(tas_hda->snd_ctls) - 1; i >= 0; i--) + snd_ctl_remove(codec->card, tas_hda->snd_ctls[i]); + + snd_ctl_remove(codec->card, tas_hda->prof_ctl); +} + +static void tasdev_fw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); + struct hda_codec *codec = tas_priv->codec; + int i, j, ret; + + pm_runtime_get_sync(tas_priv->dev); + guard(mutex)(&tas_priv->codec_lock); + + ret = tasdevice_spi_rca_parser(tas_priv, fmw); + if (ret) + goto out; + + /* Add control one time only. */ + tas_hda->prof_ctl = snd_ctl_new1(&tas2781_prof_ctrl[tas_priv->index], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->prof_ctl); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + tas2781_prof_ctrl[tas_priv->index].name, ret); + goto out; + } + j = tas_priv->index * ARRAY_SIZE(tas2781_snd_controls) / 2; + for (i = 0; i < 3; i++) { + tas_hda->snd_ctls[i] = snd_ctl_new1(&tas2781_snd_controls[i+j], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->snd_ctls[i]); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_snd_controls[i+tas_priv->index*3].name, + ret); + goto out; + } + } + + tasdevice_spi_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%08X-%01d.bin", + codec->core.subsystem_id, tas_priv->index); + ret = tasdevice_spi_dsp_parser(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dspfw load %s error\n", + tas_priv->coef_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + + /* Add control one time only. */ + tas_hda->dsp_prog_ctl = + snd_ctl_new1(&tas2781_dsp_prog_ctrl[tas_priv->index], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->dsp_prog_ctl); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_dsp_prog_ctrl[tas_priv->index].name, ret); + goto out; + } + + tas_hda->dsp_conf_ctl = + snd_ctl_new1(&tas2781_dsp_conf_ctrl[tas_priv->index], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->dsp_conf_ctl); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + tas2781_dsp_conf_ctrl[tas_priv->index].name, ret); + goto out; + } + + /* Perform AMP reset before firmware download. */ + tas_priv->rcabin.profile_cfg_id = TAS2781_PRE_POST_RESET_CFG; + tasdevice_spi_tuning_switch(tas_priv, 0); + tas2781_spi_reset(tas_priv); + tasdevice_spi_tuning_switch(tas_priv, 1); + + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + ret = tasdevice_spi_prmg_load(tas_priv, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "FW download failed = %d\n", ret); + goto out; + } + if (tas_priv->fmw->nr_programs > 0) + tas_priv->cur_prog = 0; + if (tas_priv->fmw->nr_configurations > 0) + tas_priv->cur_conf = 0; + + /* + * If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + tas_priv->save_calibration(tas_priv); + +out: + if (fmw) + release_firmware(fmw); + pm_runtime_mark_last_busy(tas_hda->priv->dev); + pm_runtime_put_autosuspend(tas_hda->priv->dev); +} + +static int tas2781_hda_bind(struct device *dev, struct device *master, + void *master_data) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct hda_component_parent *parent = master_data; + struct hda_component *comp; + struct hda_codec *codec; + int ret; + + comp = hda_component_from_index(parent, tas_hda->priv->index); + if (!comp) + return -EINVAL; + + if (comp->dev) + return -EBUSY; + + codec = parent->codec; + + pm_runtime_get_sync(dev); + + comp->dev = dev; + + strscpy(comp->name, dev_name(dev), sizeof(comp->name)); + + ret = tascodec_spi_init(tas_hda->priv, codec, THIS_MODULE, + tasdev_fw_ready); + if (!ret) + comp->playback_hook = tas2781_hda_playback_hook; + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + + return ret; +} + +static void tas2781_hda_unbind(struct device *dev, struct device *master, + void *master_data) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct hda_component_parent *parent = master_data; + struct hda_component *comp; + + comp = hda_component_from_index(parent, tas_hda->priv->index); + if (comp && (comp->dev == dev)) { + comp->dev = NULL; + memset(comp->name, 0, sizeof(comp->name)); + comp->playback_hook = NULL; + } + + tas2781_hda_remove_controls(tas_hda); + + tasdevice_spi_config_info_remove(tas_hda->priv); + tasdevice_spi_dsp_remove(tas_hda->priv); + + tas_hda->priv->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static const struct component_ops tas2781_hda_comp_ops = { + .bind = tas2781_hda_bind, + .unbind = tas2781_hda_unbind, +}; + +static void tas2781_hda_remove(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + component_del(tas_hda->priv->dev, &tas2781_hda_comp_ops); + + pm_runtime_get_sync(tas_hda->priv->dev); + pm_runtime_disable(tas_hda->priv->dev); + + pm_runtime_put_noidle(tas_hda->priv->dev); + + mutex_destroy(&tas_hda->priv->codec_lock); +} + +static int tas2781_hda_spi_probe(struct spi_device *spi) +{ + struct tasdevice_priv *tas_priv; + struct tas2781_hda *tas_hda; + const char *device_name; + int ret = 0; + + tas_hda = devm_kzalloc(&spi->dev, sizeof(*tas_hda), GFP_KERNEL); + if (!tas_hda) + return -ENOMEM; + + spi->max_speed_hz = TAS2781_SPI_MAX_FREQ; + + tas_priv = devm_kzalloc(&spi->dev, sizeof(*tas_priv), GFP_KERNEL); + if (!tas_priv) + goto err; + tas_priv->dev = &spi->dev; + tas_hda->priv = tas_priv; + tas_priv->regmap = devm_regmap_init_spi(spi, &tasdevice_regmap); + if (IS_ERR(tas_priv->regmap)) { + ret = PTR_ERR(tas_priv->regmap); + dev_err(tas_priv->dev, "Failed to allocate regmap: %d\n", + ret); + goto err; + } + if (strstr(dev_name(&spi->dev), "TXNW2781")) { + device_name = "TXNW2781"; + tas_priv->save_calibration = tas2781_save_calibration; + tas_priv->apply_calibration = tas2781_apply_calib; + } else { + goto err; + } + + tas_priv->irq = spi->irq; + dev_set_drvdata(&spi->dev, tas_hda); + ret = tas2781_read_acpi(tas_hda, device_name, + spi_get_chipselect(spi, 0)); + if (ret) + return dev_err_probe(tas_priv->dev, ret, + "Platform not supported\n"); + + tasdevice_spi_init(tas_priv); + + pm_runtime_set_autosuspend_delay(tas_priv->dev, 3000); + pm_runtime_use_autosuspend(tas_priv->dev); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_set_active(tas_priv->dev); + pm_runtime_get_noresume(tas_priv->dev); + pm_runtime_enable(tas_priv->dev); + + pm_runtime_put_autosuspend(tas_priv->dev); + + ret = component_add(tas_priv->dev, &tas2781_hda_comp_ops); + if (ret) { + dev_err(tas_priv->dev, "Register component fail: %d\n", ret); + pm_runtime_disable(tas_priv->dev); + } + +err: + if (ret) + tas2781_hda_remove(&spi->dev); + + return ret; +} + +static void tas2781_hda_spi_remove(struct spi_device *spi) +{ + tas2781_hda_remove(&spi->dev); +} + +static int tas2781_runtime_suspend(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + guard(mutex)(&tas_hda->priv->codec_lock); + + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + + tas_hda->priv->cur_book = -1; + tas_hda->priv->cur_conf = -1; + + return 0; +} + +static int tas2781_runtime_resume(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + guard(mutex)(&tas_hda->priv->codec_lock); + + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + + return 0; +} + +static int tas2781_system_suspend(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + int ret; + + ret = pm_runtime_force_suspend(dev); + if (ret) + return ret; + + /* Shutdown chip before system suspend */ + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + + /* + * Reset GPIO may be shared, so cannot reset here. + * However beyond this point, amps may be powered down. + */ + return 0; +} + +static int tas2781_system_resume(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + int ret, val; + + ret = pm_runtime_force_resume(dev); + if (ret) + return ret; + + guard(mutex)(&tas_hda->priv->codec_lock); + + ret = tasdevice_spi_dev_read(tas_hda->priv, TAS2781_REG_CLK_CONFIG, + &val); + if (ret < 0) + return ret; + + if (val == TAS2781_REG_CLK_CONFIG_RESET) { + tas_hda->priv->cur_book = -1; + tas_hda->priv->cur_conf = -1; + tas_hda->priv->cur_prog = -1; + + ret = tasdevice_spi_prmg_load(tas_hda->priv, 0); + if (ret < 0) { + dev_err(tas_hda->priv->dev, + "FW download failed = %d\n", ret); + return ret; + } + tas_hda->priv->apply_calibration(tas_hda->priv); + + if (tas_hda->priv->playback_started) + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + } + + return ret; +} + +static const struct dev_pm_ops tas2781_hda_pm_ops = { + RUNTIME_PM_OPS(tas2781_runtime_suspend, tas2781_runtime_resume, NULL) + SYSTEM_SLEEP_PM_OPS(tas2781_system_suspend, tas2781_system_resume) +}; + +static const struct spi_device_id tas2781_hda_spi_id[] = { + { "tas2781-hda", }, + {} +}; + +static const struct acpi_device_id tas2781_acpi_hda_match[] = { + {"TXNW2781", }, + {} +}; +MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); + +static struct spi_driver tas2781_hda_spi_driver = { + .driver = { + .name = "tas2781-hda", + .acpi_match_table = tas2781_acpi_hda_match, + .pm = &tas2781_hda_pm_ops, + }, + .id_table = tas2781_hda_spi_id, + .probe = tas2781_hda_spi_probe, + .remove = tas2781_hda_spi_remove, +}; +module_spi_driver(tas2781_hda_spi_driver); + +MODULE_DESCRIPTION("TAS2781 HDA SPI Driver"); +MODULE_AUTHOR("Baojun, Xu, <baojun.xug(a)ti.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c new file mode 100644 index 000000000000..c98f9beb77a0 --- /dev/null +++ b/sound/pci/hda/tas2781_spi_fwlib.c @@ -0,0 +1,2006 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA SPI driver +// +// Copyright 2024 Texas Instruments, Inc. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> + +#include <asm/unaligned.h> +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/types.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tas2781-dsp.h> +#include <sound/tlv.h> + +#include "tas2781-spi.h" + +#define OFFSET_ERROR_BIT BIT(31) + +#define ERROR_PRAM_CRCCHK 0x0000000 +#define ERROR_YRAM_CRCCHK 0x0000001 +#define PPC_DRIVER_CRCCHK 0x00000200 + +#define TAS2781_SA_COEFF_SWAP_REG TASDEVICE_REG(0, 0x35, 0x2c) +#define TAS2781_YRAM_BOOK1 140 +#define TAS2781_YRAM1_PAGE 42 +#define TAS2781_YRAM1_START_REG 88 + +#define TAS2781_YRAM2_START_PAGE 43 +#define TAS2781_YRAM2_END_PAGE 49 +#define TAS2781_YRAM2_START_REG 8 +#define TAS2781_YRAM2_END_REG 127 + +#define TAS2781_YRAM3_PAGE 50 +#define TAS2781_YRAM3_START_REG 8 +#define TAS2781_YRAM3_END_REG 27 + +/* should not include B0_P53_R44-R47 */ +#define TAS2781_YRAM_BOOK2 0 +#define TAS2781_YRAM4_START_PAGE 50 +#define TAS2781_YRAM4_END_PAGE 60 + +#define TAS2781_YRAM5_PAGE 61 +#define TAS2781_YRAM5_START_REG TAS2781_YRAM3_START_REG +#define TAS2781_YRAM5_END_REG TAS2781_YRAM3_END_REG + +#define TASDEVICE_MAXPROGRAM_NUM_KERNEL 5 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS 64 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL 10 +#define MAIN_ALL_DEVICES_1X 0x01 +#define MAIN_DEVICE_A_1X 0x02 +#define MAIN_DEVICE_B_1X 0x03 +#define MAIN_DEVICE_C_1X 0x04 +#define MAIN_DEVICE_D_1X 0x05 +#define COEFF_DEVICE_A_1X 0x12 +#define COEFF_DEVICE_B_1X 0x13 +#define COEFF_DEVICE_C_1X 0x14 +#define COEFF_DEVICE_D_1X 0x15 +#define PRE_DEVICE_A_1X 0x22 +#define PRE_DEVICE_B_1X 0x23 +#define PRE_DEVICE_C_1X 0x24 +#define PRE_DEVICE_D_1X 0x25 +#define PRE_SOFTWARE_RESET_DEVICE_A 0x41 +#define PRE_SOFTWARE_RESET_DEVICE_B 0x42 +#define PRE_SOFTWARE_RESET_DEVICE_C 0x43 +#define PRE_SOFTWARE_RESET_DEVICE_D 0x44 +#define POST_SOFTWARE_RESET_DEVICE_A 0x45 +#define POST_SOFTWARE_RESET_DEVICE_B 0x46 +#define POST_SOFTWARE_RESET_DEVICE_C 0x47 +#define POST_SOFTWARE_RESET_DEVICE_D 0x48 + +struct tas_crc { + unsigned char offset; + unsigned char len; +}; + +struct blktyp_devidx_map { + unsigned char blktyp; + unsigned char dev_idx; +}; + +/* fixed m68k compiling issue: mapping table can save code field */ +static const struct blktyp_devidx_map ppc3_tas2781_mapping_table[] = { + { MAIN_ALL_DEVICES_1X, 0x80 }, + { MAIN_DEVICE_A_1X, 0x81 }, + { COEFF_DEVICE_A_1X, 0x81 }, + { PRE_DEVICE_A_1X, 0x81 }, + { PRE_SOFTWARE_RESET_DEVICE_A, 0xC1 }, + { POST_SOFTWARE_RESET_DEVICE_A, 0xC1 }, + { MAIN_DEVICE_B_1X, 0x82 }, + { COEFF_DEVICE_B_1X, 0x82 }, + { PRE_DEVICE_B_1X, 0x82 }, + { PRE_SOFTWARE_RESET_DEVICE_B, 0xC2 }, + { POST_SOFTWARE_RESET_DEVICE_B, 0xC2 }, + { MAIN_DEVICE_C_1X, 0x83 }, + { COEFF_DEVICE_C_1X, 0x83 }, + { PRE_DEVICE_C_1X, 0x83 }, + { PRE_SOFTWARE_RESET_DEVICE_C, 0xC3 }, + { POST_SOFTWARE_RESET_DEVICE_C, 0xC3 }, + { MAIN_DEVICE_D_1X, 0x84 }, + { COEFF_DEVICE_D_1X, 0x84 }, + { PRE_DEVICE_D_1X, 0x84 }, + { PRE_SOFTWARE_RESET_DEVICE_D, 0xC4 }, + { POST_SOFTWARE_RESET_DEVICE_D, 0xC4 }, +}; + +static const struct blktyp_devidx_map ppc3_mapping_table[] = { + { MAIN_ALL_DEVICES_1X, 0x80 }, + { MAIN_DEVICE_A_1X, 0x81 }, + { COEFF_DEVICE_A_1X, 0xC1 }, + { PRE_DEVICE_A_1X, 0xC1 }, + { MAIN_DEVICE_B_1X, 0x82 }, + { COEFF_DEVICE_B_1X, 0xC2 }, + { PRE_DEVICE_B_1X, 0xC2 }, + { MAIN_DEVICE_C_1X, 0x83 }, + { COEFF_DEVICE_C_1X, 0xC3 }, + { PRE_DEVICE_C_1X, 0xC3 }, + { MAIN_DEVICE_D_1X, 0x84 }, + { COEFF_DEVICE_D_1X, 0xC4 }, + { PRE_DEVICE_D_1X, 0xC4 }, +}; + +static const struct blktyp_devidx_map non_ppc3_mapping_table[] = { + { MAIN_ALL_DEVICES, 0x80 }, + { MAIN_DEVICE_A, 0x81 }, + { COEFF_DEVICE_A, 0xC1 }, + { PRE_DEVICE_A, 0xC1 }, + { MAIN_DEVICE_B, 0x82 }, + { COEFF_DEVICE_B, 0xC2 }, + { PRE_DEVICE_B, 0xC2 }, + { MAIN_DEVICE_C, 0x83 }, + { COEFF_DEVICE_C, 0xC3 }, + { PRE_DEVICE_C, 0xC3 }, + { MAIN_DEVICE_D, 0x84 }, + { COEFF_DEVICE_D, 0xC4 }, + { PRE_DEVICE_D, 0xC4 }, +}; + +/* + * Device support different configurations for different scene, + * like voice, music, calibration, was write in regbin file. + * Will be stored into tas_priv after regbin was loaded. + */ +static struct tasdevice_config_info *tasdevice_add_config( + struct tasdevice_priv *tas_priv, unsigned char *config_data, + unsigned int config_size, int *status) +{ + struct tasdevice_config_info *cfg_info; + struct tasdev_blk_data **bk_da; + unsigned int config_offset = 0; + unsigned int i; + + /* + * In most projects are many audio cases, such as music, handfree, + * receiver, games, audio-to-haptics, PMIC record, bypass mode, + * portrait, landscape, etc. Even in multiple audios, one or + * two of the chips will work for the special case, such as + * ultrasonic application. In order to support these variable-numbers + * of audio cases, flexible configs have been introduced in the + * DSP firmware. + */ + cfg_info = kzalloc(sizeof(*cfg_info), GFP_KERNEL); + if (!cfg_info) { + *status = -ENOMEM; + return NULL; + } + + if (tas_priv->rcabin.fw_hdr.binary_version_num >= 0x105) { + if ((config_offset + 64) > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add conf: Out of boundary\n"); + goto config_err; + } + config_offset += 64; + } + + if ((config_offset + 4) > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add config: Out of boundary\n"); + goto config_err; + } + + /* + * convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + cfg_info->nblocks = get_unaligned_be32(&config_data[config_offset]); + config_offset += 4; + + /* + * Several kinds of dsp/algorithm firmwares can run on tas2781, + * the number and size of blk are not fixed and different among + * these firmwares. + */ + bk_da = cfg_info->blk_data = kcalloc(cfg_info->nblocks, + sizeof(*bk_da), GFP_KERNEL); + if (!bk_da) { + *status = -ENOMEM; + goto config_err; + } + cfg_info->real_nblocks = 0; + for (i = 0; i < cfg_info->nblocks; i++) { + if (config_offset + 12 > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d nblocks = %u!\n", + __func__, i, cfg_info->nblocks); + goto block_err; + } + bk_da[i] = kzalloc(sizeof(*bk_da[i]), GFP_KERNEL); + if (!bk_da[i]) { + *status = -ENOMEM; + goto block_err; + } + + bk_da[i]->dev_idx = config_data[config_offset]; + config_offset++; + + bk_da[i]->block_type = config_data[config_offset]; + config_offset++; + + bk_da[i]->yram_checksum = + get_unaligned_be16(&config_data[config_offset]); + config_offset += 2; + bk_da[i]->block_size = + get_unaligned_be32(&config_data[config_offset]); + config_offset += 4; + + bk_da[i]->n_subblks = + get_unaligned_be32(&config_data[config_offset]); + + config_offset += 4; + + if (config_offset + bk_da[i]->block_size > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d blks = %u!\n", + __func__, i, cfg_info->nblocks); + goto block_err; + } + /* instead of kzalloc+memcpy */ + bk_da[i]->regdata = kmemdup(&config_data[config_offset], + bk_da[i]->block_size, GFP_KERNEL); + if (!bk_da[i]->regdata) { + *status = -ENOMEM; + i++; + goto block_err; + } + + config_offset += bk_da[i]->block_size; + cfg_info->real_nblocks += 1; + } + + return cfg_info; +block_err: + for (int j = 0; j < i; j++) + kfree(bk_da[j]); + kfree(bk_da); +config_err: + kfree(cfg_info); + return NULL; +} + +/* Regbin file parser function. */ +int tasdevice_spi_rca_parser(void *context, const struct firmware *fmw) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_config_info **cfg_info; + struct tasdevice_rca_hdr *fw_hdr; + struct tasdevice_rca *rca; + unsigned int total_config_sz = 0; + int offset = 0, ret = 0, i; + unsigned char *buf; + + rca = &tas_priv->rcabin; + fw_hdr = &rca->fw_hdr; + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "Failed to read %s\n", + tas_priv->rca_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + buf = (unsigned char *)fmw->data; + fw_hdr->img_sz = get_unaligned_be32(&buf[offset]); + offset += 4; + if (fw_hdr->img_sz != fmw->size) { + dev_err(tas_priv->dev, + "File size not match, %d %u", (int)fmw->size, + fw_hdr->img_sz); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + fw_hdr->checksum = get_unaligned_be32(&buf[offset]); + offset += 4; + fw_hdr->binary_version_num = get_unaligned_be32(&buf[offset]); + if (fw_hdr->binary_version_num < 0x103) { + dev_err(tas_priv->dev, "File version 0x%04x is too low", + fw_hdr->binary_version_num); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + offset += 4; + fw_hdr->drv_fw_version = get_unaligned_be32(&buf[offset]); + offset += 8; + fw_hdr->plat_type = buf[offset++]; + fw_hdr->dev_family = buf[offset++]; + fw_hdr->reserve = buf[offset++]; + fw_hdr->ndev = buf[offset++]; + if (offset + TASDEVICE_DEVICE_SUM > fw_hdr->img_sz) { + dev_err(tas_priv->dev, "rca_ready: Out of boundary!\n"); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + for (i = 0; i < TASDEVICE_DEVICE_SUM; i++, offset++) + fw_hdr->devs[i] = buf[offset]; + + fw_hdr->nconfig = get_unaligned_be32(&buf[offset]); + offset += 4; + + for (i = 0; i < TASDEVICE_CONFIG_SUM; i++) { + fw_hdr->config_size[i] = get_unaligned_be32(&buf[offset]); + offset += 4; + total_config_sz += fw_hdr->config_size[i]; + } + + if (fw_hdr->img_sz - total_config_sz != (unsigned int)offset) { + dev_err(tas_priv->dev, "Bin file err %d - %d != %d!\n", + fw_hdr->img_sz, total_config_sz, (int)offset); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + cfg_info = kcalloc(fw_hdr->nconfig, sizeof(*cfg_info), GFP_KERNEL); + if (!cfg_info) { + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -ENOMEM; + } + rca->cfg_info = cfg_info; + rca->ncfgs = 0; + for (i = 0; i < (int)fw_hdr->nconfig; i++) { + rca->ncfgs += 1; + cfg_info[i] = tasdevice_add_config(tas_priv, &buf[offset], + fw_hdr->config_size[i], &ret); + if (ret) { + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return ret; + } + offset += (int)fw_hdr->config_size[i]; + } + + return ret; +} + +/* fixed m68k compiling issue: mapping table can save code field */ +static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block) +{ + struct blktyp_devidx_map *p = + (struct blktyp_devidx_map *)non_ppc3_mapping_table; + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &fw_hdr->fixed_hdr; + int i, n = ARRAY_SIZE(non_ppc3_mapping_table); + unsigned char dev_idx = 0; + + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; + n = ARRAY_SIZE(ppc3_tas2781_mapping_table); + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + p = (struct blktyp_devidx_map *)ppc3_mapping_table; + n = ARRAY_SIZE(ppc3_mapping_table); + } + + for (i = 0; i < n; i++) { + if (block->type == p[i].blktyp) { + dev_idx = p[i].dev_idx; + break; + } + } + + return dev_idx; +} + +/* Block parser function. */ +static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + + if (offset + 16 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + + /* + * Convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host. + */ + block->type = get_unaligned_be32(&data[offset]); + offset += 4; + + block->is_pchksum_present = data[offset++]; + block->pchksum = data[offset++]; + block->is_ychksum_present = data[offset++]; + block->ychksum = data[offset++]; + block->blk_size = get_unaligned_be32(&data[offset]); + offset += 4; + block->nr_subblocks = get_unaligned_be32(&data[offset]); + offset += 4; + + /* + * Fixed m68k compiling issue: + * 1. mapping table can save code field. + * 2. storing the dev_idx as a member of block can reduce unnecessary + * time and system resource comsumption of dev_idx mapping every + * time the block data writing to the dsp. + */ + block->dev_idx = map_dev_idx(tas_fmw, block); + + if (offset + block->blk_size > fmw->size) { + dev_err(tas_fmw->dev, "%s: nSublocks error\n", __func__); + return -EINVAL; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], block->blk_size, GFP_KERNEL); + if (!block->data) + return -ENOMEM; + + offset += block->blk_size; + + return offset; +} + +/* Data of block parser function. */ +static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = fmw->data; + struct tasdev_blk *blk; + unsigned int i; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + img_data->nr_blk = get_unaligned_be32(&data[offset]); + offset += 4; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(struct tasdev_blk), GFP_KERNEL); + if (!img_data->dev_blks) + return -ENOMEM; + + for (i = 0; i < img_data->nr_blk; i++) { + blk = &img_data->dev_blks[i]; + offset = fw_parse_block_data_kernel( + tas_fmw, blk, fmw, offset); + if (offset < 0) { + kfree(img_data->dev_blks); + return -EINVAL; + } + } + + return offset; +} + +/* Data of DSP program parser function. */ +static int fw_parse_program_data_kernel( + struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + struct tasdevice_prog *program; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &tas_fmw->programs[i]; + if (offset + 72 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + /* skip 72 unused byts */ + offset += 72; + + offset = fw_parse_data_kernel(tas_fmw, &program->dev_data, + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +/* Data of DSP configurations parser function. */ +static int fw_parse_configuration_data_kernel(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + struct tasdevice_config *config; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &tas_fmw->configs[i]; + if (offset + 80 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + memcpy(config->name, &data[offset], 64); + /* skip extra 16 bytes */ + offset += 80; + + offset = fw_parse_data_kernel(tas_fmw, &config->dev_data, + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +/* DSP firmware file header parser function for early PPC3 firmware binary. */ +static int fw_parse_variable_header_kernel(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + struct tasdevice_config *config; + struct tasdevice_prog *program; + const unsigned char *buf = fmw->data; + unsigned short max_confs; + unsigned int i; + + if (offset + 12 + 4 * TASDEVICE_MAXPROGRAM_NUM_KERNEL > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + fw_hdr->device_family = get_unaligned_be16(&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s:not TAS device\n", __func__); + return -EINVAL; + } + offset += 2; + fw_hdr->device = get_unaligned_be16(&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + return -EINVAL; + } + offset += 2; + + tas_fmw->nr_programs = get_unaligned_be32(&buf[offset]); + offset += 4; + + if (tas_fmw->nr_programs == 0 || + tas_fmw->nr_programs > TASDEVICE_MAXPROGRAM_NUM_KERNEL) { + dev_err(tas_priv->dev, "mnPrograms is invalid\n"); + return -EINVAL; + } + + tas_fmw->programs = kcalloc(tas_fmw->nr_programs, + sizeof(*tas_fmw->programs), GFP_KERNEL); + if (!tas_fmw->programs) + return -ENOMEM; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &tas_fmw->programs[i]; + program->prog_size = get_unaligned_be32(&buf[offset]); + offset += 4; + } + + /* Skip the unused prog_size */ + offset += 4 * (TASDEVICE_MAXPROGRAM_NUM_KERNEL - tas_fmw->nr_programs); + + tas_fmw->nr_configurations = get_unaligned_be32(&buf[offset]); + offset += 4; + + /* + * The max number of config in firmware greater than 4 pieces of + * tas2781s is different from the one lower than 4 pieces of + * tas2781s. + */ + max_confs = TASDEVICE_MAXCONFIG_NUM_KERNEL; + if (tas_fmw->nr_configurations == 0 || + tas_fmw->nr_configurations > max_confs) { + dev_err(tas_priv->dev, "%s: Conf is invalid\n", __func__); + kfree(tas_fmw->programs); + return -EINVAL; + } + + if (offset + 4 * max_confs > fmw->size) { + dev_err(tas_priv->dev, "%s: mpConfigurations err\n", __func__); + kfree(tas_fmw->programs); + return -EINVAL; + } + + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(*tas_fmw->configs), GFP_KERNEL); + if (!tas_fmw->configs) { + kfree(tas_fmw->programs); + return -ENOMEM; + } + + for (i = 0; i < tas_fmw->nr_programs; i++) { + config = &tas_fmw->configs[i]; + config->cfg_size = get_unaligned_be32(&buf[offset]); + offset += 4; + } + + /* Skip the unused configs */ + offset += 4 * (max_confs - tas_fmw->nr_programs); + + return offset; +} + +/* + * In sub-block data, have three type sub-block: + * 1. Single byte write. + * 2. Multi-byte write. + * 3. Delay. + * 4. Bits update. + * This function perform single byte write to device. + */ +static int tasdevice_single_byte_wr(void *context, int dev_idx, + unsigned char *data, int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned short len = get_unaligned_be16(&data[2]); + int i, subblk_offset, rc; + + subblk_offset = 4; + if (subblk_offset + 4 * len > sublocksize) { + dev_err(tas_priv->dev, "process_block: Out of boundary\n"); + return 0; + } + + for (i = 0; i < len; i++) { + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + rc = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + data[subblk_offset + 3]); + if (rc < 0) { + dev_err(tas_priv->dev, + "process_block: single write error\n"); + subblk_offset |= OFFSET_ERROR_BIT; + } + } + subblk_offset += 4; + } + + return subblk_offset; +} + +/* + * In sub-block data, have three type sub-block: + * 1. Single byte write. + * 2. Multi-byte write. + * 3. Delay. + * 4. Bits update. + * This function perform multi-write to device. + */ +static int tasdevice_burst_wr(void *context, int dev_idx, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned short len = get_unaligned_be16(&data[2]); + int subblk_offset, rc; + + subblk_offset = 4; + if (subblk_offset + 4 + len > sublocksize) { + dev_err(tas_priv->dev, "%s: BST Out of boundary\n", __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + if (len % 4) { + dev_err(tas_priv->dev, "%s:Bst-len(%u)not div by 4\n", + __func__, len); + subblk_offset |= OFFSET_ERROR_BIT; + } + + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + rc = tasdevice_spi_dev_bulk_write(tas_priv, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + &data[subblk_offset + 4], len); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: bulk_write error = %d\n", + __func__, rc); + subblk_offset |= OFFSET_ERROR_BIT; + } + } + subblk_offset += (len + 4); + + return subblk_offset; +} + +/* Just delay for ms.*/ +static int tasdevice_delay(void *context, int dev_idx, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned int sleep_time, subblk_offset = 2; + + if (subblk_offset + 2 > sublocksize) { + dev_err(tas_priv->dev, "%s: delay Out of boundary\n", + __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + sleep_time = get_unaligned_be16(&data[2]) * 1000; + fsleep(sleep_time); + } + subblk_offset += 2; + + return subblk_offset; +} + +/* + * In sub-block data, have three type sub-block: + * 1. Single byte write. + * 2. Multi-byte write. + * 3. Delay. + * 4. Bits update. + * This function perform bits update. + */ +static int tasdevice_field_wr(void *context, int dev_idx, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + int rc, subblk_offset = 2; + + if (subblk_offset + 6 > sublocksize) { + dev_err(tas_priv->dev, "%s: bit write Out of boundary\n", + __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + rc = tasdevice_spi_dev_update_bits(tas_priv, + TASDEVICE_REG(data[subblk_offset + 2], + data[subblk_offset + 3], + data[subblk_offset + 4]), + data[subblk_offset + 1], + data[subblk_offset + 5]); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: update_bits error = %d\n", + __func__, rc); + subblk_offset |= OFFSET_ERROR_BIT; + } + } + subblk_offset += 6; + + return subblk_offset; +} + +/* Data block process function. */ +static int tasdevice_process_block(void *context, unsigned char *data, + unsigned char dev_idx, int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + int blktyp = dev_idx & 0xC0, subblk_offset; + unsigned char subblk_typ = data[1]; + + switch (subblk_typ) { + case TASDEVICE_CMD_SING_W: + subblk_offset = tasdevice_single_byte_wr(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + case TASDEVICE_CMD_BURST: + subblk_offset = tasdevice_burst_wr(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + case TASDEVICE_CMD_DELAY: + subblk_offset = tasdevice_delay(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + case TASDEVICE_CMD_FIELD_W: + subblk_offset = tasdevice_field_wr(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + default: + subblk_offset = 2; + break; + } + if (((subblk_offset & OFFSET_ERROR_BIT) != 0) && blktyp != 0) { + if (blktyp == 0x80) { + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + } else + tas_priv->cur_conf = -1; + } + subblk_offset &= ~OFFSET_ERROR_BIT; + + return subblk_offset; +} + +/* + * Device support different configurations for different scene, + * this function was used for choose different config. + */ +void tasdevice_spi_select_cfg_blk(void *pContext, int conf_no, + unsigned char block_type) +{ + struct tasdevice_priv *tas_priv = pContext; + struct tasdevice_rca *rca = &tas_priv->rcabin; + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdev_blk_data **blk_data; + unsigned int j, k; + + if (conf_no >= rca->ncfgs || conf_no < 0 || !cfg_info) { + dev_err(tas_priv->dev, "conf_no should be not more than %u\n", + rca->ncfgs); + return; + } + blk_data = cfg_info[conf_no]->blk_data; + + for (j = 0; j < cfg_info[conf_no]->real_nblocks; j++) { + unsigned int length = 0, rc = 0; + + if (block_type > 5 || block_type < 2) { + dev_err(tas_priv->dev, + "block_type should be in range from 2 to 5\n"); + break; + } + if (block_type != blk_data[j]->block_type) + continue; + + for (k = 0; k < blk_data[j]->n_subblks; k++) { + tas_priv->is_loading = true; + + rc = tasdevice_process_block(tas_priv, + blk_data[j]->regdata + length, + blk_data[j]->dev_idx, + blk_data[j]->block_size - length); + length += rc; + if (blk_data[j]->block_size < length) { + dev_err(tas_priv->dev, + "%s: %u %u out of boundary\n", + __func__, length, + blk_data[j]->block_size); + break; + } + } + if (length != blk_data[j]->block_size) + dev_err(tas_priv->dev, "%s: %u %u size is not same\n", + __func__, length, blk_data[j]->block_size); + } +} + +/* Block process function. */ +static int tasdevice_load_block_kernel( + struct tasdevice_priv *tasdevice, struct tasdev_blk *block) +{ + const unsigned int blk_size = block->blk_size; + unsigned char *data = block->data; + unsigned int i, length; + + for (i = 0, length = 0; i < block->nr_subblocks; i++) { + int rc = tasdevice_process_block(tasdevice, data + length, + block->dev_idx, blk_size - length); + if (rc < 0) { + dev_err(tasdevice->dev, + "%s: %u %u sublock write error\n", + __func__, length, blk_size); + return rc; + } + length += rc; + if (blk_size < length) { + dev_err(tasdevice->dev, "%s: %u %u out of boundary\n", + __func__, length, blk_size); + rc = -ENOMEM; + return rc; + } + } + + return 0; +} + +/* DSP firmware file header parser function. */ +static int fw_parse_variable_hdr(struct tasdevice_priv *tas_priv, + struct tasdevice_dspfw_hdr *fw_hdr, + const struct firmware *fmw, int offset) +{ + const unsigned char *buf = fmw->data; + int len = strlen((char *)&buf[offset]); + + len++; + + if (offset + len + 8 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + + offset += len; + + fw_hdr->device_family = get_unaligned_be32(&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s: not TAS device\n", __func__); + return -EINVAL; + } + offset += 4; + + fw_hdr->device = get_unaligned_be32(&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + return -EINVAL; + } + offset += 4; + fw_hdr->ndev = 1; + + return offset; +} + +/* DSP firmware file header parser function for size variabled header. */ +static int fw_parse_variable_header_git(struct tasdevice_priv + *tas_priv, const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + + offset = fw_parse_variable_hdr(tas_priv, fw_hdr, fmw, offset); + + return offset; +} + +/* DSP firmware file block parser function. */ +static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + int n; + + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Type error\n", __func__); + return -EINVAL; + } + block->type = get_unaligned_be32(&data[offset]); + offset += 4; + + if (tas_fmw->fw_hdr.fixed_hdr.drv_ver >= PPC_DRIVER_CRCCHK) { + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "PChkSumPresent error\n"); + return -EINVAL; + } + block->is_pchksum_present = data[offset]; + offset++; + + block->pchksum = data[offset]; + offset++; + + block->is_ychksum_present = data[offset]; + offset++; + + block->ychksum = data[offset]; + offset++; + } else { + block->is_pchksum_present = 0; + block->is_ychksum_present = 0; + } + + block->nr_cmds = get_unaligned_be32(&data[offset]); + offset += 4; + + n = block->nr_cmds * 4; + if (offset + n > fmw->size) { + dev_err(tas_fmw->dev, + "%s: File Size(%lu) error offset = %d n = %d\n", + __func__, (unsigned long)fmw->size, offset, n); + return -EINVAL; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], n, GFP_KERNEL); + if (!block->data) + return -ENOMEM; + + offset += n; + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_data(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = (unsigned char *)fmw->data; + struct tasdev_blk *blk; + unsigned int i, n; + + if (offset + 64 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Name error\n", __func__); + return -EINVAL; + } + memcpy(img_data->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n++; + if (offset + n + 2 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Description error\n", __func__); + return -EINVAL; + } + offset += n; + img_data->nr_blk = get_unaligned_be16(&data[offset]); + offset += 2; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(*img_data->dev_blks), GFP_KERNEL); + if (!img_data->dev_blks) + return -ENOMEM; + + for (i = 0; i < img_data->nr_blk; i++) { + blk = &img_data->dev_blks[i]; + offset = fw_parse_block_data(tas_fmw, blk, fmw, offset); + if (offset < 0) + return -EINVAL; + } + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_program_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *buf = (unsigned char *)fmw->data; + struct tasdevice_prog *program; + int i; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + tas_fmw->nr_programs = get_unaligned_be16(&buf[offset]); + offset += 2; + + if (tas_fmw->nr_programs == 0) { + /* Not error in calibration Data file, return directly */ + dev_dbg(tas_priv->dev, "%s: No Programs data, maybe calbin\n", + __func__); + return offset; + } + + tas_fmw->programs = + kcalloc(tas_fmw->nr_programs, sizeof(*tas_fmw->programs), + GFP_KERNEL); + if (!tas_fmw->programs) + return -ENOMEM; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + int n = 0; + + program = &tas_fmw->programs[i]; + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + offset += 64; + + n = strlen((char *)&buf[offset]); + /* skip '\0' and 5 unused bytes */ + n += 6; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + return -EINVAL; + } + + offset += n; + + offset = fw_parse_data(tas_fmw, &program->dev_data, fmw, + offset); + if (offset < 0) + return offset; + } + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_configuration_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + struct tasdevice_config *config; + unsigned int i, n; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + tas_fmw->nr_configurations = get_unaligned_be16(&data[offset]); + offset += 2; + + if (tas_fmw->nr_configurations == 0) { + dev_err(tas_priv->dev, "%s: Conf is zero\n", __func__); + /* Not error for calibration Data file, return directly */ + return offset; + } + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(*tas_fmw->configs), GFP_KERNEL); + if (!tas_fmw->configs) + return -ENOMEM; + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &tas_fmw->configs[i]; + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "File Size err\n"); + return -EINVAL; + } + memcpy(config->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n += 15; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + return -EINVAL; + } + offset += n; + offset = fw_parse_data(tas_fmw, &config->dev_data, + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +/* yram5 page check. */ +static bool check_inpage_yram_rg(struct tas_crc *cd, + unsigned char reg, unsigned char len) +{ + bool in = false; + + if (reg <= TAS2781_YRAM5_END_REG && + reg >= TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_END_REG) + cd->len = TAS2781_YRAM5_END_REG - reg + 1; + else + cd->len = len; + cd->offset = reg; + in = true; + } else if (reg < TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_START_REG) { + cd->offset = TAS2781_YRAM5_START_REG; + cd->len = len - TAS2781_YRAM5_START_REG + reg; + in = true; + } + } + + return in; +} + +/* DSP firmware yram block check. */ +static bool check_inpage_yram_bk1(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (page == TAS2781_YRAM1_PAGE) { + if (reg >= TAS2781_YRAM1_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg + len > TAS2781_YRAM1_START_REG) { + cd->offset = TAS2781_YRAM1_START_REG; + cd->len = len - TAS2781_YRAM1_START_REG + reg; + in = true; + } + } else if (page == TAS2781_YRAM3_PAGE) { + in = check_inpage_yram_rg(cd, reg, len); + } + + return in; +} + +/* + * Return Code: + * true -- the registers are in the inpage yram + * false -- the registers are NOT in the inpage yram + */ +static bool check_inpage_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1) + in = check_inpage_yram_bk1(cd, page, reg, len); + else if (book == TAS2781_YRAM_BOOK2 && page == TAS2781_YRAM5_PAGE) + in = check_inpage_yram_rg(cd, reg, len); + + return in; +} + +/* yram4 page check. */ +static bool check_inblock_yram_bk(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if ((page >= TAS2781_YRAM4_START_PAGE && + page <= TAS2781_YRAM4_END_PAGE) || + (page >= TAS2781_YRAM2_START_PAGE && + page <= TAS2781_YRAM2_END_PAGE)) { + if (reg <= TAS2781_YRAM2_END_REG && + reg >= TAS2781_YRAM2_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg < TAS2781_YRAM2_START_REG) { + if (reg + len - 1 >= TAS2781_YRAM2_START_REG) { + cd->offset = TAS2781_YRAM2_START_REG; + cd->len = reg + len - TAS2781_YRAM2_START_REG; + in = true; + } + } + } + + return in; +} + +/* + * Return Code: + * true -- the registers are in the inblock yram + * false -- the registers are NOT in the inblock yram + */ +static bool check_inblock_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1 || book == TAS2781_YRAM_BOOK2) + in = check_inblock_yram_bk(cd, page, reg, len); + + return in; +} + +/* yram page check. */ +static bool check_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in; + + in = check_inpage_yram(cd, book, page, reg, len); + if (!in) + in = check_inblock_yram(cd, book, page, reg, len); + + return in; +} + +/* Checksum for data block. */ +static int tasdev_multibytes_chksum(struct tasdevice_priv *tasdevice, + unsigned char book, unsigned char page, + unsigned char reg, unsigned int len) +{ + struct tas_crc crc_data; + unsigned char crc_chksum = 0; + unsigned char nBuf1[128]; + int ret = 0, i; + bool in; + + if ((reg + len - 1) > 127) { + ret = -EINVAL; + dev_err(tasdevice->dev, "firmware error\n"); + goto end; + } + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg == TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (len == 4)) { + /* DSP swap command, pass */ + ret = 0; + goto end; + } + + in = check_yram(&crc_data, book, page, reg, len); + if (!in) + goto end; + + if (len == 1) { + dev_err(tasdevice->dev, "firmware error\n"); + ret = -EINVAL; + goto end; + } + + ret = tasdevice_spi_dev_bulk_read(tasdevice, + TASDEVICE_REG(book, page, crc_data.offset), + nBuf1, crc_data.len); + if (ret < 0) + goto end; + + for (i = 0; i < crc_data.len; i++) { + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + ((i + crc_data.offset) >= + TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + ((i + crc_data.offset) <= + (TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG) + 4))) + /* DSP swap command, bypass */ + continue; + else + crc_chksum += crc8(tasdevice->crc8_lkp_tbl, &nBuf1[i], + 1, 0); + } + + ret = crc_chksum; + +end: + return ret; +} + +/* Checksum for single register. */ +static int do_singlereg_checksum(struct tasdevice_priv *tasdevice, + unsigned char book, unsigned char page, + unsigned char reg, unsigned char val) +{ + struct tas_crc crc_data; + unsigned int nData1; + int ret = 0; + bool in; + + /* DSP swap command, pass */ + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg >= TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg <= (TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG) + 4))) + return 0; + + in = check_yram(&crc_data, book, page, reg, 1); + if (!in) + return 0; + ret = tasdevice_spi_dev_read(tasdevice, + TASDEVICE_REG(book, page, reg), &nData1); + if (ret < 0) + return ret; + + if (nData1 != val) { + dev_err(tasdevice->dev, + "B[0x%x]P[0x%x]R[0x%x] W[0x%x], R[0x%x]\n", + book, page, reg, val, nData1); + tasdevice->err_code |= ERROR_YRAM_CRCCHK; + return -EAGAIN; + } + + ret = crc8(tasdevice->crc8_lkp_tbl, &val, 1, 0); + + return ret; +} + +/* Block type check. */ +static void set_err_prg_cfg(unsigned int type, struct tasdevice_priv *p) +{ + if ((type == MAIN_ALL_DEVICES) || (type == MAIN_DEVICE_A) || + (type == MAIN_DEVICE_B) || (type == MAIN_DEVICE_C) || + (type == MAIN_DEVICE_D)) + p->cur_prog = -1; + else + p->cur_conf = -1; +} + +/* Checksum for data bytes. */ +static int tasdev_bytes_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, unsigned char book, + unsigned char page, unsigned char reg, unsigned int len, + unsigned char val, unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) + ret = tasdev_multibytes_chksum(tas_priv, book, page, reg, + len); + else + ret = do_singlereg_checksum(tas_priv, book, page, reg, val); + + if (ret > 0) { + *crc_chksum += ret; + goto end; + } + + if (ret != -EAGAIN) + goto end; + + block->nr_retry--; + if (block->nr_retry > 0) + goto end; + + set_err_prg_cfg(block->type, tas_priv); + +end: + return ret; +} + +/* Multi-data byte write. */ +static int tasdev_multibytes_wr(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, unsigned char book, + unsigned char page, unsigned char reg, unsigned char *data, + unsigned int len, unsigned int *nr_cmds, + unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) { + ret = tasdevice_spi_dev_bulk_write(tas_priv, + TASDEVICE_REG(book, page, reg), data + 3, len); + if (ret < 0) + return ret; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, + book, page, reg, len, 0, crc_chksum); + } else { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(book, page, reg), data[3]); + if (ret < 0) + return ret; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, book, + page, reg, 1, data[3], crc_chksum); + } + + if (!block->is_ychksum_present || ret >= 0) { + *nr_cmds += 1; + if (len >= 2) + *nr_cmds += ((len - 2) / 4) + 1; + } + + return ret; +} + +/* Checksum for block. */ +static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + unsigned int nr_value; + int ret; + + ret = tasdevice_spi_dev_read(tas_priv, TASDEVICE_CHECKSUM, &nr_value); + if (ret < 0) { + dev_err(tas_priv->dev, "%s: read error %d.\n", __func__, ret); + set_err_prg_cfg(block->type, tas_priv); + return ret; + } + + if ((nr_value & 0xff) != block->pchksum) { + dev_err(tas_priv->dev, "%s: PChkSum err %d ", __func__, ret); + dev_err(tas_priv->dev, "PChkSum = 0x%x, Reg = 0x%x\n", + block->pchksum, (nr_value & 0xff)); + tas_priv->err_code |= ERROR_PRAM_CRCCHK; + ret = -EAGAIN; + block->nr_retry--; + + if (block->nr_retry <= 0) + set_err_prg_cfg(block->type, tas_priv); + } else { + tas_priv->err_code &= ~ERROR_PRAM_CRCCHK; + } + + return ret; +} + +/* Firmware block load function. */ +static int tasdev_load_blk(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + unsigned int sleep_time, len, nr_cmds; + unsigned char offset, book, page, val; + unsigned char *data = block->data; + unsigned char crc_chksum = 0; + int ret = 0; + + while (block->nr_retry > 0) { + if (block->is_pchksum_present) { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_CHECKSUM, 0); + if (ret < 0) + break; + } + + if (block->is_ychksum_present) + crc_chksum = 0; + + nr_cmds = 0; + + while (nr_cmds < block->nr_cmds) { + data = block->data + nr_cmds * 4; + + book = data[0]; + page = data[1]; + offset = data[2]; + val = data[3]; + + nr_cmds++; + /* Single byte write */ + if (offset <= 0x7F) { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(book, page, offset), + val); + if (ret < 0) + break; + if (block->is_ychksum_present) { + ret = tasdev_bytes_chksum(tas_priv, + block, book, page, offset, + 1, val, &crc_chksum); + if (ret < 0) + break; + } + continue; + } + /* sleep command */ + if (offset == 0x81) { + /* book -- data[0] page -- data[1] */ + sleep_time = ((book << 8) + page)*1000; + fsleep(sleep_time); + continue; + } + /* Multiple bytes write */ + if (offset == 0x85) { + data += 4; + len = (book << 8) + page; + book = data[0]; + page = data[1]; + offset = data[2]; + ret = tasdev_multibytes_wr(tas_priv, + block, book, page, offset, data, + len, &nr_cmds, &crc_chksum); + if (ret < 0) + break; + } + } + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) { + /* err in current device, skip it */ + break; + } + + if (block->is_pchksum_present) { + ret = tasdev_block_chksum(tas_priv, block); + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) { + /* err in current device, skip it */ + break; + } + } + + if (block->is_ychksum_present) { + /* TBD, open it when FW ready */ + dev_err(tas_priv->dev, + "Blk YChkSum: FW = 0x%x, YCRC = 0x%x\n", + block->ychksum, crc_chksum); + + tas_priv->err_code &= + ~ERROR_YRAM_CRCCHK; + ret = 0; + } + /* skip current blk */ + break; + } + + return ret; +} + +/* Firmware block load function. */ +static int tasdevice_load_block(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + int ret = 0; + + block->nr_retry = 6; + if (tas_priv->is_loading == false) + return 0; + ret = tasdev_load_blk(tas_priv, block); + if (ret < 0) + dev_err(tas_priv->dev, "Blk (%d) load error\n", block->type); + + return ret; +} + +/* + * Select firmware binary parser & load callback functions by ppc3 version + * and firmware binary version. + */ +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) { + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + } else if (ppcver == 0x00) { + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + } else { + dev_err(tas_priv->dev, + "Wrong PPCVer :0x%08x\n", ppcver); + rc = -EINVAL; + } + } else { + dev_err(tas_priv->dev, "Wrong DrvVer : 0x%02x\n", drv_ver); + rc = -EINVAL; + } + + return rc; +} + +/* DSP firmware binary file header parser function. */ +static int fw_parse_header(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &fw_hdr->fixed_hdr; + static const unsigned char magic_number[] = {0x35, 0x35, 0x35, 0x32, }; + const unsigned char *buf = (unsigned char *)fmw->data; + + if (offset + 92 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + if (memcmp(&buf[offset], magic_number, 4)) { + dev_err(tas_priv->dev, "%s: Magic num NOT match\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 4; + + /* + * Convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + fw_fixed_hdr->fwsize = get_unaligned_be32(&buf[offset]); + offset += 4; + if (fw_fixed_hdr->fwsize != fmw->size) { + dev_err(tas_priv->dev, "File size not match, %lu %u", + (unsigned long)fmw->size, fw_fixed_hdr->fwsize); + offset = -EINVAL; + goto out; + } + offset += 4; + fw_fixed_hdr->ppcver = get_unaligned_be32(&buf[offset]); + offset += 8; + fw_fixed_hdr->drv_ver = get_unaligned_be32(&buf[offset]); + offset += 72; + +out: + return offset; +} + +/* DSP firmware binary file parser function. */ +static int tasdevice_dspfw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr; + struct tasdevice_fw *tas_fmw; + int offset = 0, ret = 0; + + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "%s: Failed to read firmware %s\n", + __func__, tas_priv->coef_binaryname); + return -EINVAL; + } + + tas_priv->fmw = kzalloc(sizeof(*tas_priv->fmw), GFP_KERNEL); + 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) + return -EINVAL; + + fw_fixed_hdr = &tas_fmw->fw_hdr.fixed_hdr; + /* Support different versions of firmware */ + switch (fw_fixed_hdr->drv_ver) { + case 0x301: + case 0x302: + case 0x502: + case 0x503: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + break; + case 0x202: + case 0x400: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + break; + default: + ret = dspfw_default_callback(tas_priv, + fw_fixed_hdr->drv_ver, fw_fixed_hdr->ppcver); + if (ret) + return ret; + break; + } + + offset = tas_priv->fw_parse_variable_header(tas_priv, fmw, offset); + if (offset < 0) + return offset; + + offset = tas_priv->fw_parse_program_data(tas_priv, tas_fmw, fmw, + offset); + if (offset < 0) + return offset; + + offset = tas_priv->fw_parse_configuration_data(tas_priv, + tas_fmw, fmw, offset); + if (offset < 0) + ret = offset; + + return ret; +} + +/* DSP firmware binary file parser function. */ +int tasdevice_spi_dsp_parser(void *context) +{ + struct tasdevice_priv *tas_priv = context; + const struct firmware *fw_entry; + int ret; + + ret = request_firmware(&fw_entry, tas_priv->coef_binaryname, + tas_priv->dev); + if (ret) { + dev_err(tas_priv->dev, "%s: load %s error\n", __func__, + tas_priv->coef_binaryname); + return ret; + } + + ret = tasdevice_dspfw_ready(fw_entry, tas_priv); + release_firmware(fw_entry); + fw_entry = NULL; + + return ret; +} + +/* DSP firmware program block data remove function. */ +static void tasdev_dsp_prog_blk_remove(struct tasdevice_prog *prog) +{ + struct tasdevice_data *tas_dt; + struct tasdev_blk *blk; + unsigned int i; + + if (!prog) + return; + + tas_dt = &prog->dev_data; + + if (!tas_dt->dev_blks) + return; + + for (i = 0; i < tas_dt->nr_blk; i++) { + blk = &tas_dt->dev_blks[i]; + kfree(blk->data); + } + kfree(tas_dt->dev_blks); +} + +/* DSP firmware program block data remove function. */ +static void tasdev_dsp_prog_remove(struct tasdevice_prog *prog, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_prog_blk_remove(&prog[i]); + kfree(prog); +} + +/* DSP firmware config block data remove function. */ +static void tasdev_dsp_cfg_blk_remove(struct tasdevice_config *cfg) +{ + struct tasdevice_data *tas_dt; + struct tasdev_blk *blk; + unsigned int i; + + if (cfg) { + tas_dt = &cfg->dev_data; + + if (!tas_dt->dev_blks) + return; + + for (i = 0; i < tas_dt->nr_blk; i++) { + blk = &tas_dt->dev_blks[i]; + kfree(blk->data); + } + kfree(tas_dt->dev_blks); + } +} + +/* DSP firmware config remove function. */ +static void tasdev_dsp_cfg_remove(struct tasdevice_config *config, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_cfg_blk_remove(&config[i]); + kfree(config); +} + +/* DSP firmware remove function. */ +void tasdevice_spi_dsp_remove(void *context) +{ + struct tasdevice_priv *tas_dev = context; + + if (!tas_dev->fmw) + return; + + if (tas_dev->fmw->programs) + tasdev_dsp_prog_remove(tas_dev->fmw->programs, + tas_dev->fmw->nr_programs); + if (tas_dev->fmw->configs) + tasdev_dsp_cfg_remove(tas_dev->fmw->configs, + tas_dev->fmw->nr_configurations); + kfree(tas_dev->fmw); + tas_dev->fmw = NULL; +} + +/* DSP firmware calibration data remove function. */ +static void tas2781_clear_calfirmware(struct tasdevice_fw *tas_fmw) +{ + struct tasdevice_calibration *calibration; + struct tasdev_blk *block; + unsigned int blks; + int i; + + if (!tas_fmw->calibrations) + goto out; + + for (i = 0; i < tas_fmw->nr_calibrations; i++) { + calibration = &tas_fmw->calibrations[i]; + if (!calibration) + continue; + + if (!calibration->dev_data.dev_blks) + continue; + + for (blks = 0; blks < calibration->dev_data.nr_blk; blks++) { + block = &calibration->dev_data.dev_blks[blks]; + if (!block) + continue; + kfree(block->data); + } + kfree(calibration->dev_data.dev_blks); + } + kfree(tas_fmw->calibrations); +out: + kfree(tas_fmw); +} + +/* Calibration data from firmware remove function. */ +void tasdevice_spi_calbin_remove(void *context) +{ + struct tasdevice_priv *tas_priv = context; + + if (tas_priv->cali_data_fmw) { + tas2781_clear_calfirmware(tas_priv->cali_data_fmw); + tas_priv->cali_data_fmw = NULL; + } +} + +/* Configuration remove function. */ +void tasdevice_spi_config_info_remove(void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_rca *rca = &tas_priv->rcabin; + struct tasdevice_config_info **ci = rca->cfg_info; + unsigned int i, j; + + if (!ci) + return; + for (i = 0; i < rca->ncfgs; i++) { + if (!ci[i]) + continue; + if (ci[i]->blk_data) { + for (j = 0; j < ci[i]->real_nblocks; j++) { + if (!ci[i]->blk_data[j]) + continue; + kfree(ci[i]->blk_data[j]->regdata); + kfree(ci[i]->blk_data[j]); + } + kfree(ci[i]->blk_data); + } + kfree(ci[i]); + } + kfree(ci); +} + +/* DSP firmware program block data load function. */ +static int tasdevice_load_data(struct tasdevice_priv *tas_priv, + struct tasdevice_data *dev_data) +{ + struct tasdev_blk *block; + unsigned int i; + int ret = 0; + + for (i = 0; i < dev_data->nr_blk; i++) { + block = &dev_data->dev_blks[i]; + ret = tas_priv->tasdevice_load_block(tas_priv, block); + if (ret < 0) + break; + } + + return ret; +} + +/* DSP firmware program load interface function. */ +int tasdevice_spi_prmg_load(void *context, int prm_no) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + struct tasdevice_config *conf; + int ret = 0; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + return -EINVAL; + } + if (prm_no >= 0 && prm_no <= tas_fmw->nr_programs) { + tas_priv->cur_conf = 0; + tas_priv->is_loading = true; + program = &tas_fmw->programs[prm_no]; + ret = tasdevice_load_data(tas_priv, &program->dev_data); + if (ret < 0) { + dev_err(tas_priv->dev, "Program failed %d.\n", ret); + return ret; + } + tas_priv->cur_prog = prm_no; + + conf = &tas_fmw->configs[tas_priv->cur_conf]; + ret = tasdevice_load_data(tas_priv, &conf->dev_data); + if (ret < 0) + dev_err(tas_priv->dev, "Config failed %d.\n", ret); + } else { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + return -EINVAL; + } + + return ret; +} + +/* RCABIN configuration switch interface function. */ +void tasdevice_spi_tuning_switch(void *context, int state) +{ + struct tasdevice_priv *tas_priv = context; + int profile_cfg_id = tas_priv->rcabin.profile_cfg_id; + + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + return; + } + + if (state == 0) + tasdevice_spi_select_cfg_blk(tas_priv, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + else + tasdevice_spi_select_cfg_blk(tas_priv, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN); +} -- 2.40.1

3 2

[PATCH] ASoC: topology: Fix route memory corruption
by Amadeusz Sławiński 15 Aug '24

15 Aug '24

It was reported that recent fix for memory corruption during topology load, causes corruption in other cases. Instead of being overeager with checking topology, assume that it is properly formatted and just duplicate strings. Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Closes: https://lore.kernel.org/linux-sound/171812236450.201359.3019210915105428447… Suggested-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 0225bc5fc425a..4b166294602fa 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1052,21 +1052,15 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, break; } - route->source = devm_kmemdup(tplg->dev, elem->source, - min(strlen(elem->source), maxlen), - GFP_KERNEL); - route->sink = devm_kmemdup(tplg->dev, elem->sink, - min(strlen(elem->sink), maxlen), - GFP_KERNEL); + route->source = devm_kstrdup(tplg->dev, elem->source, GFP_KERNEL); + route->sink = devm_kstrdup(tplg->dev, elem->sink, GFP_KERNEL); if (!route->source || !route->sink) { ret = -ENOMEM; break; } if (strnlen(elem->control, maxlen) != 0) { - route->control = devm_kmemdup(tplg->dev, elem->control, - min(strlen(elem->control), maxlen), - GFP_KERNEL); + route->control = devm_kstrdup(tplg->dev, elem->control, GFP_KERNEL); if (!route->control) { ret = -ENOMEM; break; -- 2.34.1

5 6

[PATCH v1] ASoC: codecs: ES8326: Adjust buttons
by Zhang Yi 15 Aug '24

15 Aug '24

> > To adapt to chrome, we have adjusted the buttons to match the system. > > This needs a bit more of a commit log explaining what's going on. > > > +#if IS_ENABLED(CONFIG_SND_SOC_MT8188_MT6359) > > + const int hs_keys[] = { > > + SND_JACK_BTN_0, SND_JACK_BTN_2, SND_JACK_BTN_3 > > + }; > > +#else > > + const int hs_keys[] = { > > + SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2 > > + }; > > +#endif > > This will remap the buttons no matter what system we're running on so > while it's a bit better it's still going to cause issues. We could use > a per board approach like this but it needs to be at runtime, based on > the machine compatible rather than a build time selection. > > The support for this device was added in e794a894427b1d ("ASoC: > mediatek: mt8188-mt6359: add es8326 support") which was added in v6.8 > so this would also be a potential issue in mainline, given that this is > likely only used by Chromebooks and didn't make it to a LTS yet we could > possibly give it a go but we might have to revert. Because this change is to adapt mt8188-mt6359.On mt8188-mt6359, SND_JACK_BTN_2 corresponds to KEY_VOLUMEUP and SND_JACK_BTN_3 corresponds to KEY_VOLUMEDOWN. So we modified the es8326 driver to adapt the mt8188-mt6359. The changes we are making will not affect the application on all platforms.

1 0

[PATCH] ASoC: soc-pcm: remove snd_soc_dpcm_stream_{lock/unlock}_irq()
by Kuninori Morimoto 14 Aug '24

14 Aug '24

soc-pcm.c has snd_soc_dpcm_stream_{lock/unlock}_irq() helper function, but it is almost nothing help. It just makes a code complex. Let's remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 25 +++++++------------------ 1 file changed, 7 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index bad823718ae47..58381801e41d4 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -49,21 +49,9 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, return ret; } -static inline void snd_soc_dpcm_stream_lock_irq(struct snd_soc_pcm_runtime *rtd, - int stream) -{ - snd_pcm_stream_lock_irq(snd_soc_dpcm_get_substream(rtd, stream)); -} - #define snd_soc_dpcm_stream_lock_irqsave_nested(rtd, stream, flags) \ snd_pcm_stream_lock_irqsave_nested(snd_soc_dpcm_get_substream(rtd, stream), flags) -static inline void snd_soc_dpcm_stream_unlock_irq(struct snd_soc_pcm_runtime *rtd, - int stream) -{ - snd_pcm_stream_unlock_irq(snd_soc_dpcm_get_substream(rtd, stream)); -} - #define snd_soc_dpcm_stream_unlock_irqrestore(rtd, stream, flags) \ snd_pcm_stream_unlock_irqrestore(snd_soc_dpcm_get_substream(rtd, stream), flags) @@ -260,14 +248,14 @@ static void dpcm_set_fe_update_state(struct snd_soc_pcm_runtime *fe, struct snd_pcm_substream *substream = snd_soc_dpcm_get_substream(fe, stream); - snd_soc_dpcm_stream_lock_irq(fe, stream); + snd_pcm_stream_lock_irq(substream); if (state == SND_SOC_DPCM_UPDATE_NO && fe->dpcm[stream].trigger_pending) { dpcm_fe_dai_do_trigger(substream, fe->dpcm[stream].trigger_pending - 1); fe->dpcm[stream].trigger_pending = 0; } fe->dpcm[stream].runtime_update = state; - snd_soc_dpcm_stream_unlock_irq(fe, stream); + snd_pcm_stream_unlock_irq(substream); } static void dpcm_set_be_update_state(struct snd_soc_pcm_runtime *be, @@ -1272,10 +1260,10 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, dpcm->be = be; dpcm->fe = fe; dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW; - snd_soc_dpcm_stream_lock_irq(fe, stream); + snd_pcm_stream_lock_irq(fe_substream); list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); - snd_soc_dpcm_stream_unlock_irq(fe, stream); + snd_pcm_stream_unlock_irq(fe_substream); dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, @@ -1320,11 +1308,12 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; + struct snd_pcm_substream *substream = snd_soc_dpcm_get_substream(fe, stream); LIST_HEAD(deleted_dpcms); snd_soc_dpcm_mutex_assert_held(fe); - snd_soc_dpcm_stream_lock_irq(fe, stream); + snd_pcm_stream_lock_irq(substream); for_each_dpcm_be_safe(fe, stream, dpcm, d) { dev_dbg(fe->dev, "ASoC: BE %s disconnect check for %s\n", stream ? "capture" : "playback", @@ -1343,7 +1332,7 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) list_del(&dpcm->list_be); list_move(&dpcm->list_fe, &deleted_dpcms); } - snd_soc_dpcm_stream_unlock_irq(fe, stream); + snd_pcm_stream_unlock_irq(substream); while (!list_empty(&deleted_dpcms)) { dpcm = list_first_entry(&deleted_dpcms, struct snd_soc_dpcm, -- 2.43.0

2 1

[PATCH 1/7] ASoC: fsl_audmix: Switch to RUNTIME/SYSTEM_SLEEP_PM_OPS()
by Fabio Estevam 14 Aug '24

14 Aug '24

From: Fabio Estevam <festevam(a)denx.de> Replace SET_RUNTIME_PM_OPS()/SET SYSTEM_SLEEP_PM_OPS() with their modern RUNTIME_PM_OPS() and SYSTEM_SLEEP_PM_OPS() alternatives. The combined usage of pm_ptr() and RUNTIME_PM_OPS/SYSTEM_SLEEP_PM_OPS() allows the compiler to evaluate if the runtime suspend/resume() functions are used at build time or are simply dead code. This allows removing the CONFIG_PM ifdefery from the runtime suspend/resume() functions. Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- sound/soc/fsl/fsl_audmix.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index 1671a3037c60..f3a24758aedb 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -512,7 +512,6 @@ static void fsl_audmix_remove(struct platform_device *pdev) platform_device_unregister(priv->pdev); } -#ifdef CONFIG_PM static int fsl_audmix_runtime_resume(struct device *dev) { struct fsl_audmix *priv = dev_get_drvdata(dev); @@ -540,14 +539,11 @@ static int fsl_audmix_runtime_suspend(struct device *dev) return 0; } -#endif /* CONFIG_PM */ static const struct dev_pm_ops fsl_audmix_pm = { - SET_RUNTIME_PM_OPS(fsl_audmix_runtime_suspend, - fsl_audmix_runtime_resume, - NULL) - SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, - pm_runtime_force_resume) + RUNTIME_PM_OPS(fsl_audmix_runtime_suspend, fsl_audmix_runtime_resume, + NULL) + SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) }; static struct platform_driver fsl_audmix_driver = { @@ -556,7 +552,7 @@ static struct platform_driver fsl_audmix_driver = { .driver = { .name = "fsl-audmix", .of_match_table = fsl_audmix_ids, - .pm = &fsl_audmix_pm, + .pm = pm_ptr(&fsl_audmix_pm), }, }; module_platform_driver(fsl_audmix_driver); -- 2.34.1

2 7

Re: [PATCH] ASoC: sti: add missing probe entry for player and reader
by Mark Brown 14 Aug '24

14 Aug '24

On Mon, Jul 22, 2024 at 02:31:24PM +0200, jau(a)free.fr wrote: > Thanks for the review! > > See updated changlog, if ok for you, I will send a V2 Yes, that's fine.

1 0

[PATCH v1] ASoc: tas2781: Add Calibration Kcontrols for Chromebook
by Shenghao Ding 14 Aug '24

14 Aug '24

Add calibration related kcontrol for speaker impedance calibration and speaker leakage check for Chromebook Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Add TAS2563 into the Header of tas2781-comlib.c - Add channel no into the log for error debug - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/202407131… - Create tas2563-tlv.h into include/sound - Move tas2563_dvc_table from tas2781-tlv.h to tas2563-tlv.h - Add "#include <sound/tas2563-tlv.h>" into tas2781-i2c.c - Correct the filename in the header comments of tas2781-comlib.c, remove the wrong file name. - renamae tasdevice_chn_switch to put it in a single line - With inverted check this entire function becomes neater in tasdev_chn_switch. - With all possible restrictions this can be on a single line besides the fact that the second one currently has a broken indentation. - remove unnecessary parentheses. - linux/* followed by asm/* as the latter is not so generic as the former. - leave the trailing comma - replace scnprintf with strscpy if necessary. - Add registers for TAS2563 and TAS2781 calibration. - Add cali_data_restore for regsiter restore after calibration. - Add is_user_space_calidata to store the flag where the calibrated from, user space or bin file from driver parsing. - Add TASDEVICE_RCA_FW_OK to support only register setting bin in the device. - Add tas2563_dvc_table for relationship between tas2563 digtial gains and register values. - tasdevice_chn_switch for chip switch among multiple chips(tas2563 or tas2781) - Add loading the calibrated values from user space in tasdev_load_calibrated_data - Correct no dsp no work, it can still work in bypass mode. - Add calibrated register setting for tas2563&tas2781. - Add mutex into each kcontrol. - Rename tas2781_force_fwload_get, tas2781_force_fwload_put--> tasdev_force_fwload_get, tasdev_force_fwload_put - Define tas2781_snd_control and tas2563_snd_control for volume - Define tas2781_cali_control and tas2563_cali_control for calibration --- include/sound/tas2563-tlv.h | 279 +++++++++ include/sound/tas2781-tlv.h | 260 --------- include/sound/tas2781.h | 72 +++ sound/soc/codecs/tas2781-comlib.c | 32 +- sound/soc/codecs/tas2781-fmwlib.c | 39 +- sound/soc/codecs/tas2781-i2c.c | 907 ++++++++++++++++++++++++++++-- 6 files changed, 1267 insertions(+), 322 deletions(-) create mode 100644 include/sound/tas2563-tlv.h diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h new file mode 100644 index 000000000000..2dab81a4c359 --- /dev/null +++ b/include/sound/tas2563-tlv.h @@ -0,0 +1,279 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2563 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2563 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2563 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// + +#ifndef __TAS2563_TLV_H__ +#define __TAS2563_TLV_H__ + +static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); + +/* pow(10, db/20) * pow(2,30) */ +static const unsigned char tas2563_dvc_table[][4] = { + { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ + { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ + { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ + { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ + { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ + { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ + { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ + { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ + { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ + { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ + { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ + { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ + { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ + { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ + { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ + { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ + { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ + { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ + { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ + { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ + { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ + { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ + { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ + { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ + { 0X7F, 0XFF, 0XFF, 0XFF }, /* 6.0db */ +}; +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 99c41bfc7827..5b003843752b 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -17,265 +17,5 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); -static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); -/* pow(10, db/20) * pow(2,30) */ -static const unsigned char tas2563_dvc_table[][4] = { - { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ - { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ - { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ - { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ - { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ - { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ - { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ - { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ - { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ - { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ - { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ - { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ - { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ - { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ - { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ - { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ - { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ - { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ - { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ - { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ - { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ - { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ - { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ - { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ - { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ - { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ - { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ - { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ - { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ - { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ - { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ - { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ - { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ - { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ - { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ - { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ - { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ - { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ - { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ - { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ - { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ - { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ - { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ - { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ - { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ - { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ - { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ - { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ - { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ - { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ - { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ - { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ - { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ - { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ - { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ - { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ - { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ - { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ - { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ - { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ - { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ - { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ - { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ - { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ - { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ - { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ - { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ - { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ - { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ - { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ - { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ - { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ - { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ - { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ - { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ - { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ - { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ - { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ - { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ - { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ - { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ - { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ - { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ - { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ - { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ - { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ - { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ - { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ - { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ - { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ - { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ - { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ - { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ - { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ - { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ - { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ - { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ - { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ - { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ - { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ - { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ - { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ - { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ - { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ - { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ - { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ - { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ - { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ - { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ - { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ - { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ - { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ - { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ - { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ - { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ - { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ - { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ - { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ - { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ - { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ - { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ - { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ - { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ - { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ - { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ - { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ - { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ - { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ - { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ - { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ - { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ - { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ - { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ - { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ - { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ - { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ - { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ - { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ - { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ - { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ - { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ - { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ - { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ - { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ - { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ - { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ - { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ - { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ - { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ - { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ - { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ - { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ - { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ - { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ - { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ - { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ - { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ - { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ - { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ - { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ - { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ - { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ - { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ - { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ - { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ - { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ - { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ - { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ - { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ - { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ - { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ - { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ - { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ - { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ - { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ - { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ - { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ - { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ - { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ - { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ - { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ - { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ - { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ - { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ - { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ - { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ - { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ - { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ - { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ - { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ - { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ - { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ - { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ - { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ - { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ - { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ - { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ - { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ - { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ - { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ - { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ - { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ - { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ - { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ - { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ - { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ - { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ - { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ - { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ - { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ - { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ - { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ - { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ - { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ - { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ - { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ - { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ - { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ - { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ - { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ - { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ - { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ - { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ - { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ - { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ - { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ - { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ - { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ - { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ - { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ - { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ - { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ - { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ - { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ - { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ - { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ - { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ - { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ - { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ - { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ - { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ - { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ - { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ - { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ - { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ - { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ - { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ - { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ - { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ - { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ - { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ - { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ - { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ - { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ - { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ - { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ -}; #endif diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 18161d02a96f..81fb34c9da89 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -23,6 +23,7 @@ #define SMARTAMP_MODULE_NAME "tas2781" #define TAS2781_GLOBAL_ADDR 0x40 #define TAS2563_GLOBAL_ADDR 0x48 + #define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ SNDRV_PCM_RATE_88200) @@ -49,12 +50,69 @@ /*I2C Checksum */ #define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) +/* XM_340 */ +#define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x02, 0x4c) +/* XM_341 */ +#define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x02, 0x64) + /* Volume control */ #define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) #define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) #define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) +#define TAS2563_IDLE TASDEVICE_REG(0x00, 0x00, 0x3e) +#define TAS2563_PRM_R0_REG TASDEVICE_REG(0x00, 0x0f, 0x34) +#define TAS2563_PRM_R0_LOW_REG TASDEVICE_REG(0x00, 0x0f, 0x48) +#define TAS2563_PRM_INVR0_REG TASDEVICE_REG(0x00, 0x0f, 0x40) +#define TAS2563_PRM_POW_REG TASDEVICE_REG(0x00, 0x0d, 0x3c) +#define TAS2563_PRM_TLIMIT_REG TASDEVICE_REG(0x00, 0x10, 0x14) + +#define TAS2563_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x02, 0x70) +#define TAS2563_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x02, 0x48) + +#define TAS2563_PRM_ENFF_REG TASDEVICE_REG(0x00, 0x0d, 0x54) +#define TAS2563_PRM_DISTCK_REG TASDEVICE_REG(0x00, 0x0d, 0x58) +#define TAS2563_PRM_TE_SCTHR_REG TASDEVICE_REG(0x00, 0x0f, 0x60) +#define TAS2563_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x0d, 0x74) +#define TAS2563_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x0d, 0x7c) +/* prm_Int_B0 */ +#define TAS2563_TE_TA1_REG TASDEVICE_REG(0x00, 0x10, 0x0c) +/* prm_Int_A1 */ +#define TAS2563_TE_TA1_AT_REG TASDEVICE_REG(0x00, 0x10, 0x10) +/* prm_TE_Beta */ +#define TAS2563_TE_TA2_REG TASDEVICE_REG(0x00, 0x0f, 0x64) +/* prm_TE_Beta1 */ +#define TAS2563_TE_AT_REG TASDEVICE_REG(0x00, 0x0f, 0x68) +/* prm_TE_1_Beta1 */ +#define TAS2563_TE_DT_REG TASDEVICE_REG(0x00, 0x0f, 0x70) + +#define TAS2781_PRM_R0_REG TASDEVICE_REG(0x00, 0x17, 0x74) +#define TAS2781_PRM_R0_LOW_REG TASDEVICE_REG(0x00, 0x18, 0x0c) +#define TAS2781_PRM_INVR0_REG TASDEVICE_REG(0x00, 0x18, 0x14) +#define TAS2781_PRM_POW_REG TASDEVICE_REG(0x00, 0x13, 0x70) +#define TAS2781_PRM_TLIMIT_REG TASDEVICE_REG(0x00, 0x18, 0x7c) + +#define TAS2781_PRM_INT_MASK_REG TASDEVICE_REG(0x00, 0x00, 0x3b) +#define TAS2781_PRM_CLK_CFG_REG TASDEVICE_REG(0x00, 0x00, 0x5c) +#define TAS2781_PRM_RSVD_REG TASDEVICE_REG(0x00, 0x01, 0x19) +#define TAS2781_PRM_TEST_57_REG TASDEVICE_REG(0x00, 0xfd, 0x39) +#define TAS2781_PRM_TEST_62_REG TASDEVICE_REG(0x00, 0xfd, 0x3e) +#define TAS2781_PRM_PVDD_UVLO_REG TASDEVICE_REG(0x00, 0x00, 0x71) +#define TAS2781_PRM_CHNL_0_REG TASDEVICE_REG(0x00, 0x00, 0x03) +#define TAS2781_PRM_NG_CFG0_REG TASDEVICE_REG(0x00, 0x00, 0x35) +#define TAS2781_PRM_IDLE_CH_DET_REG TASDEVICE_REG(0x00, 0x00, 0x66) +#define TAS2781_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x14, 0x38) +#define TAS2781_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x14, 0x40) +#define TAS2781_PRM_SINEGAIN2_REG TASDEVICE_REG(0x00, 0x14, 0x44) + +#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xFD, 0x0D) +#define TAS2781_TEST_PAGE_UNLOCK 0x0D + +#define TAS2781_RUNTIME_LATCH_RE_REG TASDEVICE_REG(0x00, 0x00, 0x49) +#define TAS2781_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x62, 0x48) +#define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) + #define TASDEVICE_CMD_SING_W 0x1 #define TASDEVICE_CMD_BURST 0x2 #define TASDEVICE_CMD_DELAY 0x3 @@ -70,7 +128,15 @@ enum device_catlog_id { OTHERS }; +struct bulk_reg_val { + int reg; + unsigned char val[4]; + unsigned char val_len; + bool is_locked; +}; + struct tasdevice { + struct bulk_reg_val *cali_data_restore; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -89,6 +155,9 @@ struct tasdevice_irqinfo { struct calidata { unsigned char *data; unsigned long total_sz; + unsigned int *reg_array; + unsigned int reg_array_sz; + unsigned int cali_dat_sz; }; struct tasdevice_priv { @@ -124,6 +193,7 @@ struct tasdevice_priv { bool force_fwload_status; bool playback_started; bool isacpi; + bool is_user_space_calidata; unsigned int global_addr; int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, @@ -150,6 +220,8 @@ int tasdevice_init(struct tasdevice_priv *tas_priv); void tasdevice_remove(struct tasdevice_priv *tas_priv); int tasdevice_save_calibration(struct tasdevice_priv *tas_priv); void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv); +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn); int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *value); int tasdevice_dev_write(struct tasdevice_priv *tas_priv, diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 1fbf4560f5cc..e7599f9fc125 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// TAS2781 Common functions for HDA and ASoC Audio drivers +// TAS2563/TAS2781 Common functions for HDA and ASoC Audio drivers // // Copyright 2023 - 2024 Texas Instruments, Inc. // @@ -64,8 +64,8 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, */ ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); if (ret < 0) { - dev_err(tas_priv->dev, "%s, E=%d\n", - __func__, ret); + dev_err(tas_priv->dev, "%s, E=%d channel:%d\n", + __func__, ret, chn); goto out; } } @@ -89,6 +89,32 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, return ret; } +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + struct regmap *map = tas_priv->regmap; + int ret; + + if (client->addr != tasdev->dev_addr) { + client->addr = tasdev->dev_addr; + /* All devices share the same regmap, clear the page + * inside regmap once switching to another device. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + return ret; + } + return 1; + } + return 0; +} +EXPORT_SYMBOL_GPL(tasdev_chn_switch); + int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *val) { diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 63626b982d04..2b5eb67a52e8 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2152,20 +2152,38 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) { + struct tasdevice_fw *cal_fmw = priv->tasdevice[i].cali_data_fmw; + struct calidata *cali_data = &priv->cali_data; + unsigned char *data = &cali_data->data[1]; struct tasdevice_calibration *cal; - struct tasdevice_fw *cal_fmw; + int k = i * (cali_data->cali_dat_sz + 1); + int j, rc; - cal_fmw = priv->tasdevice[i].cali_data_fmw; + /* Load the calibrated data from cal bin file */ + if (!priv->is_user_space_calidata && cal_fmw) { + cal = cal_fmw->calibrations; - /* No calibrated data for current devices, playback will go ahead. */ - if (!cal_fmw) + if (cal) + load_calib_data(priv, &cal->dev_data); return; - - cal = cal_fmw->calibrations; - if (cal) + } + if (!priv->is_user_space_calidata) return; - - load_calib_data(priv, &cal->dev_data); + /* load calibrated data from user space */ + if (data[k] != i) { + dev_err(priv->dev, "%s: no cal-data for dev %d from usr-spc\n", + __func__, i); + return; + } + k++; + for (j = 0; j < cali_data->reg_array_sz; j++) { + rc = tasdevice_dev_bulk_write(priv, i, cali_data->reg_array[j], + &(data[k + 4 * j]), 4); + if (rc < 0) + dev_err(priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + i, j, rc); + } } int tasdevice_select_tuningprm_cfg(void *context, int prm_no, @@ -2260,9 +2278,10 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, tas_priv->tasdevice[i].cur_conf = cfg_no; } } - } else + } else { dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", __func__, cfg_no); + } status |= cfg_info[rca_conf_no]->active_dev; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e79d613745b4..e36934c8eeb2 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,9 +30,141 @@ #include <sound/soc.h> #include <sound/tas2781.h> #include <sound/tlv.h> +#include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> #include <asm/unaligned.h> +static const struct bulk_reg_val tas2563_cali_start_reg[] = { + { + .reg = TAS2563_IDLE, + .val_len = 1, + .val = { 0x00 }, + }, + { + .reg = TAS2563_PRM_ENFF_REG, + .val_len = 4, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_DISTCK_REG, + .val_len = 4, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_TE_SCTHR_REG, + .val_len = 4, + .val = { 0x7f, 0xff, 0xff, 0xff }, + }, + { + .reg = TAS2563_PRM_PLT_FLAG_REG, + .val_len = 4, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_SINEGAIN_REG, + .val_len = 4, + .val = { 0x0a, 0x3d, 0x70, 0xa4 }, + }, + { + .reg = TAS2563_TE_TA1_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_TA1_AT_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_TA2_REG, + .val_len = 4, + .val = { 0x00, 0x06, 0xd3, 0x72 }, + }, + { + .reg = TAS2563_TE_AT_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_DT_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, +}; + +static const struct bulk_reg_val tas2781_cali_start_reg[] = { + { + .reg = TAS2781_PRM_INT_MASK_REG, + .val = { 0xfe }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_CLK_CFG_REG, + .val = { 0xdd }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_RSVD_REG, + .val = { 0x20 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_TEST_57_REG, + .val = { 0x14 }, + .val_len = 1, + .is_locked = true + }, + { + .reg = TAS2781_PRM_TEST_62_REG, + .val = { 0x45 }, + .val_len = 1, + .is_locked = true + }, + { + .reg = TAS2781_PRM_PVDD_UVLO_REG, + .val = { 0x03 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_CHNL_0_REG, + .val = { 0xA8 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_NG_CFG0_REG, + .val = { 0xb9 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_IDLE_CH_DET_REG, + .val = { 0x92 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_PLT_FLAG_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_SINEGAIN_REG, + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_SINEGAIN2_REG, + .val_len = 4, + .is_locked = false + }, +}; + static const struct i2c_device_id tasdevice_id[] = { { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, @@ -49,6 +181,22 @@ static const struct of_device_id tasdevice_of_match[] = { MODULE_DEVICE_TABLE(of, tasdevice_of_match); #endif +static const int tas2563_cali_data_reg[] = { + TAS2563_PRM_R0_REG, + TAS2563_PRM_R0_LOW_REG, + TAS2563_PRM_INVR0_REG, + TAS2563_PRM_POW_REG, + TAS2563_PRM_TLIMIT_REG, +}; + +static const int tas2781_cali_data_reg[] = { + TAS2781_PRM_R0_REG, + TAS2781_PRM_R0_LOW_REG, + TAS2781_PRM_INVR0_REG, + TAS2781_PRM_POW_REG, + TAS2781_PRM_TLIMIT_REG, +}; + /** * tas2781_digital_getvol - get the volum control * @kcontrol: control pointer @@ -67,8 +215,13 @@ static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_digital_getvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_digital_getvol(tas_priv, ucontrol, mc); + return rc; } static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, @@ -78,8 +231,13 @@ static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_digital_putvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_digital_putvol(tas_priv, ucontrol, mc); + return rc; } static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, @@ -89,8 +247,13 @@ static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; - return tasdevice_amp_getvol(tas_priv, ucontrol, mc); + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_amp_getvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); + + return rc; } static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, @@ -101,8 +264,13 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; - return tasdevice_amp_putvol(tas_priv, ucontrol, mc); + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_amp_putvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); + + return rc; } static int tasdev_force_fwload_get(struct snd_kcontrol *kcontrol, @@ -141,6 +309,414 @@ static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static int tasdev_cali_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned char *data = tas_priv->cali_data.data; + + mutex_lock(&tas_priv->codec_lock); + if (!tas_priv->is_user_space_calidata || + tas_priv->cali_data.total_sz != bytes_ext->max) + goto out; + + memcpy(dst, data, bytes_ext->max); +out: + mutex_unlock(&tas_priv->codec_lock); + return 0; +} + +static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, + unsigned char *dst, int data_len) +{ + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int rc = -1; + int i; + + if (data_len != 4) + return rc; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + rc = tasdevice_dev_bulk_read(tas_priv, i, reg, &dst[1], + 4); + break; + } + } + + return rc; +} + +static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, + unsigned char *dat) +{ + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int j; + + if (p == NULL) + return; + + /* Store the current setting from the chip */ + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_read(tas_priv, i, p[j].reg, + (int *)&p[j].val[0]); + } else { + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } + + /* Update the setting for calibration */ + for (j = 0; j < sum - 2; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + tas2781_cali_start_reg[j].val[0]); + } else { + 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); +} + +static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dat = ucontrol->value.bytes.data; + int rc = 1; + int i; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2781 && + bytes_ext->max != 9 * tas_priv->ndev) { + rc = 0; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + int k = i * 9; + + if (dat[k] != i) { + dev_err(tas_priv->dev, + "%s:no cal-setting for dev %d\n", __func__, i); + continue; + } + sngl_calib_start(tas_priv, i, dat + k); + } +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static void tas2781_calib_stop_put(struct tasdevice_priv *tas_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; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + } else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct bulk_reg_val *q = (struct bulk_reg_val *)tas2563_cali_start_reg; + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int rc = 1; + int i, j; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2563) { + rc = -1; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_read(tas_priv, + i, p[j].reg, + (unsigned int *)&p[j].val[0]); + else + tasdevice_dev_bulk_read(tas_priv, + i, p[j].reg, p[j].val, 4); + } + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_write(tas_priv, i, p[j].reg, + q[j].val[0]); + else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + q[j].val, 4); + } + } +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static void tas2563_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tasdev_calib_stop_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct calidata *cali_data = &tas_priv->cali_data; + unsigned char *data = cali_data->data; + int index = 0, i, j; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id == TAS2563) + tas2563_calib_stop_put(tas_priv); + else + tas2781_calib_stop_put(tas_priv); + + for (i = 0; i < tas_priv->ndev; i++) { + data[index] = i; + index++; + for (j = 0; j < cali_data->reg_array_sz; j++) { + tasdevice_dev_bulk_read(tas_priv, i, + cali_data->reg_array[j], &data[index], 4); + index += 4; + } + } + mutex_unlock(&tas_priv->codec_lock); + + return 1; +} + +static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *data = tas_priv->cali_data.data; + int rc = 1; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->cali_data.total_sz != bytes_ext->max) { + rc = 0; + goto out; + } + tas_priv->is_user_space_calidata = true; + data[0] = bytes_ext->max; + memcpy(&data[1], ucontrol->value.bytes.data, bytes_ext->max - 1); +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static int tas2781_latch_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + unsigned char *dst = ucontrol->value.bytes.data; + int i, val, rc = -1; + + mutex_lock(&tas_priv->codec_lock); + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + tasdevice_dev_read(tas_priv, i, + TAS2781_RUNTIME_LATCH_RE_REG, &val); + dst[1] = val; + rc = 0; + break; + } + } + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG_TF; + else + reg = TAS2563_RUNTIME_RE_REG_TF; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG; + else + reg = TAS2563_RUNTIME_RE_REG; + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_PRM_R0_REG; + else + reg = TAS2563_PRM_R0_REG; + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + 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; + int rc = -1; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + 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; + int rc = -1; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_nop_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + static int tas2563_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -241,6 +817,16 @@ static const struct snd_kcontrol_new tasdevice_snd_controls[] = { tasdev_force_fwload_get, tasdev_force_fwload_put), }; +static const struct snd_kcontrol_new tasdevice_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tasdev_calib_stop_put), + SND_SOC_BYTES_EXT("Amp TF Data", 5, tasdev_tf_data_get, NULL), + SND_SOC_BYTES_EXT("Amp RE Data", 5, tasdev_re_data_get, NULL), + SND_SOC_BYTES_EXT("Amp R0 Data", 5, tasdev_r0_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA1 Data", 5, tasdev_XMA1_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA2 Data", 5, tasdev_XMA2_data_get, NULL), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -250,6 +836,10 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, dvc_tlv), }; +static const struct snd_kcontrol_new tas2781_cali_controls[] = { + SND_SOC_BYTES_EXT("Amp Latch Data", 2, tas2781_latch_reg_get, NULL), +}; + static const struct snd_kcontrol_new tas2563_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2563_DVC_LVL, 0, 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, @@ -257,6 +847,11 @@ static const struct snd_kcontrol_new tas2563_snd_controls[] = { tas2563_dvc_tlv), }; +static const struct snd_kcontrol_new tas2563_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Start", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tas2563_calib_start_put), +}; + static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -274,6 +869,31 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, return ret; } +static int tasdevice_info_active_num(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->ndev - 1; + + return 0; +} + +static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = TAS2563; + uinfo->value.integer.max = TAS2781; + + return 0; +} + static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -330,29 +950,37 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, return 0; } +static int tasdevice_get_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->chip_id; + + return 0; +} + static int tasdevice_create_control(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *prof_ctrls; - int nr_controls = 1; + char *name; + int nr_controls = 2; int mix_index = 0; int ret; - char *name; prof_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, sizeof(prof_ctrls[0]), GFP_KERNEL); - if (!prof_ctrls) { - ret = -ENOMEM; - goto out; - } + if (!prof_ctrls) + return -ENOMEM; /* Create a mixer item for selecting the active profile */ name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - if (!name) { - ret = -ENOMEM; - goto out; - } - scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "Speaker Profile Id"); + if (!name) + return -ENOMEM; + + strscpy(name, "Speaker Profile Id", SNDRV_CTL_ELEM_ID_NAME_MAXLEN); prof_ctrls[mix_index].name = name; prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; prof_ctrls[mix_index].info = tasdevice_info_profile; @@ -363,7 +991,6 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) ret = snd_soc_add_component_controls(tas_priv->codec, prof_ctrls, nr_controls < mix_index ? nr_controls : mix_index); -out: return ret; } @@ -423,12 +1050,48 @@ static int tasdevice_configuration_put( return ret; } -static int tasdevice_dsp_create_ctrls( - struct tasdevice_priv *tas_priv) +static int tasdevice_active_num_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + ucontrol->value.integer.value[0] = i; + return 0; + } + } + + return -1; +} + +static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int dev_id = ucontrol->value.integer.value[0]; + int max = tas_priv->ndev - 1, rc; + + dev_id = clamp(dev_id, 0, max); + + mutex_lock(&tas_priv->codec_lock); + rc = tasdev_chn_switch(tas_priv, dev_id); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *prog_name, *conf_name; - int nr_controls = 2; + char *active_dev_num, *chip_id; + char *conf_name, *prog_name; + int nr_controls = 4; int mix_index = 0; int ret; @@ -437,23 +1100,23 @@ static int tasdevice_dsp_create_ctrls( */ dsp_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, sizeof(dsp_ctrls[0]), GFP_KERNEL); - if (!dsp_ctrls) { - ret = -ENOMEM; - goto out; - } + if (!dsp_ctrls) + return -ENOMEM; /* Create a mixer item for selecting the active profile */ - prog_name = devm_kzalloc(tas_priv->dev, + active_dev_num = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + conf_name = devm_kzalloc(tas_priv->dev, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); + prog_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - if (!prog_name || !conf_name) { - ret = -ENOMEM; - goto out; - } + chip_id = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + if (!active_dev_num || !conf_name || !chip_id || !prog_name) + return -ENOMEM; - scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Program Id"); + strscpy(prog_name, "Speaker Program Id", + SNDRV_CTL_ELEM_ID_NAME_MAXLEN); dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -461,8 +1124,7 @@ static int tasdevice_dsp_create_ctrls( dsp_ctrls[mix_index].put = tasdevice_program_put; mix_index++; - scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Config Id"); + strscpy(conf_name, "Speaker Config Id", SNDRV_CTL_ELEM_ID_NAME_MAXLEN); dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; @@ -470,13 +1132,154 @@ static int tasdevice_dsp_create_ctrls( dsp_ctrls[mix_index].put = tasdevice_configuration_put; mix_index++; + strscpy(active_dev_num, "Activate Tasdevice Num", + SNDRV_CTL_ELEM_ID_NAME_MAXLEN); + dsp_ctrls[mix_index].name = active_dev_num; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_active_num; + dsp_ctrls[mix_index].get = tasdevice_active_num_get; + dsp_ctrls[mix_index].put = tasdevice_active_num_put; + mix_index++; + + strscpy(chip_id, "Tasdevice Chip Id", SNDRV_CTL_ELEM_ID_NAME_MAXLEN); + dsp_ctrls[mix_index].name = chip_id; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_chip_id; + dsp_ctrls[mix_index].get = tasdevice_get_chip_id; + mix_index++; + ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); -out: return ret; } +static int tasdevice_create_cali_ctrls(struct tasdevice_priv *tas_priv) +{ + struct calidata *cali_data = &tas_priv->cali_data; + struct tasdevice *tasdev = tas_priv->tasdevice; + struct soc_bytes_ext *ext_cali_data; + struct snd_kcontrol_new *cali_ctrls; + unsigned int num_controls; + char *cali_name; + int rc, i; + + rc = snd_soc_add_component_controls(tas_priv->codec, + tasdevice_cali_controls, ARRAY_SIZE(tasdevice_cali_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add cali controls err rc = %d", + __func__, rc); + return rc; + } + + if (tas_priv->chip_id == TAS2781) { + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; + num_controls = ARRAY_SIZE(tas2781_cali_controls); + cali_data->reg_array = (unsigned int *)tas2781_cali_data_reg; + cali_data->reg_array_sz = ARRAY_SIZE(tas2781_cali_data_reg); + for (i = 0; i < tas_priv->ndev; i++) { + tasdev[i].cali_data_restore = + kmemdup(tas2781_cali_start_reg, + sizeof(tas2781_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_restore) + return -ENOMEM; + } + } else { + cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; + num_controls = ARRAY_SIZE(tas2563_cali_controls); + cali_data->reg_array = (unsigned int *)tas2563_cali_data_reg; + cali_data->reg_array_sz = ARRAY_SIZE(tas2563_cali_data_reg); + for (i = 0; i < tas_priv->ndev; i++) { + tasdev[i].cali_data_restore = + kmemdup(tas2563_cali_start_reg, + sizeof(tas2563_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_restore) + return -ENOMEM; + } + } + + rc = snd_soc_add_component_controls(tas_priv->codec, cali_ctrls, + num_controls); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add chip cali ctrls err rc = %d", + __func__, rc); + return rc; + } + + /* index for cali_ctrls */ + i = 0; + if (tas_priv->chip_id == TAS2781) + num_controls = 2; + else + num_controls = 1; + + /* + * Alloc kcontrol via devm_kzalloc, which don't manually + * free the kcontrol + */ + cali_ctrls = devm_kcalloc(tas_priv->dev, num_controls, + sizeof(cali_ctrls[0]), GFP_KERNEL); + if (!cali_ctrls) + return -ENOMEM; + + cali_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + ext_cali_data = devm_kzalloc(tas_priv->dev, sizeof(*ext_cali_data), + GFP_KERNEL); + if (!cali_name || !ext_cali_data) + return -ENOMEM; + + strscpy(cali_name, "Speaker Calibrated Data", + SNDRV_CTL_ELEM_ID_NAME_MAXLEN); + /* the number of calibrated data per tas2563/tas2781 */ + cali_data->cali_dat_sz = cali_data->reg_array_sz * 4; + ext_cali_data->max = tas_priv->ndev * + (cali_data->cali_dat_sz + 1) + 1; + tas_priv->cali_data.total_sz = ext_cali_data->max; + tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + ext_cali_data->max, GFP_KERNEL); + cali_ctrls[i].name = cali_name; + cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cali_ctrls[i].info = snd_soc_bytes_info_ext; + cali_ctrls[i].get = tasdev_cali_data_get; + cali_ctrls[i].put = tasdev_cali_data_put; + cali_ctrls[i].private_value = (unsigned long)ext_cali_data; + i++; + + cali_data->data = devm_kzalloc(tas_priv->dev, cali_data->total_sz, + GFP_KERNEL); + if (!cali_data->data) + return -ENOMEM; + + if (tas_priv->chip_id == TAS2781) { + struct soc_bytes_ext *ext_cali_start; + char *cali_start_name; + + cali_start_name = devm_kzalloc(tas_priv->dev, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); + ext_cali_start = devm_kzalloc(tas_priv->dev, + sizeof(*ext_cali_start), GFP_KERNEL); + if (!cali_start_name || !ext_cali_start) + return -ENOMEM; + + strscpy(cali_start_name, "Calibration Start", + SNDRV_CTL_ELEM_ID_NAME_MAXLEN); + ext_cali_data->max = tas_priv->ndev * 9; + cali_ctrls[i].name = cali_start_name; + cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cali_ctrls[i].info = snd_soc_bytes_info_ext; + cali_ctrls[i].put = tas2781_calib_start_put; + cali_ctrls[i].get = tasdev_nop_get; + cali_ctrls[i].private_value = (unsigned long)ext_cali_start; + i++; + } + + rc = snd_soc_add_component_controls(tas_priv->codec, cali_ctrls, + num_controls < i ? num_controls : i); + + return rc; +} + static void tasdevice_fw_ready(const struct firmware *fmw, void *context) { @@ -523,6 +1326,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, goto out; } + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; /* If calibrated data occurs error, dsp will still works with default @@ -558,18 +1367,14 @@ static void tasdevice_fw_ready(const struct firmware *fmw, release_firmware(fmw); } -static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, int stream) { - struct snd_soc_component *codec = snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *codec = dai->component; struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); - int state = 0; /* Codec Lock Hold */ mutex_lock(&tas_priv->codec_lock); - if (event == SND_SOC_DAPM_PRE_PMD) - state = 1; - tasdevice_tuning_switch(tas_priv, state); + tasdevice_tuning_switch(tas_priv, mute); /* Codec Lock Release*/ mutex_unlock(&tas_priv->codec_lock); @@ -578,18 +1383,16 @@ static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_AIF_OUT_E("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, - 0, 0, tasdevice_dapm_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SPK("SPK", NULL), SND_SOC_DAPM_OUTPUT("OUT"), - SND_SOC_DAPM_INPUT("DMIC") + SND_SOC_DAPM_INPUT("DMIC"), }; static const struct snd_soc_dapm_route tasdevice_audio_map[] = { {"SPK", NULL, "ASI"}, {"OUT", NULL, "SPK"}, - {"ASI OUT", NULL, "DMIC"} + {"ASI OUT", NULL, "DMIC"}, }; static int tasdevice_startup(struct snd_pcm_substream *substream, @@ -668,11 +1471,12 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { .startup = tasdevice_startup, .hw_params = tasdevice_hw_params, .set_sysclk = tasdevice_set_dai_sysclk, + .mute_stream = tasdevice_mute, }; static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", @@ -724,6 +1528,11 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) static void tasdevice_deinit(void *context) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) + kfree(tasdev[i].cali_data_restore); tasdevice_config_info_remove(tas_priv); tasdevice_dsp_remove(tas_priv); -- 2.34.1

2 1

Re: [RFC PATCH 1/6] ALSA: compress: add Sample Rate Converter codec support
by Shengjiu Wang 14 Aug '24

14 Aug '24

On Wed, Aug 14, 2024 at 5:40 PM Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> wrote: > > > > Yes, to go further, I think we can use SND_AUDIOCODEC_PCM, then > > the SRC type will be dropped. > > sounds good. > > > But my understanding of the control means the .set_metadata() API, right? > > As I said, the output rate, output format, and ratio modifier are applied to > > the instances of ASRC, which is the snd_compr_stream in driver. > > so only the .set_metadata() API can be used for these purposes. > > Humm, this is more controversial. > > The term 'metadata' really referred to known information present in > headers or additional ID3 tags and not in the compressed file itself. > The .set_metadata was assumed to be called ONCE before decoding. > > But here you have a need to update the ratio modifier on a regular basis > to compensate for the drift. This isn't what this specific callback was > designed for. We could change and allow this callback to be used > multiple times, but then this could create problems for existing > implementations which cannot deal with modified metadata on the fly. .set_metadata can be called multi times now, no need to change currently. > > And then there's the problem of defining a 'key' for the metadata. the > definition of the key is a u32, so there's plenty of space for different > implementations, but a collision is possible. We'd need an agreement on > how to allocate keys to different solutions without changing the header > file for every implementation. Can we define a private space for each case? For example the key larger than 0x80000000 is private, each driver can define it by themself? > > It sounds like we'd need a 'runtime params' callback - unless there's a > better trick to tie the control and compress layers? >

3 2

[PATCH v1] ALSA: hda: tas2563: mark const variables as __maybe_unused
by Shenghao Ding 14 Aug '24

14 Aug '24

From: Baojun Xu <baojun.xu(a)ti.com> An earlier patch changed the DECLARE_TLV_DB_SCALE declaration, but now there are additional static const variables that cause the same build warnings: In file included from sound/pci/hda/tas2781_hda_i2c.c:23: include/sound/tas2563-tlv.h:21:28: error: 'tas2563_dvc_table' defined but not used [-Werror=unused-const-variable=] 21 | static const unsigned char tas2563_dvc_table[][4] = { | ^~~~~~~~~~~~~~~~~ ^~~~~~~~~~~~~~~ Mark it as unused as well. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: --- include/sound/tas2563-tlv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h index faa3e194f73b..bb269b21f460 100644 --- a/include/sound/tas2563-tlv.h +++ b/include/sound/tas2563-tlv.h @@ -18,7 +18,7 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); /* pow(10, db/20) * pow(2,30) */ -static const unsigned char tas2563_dvc_table[][4] = { +static const __maybe_unused unsigned char tas2563_dvc_table[][4] = { { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ -- 2.43.0

2 1

[PATCH v1] ASoC: codecs: ES8326: Adjust buttons
by Zhang Yi 14 Aug '24

14 Aug '24

To adapt to chrome, we have adjusted the buttons to match the system. Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 24 +++++++++++++++++------- 1 file changed, 17 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index d5362b3be484..cf22d3b745d5 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -752,6 +752,16 @@ static void es8326_jack_button_handler(struct work_struct *work) static int button_to_report, press_count; static int prev_button, cur_button; +#if IS_ENABLED(CONFIG_SND_SOC_MT8188_MT6359) + const int hs_keys[] = { + SND_JACK_BTN_0, SND_JACK_BTN_2, SND_JACK_BTN_3 + }; +#else + const int hs_keys[] = { + SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2 + }; +#endif + if (!(es8326->jack->status & SND_JACK_HEADSET)) /* Jack unplugged */ return; @@ -760,16 +770,16 @@ static void es8326_jack_button_handler(struct work_struct *work) switch (iface) { case 0x93: /* pause button detected */ - cur_button = SND_JACK_BTN_0; + cur_button = hs_keys[0]; break; case 0x6f: case 0x4b: /* button volume up */ - cur_button = SND_JACK_BTN_1; + cur_button = hs_keys[1]; break; case 0x27: /* button volume down */ - cur_button = SND_JACK_BTN_2; + cur_button = hs_keys[2]; break; case 0x1e: case 0xe2: @@ -785,7 +795,7 @@ static void es8326_jack_button_handler(struct work_struct *work) if (press_count > 3) { /* report a press every 120ms */ snd_soc_jack_report(es8326->jack, cur_button, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); press_count = 0; } button_to_report = cur_button; @@ -800,9 +810,9 @@ static void es8326_jack_button_handler(struct work_struct *work) /* released or no pressed */ if (button_to_report != 0) { snd_soc_jack_report(es8326->jack, button_to_report, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); snd_soc_jack_report(es8326->jack, 0, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); button_to_report = 0; } es8326_disable_micbias(es8326->component); @@ -845,7 +855,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) if (es8326->jack->status & SND_JACK_HEADPHONE) { dev_dbg(comp->dev, "Report hp remove event\n"); snd_soc_jack_report(es8326->jack, 0, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); /* mute adc when mic path switch */ regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); -- 2.17.1

1 0

Behringer UCM204HD shows only both lines (4 speakers)
by GitHub issues - edited 14 Aug '24

14 Aug '24

alsa-project/alsa-ucm-conf issue #436 was edited from Udrys: Hi, I'm noob on linux so please don't beat me very hard :) I have Linux Mint 22 (Kernel 6.8.0-40-generic and Mint 21.3 had the same problem). So my UCM204HD shows only one 4 speaker channel. Not separate like on Windows or Debian. I tried to update configuration from this page but there are no difference. Maybe I'm missing something? Could you please help me? Thank you very much. P.s. Apparently it's absolutely doesn't matter what I do with /usr/share/alsa files, they don't change anything. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/436 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[tiwai-sound:for-linus 2/7] ld.lld: error: undefined symbol: i2c_acpi_client_count
by kernel test robot 14 Aug '24

14 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: 8475a1d9bb7acf1cb15842dd24baab0e8ea4e4ff commit: 03c5c350e38d9346b69357d0e52c3c40495c14a0 [2/7] ALSA: hda/realtek: Add support for new HP G12 laptops config: i386-randconfig-013-20240811 (https://download.01.org/0day-ci/archive/20240814/202408141333.gYnaitcV-lkp@…) compiler: clang version 18.1.5 (https://github.com/llvm/llvm-project 617a15a9eac96088ae5e9134248d8236e34b91b1) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240814/202408141333.gYnaitcV-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408141333.gYnaitcV-lkp@intel.com/ All errors (new ones prefixed by >>): >> ld.lld: error: undefined symbol: i2c_acpi_client_count >>> referenced by patch_realtek.c:6891 (sound/pci/hda/patch_realtek.c:6891) >>> sound/pci/hda/patch_realtek.o:(cs35lxx_autodet_fixup) in archive vmlinux.a -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH] ASoC: codecs: ES8326: Adjust buttons
by Zhang Yi 14 Aug '24

14 Aug '24

> > To adapt to chrome, we have adjusted the buttons to match the system. > > > case 0x4b: > > /* button volume up */ > > - cur_button = SND_JACK_BTN_1; > > + cur_button = SND_JACK_BTN_2; > > break; > > case 0x27: > > /* button volume down */ > > - cur_button = SND_JACK_BTN_2; > > + cur_button = SND_JACK_BTN_3; > > break; > > This is really problematic, we're changing the keys reported here in a > way that's not ABI compatible. If this was a new driver then fine but > this is quite old, it's in LTS releases, so we might be disrupting > people's existing systems. I'm not sure we have a good solution for > this at this point, we're not even swapping the buttons here but rather > using a different set of buttons. Probably ChromeOS needs to work > around this somehow, or we need to add some mechanism which allows > userspace to configure the driver mapping at runtime. We'll use a different method to solve this issue.thank you

1 0

[RESEND PATCH] ASoc: tas2781: Rename dai_driver name to unify the name between TAS2563 and TAS2781
by Shenghao Ding 13 Aug '24

13 Aug '24

Rename dai_driver name to unify the name between TAS2563 and TAS2781. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 313c17f76ca4..8c9efd9183ff 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -672,7 +672,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", -- 2.34.1

2 1

[PATCH] ASoC: meson: axg-fifo: fix irq scheduling issue with PREEMPT_RT
by Jerome Brunet 13 Aug '24

13 Aug '24

With PREEMPT_RT enabled, spinlocks become preemptible. This is usually not a problem with spinlocks used in IRQ context since IRQ handlers get threaded. However, if IRQF_ONESHOT is set, the upper half of a threaded irq handler won't be threaded and this causes scheduling problems if spinlocks are used in the upper half, like with regmap when '.fast_io' is set. In this particular instance, it is actually better to do everything in the bottom half and it solves the problem with PREEMPT_RT. Reported-by: Arseniy Krasnov <avkrasnov(a)salutedevices.com> Closes: https://lore.kernel.org/linux-amlogic/20240729131652.3012327-1-avkrasnov@sa… Suggested-by: Sebastian Andrzej Siewior <bigeasy(a)linutronix.de> Fixes: b11d26660dff ("ASoC: meson: axg-fifo: use threaded irq to check periods") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-fifo.c | 26 ++++++++++---------------- 1 file changed, 10 insertions(+), 16 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 7e6090af720b..d0d05fa44d66 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -207,25 +207,18 @@ static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) status = FIELD_GET(STATUS1_INT_STS, status); axg_fifo_ack_irq(fifo, status); - /* Use the thread to call period elapsed on nonatomic links */ - if (status & FIFO_INT_COUNT_REPEAT) - return IRQ_WAKE_THREAD; + if (status & ~FIFO_INT_COUNT_REPEAT) + dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", + status); - dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", - status); + if (status & FIFO_INT_COUNT_REPEAT) { + snd_pcm_period_elapsed(ss); + return IRQ_HANDLED; + } return IRQ_NONE; } -static irqreturn_t axg_fifo_pcm_irq_block_thread(int irq, void *dev_id) -{ - struct snd_pcm_substream *ss = dev_id; - - snd_pcm_period_elapsed(ss); - - return IRQ_HANDLED; -} - int axg_fifo_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *ss) { @@ -251,8 +244,9 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, if (ret) return ret; - ret = request_threaded_irq(fifo->irq, axg_fifo_pcm_irq_block, - axg_fifo_pcm_irq_block_thread, + /* Use the bottom half of a threaded irq with non-atomic links */ + ret = request_threaded_irq(fifo->irq, NULL, + axg_fifo_pcm_irq_block, IRQF_ONESHOT, dev_name(dev), ss); if (ret) return ret; -- 2.43.0

3 4

[PATCH] ASoC: amd: acp: add legacy driver support for ACP7.1 based platforms
by Vijendar Mukunda 13 Aug '24

13 Aug '24

Add acp pci driver and machine driver changes for ACP7.1 based platforms for legacy stack. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-legacy-common.c | 7 ++++--- sound/soc/amd/acp/acp-mach-common.c | 2 +- sound/soc/amd/acp/acp-mach.h | 1 + sound/soc/amd/acp/acp-pci.c | 4 ++++ sound/soc/amd/acp/acp70.c | 12 ++++++++++-- sound/soc/amd/acp/amd.h | 1 + 6 files changed, 21 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index 04bd605fdce3..3cc083fac837 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -270,6 +270,7 @@ static int acp_power_on(struct acp_chip_info *chip) acp_pgfsm_ctrl_reg = ACP63_PGFSM_CONTROL; break; case ACP70_DEV: + case ACP71_DEV: acp_pgfsm_stat_reg = ACP70_PGFSM_STATUS; acp_pgfsm_ctrl_reg = ACP70_PGFSM_CONTROL; break; @@ -336,10 +337,9 @@ int acp_deinit(struct acp_chip_info *chip) if (ret) return ret; - if (chip->acp_rev != ACP70_DEV) + if (chip->acp_rev < ACP70_DEV) writel(0, chip->base + ACP_CONTROL); - - if (chip->acp_rev >= ACP70_DEV) + else writel(0x01, chip->base + ACP_ZSC_DSP_CTRL); return 0; } @@ -461,6 +461,7 @@ void check_acp_config(struct pci_dev *pci, struct acp_chip_info *chip) check_acp6x_config(chip); break; case ACP70_DEV: + case ACP71_DEV: pdm_addr = ACP70_PDM_ADDR; check_acp70_config(chip); break; diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index a36300a4ed8a..e9ff4815c12c 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -1766,7 +1766,7 @@ int acp_legacy_dai_links_create(struct snd_soc_card *card) } else if (drv_data->platform == ACP63) { links[i].platforms = platform_acp63_component; links[i].num_platforms = ARRAY_SIZE(platform_acp63_component); - } else if (drv_data->platform == ACP70) { + } else if ((drv_data->platform == ACP70) || (drv_data->platform == ACP71)) { links[i].platforms = platform_acp70_component; links[i].num_platforms = ARRAY_SIZE(platform_acp70_component); } else { diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index a48546d8d407..93d9e3886b7e 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -56,6 +56,7 @@ enum platform_end_point { REMBRANDT, ACP63, ACP70, + ACP71, }; struct acp_mach_ops { diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index b0304b813cad..f7450a5bd103 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -95,6 +95,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id chip->name = "acp_asoc_acp70"; chip->acp_rev = ACP70_DEV; break; + case 0x71: + chip->name = "acp_asoc_acp70"; + chip->acp_rev = ACP71_DEV; + break; default: dev_err(dev, "Unsupported device revision:0x%x\n", pci->revision); ret = -EINVAL; diff --git a/sound/soc/amd/acp/acp70.c b/sound/soc/amd/acp/acp70.c index a2cbdcca4313..81636faa22cd 100644 --- a/sound/soc/amd/acp/acp70.c +++ b/sound/soc/amd/acp/acp70.c @@ -147,7 +147,11 @@ static int acp_acp70_audio_probe(struct platform_device *pdev) return -ENODEV; } - if (chip->acp_rev != ACP70_DEV) { + switch (chip->acp_rev) { + case ACP70_DEV: + case ACP71_DEV: + break; + default: dev_err(&pdev->dev, "Un-supported ACP Revision %d\n", chip->acp_rev); return -ENODEV; } @@ -178,7 +182,11 @@ static int acp_acp70_audio_probe(struct platform_device *pdev) adata->num_dai = ARRAY_SIZE(acp70_dai); adata->rsrc = &rsrc; adata->machines = snd_soc_acpi_amd_acp70_acp_machines; - adata->platform = ACP70; + if (chip->acp_rev == ACP70_DEV) + adata->platform = ACP70; + else + adata->platform = ACP71; + adata->flag = chip->flag; acp_machine_select(adata); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index c095a34a7229..4f0cd7dd04e5 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -22,6 +22,7 @@ #define ACP6X_DEV 6 #define ACP63_DEV 0x63 #define ACP70_DEV 0x70 +#define ACP71_DEV 0x71 #define DMIC_INSTANCE 0x00 #define I2S_SP_INSTANCE 0x01 -- 2.34.1

2 1

[PATCH] ASoC: amd: acp: Convert comma to semicolon
by Vijendar Mukunda 13 Aug '24

13 Aug '24

Replace a comma between expression statements by a semicolon. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/amd/acp/acp-sdw-sof-mach.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index f7e4af850309..08f368b3bbc8 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -657,9 +657,9 @@ static int mc_probe(struct platform_device *pdev) ctx->private = amd_ctx; card = &ctx->card; card->dev = &pdev->dev; - card->name = "amd-soundwire", - card->owner = THIS_MODULE, - card->late_probe = asoc_sdw_card_late_probe, + card->name = "amd-soundwire"; + card->owner = THIS_MODULE; + card->late_probe = asoc_sdw_card_late_probe; snd_soc_card_set_drvdata(card, ctx); -- 2.34.1

2 1

[PATCH] ASoC: cs43130: Constify struct reg_sequence and reg_sequences
by Christophe JAILLET 13 Aug '24

13 Aug '24

'struct reg_sequence' and 'struct reg_sequences' are not modified in this drivers. Constifying these structures moves some data to a read-only section, so increase overall security. On a x86_64, with allmodconfig: Before: ====== text data bss dec hex filename 54409 7881 64 62354 f392 sound/soc/codecs/cs43130.o After: ===== text data bss dec hex filename 55562 6729 64 62355 f393 sound/soc/codecs/cs43130.o Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Compile tested-only. --- sound/soc/codecs/cs43130.c | 40 +++++++++++++++++++------------------- 1 file changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index cb4ca80f36d2..f8e2fb69ada2 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1805,7 +1805,7 @@ static struct attribute *hpload_attrs[] = { }; ATTRIBUTE_GROUPS(hpload); -static struct reg_sequence hp_en_cal_seq[] = { +static const struct reg_sequence hp_en_cal_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, {CS43130_HP_MEAS_LOAD_2, 0}, @@ -1820,7 +1820,7 @@ static struct reg_sequence hp_en_cal_seq[] = { {CS43130_HP_LOAD_1, 0x80}, }; -static struct reg_sequence hp_en_cal_seq2[] = { +static const struct reg_sequence hp_en_cal_seq2[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, {CS43130_HP_MEAS_LOAD_2, 0}, @@ -1828,7 +1828,7 @@ static struct reg_sequence hp_en_cal_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, }; -static struct reg_sequence hp_dis_cal_seq[] = { +static const struct reg_sequence hp_dis_cal_seq[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_DXD1, 0x99}, {CS43130_DXD12, 0}, @@ -1836,12 +1836,12 @@ static struct reg_sequence hp_dis_cal_seq[] = { {CS43130_HP_LOAD_1, 0}, }; -static struct reg_sequence hp_dis_cal_seq2[] = { +static const struct reg_sequence hp_dis_cal_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_HP_LOAD_1, 0}, }; -static struct reg_sequence hp_dc_ch_l_seq[] = { +static const struct reg_sequence hp_dc_ch_l_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x0A}, {CS43130_DXD17, 0x93}, @@ -1851,12 +1851,12 @@ static struct reg_sequence hp_dc_ch_l_seq[] = { {CS43130_HP_LOAD_1, 0x81}, }; -static struct reg_sequence hp_dc_ch_l_seq2[] = { +static const struct reg_sequence hp_dc_ch_l_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_HP_LOAD_1, 0x81}, }; -static struct reg_sequence hp_dc_ch_r_seq[] = { +static const struct reg_sequence hp_dc_ch_r_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x8A}, {CS43130_DXD17, 0x15}, @@ -1866,12 +1866,12 @@ static struct reg_sequence hp_dc_ch_r_seq[] = { {CS43130_HP_LOAD_1, 0x91}, }; -static struct reg_sequence hp_dc_ch_r_seq2[] = { +static const struct reg_sequence hp_dc_ch_r_seq2[] = { {CS43130_HP_LOAD_1, 0x90}, {CS43130_HP_LOAD_1, 0x91}, }; -static struct reg_sequence hp_ac_ch_l_seq[] = { +static const struct reg_sequence hp_ac_ch_l_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x0A}, {CS43130_DXD17, 0x93}, @@ -1881,12 +1881,12 @@ static struct reg_sequence hp_ac_ch_l_seq[] = { {CS43130_HP_LOAD_1, 0x82}, }; -static struct reg_sequence hp_ac_ch_l_seq2[] = { +static const struct reg_sequence hp_ac_ch_l_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_HP_LOAD_1, 0x82}, }; -static struct reg_sequence hp_ac_ch_r_seq[] = { +static const struct reg_sequence hp_ac_ch_r_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x8A}, {CS43130_DXD17, 0x15}, @@ -1896,24 +1896,24 @@ static struct reg_sequence hp_ac_ch_r_seq[] = { {CS43130_HP_LOAD_1, 0x92}, }; -static struct reg_sequence hp_ac_ch_r_seq2[] = { +static const struct reg_sequence hp_ac_ch_r_seq2[] = { {CS43130_HP_LOAD_1, 0x90}, {CS43130_HP_LOAD_1, 0x92}, }; -static struct reg_sequence hp_cln_seq[] = { +static const struct reg_sequence hp_cln_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, {CS43130_HP_MEAS_LOAD_2, 0}, }; struct reg_sequences { - struct reg_sequence *seq; - int size; - unsigned int msk; + const struct reg_sequence *seq; + int size; + unsigned int msk; }; -static struct reg_sequences hpload_seq1[] = { +static const struct reg_sequences hpload_seq1[] = { { .seq = hp_en_cal_seq, .size = ARRAY_SIZE(hp_en_cal_seq), @@ -1951,7 +1951,7 @@ static struct reg_sequences hpload_seq1[] = { }, }; -static struct reg_sequences hpload_seq2[] = { +static const struct reg_sequences hpload_seq2[] = { { .seq = hp_en_cal_seq2, .size = ARRAY_SIZE(hp_en_cal_seq2), @@ -2041,7 +2041,7 @@ static int cs43130_update_hpload(unsigned int msk, int ac_idx, } static int cs43130_hpload_proc(struct cs43130_private *cs43130, - struct reg_sequence *seq, int seq_size, + const struct reg_sequence *seq, int seq_size, unsigned int rslt_msk, int ac_idx) { int ret; @@ -2122,7 +2122,7 @@ static void cs43130_imp_meas(struct work_struct *wk) int i, ret, ac_idx; struct cs43130_private *cs43130; struct snd_soc_component *component; - struct reg_sequences *hpload_seq; + const struct reg_sequences *hpload_seq; cs43130 = container_of(wk, struct cs43130_private, work); component = cs43130->component; -- 2.45.2

2 1

[PATCH V2 1/2] ASoC: SOF: amd: move iram-dram fence register programming sequence
by Vijendar Mukunda 13 Aug '24

13 Aug '24

The existing code modifies IRAM and DRAM size after sha dma start for vangogh platform. The problem with this sequence is that it might cause sha dma failure when firmware code binary size is greater than the default IRAM size. To fix this issue, Move the iram-dram fence register sequence prior to sha dma start. Fixes: 094d11768f74 ("ASoC: SOF: amd: Skip IRAM/DRAM size modification for Steam Deck OLED") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- changes since v1: - Rephrase commit description sound/soc/sof/amd/acp.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 7b122656efd1..b8d4f986f89d 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -263,6 +263,17 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_STRT_ADDR, start_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_DESTINATION_ADDR, dest_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_MSG_LENGTH, image_length); + + /* psp_send_cmd only required for vangogh platform (rev - 5) */ + if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { + /* Modify IRAM and DRAM size */ + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); + if (ret) + return ret; + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); + if (ret) + return ret; + } snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_CMD, ACP_SHA_RUN); ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_TRANSFER_BYTE_CNT, @@ -280,17 +291,6 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return ret; } - /* psp_send_cmd only required for vangogh platform (rev - 5) */ - if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { - /* Modify IRAM and DRAM size */ - ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); - if (ret) - return ret; - ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); - if (ret) - return ret; - } - ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_DSP_FW_QUALIFIER, fw_qualifier, fw_qualifier & DSP_FW_RUN_ENABLE, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); -- 2.34.1

4 4

[PATCH 1/2] ASoC: SOF: amd: move iram-dram fence register programming sequence
by Vijendar Mukunda 13 Aug '24

13 Aug '24

As per design, ACP iram-dram fence register sequence should be initiated before triggering SHA dma. This ensures that IRAM size will programmed correctly before initiaing SHA dma. Fixes: 094d11768f74 ("ASoC: SOF: amd: Skip IRAM/DRAM size modification for Steam Deck OLED") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/amd/acp.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 7b122656efd1..b8d4f986f89d 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -263,6 +263,17 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_STRT_ADDR, start_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_DESTINATION_ADDR, dest_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_MSG_LENGTH, image_length); + + /* psp_send_cmd only required for vangogh platform (rev - 5) */ + if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { + /* Modify IRAM and DRAM size */ + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); + if (ret) + return ret; + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); + if (ret) + return ret; + } snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_CMD, ACP_SHA_RUN); ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_TRANSFER_BYTE_CNT, @@ -280,17 +291,6 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return ret; } - /* psp_send_cmd only required for vangogh platform (rev - 5) */ - if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { - /* Modify IRAM and DRAM size */ - ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); - if (ret) - return ret; - ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); - if (ret) - return ret; - } - ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_DSP_FW_QUALIFIER, fw_qualifier, fw_qualifier & DSP_FW_RUN_ENABLE, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); -- 2.34.1

4 4

[PATCH] ASoC: MAINTAINERS: Drop Banajit Goswami from Qualcomm sound drivers
by Krzysztof Kozlowski 13 Aug '24

13 Aug '24

There was no active maintenance from Banajit Goswami - last email is from 2019 - so make obvious that Qualcomm sound drivers are maintained by only one person. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Not moving to credits because lore does not point to significant contributions. Just ~140 emails: https://lore.kernel.org/all/?q=f%3A%22Banajit+Goswami%22 --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index c8e16a649a0e..43e7668aacb0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -18419,7 +18419,6 @@ F: drivers/crypto/intel/qat/ QCOM AUDIO (ASoC) DRIVERS M: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> -M: Banajit Goswami <bgoswami(a)quicinc.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: linux-arm-msm(a)vger.kernel.org S: Supported -- 2.43.0

2 2

[PATCH] ASoc: tas2781: Rename dai_driver name to unify the name between TAS2563 and TAS2781
by Shenghao Ding 13 Aug '24

13 Aug '24

Rename dai_driver name to unify the name between TAS2563 and TAS2781. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8a57c045afdd..8f9574189e1f 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -673,7 +673,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", -- 2.34.1

3 2

[PATCH] ASoC: codecs: ES8326: Adjust buttons
by Zhang Yi 13 Aug '24

13 Aug '24

To adapt to chrome, we have adjusted the buttons to match the system. Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index d5362b3be484..5b4586d9e67a 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -765,11 +765,11 @@ static void es8326_jack_button_handler(struct work_struct *work) case 0x6f: case 0x4b: /* button volume up */ - cur_button = SND_JACK_BTN_1; + cur_button = SND_JACK_BTN_2; break; case 0x27: /* button volume down */ - cur_button = SND_JACK_BTN_2; + cur_button = SND_JACK_BTN_3; break; case 0x1e: case 0xe2: @@ -785,7 +785,7 @@ static void es8326_jack_button_handler(struct work_struct *work) if (press_count > 3) { /* report a press every 120ms */ snd_soc_jack_report(es8326->jack, cur_button, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); press_count = 0; } button_to_report = cur_button; @@ -800,9 +800,9 @@ static void es8326_jack_button_handler(struct work_struct *work) /* released or no pressed */ if (button_to_report != 0) { snd_soc_jack_report(es8326->jack, button_to_report, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); snd_soc_jack_report(es8326->jack, 0, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); button_to_report = 0; } es8326_disable_micbias(es8326->component); @@ -845,7 +845,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) if (es8326->jack->status & SND_JACK_HEADPHONE) { dev_dbg(comp->dev, "Report hp remove event\n"); snd_soc_jack_report(es8326->jack, 0, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); /* mute adc when mic path switch */ regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); -- 2.17.1

1 0

[PATCH v2] ALSA: ASoC/tas2781: fix wrong calibrated data order
by Shenghao Ding 13 Aug '24

13 Aug '24

From: Baojun Xu <baojun.xu(a)ti.com> Wrong calibration data order cause sound too low in some device. Fix wrong calibrated data order, add calibration data converssion by get_unaligned_be32() after reading from UEFI. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v2: - Change data type from int to __be32. v1: - Change copyright date, and add new maintainer. - Add unaligned.h included for get_unaligned_be32(). - In tas2781_apply_calib(), change data address transfer directly to get data by get_unaligned_be32(), and send address to device. --- sound/pci/hda/tas2781_hda_i2c.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 8a7fe48043d2..5a8193285f9f 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -2,10 +2,12 @@ // // TAS2781 HDA I2C driver // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Current maintainer: Baojun Xu <baojun.xu(a)ti.com> +#include <asm/unaligned.h> #include <linux/acpi.h> #include <linux/crc8.h> #include <linux/crc32.h> @@ -519,20 +521,22 @@ static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) static const unsigned char rgno_array[CALIB_MAX] = { 0x74, 0x0c, 0x14, 0x70, 0x7c, }; - unsigned char *data; + int offset = 0; int i, j, rc; + __be32 data; for (i = 0; i < tas_priv->ndev; i++) { - data = tas_priv->cali_data.data + - i * TASDEVICE_SPEAKER_CALIBRATION_SIZE; for (j = 0; j < CALIB_MAX; j++) { + data = get_unaligned_be32( + &tas_priv->cali_data.data[offset]); rc = tasdevice_dev_bulk_write(tas_priv, i, TASDEVICE_REG(0, page_array[j], rgno_array[j]), - &(data[4 * j]), 4); + (unsigned char *)&data, 4); if (rc < 0) dev_err(tas_priv->dev, "chn %d calib %d bulk_wr err = %d\n", i, j, rc); + offset += 4; } } } -- 2.43.0

3 2

[PATCH 1/8] ASoC: SOF: amd: Fix for incorrect acp error satus register offset
by Vijendar Mukunda 13 Aug '24

13 Aug '24

Adding 'dsp_intr_base' to ACP error status register offset in irq handler points to wrong register offset. ACP error status register offset got changed from ACP 6.0 onwards. Add 'acp_error_stat' descriptor field and update the value based on the ACP variant. Fixes: 0e44572a28a4 ("ASoC: SOF: amd: Add helper callbacks for ACP's DMA configuration") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/amd/acp-dsp-offset.h | 3 ++- sound/soc/sof/amd/acp.c | 5 +++-- sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-acp63.c | 1 + sound/soc/sof/amd/pci-rmb.c | 1 + sound/soc/sof/amd/pci-rn.c | 1 + 6 files changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/amd/acp-dsp-offset.h b/sound/soc/sof/amd/acp-dsp-offset.h index 59afbe2e0f42..66968efda869 100644 --- a/sound/soc/sof/amd/acp-dsp-offset.h +++ b/sound/soc/sof/amd/acp-dsp-offset.h @@ -76,7 +76,8 @@ #define DSP_SW_INTR_CNTL_OFFSET 0x0 #define DSP_SW_INTR_STAT_OFFSET 0x4 #define DSP_SW_INTR_TRIG_OFFSET 0x8 -#define ACP_ERROR_STATUS 0x18C4 +#define ACP3X_ERROR_STATUS 0x18C4 +#define ACP6X_ERROR_STATUS 0x1A4C #define ACP3X_AXI2DAGB_SEM_0 0x1880 #define ACP5X_AXI2DAGB_SEM_0 0x1884 #define ACP6X_AXI2DAGB_SEM_0 0x1874 diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 7b122656efd1..d0b7d1c54248 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -92,6 +92,7 @@ static int config_dma_channel(struct acp_dev_data *adata, unsigned int ch, unsigned int idx, unsigned int dscr_count) { struct snd_sof_dev *sdev = adata->dev; + const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int val, status; int ret; @@ -102,7 +103,7 @@ static int config_dma_channel(struct acp_dev_data *adata, unsigned int ch, val & (1 << ch), ACP_REG_POLL_INTERVAL, ACP_REG_POLL_TIMEOUT_US); if (ret < 0) { - status = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_ERROR_STATUS); + status = snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->acp_error_stat); val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_DMA_ERR_STS_0 + ch * sizeof(u32)); dev_err(sdev->dev, "ACP_DMA_ERR_STS :0x%x ACP_ERROR_STATUS :0x%x\n", val, status); @@ -404,7 +405,7 @@ static irqreturn_t acp_irq_handler(int irq, void *dev_id) snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_stat, ACP_ERROR_IRQ_MASK); snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_SW0_I2S_ERROR_REASON, 0); snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_SW1_I2S_ERROR_REASON, 0); - snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_ERROR_STATUS, 0); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->acp_error_stat, 0); irq_flag = 1; } diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index ec9170b3f068..6ac853ff6093 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -203,6 +203,7 @@ struct sof_amd_acp_desc { u32 probe_reg_offset; u32 reg_start_addr; u32 reg_end_addr; + u32 acp_error_stat; u32 sdw_max_link_count; u64 sdw_acpi_dev_addr; }; diff --git a/sound/soc/sof/amd/pci-acp63.c b/sound/soc/sof/amd/pci-acp63.c index 54d42f83ce9e..c3da70549995 100644 --- a/sound/soc/sof/amd/pci-acp63.c +++ b/sound/soc/sof/amd/pci-acp63.c @@ -35,6 +35,7 @@ static const struct sof_amd_acp_desc acp63_chip_info = { .ext_intr_cntl = ACP6X_EXTERNAL_INTR_CNTL, .ext_intr_stat = ACP6X_EXT_INTR_STAT, .ext_intr_stat1 = ACP6X_EXT_INTR_STAT1, + .acp_error_stat = ACP6X_ERROR_STATUS, .dsp_intr_base = ACP6X_DSP_SW_INTR_BASE, .sram_pte_offset = ACP6X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP6X_AXI2DAGB_SEM_0, diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index 4bc30951f8b0..194b7ff37e9e 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -33,6 +33,7 @@ static const struct sof_amd_acp_desc rembrandt_chip_info = { .pgfsm_base = ACP6X_PGFSM_BASE, .ext_intr_stat = ACP6X_EXT_INTR_STAT, .dsp_intr_base = ACP6X_DSP_SW_INTR_BASE, + .acp_error_stat = ACP6X_ERROR_STATUS, .sram_pte_offset = ACP6X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP6X_AXI2DAGB_SEM_0, .fusion_dsp_offset = ACP6X_DSP_FUSION_RUNSTALL, diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index e08875bdfa8b..bff2d979ea6a 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -33,6 +33,7 @@ static const struct sof_amd_acp_desc renoir_chip_info = { .pgfsm_base = ACP3X_PGFSM_BASE, .ext_intr_stat = ACP3X_EXT_INTR_STAT, .dsp_intr_base = ACP3X_DSP_SW_INTR_BASE, + .acp_error_stat = ACP3X_ERROR_STATUS, .sram_pte_offset = ACP3X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP3X_AXI2DAGB_SEM_0, .acp_clkmux_sel = ACP3X_CLKMUX_SEL, -- 2.34.1

4 16

[PATCH 0/2] Add a driver for the Iron Device SMA1307 Amp
by Kiseok Jo 13 Aug '24

13 Aug '24

This adds basic audio support for the Iron Device SMA1307 amplifier Kiseok Jo (2): ASoC: sma1307: Add driver for Iron Device SMA1307 ASoC: sma1307: Add bindings for Iron Device SMA1307 amplifier .../bindings/sound/irondevice,sma1307.yaml | 66 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2630 +++++++++++++++++ sound/soc/codecs/sma1307.h | 456 +++ 5 files changed, 3162 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h -- 2.39.2

4 8

[PATCH RESEND 0/8] AMD SOF stack fixes and improvements
by Vijendar Mukunda 12 Aug '24

12 Aug '24

This patch series consists of few fixes and improvments for AMD SOF stack. Link: https://github.com/thesofproject/linux/pull/5103 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Vijendar Mukunda (8): ASoC: SOF: amd: Fix for incorrect acp error satus register offset ASoC: SOF: amd: fix for acp error reason registers wrong offset ASoC: SOF: amd: move iram-dram fence register programming sequence ASoC: SOF: amd: fix for acp init sequence ASoC: SOF: amd: update conditional check for cache register update ASoC: SOF: amd: modify psp send command conditional checks ASoC: SOF: amd: remove unused variable from sof_amd_acp_desc structure ASoC: amd: acp: Convert comma to semicolon sound/soc/amd/acp/acp-sdw-sof-mach.c | 6 +-- sound/soc/sof/amd/acp-dsp-offset.h | 6 ++- sound/soc/sof/amd/acp-loader.c | 2 +- sound/soc/sof/amd/acp.c | 59 ++++++++++++++++++---------- sound/soc/sof/amd/acp.h | 10 +++-- sound/soc/sof/amd/pci-acp63.c | 3 +- sound/soc/sof/amd/pci-rmb.c | 3 +- sound/soc/sof/amd/pci-rn.c | 3 +- sound/soc/sof/amd/pci-vangogh.c | 1 - 9 files changed, 60 insertions(+), 33 deletions(-) -- 2.34.1

2 9

[PATCH v1] ASoc: tas2781: Add new Kontrol to set tas2563 digital gain
by Shenghao Ding 11 Aug '24

11 Aug '24

Requriment from customer to add new kcontrol to set tas2563 digital gain and set "Speaker Force Firmware Load" as the common kcontrol for both tas27871 and tas2563. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Changed the copyright year to 2024 in tas2781-tlv.h. - Changed the copyright year to 2024 in tas2781.h. - Set "Speaker Force Firmware Load" as the common kcontrol for for both tas27871 and tas2563, that is tasdevice_snd_controls. - Only tas2781_snd_controls only for gain setting. - Add tas2563_dvc_tlv kcontrol for tas2563 digital gain setting. - In codec probe, for tas2563, loading tas2563 gain control; for tas2781, loading tas2781 gain control. --- include/sound/tas2781-tlv.h | 262 ++++++++++++++++++++++++++++++++- include/sound/tas2781.h | 3 +- sound/soc/codecs/tas2781-i2c.c | 135 ++++++++++++++++- 3 files changed, 392 insertions(+), 8 deletions(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 1dc59005d241..99c41bfc7827 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -17,5 +17,265 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); +static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); +/* pow(10, db/20) * pow(2,30) */ +static const unsigned char tas2563_dvc_table[][4] = { + { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ + { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ + { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ + { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ + { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ + { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ + { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ + { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ + { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ + { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ + { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ + { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ + { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ + { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ + { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ + { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ + { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ + { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ + { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ + { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ + { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ + { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ + { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ + { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ + { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ +}; #endif diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index cd8ce522b78e..3275575f5bfb 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -50,6 +50,7 @@ #define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) /* Volume control */ +#define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) #define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) #define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 4d1a0d836e77..4960ab6577f7 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,6 +30,7 @@ #include <sound/tas2781.h> #include <sound/tlv.h> #include <sound/tas2781-tlv.h> +#include <asm/unaligned.h> static const struct i2c_device_id tasdevice_id[] = { { "tas2563", TAS2563 }, @@ -103,7 +104,7 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, return tasdevice_amp_putvol(tas_priv, ucontrol, mc); } -static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -118,7 +119,7 @@ static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, return 0; } -static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -127,18 +128,114 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, snd_soc_component_get_drvdata(component); bool change, val = (bool)ucontrol->value.integer.value[0]; + mutex_lock(&tas_priv->codec_lock); if (tas_priv->force_fwload_status == val) change = false; else { change = true; tas_priv->force_fwload_status = val; } + mutex_unlock(&tas_priv->codec_lock); dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, tas_priv->force_fwload_status ? "ON" : "OFF"); return change; } +static int tas2563_digital_gain_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int l = 0, r = mc->max; + unsigned int target, ar_mid, mid, ar_l, ar_r; + unsigned int reg = mc->reg; + unsigned char data[4]; + int ret; + + mutex_lock(&tas_dev->codec_lock); + /* Read the primary device */ + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + if (ret) { + dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + target = get_unaligned_be32(&data[0]); + + while (r > 1 + l) { + mid = (l + r) / 2; + ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + if (target < ar_mid) + r = mid; + else + l = mid; + } + + ar_l = get_unaligned_be32(tas2563_dvc_table[l]); + ar_r = get_unaligned_be32(tas2563_dvc_table[r]); + + /* find out the member same as or closer to the current volume */ + ucontrol->value.integer.value[0] = + abs(target - ar_l) <= abs(target - ar_r) ? l : r; +out: + mutex_unlock(&tas_dev->codec_lock); + return 0; +} + +static int tas2563_digital_gain_put( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int reg = mc->reg; + unsigned int volrd, volwr; + int vol = ucontrol->value.integer.value[0]; + int max = mc->max, i, ret = 1; + unsigned char data[4]; + + vol = clamp(vol, 0, max); + mutex_lock(&tas_dev->codec_lock); + /* Read the primary device */ + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + if (ret) { + dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + volrd = get_unaligned_be32(&data[0]); + volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + + if (volrd == volwr) { + ret = 0; + goto out; + } + + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_bulk_write(tas_dev, i, reg, + (unsigned char *)tas2563_dvc_table[vol], 4); + if (ret) + dev_err(tas_dev->dev, + "%s, set digital vol error in device %d\n", + __func__, i); + } + +out: + mutex_unlock(&tas_dev->codec_lock); + return ret; +} + +static const struct snd_kcontrol_new tasdevice_snd_controls[] = { + SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tasdev_force_fwload_get, tasdev_force_fwload_put), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -146,8 +243,13 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), - SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, - tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static const struct snd_kcontrol_new tas2563_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2563_DVC_LVL, 0, + 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, + tas2563_digital_gain_get, tas2563_digital_gain_put, + tas2563_dvc_tlv), }; static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, @@ -578,6 +680,27 @@ static struct snd_soc_dai_driver tasdevice_dai_driver[] = { static int tasdevice_codec_probe(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int rc; + + if (tas_priv->chip_id == TAS2781) { + rc = snd_soc_add_component_controls(codec, + tas2781_snd_controls, + ARRAY_SIZE(tas2781_snd_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + } else { + rc = snd_soc_add_component_controls(codec, + tas2563_snd_controls, + ARRAY_SIZE(tas2563_snd_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + } tas_priv->name_prefix = codec->name_prefix; return tascodec_init(tas_priv, codec, THIS_MODULE, tasdevice_fw_ready); @@ -605,8 +728,8 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .controls = tas2781_snd_controls, - .num_controls = ARRAY_SIZE(tas2781_snd_controls), + .controls = tasdevice_snd_controls, + .num_controls = ARRAY_SIZE(tasdevice_snd_controls), .dapm_widgets = tasdevice_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), .dapm_routes = tasdevice_audio_map, -- 2.34.1

4 3

[PATCH v2] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 11 Aug '24

11 Aug '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job change to Shenghao Ding and Baojun Xu, drop entries for the audio converter and add entries for both haptics drivers and middle- power audio amplifier drivers. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 29 +++++++++++------------------ 1 file changed, 11 insertions(+), 18 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index f328373463b0..d1f7e096b270 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22569,37 +22569,30 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml -F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml -F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml -F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml -F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: drivers/input/misc/drv2*.c F: include/sound/tas2*.h -F: include/sound/tlv320*.h +F: include/sound/tas5086.h F: include/sound/tpa6130a2-plat.h F: sound/pci/hda/tas2781_hda_i2c.c -F: sound/soc/codecs/pcm1681.c -F: sound/soc/codecs/pcm1789*.* -F: sound/soc/codecs/pcm179x*.* -F: sound/soc/codecs/pcm186x*.* -F: sound/soc/codecs/pcm3008.* -F: sound/soc/codecs/pcm3060*.* -F: sound/soc/codecs/pcm3168a*.* -F: sound/soc/codecs/pcm5102a.c -F: sound/soc/codecs/pcm512x*.* F: sound/soc/codecs/tas2*.* -F: sound/soc/codecs/tlv320*.* +F: sound/soc/codecs/tas5*.* F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS -- 2.34.1

1 0

[PATCH 0/3] Add DisplayPort sound support for Fairphone 5 smartphone
by Luca Weiss 10 Aug '24

10 Aug '24

Add the necessary sound card bits and some dts additions to enable sound over DisplayPort-over-USB-C, e.g. to a connected TV or monitor. The UCM files can be found here: https://gitlab.com/postmarketOS/pmaports/-/tree/master/device/testing/devic… Two extra notes: 1. I don't quite understand whether the sound driver should have SoC-specific compatible or device-specific compatible. Some earlier patches by another author for a QCM6490 board and a QCS6490 board use device-specific compatible - but from what I can tell this is not how it's generally done for other sound drivers? 2. Unfortunately DisplayPort enablement itself for Fairphone 5 is not upstream yet. This is blocked by DSI display bringup upstream which is blocked by DSC 1:1:1 not being supported upstream yet and just working with a hacky patch. Nevertheless, DisplayPort audio was validated working with no additional sound-related changes so once DisplayPort gets enabled, sound should also just work upstream. Signed-off-by: Luca Weiss <luca.weiss(a)fairphone.com> --- Luca Weiss (3): ASoC: dt-bindings: qcom,sm8250: Add generic QCM6490 sound card ASoC: qcom: sc8280xp: Add support for generic QCM6490 arm64: dts: qcom: qcm6490-fairphone-fp5: Add DisplayPort sound support .../devicetree/bindings/sound/qcom,sm8250.yaml | 1 + arch/arm64/boot/dts/qcom/qcm6490-fairphone-fp5.dts | 37 ++++++++++++++++++++++ sound/soc/qcom/sc8280xp.c | 1 + 3 files changed, 39 insertions(+) --- base-commit: 6cdb38a56eaf615abc60fbeec0e4ccbdf93468e3 change-id: 20240809-fp5-dp-sound-b3768f3019bd Best regards, -- Luca Weiss <luca.weiss(a)fairphone.com>

4 9

[PATCH AUTOSEL 5.10 02/11] ASoC: Intel: sof_sdw: Add quirks for some new Dell laptops
by Sasha Levin 10 Aug '24

10 Aug '24

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91cdecaba791c74df6da0650e797fe1192cf2700 ] Add quirks for some new Dell laptops using Cirrus amplifiers in a bridge configuration. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://msgid.link/r/20240527193552.165567-11-pierre-louis.bossart@linux.in… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 25bf73a7e7bfa..ad3694d36d969 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -234,6 +234,22 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD2_100K), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE3") + }, + .driver_data = (void *)(SOF_SIDECAR_AMPS), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE4") + }, + .driver_data = (void *)(SOF_SIDECAR_AMPS), + }, {} }; -- 2.43.0

2 2

[PATCH] ALSA: pcm: Add xrun counter for snd_pcm_substream
by Norman Bintang 10 Aug '24

10 Aug '24

This patch adds an xrun counter to snd_pcm_substream as an alternative to using logs from XRUN_DEBUG_BASIC. The counter provides a way to track the number of xrun occurences, accessible through the /proc interface. The counter is enabled when CONFIG_SND_PCM_XRUN_DEBUG is set. Example output: $ cat /proc/asound/card0/pcm9p/sub0/status owner_pid : 1425 trigger_time: 235.248957291 tstamp : 0.000000000 delay : 1912 avail : 480 avail_max : 1920 ----- hw_ptr : 672000 appl_ptr : 673440 xrun_counter: 3 # (new row) Signed-off-by: Norman Bintang <normanbt(a)chromium.org> Reviewed-by: Chih-Yang Hsia <paulhsia(a)chromium.org> Tested-by: Chih-Yang Hsia <paulhsia(a)chromium.org> Reviewed-by: David Riley <davidriley(a)chromium.org> --- include/sound/pcm.h | 3 +++ sound/core/pcm.c | 6 ++++++ sound/core/pcm_lib.c | 3 +++ 3 files changed, 12 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index ac8f3aef9205..384032b6c59c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -498,6 +498,9 @@ struct snd_pcm_substream { /* misc flags */ unsigned int hw_opened: 1; unsigned int managed_buffer_alloc:1; +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + unsigned int xrun_counter; /* number of times xrun happens */ +#endif /* CONFIG_SND_PCM_XRUN_DEBUG */ }; #define SUBSTREAM_BUSY(substream) ((substream)->ref_count > 0) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index dc37f3508dc7..290690fc2abc 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -462,6 +462,9 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, snd_iprintf(buffer, "-----\n"); snd_iprintf(buffer, "hw_ptr : %ld\n", runtime->status->hw_ptr); snd_iprintf(buffer, "appl_ptr : %ld\n", runtime->control->appl_ptr); +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + snd_iprintf(buffer, "xrun_counter: %d\n", substream->xrun_counter); +#endif } #ifdef CONFIG_SND_PCM_XRUN_DEBUG @@ -970,6 +973,9 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, substream->pid = get_pid(task_pid(current)); pstr->substream_opened++; *rsubstream = substream; +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + substream->xrun_counter = 0; +#endif /* CONFIG_SND_PCM_XRUN_DEBUG */ return 0; } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6e7905749c4a..6eaa950504cf 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -184,6 +184,9 @@ void __snd_pcm_xrun(struct snd_pcm_substream *substream) pcm_warn(substream->pcm, "XRUN: %s\n", name); dump_stack_on_xrun(substream); } +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + substream->xrun_counter++; +#endif } #ifdef CONFIG_SND_PCM_XRUN_DEBUG -- 2.46.0.76.ge559c4bf1a-goog

1 0

[PATCH 0/3] ALSA: update sample rate definition for eARC
by Jerome Brunet 10 Aug '24

10 Aug '24

This patchset updates some sample rate definition for eARC support. Notably it: * adds the definition for 128kHz * update the ASoC spdif codecs to include all supported rates, up to 768kHz * adds missing IEC958 definition for 128k, 352.4k, 384k and 705.6kHz On that last point, as noted in the related commit description, I'd like to stress that I have found the definitions using a testing equipement. I do not have access to the IEC958 specification. It would be best if someone with access to this specification could review/ack patch #2, before applying it. Jerome Brunet (3): ALSA: pcm: add support for 128kHz sample rate ALSA: IEC958 definition for consumer status channel update ASoC: spdif: extend supported rates to 768kHz include/sound/asoundef.h | 6 +++++- include/sound/pcm.h | 13 +++++++------ sound/core/pcm_native.c | 6 +++--- sound/soc/codecs/spdif_receiver.c | 3 ++- sound/soc/codecs/spdif_transmitter.c | 3 ++- 5 files changed, 19 insertions(+), 12 deletions(-) -- 2.43.0

5 15

(Question) Difference between defaults.pcm.srate and defaults.pcm.rate
by GitHub issues - opened 09 Aug '24

09 Aug '24

alsa-project/alsa-lib issue #406 was opened from ViNi-Arco: What is the difference between `defaults.pcm.srate` and `defaults.pcm.rate`? I know there's `defaults.pcm.!rate`, but I couldn't find any information about defaults.pcm.srate I'm asking because I was given this configuration: ![IMG](https://github.com/user-attachments/assets/84316655-2d44-4f54-a842-86bd60a780e3) Issue URL : https://github.com/alsa-project/alsa-lib/issues/406 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

(Question) Difference between defaults.pcm.srate and defaults.pcm.rate
by GitHub issues - edited 09 Aug '24

09 Aug '24

alsa-project/alsa-lib issue #406 was edited from ViNi-Arco: What is the difference between `defaults.pcm.srate`? I know there's `defaults.pcm.!rate` and `defaults.pcm.rate`, but I couldn't find any information about defaults.pcm.srate I'm asking because I was given this configuration: ![IMG](https://github.com/user-attachments/assets/84316655-2d44-4f54-a842-86bd60a780e3) Issue URL : https://github.com/alsa-project/alsa-lib/issues/406 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] ASoc: tas2781: Add Calibration Kcontrols and tas2563 digtial gain for Chromebook
by Shenghao Ding 09 Aug '24

09 Aug '24

Calibrated data will be set to default after loading DSP config params, which will cause speaker protection work abnormally. Reload calibrated data after loading DSP config params. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Changed the copyright year to 2024 in the related files. - Add CAL_DAT_SZ for calibrated data size. - Add TAS2563_DVC_LVL for digtial gain kcontrol. - Add registers for TAS2563 and TAS2781 calibration. - Add cali_data_restore for regsiter restore after calibration. - Add is_user_space_calidata to store the flag where the calibrated from, user space or bin file from driver parsing. - Add TASDEVICE_RCA_FW_OK to support only register setting bin in the device. - Add tas2563_dvc_table for relationship between tas2563 digtial gains and register values. - Correct the filename in the header comments of tas2781-comlib.c, tas2781-lib.c --> tas2781-comlib.c. - tasdevice_chn_switch for chip switch among multiple chips(tas2563 or tas2781) - Add loading the calibrated values from user space in tasdev_load_calibrated_data - Correct no dsp no work, it can still work in bypass mode. - Add calibrated register setting for tas2563&tas2781. - Add mutex into each kcontrol. - rename tas2781_force_fwload_get, tas2781_force_fwload_put--> tasdev_force_fwload_get, tasdev_force_fwload_put - rename tas2781_codec --> tasdev_codec - define tas2781_snd_control and tas2563_snd_control for volume - define tas2781_cali_control and tas2563_cali_control for calibration --- include/sound/tas2781-dsp.h | 1 + include/sound/tas2781.h | 92 ++- sound/soc/codecs/tas2781-comlib.c | 35 +- sound/soc/codecs/tas2781-fmwlib.c | 51 +- sound/soc/codecs/tas2781-i2c.c | 941 +++++++++++++++++++++++++++++- 5 files changed, 1061 insertions(+), 59 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 7fba7ea26a4b..74ab6f113029 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -121,6 +121,7 @@ enum tasdevice_dsp_fw_state { TASDEVICE_DSP_FW_NONE = 0, TASDEVICE_DSP_FW_PENDING, TASDEVICE_DSP_FW_FAIL, + TASDEVICE_RCA_FW_OK, TASDEVICE_DSP_FW_ALL_OK, }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 99ca3e401fd1..2fb3d03d0040 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -23,6 +23,8 @@ #define SMARTAMP_MODULE_NAME "tas2781" #define TAS2781_GLOBAL_ADDR 0x40 #define TAS2563_GLOBAL_ADDR 0x48 +#define CAL_DAT_SZ 20 + #define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ SNDRV_PCM_RATE_88200) @@ -31,6 +33,11 @@ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + /*PAGE Control Register (available in page0 of each book) */ #define TASDEVICE_PAGE_SELECT 0x00 #define TASDEVICE_BOOKCTL_PAGE 0x00 @@ -43,21 +50,73 @@ (page * 128)) + reg) /*Software Reset */ -#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) -#define TAS2781_REG_SWRESET_RESET BIT(0) +#define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x00, 0X00, 0x01) +#define TASDEVICE_REG_SWRESET_RESET BIT(0) /*I2C Checksum */ -#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) +#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x00, 0x00, 0x7E) + +/* XM_340 */ +#define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x02, 0x4c) +/* XM_341 */ +#define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x02, 0x64) /* Volume control */ -#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) -#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) +#define TAS2781_DVC_LVL TASDEVICE_REG(0x00, 0x00, 0x1A) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x00, 0x00, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) -#define TASDEVICE_CMD_SING_W 0x1 -#define TASDEVICE_CMD_BURST 0x2 -#define TASDEVICE_CMD_DELAY 0x3 -#define TASDEVICE_CMD_FIELD_W 0x4 +#define TAS2563_PRM_R0_REG TASDEVICE_REG(0x00, 0x0f, 0x34) +#define TAS2563_PRM_R0_LOW_REG TASDEVICE_REG(0x00, 0x0f, 0x48) +#define TAS2563_PRM_INVR0_REG TASDEVICE_REG(0x00, 0x0f, 0x40) +#define TAS2563_PRM_POW_REG TASDEVICE_REG(0x00, 0x0d, 0x3c) +#define TAS2563_PRM_TLIMIT_REG TASDEVICE_REG(0x00, 0x10, 0x14) + +#define TAS2563_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x02, 0x70) +#define TAS2563_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x02, 0x48) + +#define TAS2563_PRM_ENFF_REG TASDEVICE_REG(0x00, 0x0d, 0x54) +#define TAS2563_PRM_DISTCK_REG TASDEVICE_REG(0x00, 0x0d, 0x58) +#define TAS2563_PRM_TE_SCTHR_REG TASDEVICE_REG(0x00, 0x0f, 0x60) +#define TAS2563_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x0d, 0x74) +#define TAS2563_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x0d, 0x7c) +/* prm_Int_B0 */ +#define TAS2563_TE_TA1_REG TASDEVICE_REG(0x00, 0x10, 0x0c) +/* prm_Int_A1 */ +#define TAS2563_TE_TA1_AT_REG TASDEVICE_REG(0x00, 0x10, 0x10) +/* prm_TE_Beta */ +#define TAS2563_TE_TA2_REG TASDEVICE_REG(0x00, 0x0f, 0x64) +/* prm_TE_Beta1 */ +#define TAS2563_TE_AT_REG TASDEVICE_REG(0x00, 0x0f, 0x68) +/* prm_TE_1_Beta1 */ +#define TAS2563_TE_DT_REG TASDEVICE_REG(0x00, 0x0f, 0x70) + +#define TAS2781_PRM_R0_REG TASDEVICE_REG(0x00, 0x17, 0x74) +#define TAS2781_PRM_R0_LOW_REG TASDEVICE_REG(0x00, 0x18, 0x0c) +#define TAS2781_PRM_INVR0_REG TASDEVICE_REG(0x00, 0x18, 0x14) +#define TAS2781_PRM_POW_REG TASDEVICE_REG(0x00, 0x13, 0x70) +#define TAS2781_PRM_TLIMIT_REG TASDEVICE_REG(0x00, 0x18, 0x7c) + +#define TAS2781_PRM_INT_MASK_REG TASDEVICE_REG(0x00, 0x00, 0x3b) +#define TAS2781_PRM_CLK_CFG_REG TASDEVICE_REG(0x00, 0x00, 0x5c) +#define TAS2781_PRM_RSVD_REG TASDEVICE_REG(0x00, 0x01, 0x19) +#define TAS2781_PRM_TEST_57_REG TASDEVICE_REG(0x00, 0xfd, 0x39) +#define TAS2781_PRM_TEST_62_REG TASDEVICE_REG(0x00, 0xfd, 0x3e) +#define TAS2781_PRM_PVDD_UVLO_REG TASDEVICE_REG(0x00, 0x00, 0x71) +#define TAS2781_PRM_CHNL_0_REG TASDEVICE_REG(0x00, 0x00, 0x03) +#define TAS2781_PRM_NG_CFG0_REG TASDEVICE_REG(0x00, 0x00, 0x35) +#define TAS2781_PRM_IDLE_CH_DET_REG TASDEVICE_REG(0x00, 0x00, 0x66) +#define TAS2781_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x14, 0x38) +#define TAS2781_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x14, 0x40) +#define TAS2781_PRM_SINEGAIN2_REG TASDEVICE_REG(0x00, 0x14, 0x44) + +#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xFD, 0x0D) +#define TAS2781_TEST_PAGE_UNLOCK 0x0D + +#define TAS2781_RUNTIME_LATCH_RE_REG TASDEVICE_REG(0x00, 0x00, 0x49) +#define TAS2781_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x62, 0x48) +#define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) enum audio_device { TAS2563, @@ -69,7 +128,15 @@ enum device_catlog_id { OTHERS }; +struct bulk_reg_val { + int reg; + unsigned char val[4]; + unsigned char val_len; + bool is_locked; +}; + struct tasdevice { + struct bulk_reg_val *cali_data_restore; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -88,6 +155,8 @@ struct tasdevice_irqinfo { struct calidata { unsigned char *data; unsigned long total_sz; + unsigned int *reg_array; + unsigned int reg_array_sz; }; struct tasdevice_priv { @@ -122,6 +191,7 @@ struct tasdevice_priv { bool force_fwload_status; bool playback_started; bool isacpi; + bool is_user_space_calidata; unsigned int global_addr; int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, @@ -148,6 +218,8 @@ int tasdevice_init(struct tasdevice_priv *tas_priv); void tasdevice_remove(struct tasdevice_priv *tas_priv); int tasdevice_save_calibration(struct tasdevice_priv *tas_priv); void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv); +int tasdevice_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn); int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *value); int tasdevice_dev_write(struct tasdevice_priv *tas_priv, diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 3aa81514dad7..9d7b56663b6b 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -1,8 +1,8 @@ // SPDX-License-Identifier: GPL-2.0 // -// tas2781-lib.c -- TAS2781 Common functions for HDA and ASoC Audio drivers +// tas2781-comlib.c -- TAS2781 Common functions for HDA and ASoC Audio drivers // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -89,6 +89,33 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, return ret; } +int tasdevice_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + struct regmap *map = tas_priv->regmap; + int ret; + + if (client->addr != tasdev->dev_addr) { + client->addr = tasdev->dev_addr; + /* All devices share the same regmap, clear the page + * inside regmap once switching to another device. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", + __func__, ret); + return ret; + } + return 1; + } + return 0; +} +EXPORT_SYMBOL_GPL(tasdevice_chn_switch); + int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *val) { @@ -254,8 +281,8 @@ void tas2781_reset(struct tasdevice_priv *tas_dev) } else { for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_write(tas_dev, i, - TAS2781_REG_SWRESET, - TAS2781_REG_SWRESET_RESET); + TASDEVICE_REG_SWRESET, + TASDEVICE_REG_SWRESET_RESET); if (ret < 0) dev_err(tas_dev->dev, "dev %d swreset fail, %d\n", diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 265a8ca25cbb..19a5133083dc 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2153,20 +2153,39 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) { + struct tasdevice_fw *cal_fmw = priv->tasdevice[i].cali_data_fmw; + struct calidata *cali_data = &priv->cali_data; + unsigned char *data = cali_data->data; struct tasdevice_calibration *cal; - struct tasdevice_fw *cal_fmw; + int k = i * (CAL_DAT_SZ + 1); + int j, rc; - cal_fmw = priv->tasdevice[i].cali_data_fmw; + if (!priv->is_user_space_calidata && + cal_fmw) { + cal = cal_fmw->calibrations; - /* No calibrated data for current devices, playback will go ahead. */ - if (!cal_fmw) + if (cal) + load_calib_data(priv, &cal->dev_data); return; - - cal = cal_fmw->calibrations; - if (cal) + } + if (!priv->is_user_space_calidata) return; - - load_calib_data(priv, &cal->dev_data); + /* load calibrated data from user space */ + if (data[k] != i) { + dev_err(priv->dev, "%s: no cal-data for dev %d from usr-spc\n", + __func__, i); + return; + } + for (j = 0; j < cali_data->reg_array_sz; j++) { + if (data[k] != j) + rc = tasdevice_dev_bulk_write(priv, i, + cali_data->reg_array[j], + &(data[k + 4 * j]), 4); + if (rc < 0) + dev_err(priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + i, j, rc); + } } int tasdevice_select_tuningprm_cfg(void *context, int prm_no, @@ -2261,9 +2280,10 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, tas_priv->tasdevice[i].cur_conf = cfg_no; } } - } else + } else { dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", __func__, cfg_no); + } status |= cfg_info[rca_conf_no]->active_dev; @@ -2324,13 +2344,15 @@ void tasdevice_tuning_switch(void *context, int state) struct tasdevice_fw *tas_fmw = tas_priv->fmw; int profile_cfg_id = tas_priv->rcabin.profile_cfg_id; - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + /* Only RCA file loaded still can work without speaker protection */ + if (!(tas_priv->fw_state == TASDEVICE_RCA_FW_OK || + tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK)) { + dev_err(tas_priv->dev, "No firmware loaded\n"); return; } if (state == 0) { - if (tas_priv->cur_prog < tas_fmw->nr_programs) { + if (tas_fmw && tas_priv->cur_prog < tas_fmw->nr_programs) { /*dsp mode or tuning mode*/ profile_cfg_id = tas_priv->rcabin.profile_cfg_id; tasdevice_select_tuningprm_cfg(tas_priv, @@ -2340,9 +2362,10 @@ void tasdevice_tuning_switch(void *context, int state) tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, TASDEVICE_BIN_BLK_PRE_POWER_UP); - } else + } else { tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, TASDEVICE_BIN_BLK_PRE_SHUTDOWN); + } } EXPORT_SYMBOL_NS_GPL(tasdevice_tuning_switch, SND_SOC_TAS2781_FMWLIB); diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9350972dfefe..239f3538c558 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -13,6 +13,7 @@ // Author: Kevin Lu <kevin-lu(a)ti.com> // +#include <asm/unaligned.h> #include <linux/crc8.h> #include <linux/firmware.h> #include <linux/gpio/consumer.h> @@ -31,6 +32,122 @@ #include <sound/tlv.h> #include <sound/tas2781-tlv.h> +static const struct bulk_reg_val tas2563_cali_start_reg[] = { + { + .reg = TAS2563_PRM_ENFF_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_DISTCK_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_TE_SCTHR_REG, + .val = { 0x7f, 0xff, 0xff, 0xff }, + }, + { + .reg = TAS2563_PRM_PLT_FLAG_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_SINEGAIN_REG, + .val = { 0x0a, 0x3d, 0x70, 0xa4 }, + }, + { + .reg = TAS2563_TE_TA1_REG, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_TA1_AT_REG, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_TA2_REG, + .val = { 0x00, 0x06, 0xd3, 0x72 }, + }, + { + .reg = TAS2563_TE_AT_REG, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_DT_REG, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, +}; + +static const struct bulk_reg_val tas2781_cali_start_reg[] = { + { + .reg = TAS2781_PRM_INT_MASK_REG, + .val = { 0xfe }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_CLK_CFG_REG, + .val = { 0xdd }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_RSVD_REG, + .val = { 0x20 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_TEST_57_REG, + .val = { 0x14 }, + .val_len = 1, + .is_locked = true + }, + { + .reg = TAS2781_PRM_TEST_62_REG, + .val = { 0x45 }, + .val_len = 1, + .is_locked = true + }, + { + .reg = TAS2781_PRM_PVDD_UVLO_REG, + .val = { 0x03 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_CHNL_0_REG, + .val = { 0xA8 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_NG_CFG0_REG, + .val = { 0xb9 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_IDLE_CH_DET_REG, + .val = { 0x92 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_PLT_FLAG_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_SINEGAIN_REG, + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_SINEGAIN2_REG, + .val_len = 4, + .is_locked = false + }, +}; + static const struct i2c_device_id tasdevice_id[] = { { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, @@ -47,6 +164,22 @@ static const struct of_device_id tasdevice_of_match[] = { MODULE_DEVICE_TABLE(of, tasdevice_of_match); #endif +static const int tas2563_cali_data_reg[] = { + TAS2563_PRM_R0_REG, + TAS2563_PRM_R0_LOW_REG, + TAS2563_PRM_INVR0_REG, + TAS2563_PRM_POW_REG, + TAS2563_PRM_TLIMIT_REG, +}; + +static const int tas2781_cali_data_reg[] = { + TAS2781_PRM_R0_REG, + TAS2781_PRM_R0_LOW_REG, + TAS2781_PRM_INVR0_REG, + TAS2781_PRM_POW_REG, + TAS2781_PRM_TLIMIT_REG, +}; + /** * tas2781_digital_getvol - get the volum control * @kcontrol: control pointer @@ -65,8 +198,13 @@ static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_digital_getvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_digital_getvol(tas_priv, ucontrol, mc); + return rc; } static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, @@ -76,8 +214,13 @@ static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_digital_putvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_digital_putvol(tas_priv, ucontrol, mc); + return rc; } static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, @@ -87,8 +230,13 @@ static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_amp_getvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_amp_getvol(tas_priv, ucontrol, mc); + return rc; } static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, @@ -99,11 +247,16 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_amp_putvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_amp_putvol(tas_priv, ucontrol, mc); + return rc; } -static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -118,7 +271,7 @@ static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, return 0; } -static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -127,18 +280,496 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, snd_soc_component_get_drvdata(component); bool change, val = (bool)ucontrol->value.integer.value[0]; + mutex_lock(&tas_priv->codec_lock); if (tas_priv->force_fwload_status == val) change = false; else { change = true; tas_priv->force_fwload_status = val; } + mutex_unlock(&tas_priv->codec_lock); dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, tas_priv->force_fwload_status ? "ON" : "OFF"); return change; } +static int tasdev_cali_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned char *data = tas_priv->cali_data.data; + + mutex_lock(&tas_priv->codec_lock); + if (!tas_priv->is_user_space_calidata || + tas_priv->cali_data.total_sz != bytes_ext->max) { + goto out; + } + memcpy(dst, data, bytes_ext->max); +out: + mutex_unlock(&tas_priv->codec_lock); + return 0; +} + +static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, + unsigned char *dst, int data_len) +{ + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int rc = -1; + int i; + + if (data_len != 4) + return rc; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + tasdevice_dev_bulk_read(tas_priv, i, reg, &dst[1], + 4); + rc = 0; + break; + } + } + + return rc; +} + +static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + unsigned char *dat = ucontrol->value.bytes.data; + int rc = 1; + int i, j; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2781 && + bytes_ext->max != 8 * tas_priv->ndev) { + rc = -1; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + int k = i * 9; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_read(tas_priv, i, p[j].reg, + (int *)&p[j].val[0]); + } else + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); + } + + for (j = 0; j < sum - 2; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + tas2781_cali_start_reg[j].val[0]); + } else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + (unsigned char *) + tas2781_cali_start_reg[j].val, 4); + } + + if (dat[k] != i) { + dev_err(tas_priv->dev, + "%s: no cal-setting for dev %d\n", __func__, + i); + continue; + } + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + &dat[i * 9 + 1], 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, + &dat[i * 9 + 5], 4); + } +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static void tas2781_calib_stop_put(struct tasdevice_priv *tas_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; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + } else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int rc = 1; + int i, j; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2563) { + rc = -1; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + for (j = 0; j < sum; j++) { + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); + } + + for (j = 0; j < sum; j++) { + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + (unsigned char *)tas2563_cali_start_reg[j].val, + 4); + } + } +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static void tas2563_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tasdev_calib_stop_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id == TAS2563) + tas2563_calib_stop_put(tas_priv); + else + tas2781_calib_stop_put(tas_priv); + mutex_unlock(&tas_priv->codec_lock); + + return 1; +} + +static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *data = tas_priv->cali_data.data; + int rc = 1; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->cali_data.total_sz != bytes_ext->max) { + rc = -1; + goto out; + } + tas_priv->is_user_space_calidata = true; + memcpy(data, ucontrol->value.bytes.data, bytes_ext->max); +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static int tas2781_latch_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + unsigned char *dst = ucontrol->value.bytes.data; + int i, val, rc = -1; + + mutex_lock(&tas_priv->codec_lock); + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + tasdevice_dev_read(tas_priv, i, + TAS2781_RUNTIME_LATCH_RE_REG, &val); + dst[1] = val; + rc = 0; + break; + } + } + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG_TF; + else + reg = TAS2563_RUNTIME_RE_REG_TF; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG; + else + reg = TAS2563_RUNTIME_RE_REG; + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_PRM_R0_REG; + else + reg = TAS2563_PRM_R0_REG; + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + 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; + int rc = -1; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + 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; + int rc = -1; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_nop_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + +static int tas2563_digital_gain_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int l = 0, r = mc->max; + unsigned int target, ar_mid, mid, ar_l, ar_r; + unsigned int reg = mc->reg; + unsigned char data[4]; + int ret; + + mutex_lock(&tas_dev->codec_lock); + /* Read the primary device */ + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + if (ret) { + dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + target = get_unaligned_be32(&data[0]); + + while (r > 1 + l) { + mid = (l + r) / 2; + ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + if (target < ar_mid) + r = mid; + else + l = mid; + } + + ar_l = get_unaligned_be32(tas2563_dvc_table[l]); + ar_r = get_unaligned_be32(tas2563_dvc_table[r]); + + ucontrol->value.integer.value[0] = + abs(target - ar_l) <= abs(target - ar_r) ? l : r; +out: + mutex_unlock(&tas_dev->codec_lock); + return 0; +} + +static int tas2563_digital_gain_put( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int reg = mc->reg; + unsigned int volrd, volwr; + int vol = ucontrol->value.integer.value[0]; + int max = mc->max, i, ret = 1; + unsigned char data[4]; + + vol = clamp(vol, 0, max); + mutex_lock(&tas_dev->codec_lock); + /* Read the primary device */ + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + if (ret) { + dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + volrd = get_unaligned_be32(&data[0]); + volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + + if (volrd == volwr) { + ret = 0; + goto out; + } + + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_bulk_write(tas_dev, i, reg, + (unsigned char *)tas2563_dvc_table[vol], 4); + if (ret) + dev_err(tas_dev->dev, + "%s, set digital vol error in device %d\n", + __func__, i); + } + +out: + mutex_unlock(&tas_dev->codec_lock); + return ret; +} + +static const struct snd_kcontrol_new tasdevice_snd_controls[] = { + SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tasdev_force_fwload_get, tasdev_force_fwload_put), +}; + +static const struct snd_kcontrol_new tasdevice_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tasdev_calib_stop_put), + SND_SOC_BYTES_EXT("Amp TF Data", 5, tasdev_tf_data_get, NULL), + SND_SOC_BYTES_EXT("Amp RE Data", 5, tasdev_re_data_get, NULL), + SND_SOC_BYTES_EXT("Amp R0 Data", 5, tasdev_r0_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA1 Data", 5, tasdev_XMA1_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA2 Data", 5, tasdev_XMA2_data_get, NULL), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -146,8 +777,22 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), - SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, - tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static const struct snd_kcontrol_new tas2781_cali_controls[] = { + SND_SOC_BYTES_EXT("Amp Latch Data", 2, tas2781_latch_reg_get, NULL), +}; + +static const struct snd_kcontrol_new tas2563_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2563_DVC_LVL, 0, + 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, + tas2563_digital_gain_get, tas2563_digital_gain_put, + tas2563_dvc_tlv), +}; + +static const struct snd_kcontrol_new tas2563_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Start", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tas2563_calib_start_put), }; static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, @@ -167,6 +812,31 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, return ret; } +static int tasdevice_info_active_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_priv->ndev - 1; + + return 0; +} + +static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = TAS2563; + uinfo->value.integer.max = TAS2781; + + return 0; +} + static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -223,6 +893,17 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, return 0; } +static int tasdevice_get_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->chip_id; + + return 0; +} + static int tasdevice_create_control(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *prof_ctrls; @@ -316,15 +997,54 @@ static int tasdevice_configuration_put( return ret; } -static int tasdevice_dsp_create_ctrls( - struct tasdevice_priv *tas_priv) +static int tasdevice_active_id_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + ucontrol->value.integer.value[0] = i; + return 0; + } + } + + return -1; +} + +static int tasdevice_active_id_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int nr_dev = ucontrol->value.integer.value[0]; + int max = tas_priv->ndev - 1, rc; + + nr_dev = clamp(nr_dev, 0, max); + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_chn_switch(tas_priv, nr_dev); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { + char *active_dev_name, *cali_name, *chip_id, *conf_name, *prog_name; + struct calidata *cali_data = &tas_priv->cali_data; struct snd_kcontrol_new *dsp_ctrls; - char *prog_name, *conf_name; - int nr_controls = 2; + struct soc_bytes_ext *ext_cali_data; + int nr_controls = 5; int mix_index = 0; int ret; + if (tas_priv->chip_id == TAS2781) + nr_controls++; /* Alloc kcontrol via devm_kzalloc, which don't manually * free the kcontrol */ @@ -336,11 +1056,20 @@ static int tasdevice_dsp_create_ctrls( } /* Create a mixer item for selecting the active profile */ - prog_name = devm_kzalloc(tas_priv->dev, + active_dev_name = devm_kzalloc(tas_priv->dev, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); + cali_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + chip_id = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + prog_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - if (!prog_name || !conf_name) { + ext_cali_data = devm_kzalloc(tas_priv->dev, sizeof(*ext_cali_data), + GFP_KERNEL); + if (!active_dev_name || !cali_name || !conf_name || !chip_id || + !ext_cali_data || !prog_name) { ret = -ENOMEM; goto out; } @@ -363,6 +1092,68 @@ static int tasdevice_dsp_create_ctrls( dsp_ctrls[mix_index].put = tasdevice_configuration_put; mix_index++; + scnprintf(active_dev_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "Activate Tasdevice Id"); + dsp_ctrls[mix_index].name = active_dev_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_active_id; + dsp_ctrls[mix_index].get = tasdevice_active_id_get; + dsp_ctrls[mix_index].put = tasdevice_active_id_put; + mix_index++; + + scnprintf(chip_id, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "Tasdevice Chip Id"); + dsp_ctrls[mix_index].name = chip_id; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_chip_id; + dsp_ctrls[mix_index].get = tasdevice_get_chip_id; + mix_index++; + + scnprintf(cali_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "Speaker Calibrated Data"); + ext_cali_data->max = tas_priv->ndev * CAL_DAT_SZ; + tas_priv->cali_data.total_sz = ext_cali_data->max; + tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + ext_cali_data->max, GFP_KERNEL); + dsp_ctrls[mix_index].name = cali_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_bytes_info_ext; + dsp_ctrls[mix_index].get = tasdev_cali_data_get; + dsp_ctrls[mix_index].put = tasdev_cali_data_put; + dsp_ctrls[mix_index].private_value = (unsigned long)ext_cali_data; + mix_index++; + + cali_data->data = devm_kzalloc(tas_priv->dev, tas_priv->ndev * + (cali_data->reg_array_sz * 4 + 1), GFP_KERNEL); + if (!cali_data->data) + return -ENOMEM; + + if (tas_priv->chip_id == TAS2781) { + struct soc_bytes_ext *ext_cali_start; + char *cali_start_name; + + cali_start_name = devm_kzalloc(tas_priv->dev, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); + ext_cali_start = devm_kzalloc(tas_priv->dev, + sizeof(*ext_cali_start), GFP_KERNEL); + if (!cali_start_name || !ext_cali_start) { + ret = -ENOMEM; + goto out; + } + + scnprintf(cali_start_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "Calibration Start"); + ext_cali_data->max = tas_priv->ndev * 9; + dsp_ctrls[mix_index].name = cali_start_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_bytes_info_ext; + dsp_ctrls[mix_index].put = tas2781_calib_start_put; + dsp_ctrls[mix_index].get = tasdev_nop_get; + dsp_ctrls[mix_index].private_value = + (unsigned long)ext_cali_start; + mix_index++; + } + ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); @@ -370,6 +1161,54 @@ static int tasdevice_dsp_create_ctrls( return ret; } +static int tasdevice_create_cali_ctrls(struct tasdevice_priv *tas_priv) +{ + struct calidata *cali_data = &tas_priv->cali_data; + struct snd_kcontrol_new *cali_ctrls; + unsigned int num_controls; + int rc; + + rc = snd_soc_add_component_controls(tas_priv->codec, + tasdevice_cali_controls, ARRAY_SIZE(tasdevice_cali_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add cali control err rc = %d", + __func__, rc); + return rc; + } + + if (tas_priv->chip_id == TAS2781) { + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; + num_controls = ARRAY_SIZE(tas2781_cali_controls); + cali_data->reg_array = (unsigned int *)tas2781_cali_data_reg; + cali_data->reg_array_sz = ARRAY_SIZE(tas2781_cali_data_reg); + } else { + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; + num_controls = ARRAY_SIZE(tas2563_cali_controls); + cali_data->reg_array = (unsigned int *)tas2563_cali_data_reg; + cali_data->reg_array_sz = sizeof(tas2563_cali_data_reg); + for (i = 0; i < tas_priv->ndev; i++) { + tasdev[i].cali_data_restore = + kmemdup(tas2563_cali_start_reg, + sizeof(tas2563_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_restore) + return -ENOMEM; + } + } + + rc = snd_soc_add_component_controls(tas_priv->codec, cali_ctrls, + num_controls); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + + return rc; +} + static void tasdevice_fw_ready(const struct firmware *fmw, void *context) { @@ -380,23 +1219,36 @@ static void tasdevice_fw_ready(const struct firmware *fmw, mutex_lock(&tas_priv->codec_lock); ret = tasdevice_rca_parser(tas_priv, fmw); - if (ret) + if (ret) { + tasdevice_config_info_remove(tas_priv); goto out; + } tasdevice_create_control(tas_priv); tasdevice_dsp_remove(tas_priv); tasdevice_calbin_remove(tas_priv); - tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + tas_priv->fw_state = TASDEVICE_RCA_FW_OK; scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", tas_priv->dev_name); + ret = tasdevice_dsp_parser(tas_priv); if (ret) { dev_err(tas_priv->dev, "dspfw load %s error\n", tas_priv->coef_binaryname); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; goto out; } - tasdevice_dsp_create_ctrls(tas_priv); + + ret = tasdevice_dsp_create_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dsp controls error\n"); + goto out; + } + + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; @@ -417,9 +1269,8 @@ static void tasdevice_fw_ready(const struct firmware *fmw, tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; out: - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - /*If DSP FW fail, kcontrol won't be created */ - tasdevice_config_info_remove(tas_priv); + if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { + /*If DSP FW fail, DSP kcontrol won't be created */ tasdevice_dsp_remove(tas_priv); } mutex_unlock(&tas_priv->codec_lock); @@ -466,14 +1317,14 @@ static int tasdevice_startup(struct snd_pcm_substream *substream, { struct snd_soc_component *codec = dai->component; struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); - int ret = 0; - if (tas_priv->fw_state != TASDEVICE_DSP_FW_ALL_OK) { - dev_err(tas_priv->dev, "DSP bin file not loaded\n"); - ret = -EINVAL; + if (!(tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK || + tas_priv->fw_state == TASDEVICE_RCA_FW_OK)) { + dev_err(tas_priv->dev, "Bin file not loaded\n"); + return -EINVAL; } - return ret; + return 0; } static int tasdevice_hw_params(struct snd_pcm_substream *substream, @@ -541,7 +1392,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", @@ -565,13 +1416,41 @@ static struct snd_soc_dai_driver tasdevice_dai_driver[] = { static int tasdevice_codec_probe(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int rc; + + if (tas_priv->chip_id == TAS2781) { + rc = snd_soc_add_component_controls(codec, + tas2781_snd_controls, + ARRAY_SIZE(tas2781_snd_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + } else { + rc = snd_soc_add_component_controls(codec, + tas2563_snd_controls, + ARRAY_SIZE(tas2563_snd_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + } + + rc = tascodec_init(tas_priv, codec, THIS_MODULE, tasdevice_fw_ready); - return tascodec_init(tas_priv, codec, THIS_MODULE, tasdevice_fw_ready); + return rc; } static void tasdevice_deinit(void *context) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) + kfree(tasdev[i].cali_data_restore); tasdevice_config_info_remove(tas_priv); tasdevice_dsp_remove(tas_priv); @@ -591,8 +1470,8 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .controls = tas2781_snd_controls, - .num_controls = ARRAY_SIZE(tas2781_snd_controls), + .controls = tasdevice_snd_controls, + .num_controls = ARRAY_SIZE(tasdevice_snd_controls), .dapm_widgets = tasdevice_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), .dapm_routes = tasdevice_audio_map, -- 2.34.1

5 6

[GIT PULL] ASoC fixes for v6.11-rc2
by Mark Brown 09 Aug '24

09 Aug '24

The following changes since commit 8400291e289ee6b2bf9779ff1c83a291501f017b: Linux 6.11-rc1 (2024-07-28 14:19:55 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.11-rc2 for you to fetch changes up to 72776774b55bb59b7b1b09117e915a5030110304: ASoC: cs35l56: Patch CS35L56_IRQ1_MASK_18 to the default value (2024-08-08 20:34:56 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.11 Quite a lot of fixes have come in since the merge window, there's some repetitive fixes over the Qualcomm drivers increasing the patch count, along with a large batch of fixes from Cirrus. We also have some quirks and some individual fixes. ---------------------------------------------------------------- Bruno Ancona (1): ASoC: amd: yc: Support mic on HP 14-em0002la Charles Keepax (5): ASoC: cs42l43: Remove redundant semi-colon at end of function ASoC: soc-component: Add new snd_soc_component_get_kcontrol() helpers ASoC: cs35l45: Use new snd_soc_component_get_kcontrol_locked() helper ASoC: cs42l43: Cache shutter IRQ control pointers MAINTAINERS: Update Cirrus Logic parts to linux-sound mailing list Curtis Malainey (1): ASoC: SOF: Remove libraries from topology lookups Francesco Dolcini (1): ASoC: nau8822: Lower debug print priority Jerome Audu (1): ASoC: sti: add missing probe entry for player and reader Jerome Brunet (1): ASoC: meson: axg-fifo: fix irq scheduling issue with PREEMPT_RT Johan Hovold (1): ASoC: codecs: lpass-macro: fix missing codec version Krzysztof Kozlowski (10): ASoC: codecs: wcd937x-sdw: Correct Soundwire ports mask ASoC: codecs: wcd938x-sdw: Correct Soundwire ports mask ASoC: codecs: wcd939x-sdw: Correct Soundwire ports mask ASoC: codecs: wsa881x: Correct Soundwire ports mask ASoC: codecs: wsa883x: Correct Soundwire ports mask ASoC: codecs: wsa884x: Correct Soundwire ports mask ASoC: dt-bindings: qcom,wcd937x: Correct reset GPIO polarity in example ASoC: dt-bindings: qcom,wcd934x: Correct reset GPIO polarity in example ASoC: dt-bindings: qcom,wcd938x: Correct reset GPIO polarity in example ASoC: dt-bindings: qcom,wcd939x: Correct reset GPIO polarity in example Krzysztof Stępniak (1): ASoC: amd: yc: Support mic on Lenovo Thinkpad E14 Gen 6 Mark Brown (3): ASoC: codecs: wcd93xx/wsa88xx: Correct Soundwire ports ASoC: fsl_micfil: Check the difference for i.MX8 and ASoC: Extend wm_adsp so cs35l56 can suppress controls Paul Handrigan (1): ASoC: cs530x: Change IN HPF Select kcontrol name Richard Fitzgerald (2): ASoC: cs-amp-lib: Fix NULL pointer crash if efi.get_variable is NULL ASoC: cs35l56: Handle OTP read latency over SoundWire Shengjiu Wang (2): ASoC: fsl_micfil: Expand the range of FIFO watermark mask ASoC: fsl_micfil: Differentiate register access permission for platforms Simon Trimmer (3): ASoC: wm_adsp: Add control_add callback and export wm_adsp_control_add() ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients ASoC: cs35l56: Patch CS35L56_IRQ1_MASK_18 to the default value Takashi Iwai (1): ASoC: amd: yc: Add quirk entry for OMEN by HP Gaming Laptop 16-n0xxx Zhang Yi (1): ASoC: codecs: ES8326: button detect issue .../devicetree/bindings/sound/qcom,wcd934x.yaml | 3 +- .../devicetree/bindings/sound/qcom,wcd937x.yaml | 2 +- .../devicetree/bindings/sound/qcom,wcd938x.yaml | 3 +- .../devicetree/bindings/sound/qcom,wcd939x.yaml | 4 +- MAINTAINERS | 4 +- include/sound/cs35l56.h | 5 ++ include/sound/soc-component.h | 5 ++ sound/soc/amd/yc/acp6x-mach.c | 21 ++++++ sound/soc/codecs/cs-amp-lib.c | 2 +- sound/soc/codecs/cs35l45.c | 11 +--- sound/soc/codecs/cs35l56-sdw.c | 77 ++++++++++++++++++++++ sound/soc/codecs/cs35l56-shared.c | 1 + sound/soc/codecs/cs35l56.c | 11 ++++ sound/soc/codecs/cs42l43.c | 75 ++++++++++++++++----- sound/soc/codecs/cs42l43.h | 2 + sound/soc/codecs/cs530x.c | 8 +-- sound/soc/codecs/es8326.c | 2 + sound/soc/codecs/lpass-va-macro.c | 2 + sound/soc/codecs/nau8822.c | 2 +- sound/soc/codecs/wcd937x-sdw.c | 4 +- sound/soc/codecs/wcd938x-sdw.c | 4 +- sound/soc/codecs/wcd939x-sdw.c | 4 +- sound/soc/codecs/wm_adsp.c | 17 ++++- sound/soc/codecs/wm_adsp.h | 3 + sound/soc/codecs/wsa881x.c | 2 +- sound/soc/codecs/wsa883x.c | 2 +- sound/soc/codecs/wsa884x.c | 2 +- sound/soc/fsl/fsl_micfil.c | 20 ++++-- sound/soc/fsl/fsl_micfil.h | 2 +- sound/soc/meson/axg-fifo.c | 26 +++----- sound/soc/soc-component.c | 42 +++++++++--- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- sound/soc/sti/sti_uniperif.c | 2 +- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 1 + sound/soc/sti/uniperif_reader.c | 1 + 36 files changed, 291 insertions(+), 84 deletions(-)

2 1

[PATCH 0/2] ASoC: codecs: warn on unknown codec version
by Dmitry Baryshkov 09 Aug '24

09 Aug '24

While debugging broken audio issues on some of Qualcomm platforms I stumbled upon the kernel not providing the actual error information. It prints an error from the wsa_macro driver, but the actual issue is in the VA macro driver. Add error message to point to the actual error location. va_macro 3370000.codec: Unknown VA Codec version, ID: 00 / 0f / 00 wsa_macro 3240000.codec: Unsupported Codec version (0) Signed-off-by: Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org> --- Dmitry Baryshkov (2): ASoC: codecs: lpass-macro: fix version strings returned for 1.x codecs ASoC: codecs: lpass-va-macro: warn on unknown version sound/soc/codecs/lpass-macro-common.h | 6 ++++++ sound/soc/codecs/lpass-va-macro.c | 4 ++++ 2 files changed, 10 insertions(+) --- base-commit: 668d33c9ff922c4590c58754ab064aaf53c387dd change-id: 20240803-codec-version-db6f65ab51ce Best regards, -- Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org>

2 3

[tiwai-sound:for-next 69/83] sound/pcmcia/vx/vxpocket.c:273:12: error: use of undeclared identifier 'pdev'; did you mean 'p_dev'?
by kernel test robot 09 Aug '24

09 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: c46fc83e3f3c89f18962e43890de90b1c304747a commit: 2acbb5e57230830016e0fb2d61886e7a8c7f59ce [69/83] ALSA: vxpocket: Use standard print API config: x86_64-allyesconfig (https://download.01.org/0day-ci/archive/20240809/202408090809.TAYiyrqJ-lkp@…) compiler: clang version 18.1.5 (https://github.com/llvm/llvm-project 617a15a9eac96088ae5e9134248d8236e34b91b1) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090809.TAYiyrqJ-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090809.TAYiyrqJ-lkp@intel.com/ Note: the tiwai-sound/for-next HEAD c46fc83e3f3c89f18962e43890de90b1c304747a builds fine. It only hurts bisectability. All errors (new ones prefixed by >>): sound/pcmcia/vx/vxpocket.c:207:8: error: no member named 'dev' in 'struct vx_core' 207 | chip->dev = &link->dev; | ~~~~ ^ >> sound/pcmcia/vx/vxpocket.c:273:12: error: use of undeclared identifier 'pdev'; did you mean 'p_dev'? 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~ | p_dev include/linux/dev_printk.h:154:44: note: expanded from macro 'dev_err' 154 | dev_printk_index_wrap(_dev_err, KERN_ERR, dev, dev_fmt(fmt), ##__VA_ARGS__) | ^ include/linux/dev_printk.h:110:11: note: expanded from macro 'dev_printk_index_wrap' 110 | _p_func(dev, fmt, ##__VA_ARGS__); \ | ^ sound/pcmcia/vx/vxpocket.c:251:49: note: 'p_dev' declared here 251 | static int vxpocket_probe(struct pcmcia_device *p_dev) | ^ 2 errors generated. vim +273 sound/pcmcia/vx/vxpocket.c 247 248 249 /* 250 */ 251 static int vxpocket_probe(struct pcmcia_device *p_dev) 252 { 253 struct snd_card *card; 254 struct snd_vxpocket *vxp; 255 int i, err; 256 257 /* find an empty slot from the card list */ 258 for (i = 0; i < SNDRV_CARDS; i++) { 259 if (!(card_alloc & (1 << i))) 260 break; 261 } 262 if (i >= SNDRV_CARDS) { 263 dev_err(&p_dev->dev, "vxpocket: too many cards found\n"); 264 return -EINVAL; 265 } 266 if (! enable[i]) 267 return -ENODEV; /* disabled explicitly */ 268 269 /* ok, create a card instance */ 270 err = snd_card_new(&p_dev->dev, index[i], id[i], THIS_MODULE, 271 0, &card); 272 if (err < 0) { > 273 dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); 274 return err; 275 } 276 277 err = snd_vxpocket_new(card, ibl[i], p_dev, &vxp); 278 if (err < 0) { 279 snd_card_free(card); 280 return err; 281 } 282 card->private_data = vxp; 283 284 vxp->index = i; 285 card_alloc |= 1 << i; 286 287 vxp->p_dev = p_dev; 288 289 return vxpocket_config(p_dev); 290 } 291 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0