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Segfault in snd_config_delete due to incorrect node deletion in parse_array_def / parse_def
by GitHub issues - opened 09 Sep '25

09 Sep '25

alsa-project/alsa-lib issue #477 was opened from midzishi: While fuzzing snd_config_load() a reproducible segmentation fault inside snd_config_delete() was discovered. The root cause is incorrect cleanup logic in the configuration parser functions (parse_array_def() and parse_def()). In multiple branches, existing configuration nodes are replaced incorrectly: - The parser sometimes calls snd_config_delete(n) where n is NULL or a newly allocated node, instead of deleting the existing node returned by _snd_config_search() (variable g). - In other cases, a newly created node is added to the parent before its contents are successfully parsed. If parsing fails, the rollback path deletes this node explicitly, and later the parent deletes it again during recursive cleanup, so it causes double-free or corrupted linked list. This results in corrupted parent–child relationships and eventually SEGV when snd_config_delete() tries to read config->refcount of an already freed node. In both parse_array_def() and parse_def(), _snd_config_make_add() is called too early (before parsing child contents). On error, the parser deletes the child manually, but the parent will later delete it again during recursive cleanup. Proposed fix: ``` diff --git a/src/conf.c b/src/conf.c index 905c8f4d..7e403192 100644 --- a/src/conf.c +++ b/src/conf.c @@ -1274,7 +1274,8 @@ static int parse_array_def(snd_config_t *parent, input_t *input, int *idx, int s snprintf(static_id, sizeof(static_id), "%i", *idx); if (_snd_config_search(parent, static_id, -1, &g) == 0) { if (override) { - snd_config_delete(n); + snd_config_delete(g); + g = NULL; } else { /* merge */ (*idx)++; @@ -1305,7 +1306,7 @@ static int parse_array_def(snd_config_t *parent, input_t *input, int *idx, int s goto __end; } } else { - err = _snd_config_make_add(&n, &id, SND_CONFIG_TYPE_COMPOUND, parent); + err = _snd_config_make(&n, &id, SND_CONFIG_TYPE_COMPOUND); if (err < 0) goto __end; } @@ -1328,6 +1329,10 @@ static int parse_array_def(snd_config_t *parent, input_t *input, int *idx, int s err = LOCAL_UNEXPECTED_CHAR; goto __end; } + if (!skip && n && !n->parent) { + err = snd_config_add(parent, n); + if (err < 0) { snd_config_delete(n); goto __end; } + } break; } default: @@ -1422,7 +1427,7 @@ static int parse_def(snd_config_t *parent, input_t *input, int skip, int overrid err = -ENOENT; goto __end; } - err = _snd_config_make_add(&n, &id, SND_CONFIG_TYPE_COMPOUND, parent); + err = _snd_config_make(&n, &id, SND_CONFIG_TYPE_COMPOUND); if (err < 0) goto __end; n->u.compound.join = true; @@ -1464,7 +1469,7 @@ static int parse_def(snd_config_t *parent, input_t *input, int skip, int overrid goto __end; } } else { - err = _snd_config_make_add(&n, &id, SND_CONFIG_TYPE_COMPOUND, parent); + err = _snd_config_make(&n, &id, SND_CONFIG_TYPE_COMPOUND); if (err < 0) goto __end; } @@ -1483,6 +1488,10 @@ static int parse_def(snd_config_t *parent, input_t *input, int skip, int overrid err = LOCAL_UNEXPECTED_CHAR; goto __end; } + if (!skip && n && !n->parent) { + err = snd_config_add(parent, n); + if (err < 0) { snd_config_delete(n); goto __end; } + } break; } default: ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/477 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v4 1/2] ASoC: tas2781: Add tas2118, tas2x20, tas5825 support
by Baojun Xu 09 Sep '25

09 Sep '25

Add tas2020, tas2118, tas2120, tas2320, tas2570, tas2572, tas5825 tas5827 support in tas2781 driver. Tas2118, tas2x20, tas257x have no on-chip DSP, tas582x have on-chip DSP but have no calibration required stereo smart amplifier. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v4: - Change description for adding tas2570, tas2572 and tas5827 - Add tas2570, tas2572, tas5827 in device enumerate - Change the check for reset value, from "== TAS5825" to ">= TAS5825" as TAS5827 was added. - Add tas2570, tas2572, tas5827 in i2c_device_id list - Add tas2570, tas2572, tas5827 in of_device_id list - Change max of device ID from TAS2781 to TAS_OTHERS - Change the check for non-DSP chip, from "< TAS2563" to switch case, as TAS2570 and TAS2572 also have no on-chip DSP - Add TAS5827 in comments for calibration check - Change the check for on-chip DSP, from ">= TAS2563" to switch case, as TAS2570 and TAS2572 also have no on-chip DSP - Add tas2570, tas2572, tas5827 initialyze in codec probe - Add tas2570, tas2572, tas5827 in ACPI device id list v3: - No update. v2: - Update the mail list for maintainers of yaml file --- include/sound/tas2781.h | 14 +- include/sound/tas2x20-tlv.h | 259 ++++++++++++++++++++++++++ sound/soc/codecs/tas2781-comlib-i2c.c | 2 + sound/soc/codecs/tas2781-i2c.c | 184 +++++++++++++----- 4 files changed, 410 insertions(+), 49 deletions(-) create mode 100644 include/sound/tas2x20-tlv.h diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index f0aefc04a957..ddd997ac3216 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -51,7 +51,9 @@ /* Software Reset, compatble with new device (TAS5825). */ #define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) -#define TASDEVICE_REG_SWRESET_RESET (BIT(0) | BIT(4)) +#define TASDEVICE_REG_SWRESET_RESET BIT(0) + +#define TAS5825_REG_SWRESET_RESET (BIT(0) | BIT(4)) /* Checksum */ #define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) @@ -110,8 +112,17 @@ #define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) enum audio_device { + TAS2020, + TAS2118, + TAS2120, + TAS2320, TAS2563, + TAS2570, + TAS2572, TAS2781, + TAS5825, + TAS5827, + TAS_OTHERS, }; enum dspbin_type { @@ -194,6 +205,7 @@ struct tasdevice_priv { unsigned char coef_binaryname[64]; unsigned char rca_binaryname[64]; unsigned char dev_name[32]; + const unsigned char (*dvc_tlv_table)[4]; const char *name_prefix; unsigned char ndev; unsigned int dspbin_typ; diff --git a/include/sound/tas2x20-tlv.h b/include/sound/tas2x20-tlv.h new file mode 100644 index 000000000000..6e6bcec4a0a1 --- /dev/null +++ b/include/sound/tas2x20-tlv.h @@ -0,0 +1,259 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2x20/TAS2118 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2x20/TAS2118 hda driver implements for one, two, or even multiple +// TAS2x20/TAS2118 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2X20_TLV_H__ +#define __TAS2X20_TLV_H__ + +#define TAS2X20_DVC_LEVEL TASDEVICE_REG(0x0, 0x2, 0x0c) +#define TAS2X20_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x07) + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_amp_tlv, 2100, 50, 0); + +/* pow(10, db/20) * pow(2,22) */ +static const __maybe_unused unsigned char tas2x20_dvc_table[][4] = { + { 0X00, 0X00, 0X0D, 0X00 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0X00 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0X00 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0X00 }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0X00 }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0X00 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X00 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X00 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X00 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0X00 }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X00 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X00 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0X00 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X00 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X00 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X00 }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X00 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X00 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0X00 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X00 }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X00 }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0X00 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0X00 }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0X00 }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X00 }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0X00 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X00 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X00 }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X00 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X00 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0X00 }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0X00 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0X00 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X00 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X00 }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X00 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X00 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X00 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0X00 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X00 }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0X00 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0X00 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0X00 }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0X00 }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X00 }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X00 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X00 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0X00 }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0X00 }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0X00 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0X00 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X00 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0X00 }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X00 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X00 }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0X00 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0X00 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0X00 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X00 }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X00 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X00 }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X00 }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X00 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X00 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X00 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X00 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0X00 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X00 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0X00 }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X00 }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0X00 }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X00 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0X00 }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0X00 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X00 }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0X00 }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X00 }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X00 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X00 }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0X00 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X00 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X00 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0X00 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0X00 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X00 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X00 }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X00 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0X00 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0X00 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X00 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X00 }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X00 }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X00 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X00 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0X00 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X00 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X00 }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0X00 }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X00 }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0X00 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X00 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0X00 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X00 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X00 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X00 }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0X00 }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X00 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X00 }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0X00 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X00 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0X00 }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X00 }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0X00 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X00 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X00 }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0X00 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X00 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X00 }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X00 }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X00 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0X00 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0X00 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0X00 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X00 }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X00 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X00 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X00 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0X00 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X00 }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0X00 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X00 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X00 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X00 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0X00 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X00 }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0X00 }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0X00 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0X00 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X00 }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X00 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0X00 }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X00 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X00 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0X00 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X00 }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0X00 }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X00 }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X00 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X00 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0X00 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X00 }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X00 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0X00 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X00 }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X00 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0X00 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X00 }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X00 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X00 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0X00 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X00 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0X00 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X00 }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X00 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0X00 }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X00 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X00 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X00 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X00 }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0X00 }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0X00 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0X00 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0X00 }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X00 }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X00 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0X00 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0X00 }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0X00 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X00 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X00 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0X00 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0X00 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X00 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X00 }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X00 }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0X00 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X00 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X00 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X00 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0X00 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0X00 }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X00 }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0X00 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X00 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X00 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0X00 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X00 }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X00 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X00 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0X00 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X00 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X00 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X00 }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X00 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0X00 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X00 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X00 }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0X00 }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X00 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X00 }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0X00 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X00 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0X00 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X00 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X00 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X00 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X00 }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X00 }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X00 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X00 }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0X00 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0X00 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X00 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0X00 }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X00 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0X00 }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X00 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X00 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0X00 }, /* 5.5db */ + { 0X7F, 0XB2, 0X61, 0X00 }, /* 6.0db */ +}; +#endif diff --git a/sound/soc/codecs/tas2781-comlib-i2c.c b/sound/soc/codecs/tas2781-comlib-i2c.c index c078bb0a8437..b3fd7350143b 100644 --- a/sound/soc/codecs/tas2781-comlib-i2c.c +++ b/sound/soc/codecs/tas2781-comlib-i2c.c @@ -320,6 +320,8 @@ void tasdevice_reset(struct tasdevice_priv *tas_dev) for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_write(tas_dev, i, TASDEVICE_REG_SWRESET, + tas_dev->chip_id >= TAS5825 ? + TAS5825_REG_SWRESET_RESET : TASDEVICE_REG_SWRESET_RESET); if (ret < 0) dev_err(tas_dev->dev, diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index ea3cdb8553de..1539b70881d1 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,8 +30,10 @@ #include <sound/tas2781.h> #include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> +#include <sound/tas2x20-tlv.h> #include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> +#include <sound/tas5825-tlv.h> #include <linux/unaligned.h> #define X2563_CL_STT_VAL(xreg, xval) \ @@ -98,16 +100,32 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { }; static const struct i2c_device_id tasdevice_id[] = { + { "tas2020", TAS2020 }, + { "tas2118", TAS2118 }, + { "tas2120", TAS2120 }, + { "tas2320", TAS2320 }, { "tas2563", TAS2563 }, + { "tas2570", TAS2570 }, + { "tas2572", TAS2572 }, { "tas2781", TAS2781 }, + { "tas5825", TAS5825 }, + { "tas5827", TAS5827 }, {} }; MODULE_DEVICE_TABLE(i2c, tasdevice_id); #ifdef CONFIG_OF static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2020" }, + { .compatible = "ti,tas2118" }, + { .compatible = "ti,tas2120" }, + { .compatible = "ti,tas2320" }, { .compatible = "ti,tas2563" }, + { .compatible = "ti,tas2570" }, + { .compatible = "ti,tas2572" }, { .compatible = "ti,tas2781" }, + { .compatible = "ti,tas5825" }, + { .compatible = "ti,tas5827" }, {}, }; MODULE_DEVICE_TABLE(of, tasdevice_of_match); @@ -797,7 +815,7 @@ static int tasdev_nop_get( return 0; } -static int tas2563_digital_gain_get( +static int tasdevice_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -823,15 +841,15 @@ static int tas2563_digital_gain_get( while (r > 1 + l) { mid = (l + r) / 2; - ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + ar_mid = get_unaligned_be32(tas_dev->dvc_tlv_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]); + ar_l = get_unaligned_be32(tas_dev->dvc_tlv_table[l]); + ar_r = get_unaligned_be32(tas_dev->dvc_tlv_table[r]); /* find out the member same as or closer to the current volume */ ucontrol->value.integer.value[0] = @@ -841,7 +859,7 @@ static int tas2563_digital_gain_get( return 0; } -static int tas2563_digital_gain_put( +static int tasdevice_digital_gain_put( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -867,7 +885,7 @@ static int tas2563_digital_gain_put( } volrd = get_unaligned_be32(&data[0]); - volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + volwr = get_unaligned_be32(tas_dev->dvc_tlv_table[vol]); if (volrd == volwr) { rc = 0; @@ -876,7 +894,7 @@ static int tas2563_digital_gain_put( for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_bulk_write(tas_dev, i, reg, - (unsigned char *)tas2563_dvc_table[vol], 4); + (unsigned char *)tas_dev->dvc_tlv_table[vol], 4); if (ret) { dev_err(tas_dev->dev, "%s, set digital vol error in dev %d\n", @@ -892,11 +910,6 @@ static int tas2563_digital_gain_put( return rc; } -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), @@ -907,6 +920,16 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { SND_SOC_BYTES_EXT("Amp XMA2 Data", 6, tasdev_XMA2_data_get, NULL), }; +static const struct snd_kcontrol_new tas2x20_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2X20_AMP_LEVEL, + 0, 0, 42, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas2x20_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2X20_DVC_LEVEL, + 0, 0, ARRAY_SIZE(tas2x20_dvc_table) - 1, 0, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, + tas2x20_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -916,6 +939,15 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, tas2781_dvc_tlv), }; +static const struct snd_kcontrol_new tas5825_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS5825_AMP_LEVEL, + 0, 0, 31, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS5825_DVC_LEVEL, + 0, 0, 254, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_cali_controls[] = { SND_SOC_BYTES_EXT("Amp Latch Data", 3, tas2781_latch_reg_get, NULL), }; @@ -923,7 +955,7 @@ static const struct snd_kcontrol_new tas2781_cali_controls[] = { 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, - tas2563_digital_gain_get, tas2563_digital_gain_put, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, tas2563_dvc_tlv), }; @@ -968,8 +1000,8 @@ static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; - uinfo->value.integer.min = TAS2563; - uinfo->value.integer.max = TAS2781; + uinfo->value.integer.min = TAS2020; + uinfo->value.integer.max = TAS_OTHERS; return 0; } @@ -1168,9 +1200,9 @@ static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *active_dev_num, *chip_id; + char *active_dev_num, *chip_id, *fw_load; char *conf_name, *prog_name; - int nr_controls = 4; + int nr_controls = 5; int mix_index = 0; /* Alloc kcontrol via devm_kzalloc, which don't manually @@ -1228,6 +1260,19 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].get = tasdevice_get_chip_id; mix_index++; + fw_load = devm_kstrdup(tas_priv->dev, "Speaker Force Firmware Load", + GFP_KERNEL); + if (!fw_load) + return -ENOMEM; + + dsp_ctrls[mix_index].name = fw_load; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_info_bool_ext; + dsp_ctrls[mix_index].put = tasdev_force_fwload_put; + dsp_ctrls[mix_index].get = tasdev_force_fwload_get; + dsp_ctrls[mix_index].private_value = 0UL; + mix_index++; + return snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); } @@ -1587,6 +1632,16 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * failing to load DSP firmware is NOT an error. */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; + /* There is no DSP firmware required for TAS2118/2X20/257X. */ + switch (tas_priv->chip_id) { + case TAS2020: + case TAS2118: + case TAS2120: + case TAS2320: + case TAS2570: + case TAS2572: + goto out; + } if (tas_priv->name_prefix) scnprintf(tas_priv->coef_binaryname, 64, "%s-%s_coef.bin", tas_priv->name_prefix, tas_priv->dev_name); @@ -1608,34 +1663,37 @@ static void tasdevice_fw_ready(const struct firmware *fmw, 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; - /* If calibrated data occurs error, dsp will still works with default - * calibrated data inside algo. - */ - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->name_prefix) - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s-%s_cal_0x%02x.bin", tas_priv->name_prefix, - tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - else - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s_cal_0x%02x.bin", tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - ret = tas2781_load_calibration(tas_priv, - tas_priv->cal_binaryname[i], i); - if (ret != 0) - dev_err(tas_priv->dev, - "%s: load %s error, default will effect\n", - __func__, tas_priv->cal_binaryname[i]); + /* There is no calibration required for TAS5825/TAS5827. */ + if (tas_priv->chip_id < TAS5825) { + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + /* If calibrated data occurs error, dsp will still works + * with default calibrated data inside algo. + */ + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->name_prefix) + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s-%s_cal_0x%02x.bin", + tas_priv->name_prefix, + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + else + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s_cal_0x%02x.bin", + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_priv, + tas_priv->cal_binaryname[i], i); + if (ret != 0) + dev_err(tas_priv->dev, + "%s: load %s error, keep default.\n", + __func__, tas_priv->cal_binaryname[i]); + } } tasdevice_prmg_load(tas_priv, 0); @@ -1659,8 +1717,14 @@ static void tasdevice_fw_ready(const struct firmware *fmw, #endif out: if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { - /* If DSP FW fail, DSP kcontrol won't be created. */ - tasdevice_dsp_remove(tas_priv); + switch (tas_priv->chip_id) { + case TAS2563: + case TAS2781: + case TAS5825: + case TAS5827: + /* If DSP FW fail, DSP kcontrol won't be created. */ + tasdevice_dsp_remove(tas_priv); + } } mutex_unlock(&tas_priv->codec_lock); release_firmware(fmw); @@ -1804,13 +1868,30 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) int rc; switch (tas_priv->chip_id) { + case TAS2020: + case TAS2118: + case TAS2120: + case TAS2320: + case TAS2570: + case TAS2572: + p = (struct snd_kcontrol_new *)tas2x20_snd_controls; + size = ARRAY_SIZE(tas2x20_snd_controls); + tas_priv->dvc_tlv_table = tas2x20_dvc_table; + break; case TAS2781: p = (struct snd_kcontrol_new *)tas2781_snd_controls; size = ARRAY_SIZE(tas2781_snd_controls); break; + case TAS5825: + case TAS5827: + p = (struct snd_kcontrol_new *)tas5825_snd_controls; + size = ARRAY_SIZE(tas5825_snd_controls); + break; default: p = (struct snd_kcontrol_new *)tas2563_snd_controls; size = ARRAY_SIZE(tas2563_snd_controls); + tas_priv->dvc_tlv_table = tas2563_dvc_table; + break; } rc = snd_soc_add_component_controls(codec, p, size); @@ -1850,8 +1931,6 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .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, @@ -1967,7 +2046,16 @@ static void tasdevice_i2c_remove(struct i2c_client *client) #ifdef CONFIG_ACPI static const struct acpi_device_id tasdevice_acpi_match[] = { - { "TAS2781", TAS2781 }, + { "TXNW2020", TAS2020 }, + { "TXNW2118", TAS2118 }, + { "TXNW2120", TAS2120 }, + { "TXNW2320", TAS2320 }, + { "TXNW2563", TAS2563 }, + { "TXNW2570", TAS2570 }, + { "TXNW2572", TAS2572 }, + { "TXNW2781", TAS2781 }, + { "TXNW5825", TAS5825 }, + { "TXNW5827", TAS5827 }, {}, }; -- 2.43.0

4 6

ASRock X870 Riptide: extremely low mic capture volume
by GitHub issues - opened 09 Sep '25

09 Sep '25

alsa-project/alsa-ucm-conf issue #611 was opened from jktomer: I'm running with an ASRock X870 Riptide Wifi, which seems to use the same ALC4082 as other ASRock socket AM5 boards: ``` :) 18:00 viola:jkt[3828]alsa% lsusb | grep audio Bus 001 Device 008: ID 26ce:0a08 Generic USB Audio ``` Playback on this device seems to work and the various ports are recognized apparently correctly, but capture from both the front microphone and rear line-input ports is excessively low volume even with the mixer turned up as high as it goes. (The card does still show in `/proc/asound/cards` as "Generic USB Audio", but editing the relevant section in `ALC4080-HiFi.conf` seems to influence the output of `alsaucm -c hw:1 dump text`, so I'm assuming the recognition part works correctly.) Output of `alsa-info.sh` and `alsaucm -c hw:1 dump text` attached. [alsa-info.txt](https://github.com/user-attachments/files/22223994/alsa-info… [alsaucm.txt](https://github.com/user-attachments/files/22223995/alsaucm.txt) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/611 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

UCM2: Intel: sof-hda-dsp: HiFi: IPC3 mono DMIC is exposed as stereo PCM
by GitHub pull_request - opened 08 Sep '25

08 Sep '25

alsa-project/alsa-ucm-conf pull request #610 was opened from ujfalusi: With IPC3 the DMIC PCM is stereo even in case of a single (mono) DMIC connection. Make the `CaptureChannels 1` conditional to IPC4 only. Closes: https://bugzilla.redhat.com/show_bug.cgi?id=2393552 Closes: https://bugzilla.redhat.com/show_bug.cgi?id=2384324 Closes: https://github.com/thesofproject/linux/issues/5528 Fixes: 56cbdfd04339 ("UCM2: Intel: sof-hda-dsp: HiFi: Fix handling of mono DMICs") Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/610 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/610.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v4] ALSA: hda/tas2781: Fix the order of TAS2781 calibrated-data
by Shenghao Ding 08 Sep '25

08 Sep '25

A bug reported by one of my customers that the order of TAS2781 calibrated-data is incorrect, the correct way is to move R0_Low and insert it between R0 and InvR0. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v4: - Add missing base into cali_cnv(). v3: - Take Tiwai's advice on cali_cnv() to make it more simpler. v2: - Submit to the sound branch maintained by Tiwai instead of linux-next branch - Drop other fix --- sound/hda/codecs/side-codecs/tas2781_hda.c | 25 +++++++++++++++++----- 1 file changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index f46d2e06c64f..536940c78f00 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -33,6 +33,23 @@ const efi_guid_t tasdev_fct_efi_guid[] = { }; EXPORT_SYMBOL_NS_GPL(tasdev_fct_efi_guid, "SND_HDA_SCODEC_TAS2781"); +/* + * The order of calibrated-data writing function is a bit different from the + * order in UEFI. Here is the conversion to match the order of calibrated-data + * writing function. + */ +static void cali_cnv(unsigned char *data, unsigned int base, int offset) +{ + struct cali_reg reg_data; + + memcpy(&reg_data, &data[base], sizeof(reg_data)); + /* the data order has to be swapped between r0_low_reg and inv0_reg */ + swap(reg_data.r0_low_reg, reg_data.invr0_reg); + + cpu_to_be32_array((__force __be32 *)(data + offset + 1), + (u32 *)&reg_data, TASDEV_CALIB_N); +} + static void tas2781_apply_calib(struct tasdevice_priv *p) { struct calidata *cali_data = &p->cali_data; @@ -103,8 +120,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) data[l] = k; oft++; - for (i = 0; i < TASDEV_CALIB_N * 4; i++) - data[l + i + 1] = data[4 * oft + i]; + cali_cnv(data, 4 * oft, l); k++; } } @@ -130,9 +146,8 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) for (j = p->ndev - 1; j >= 0; j--) { l = j * (cali_data->cali_dat_sz_per_dev + 1); - for (i = TASDEV_CALIB_N * 4; i > 0 ; i--) - data[l + i] = data[p->index * 5 + i]; - data[l+i] = j; + cali_cnv(data, cali_data->cali_dat_sz_per_dev * j, l); + data[l] = j; } } -- 2.43.0

2 1

Feature Request: conf.d for USB-Audio
by GitHub issues - edited 08 Sep '25

08 Sep '25

alsa-project/alsa-ucm-conf issue #609 was edited from FrostyCoolSlug: The release cadence of `alsa-ucm-conf` tends to be a little on the slow side (generally around 6 months between releases), but the problem I'm hitting currently is the requirement for users to manually edit `USB-Audio.conf` in order to get earlier support for their devices (and depending on distribution, support could be delayed upwards of two years). I think a method where a package could be created to drop files into a conf.d style directory would allow for a users device to be configured, without the need for manually editing `USB-Audio.conf` to add support. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/609 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Feature Request: conf.d for USB-Audio
by GitHub issues - opened 08 Sep '25

08 Sep '25

alsa-project/alsa-ucm-conf issue #609 was opened from FrostyCoolSlug: The release cadence of `alsa-ucm-conf` tends to be a little on the slow side (generally around 6 months between releases), but the problem I'm hitting currently is the requirement for users to manually edit `USB-Audio.conf` in order to get support for their devices (and depending on distribution, support could be delayed upwards of two years). I think a method where a package could be created to drop files into a conf.d style directory would allow for a users device to be configured, without the need for manually editing `USB-Audio.conf` to add support. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/609 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Mic from audio jack not work on Dell Precision M4800. Please add mic drivers
by GitHub issues - edited 07 Sep '25

07 Sep '25

alsa-project/alsa-lib issue #476 was edited from Slavomir9: Hello, on my Dell Precision M4800 not work mic on headset Koss CS100 audio jack (non USB) version. Please add mic drivers. [Dell Precision M4800 ](https://www.google.com/search?q=%0D%0ADell+Precision+M4800%0D%0A&sca_esv=e618e5e4341fb9f4&) [Headset Koss CS100](https://www.kaufland.sk/product/5373091/) I use dual boot. On Windows 11 work mic only after install drivers from Dell website. On Dell website are drivers for Linux for extremely old version of Ubuntu: Ubuntu® 12.04 More information about bug here https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/2122269 Please add mic drivers. Thanks Issue URL : https://github.com/alsa-project/alsa-lib/issues/476 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Mic from audio jack not work on Dell Precision M4800. Please add mic drivers
by GitHub issues - edited 07 Sep '25

07 Sep '25

alsa-project/alsa-lib issue #476 was edited from Slavomir9: Hello, on my Dell Precision M4800 not work mic on headset Koss CS100 audio jack (non USB) version. Please add mic drivers. [Dell Precision M4800 ](https://www.google.com/search?q=%0D%0ADell+Precision+M4800%0D%0A&sca_esv=e618e5e4341fb9f4&) [Headset Koss CS100](https://www.kaufland.sk/product/5373091/) I use dual boot. On Windows 11 work mic only after install drivers from Dell website. More information about bug here https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/2122269 Please add mic drivers. Thanks Issue URL : https://github.com/alsa-project/alsa-lib/issues/476 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Mic from audio jack not work on Dell Precision M4800. Please add mic drivers
by GitHub issues - edited 07 Sep '25

07 Sep '25

alsa-project/alsa-lib issue #476 was edited from Slavomir9: Hello, on my Dell Precision M4800 not work mic on headset Koss CS100 audio jack (non USB) version. Please add mic drivers. [Dell Precision M4800 ](https://www.google.com/search?q=%0D%0ADell+Precision+M4800%0D%0A&sca_esv=e618e5e4341fb9f4&) [Headset Koss CS100](https://www.kaufland.sk/product/5373091/) I use dual boot. On Windows 11 work mic only after install drivers from Dell website. Please add mic drivers. Thanks Issue URL : https://github.com/alsa-project/alsa-lib/issues/476 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Mic from audio jack not work on Dell Precision M4800
by GitHub issues - edited 07 Sep '25

07 Sep '25

alsa-project/alsa-lib issue #476 was edited from Slavomir9: Hello, on my Dell Precision M4800 not work mic on headset Koss CS100 audio jack (non USB) version. Please add mic drivers. [Dell Precision M4800 ](https://www.google.com/search?q=%0D%0ADell+Precision+M4800%0D%0A&sca_esv=e618e5e4341fb9f4&) [Headset Koss CS100](https://www.kaufland.sk/product/5373091/) I use dual boot. On Windows 11 work mic only after install drivers from Dell website. Please add mic drivers. Thanks Issue URL : https://github.com/alsa-project/alsa-lib/issues/476 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Mic from audio jack not work on Dell Precision M4800
by GitHub issues - opened 07 Sep '25

07 Sep '25

alsa-project/alsa-lib issue #476 was opened from Slavomir9: Hello, on my Dell Precision M4800 not work mic on headset Koss CS100 audio jack (non USB) version. [Dell Precision M4800 ](https://www.google.com/search?q=%0D%0ADell+Precision+M4800%0D%0A&sca_esv=e618e5e4341fb9f4&) [Headset Koss CS100](https://www.kaufland.sk/product/5373091/) I use dual boot. On Windows 11 work mic only after install drivers from Dell website Issue URL : https://github.com/alsa-project/alsa-lib/issues/476 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v3] ALSA: hda/tas2781: Fix the order of TAS2781 calibrated-data
by Shenghao Ding 07 Sep '25

07 Sep '25

A bug reported by one of my customers that the order of TAS2781 calibrated-data is incorrect, the correct way is to move R0_Low and insert it between R0 and InvR0. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - Take Tiwai's advice on cali_cnv() to make it more simpler. v2: - Submit to sound branch maintianed by Tiwai instead of linux-next branch - Drop other fix --- sound/hda/codecs/side-codecs/tas2781_hda.c | 25 +++++++++++++++++----- 1 file changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index f46d2e06c64f..f4a44c6b0234 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -33,6 +33,23 @@ const efi_guid_t tasdev_fct_efi_guid[] = { }; EXPORT_SYMBOL_NS_GPL(tasdev_fct_efi_guid, "SND_HDA_SCODEC_TAS2781"); +/* + * The order of calibrated-data writing function is a bit different from the + * order in UEFI. Here is the conversion to match the order of calibrated-data + * writing function. + */ +static void cali_cnv(unsigned char *data, unsigned int base, int offset) +{ + struct cali_reg reg_data; + + memcpy(&reg_data, data, sizeof(reg_data)); + /* the data order has to be swapped between r0_low_reg and inv0_reg */ + swap(reg_data.r0_low_reg, reg_data.invr0_reg); + + cpu_to_be32_array((__force __be32 *)(data + offset + 1), + (u32 *)&reg_data, TASDEV_CALIB_N); +} + static void tas2781_apply_calib(struct tasdevice_priv *p) { struct calidata *cali_data = &p->cali_data; @@ -103,8 +120,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) data[l] = k; oft++; - for (i = 0; i < TASDEV_CALIB_N * 4; i++) - data[l + i + 1] = data[4 * oft + i]; + cali_cnv(data, 4 * oft, l); k++; } } @@ -130,9 +146,8 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) for (j = p->ndev - 1; j >= 0; j--) { l = j * (cali_data->cali_dat_sz_per_dev + 1); - for (i = TASDEV_CALIB_N * 4; i > 0 ; i--) - data[l + i] = data[p->index * 5 + i]; - data[l+i] = j; + cali_cnv(data, cali_data->cali_dat_sz_per_dev * j, l); + data[l] = j; } } -- 2.43.0

3 2

No sound, only dummy output since updating alsa-lib to 1.2.14
by GitHub issues - opened 06 Sep '25

06 Sep '25

alsa-project/alsa-lib issue #475 was opened from aaaaadrien: Hello, Since an update on Fedora Linux 42, Audio card not detected, GNOME only shows "Dummy Output". No sound over speakers or HDMI. I downgraded the package alsa-lib (from alsa-lib-0:1.2.14-3.fc42.x86_64 to alsa-lib-0:1.2.13-4.fc42.x86_64) and now i have sound again. Tested with OpenSuse Tumbleweed Live ISO GNOME, with libasound2-1.2.14, i reproduce the issue. So i think it's not a distro issue but an ALSA Issue. That's why i open a bug here. Laptop model : Dell Latitude 3510 Audio Device : Intel Comet Lake PCH-LP cAVS driver: sof-audio-pci-intel-cnl API: ALSA version: k6.16.4-200.fc42.x86_64 status: kernel-api Reproducible: Always Steps to Reproduce: 1. Update alsa-lib to 1.2.14-3.fc42.x86_64 2. Reboot 3. See you only have dummy output Actual Results: No sound Expected Results: Sound always here ! Additional Information: Laptop model : Dell Latitude 3510 Audio Device : Intel Comet Lake PCH-LP cAVS driver: sof-audio-pci-intel-cnl API: ALSA version: k6.16.4-200.fc42.x86_64 status: kernel-api I opened a bug on Red Hat bugzilla yesterday but i enclose the link if you want to see the commands and checks i tested : https://bugzilla.redhat.com/show_bug.cgi?id=2393552 Issue URL : https://github.com/alsa-project/alsa-lib/issues/475 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSAfirewire broken
by M. Armsby 04 Sep '25

04 Sep '25

Hello everyone. 2 Firewire problems with ALSAfirewire. 1. Firewire Midi was working fine with kernel 5.10 with Reaper. Then obviously a lot changed with the Firewire Stack. The midi output from Reaper has not changed according to it's developer - Justin. The problem is direct ALSA-Midi. We tested the midi app from www.alsa-project.org/alsa-doc/alsa-lib/_2test_2rawmidi_8c-example.html#a0 yesterday and it freezes Debian and just breaks Arch with kernel 6.15 Thread at Reaper Bugs forum: https://forum.cockos.com/showthread.php?t=300278 Has low level ALSA-Midi API changed? 2. If using Pipewire-Jack with ALSAfirewire the Reaper midi output works fine but unfortunately Pipewire adds 90ms delay when playing back through any Firewire interface compared with an onboard soundcard. The pipewire dev Wim Tayman, says it's ALSAfirewire fault but I proved it is not - see thread: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues?show=eyJpaWQiOiI0… The ALSA recording and playback is confirmed by my tests to be working under 10ms RTT with Echo, Motu and Bebop interfaces. Is this problem related to first? Low level communication error? Cheers - and let me know if I can help in any way, although I'm not a programmer. P. S. I hope you can read my post as I loaded up K-9 eMail client so as to deliver text only. -- Martin Armsby

3 11

[PATCH v2] ALSA: hda/tas2781: Fix the order of TAS2781 calibrated-data
by Shenghao Ding 04 Sep '25

04 Sep '25

A bug reported by one of my customers that the order of TAS2781 calibrated-data is incorrect, the correct way is to move R0_Low and insert it between R0 and InvR0. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Submit to sound branch maintianed by Tiwai instead of linux-next branch - drop other fix --- sound/hda/codecs/side-codecs/tas2781_hda.c | 38 ++++++++++++++++++---- 1 file changed, 32 insertions(+), 6 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index f46d2e06c64f..cd9990869e18 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -33,6 +33,32 @@ const efi_guid_t tasdev_fct_efi_guid[] = { }; EXPORT_SYMBOL_NS_GPL(tasdev_fct_efi_guid, "SND_HDA_SCODEC_TAS2781"); +/* + * The order of calibrated-data writing is a bit different from the order + * in UEFI. Here is the conversion to match the order of calibrated-data + * writing. + */ +static void cali_cnv(unsigned char *data, unsigned int base, int offset) +{ + __be32 bedata[TASDEV_CALIB_N]; + int i; + + /* r0_reg */ + bedata[0] = cpu_to_be32(*(uint32_t *)&data[base]); + /* r0_low_reg */ + bedata[1] = cpu_to_be32(*(uint32_t *)&data[base + 8]); + /* invr0_reg */ + bedata[2] = cpu_to_be32(*(uint32_t *)&data[base + 4]); + /* pow_reg */ + bedata[3] = cpu_to_be32(*(uint32_t *)&data[base + 12]); + /* tlimit_reg */ + bedata[4] = cpu_to_be32(*(uint32_t *)&data[base + 16]); + + for (i = 0; i < TASDEV_CALIB_N; i++) + memcpy(&data[offset + i * 4 + 1], &bedata[i], + sizeof(bedata[i])); +} + static void tas2781_apply_calib(struct tasdevice_priv *p) { struct calidata *cali_data = &p->cali_data; @@ -86,6 +112,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) for (j = 0, k = 0; j < node_num; j++) { oft = j * 6 + 3; + /* Calibration registers address */ if (tmp_val[oft] == TASDEV_UEFI_CALI_REG_ADDR_FLG) { for (i = 0; i < TASDEV_CALIB_N; i++) { buf = &data[(oft + i + 1) * 4]; @@ -93,6 +120,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) buf[2], buf[3]); } } else { + /* Calibrated data */ l = j * (cali_data->cali_dat_sz_per_dev + 1); if (k >= p->ndev || l > oft * 4) { dev_err(p->dev, "%s: dev sum error\n", @@ -103,8 +131,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) data[l] = k; oft++; - for (i = 0; i < TASDEV_CALIB_N * 4; i++) - data[l + i + 1] = data[4 * oft + i]; + cali_cnv(data, 4 * oft, l); k++; } } @@ -127,12 +154,11 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) dev_err(p->dev, "%s: V1 CRC error\n", __func__); return; } - + /* reverse rearrangement in case of overlap */ for (j = p->ndev - 1; j >= 0; j--) { l = j * (cali_data->cali_dat_sz_per_dev + 1); - for (i = TASDEV_CALIB_N * 4; i > 0 ; i--) - data[l + i] = data[p->index * 5 + i]; - data[l+i] = j; + cali_cnv(data, cali_data->cali_dat_sz_per_dev * j, l); + data[l] = j; } } -- 2.43.0

3 4

[PATCH] ASoC: amd: amd_sdw: Add quirks for some new Dell laptops
by Syed Saba Kareem 03 Sep '25

03 Sep '25

Add a quirk to include the codec amplifier function for Dell SKU's listed in quirk table. Note: In these SKU's, the RT722 codec amplifier is excluded, and an external amplifier is used instead. Signed-off-by: Syed Saba Kareem <syed.sabakareem(a)amd.com> --- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/amd/acp/acp-sdw-legacy-mach.c b/sound/soc/amd/acp/acp-sdw-legacy-mach.c index c2197b75a7dd..5a3cfedacbaf 100644 --- a/sound/soc/amd/acp/acp-sdw-legacy-mach.c +++ b/sound/soc/amd/acp/acp-sdw-legacy-mach.c @@ -79,6 +79,22 @@ static const struct dmi_system_id soc_sdw_quirk_table[] = { }, .driver_data = (void *)(ASOC_SDW_CODEC_SPKR), }, + { + .callback = soc_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0DD3"), + }, + .driver_data = (void *)(ASOC_SDW_CODEC_SPKR), + }, + { + .callback = soc_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0DD4"), + }, + .driver_data = (void *)(ASOC_SDW_CODEC_SPKR), + }, {} }; -- 2.43.0

2 1

ucm2: Qualcomm: x1e80100: Add X1E80100-EVK configuration
by GitHub pull_request - opened 03 Sep '25

03 Sep '25

alsa-project/alsa-ucm-conf pull request #608 was opened from ReaLcdcLeQi: The X1E80100-EVK needs basically the same configuration as the X1E80100-CRD. The dts upstream is already in progress. https://lore.kernel.org/all/20250828-hamoa_initial-v8-3-c9d173072a5c@oss.qu… message id <20250828-hamoa_initial-v8-3-c9d173072a5c(a)oss.qualcomm.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/608 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/608.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

A52 producing static noise
by GitHub issues - edited 03 Sep '25

03 Sep '25

alsa-project/alsa-plugins issue #63 was edited from mingww64: Setup is GPU----(HDMI)----TV----(eARC)----Sonos Arc Playing AC3 audio sample works and can confirm the speakers is receiving in AC3 format (Sonos app displaying Dolby Digital 5.1), however using the a52 plugin with `speaker-test` only static noise was coming out intermittently and Sonos app doesn't show any audio format. The GPU is a rx9070xt , I'm unsure if the problem is on the plugin or on AMD's side. ```log Slave: Hardware PCM card 0 'HDA ATI HDMI' device 9 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 3072 period_size : 768 period_time : 16000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 768 period_event : 0 start_threshold : 3072 stop_threshold : 3072 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 appl_ptr : 0 hw_ptr : 0 0 - Front Left 4 - Front Center 1 - Front Right 3 - Rear Right 2 - Rear Left 5 - LFE Time per period = 17.888669 0 - Front Left 4 - Front Center 1 - Front Right ^C 3 - Rear Right 2 - Rear Left 5 - LFE Time per period = 8.512034 ``` Issue URL : https://github.com/alsa-project/alsa-plugins/issues/63 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

A52 producing static noise
by GitHub issues - opened 03 Sep '25

03 Sep '25

alsa-project/alsa-plugins issue #63 was opened from mingww64: Setup is GPU----(HDMI)----TV----(eARC)----Sonos Arc Playing AC3 audio sample works and can confirm the speakers is receiving in AC3 format (Sonos app displaying Dolby Digital 5.1), however using the a52 plugin with `speaker-test` only static noise was coming out intermittently and Sonos app doesn't show any audio format. ```log Slave: Hardware PCM card 0 'HDA ATI HDMI' device 9 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 3072 period_size : 768 period_time : 16000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 768 period_event : 0 start_threshold : 3072 stop_threshold : 3072 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 appl_ptr : 0 hw_ptr : 0 0 - Front Left 4 - Front Center 1 - Front Right 3 - Rear Right 2 - Rear Left 5 - LFE Time per period = 17.888669 0 - Front Left 4 - Front Center 1 - Front Right ^C 3 - Rear Right 2 - Rear Left 5 - LFE Time per period = 8.512034 ``` Issue URL : https://github.com/alsa-project/alsa-plugins/issues/63 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[BUG] ALSA FireWire: poll timeout causes JACK to stop (6.10 – 6.17-rc2)
by Илья Кузнецов 03 Sep '25

03 Sep '25

________________________________ Hello ALSA developers, I would like to report a reproducible issue with snd-fireface on Linux kernels 6.10 through 6.17-rc2. ________________________________ System Information Kernel versions tested: 6.10 → 6.17-rc2 Audio interface: RME Fireface 800 Driver in use: snd-fireface Audio server: JACK (tested with jackd and QJackCtl) Distribution: Debian 12 Kernel config: ALSA FireWire drivers enabled, no FFADO in use Chip 1394: Texas Instrumets ________________________________ Steps to Reproduce Boot kernel 6.10 … 6.17-rc2 with snd-fireface loaded. Start JACK with ALSA backend, for example: jackd -d alsa -d hw:Fireface -p 64 -n 3 Use the system normally — during playback, recording, or even when idle. The issue may occur at any time without apparent external cause. Actual Behavior After some idle time, ALSA stops responding and JACK prints: ERROR: ALSA: poll time out, polled for ... usecs ERROR: JackAudioDriver::ProcessAsync: read error, stopping... JACK immediately terminates. The Fireface 800 does not recover until JACK is restarted. ________________________________ Expected Behavior The ALSA FireWire driver should maintain the stream and allow JACK to continue running without manual restart, even when idle. ________________________________ Notes With FFADO backend (jackd -d firewire), the same hardware works reliably and does not exhibit this issue. This suggests that the problem lies in the ALSA FireWire streaming engine rather than the hardware itself. The problem has been reproducible across all tested kernels (vanila, rt, liquorix) from 6.10 through 6.17-rc2. ________________________________ Request Could you please confirm if this issue is known? Is there any ongoing work on the ALSA FireWire streaming engine (snd-fireface) to address these poll timeout conditions? I can provide additional logs (dmesg, journalctl) or test patches if needed. Thank you, Ilya Kuznetsov.

2 2

I2S Stereo ADC Soundcard Hacking
by sylvain.gre＠proton.me 02 Sep '25

02 Sep '25

My idea is to exchange two input signals (CH_L1_IN and CH_L2_IN) at the sampling rate (192kHz) to produce one signal (CH_L_OUT), sampled at 192kHz (small electronic cicuit cf https://electronics.stackexchange.com/questions/754648/i2s-stereo-adc-sound…). In this way, in the digital domain I will receive CH_L1_IN, CH_R1_IN, CH_L2_IN, CH_R2_IN... Of course, the sampling frequency will be decreased by 2 using this method (96kHz instead of 192kHz). But the number of channels is increased by 2 also (4 instead of 2). Where can I begin with alsa to increase the number of channels and adjust the sampling frequency ? Something like a virtual soundcard maybe ? I have skills in dev but don't know where I have to start with alsa fot this hack...

1 0

[PATCH v1] ALSA: hda/tas2781: Fix the order of TAS2781 calibrated-data
by Shenghao Ding 02 Sep '25

02 Sep '25

A bug reported by one of my customers that the order of TAS2781 calibrated-data is incorrect, the correct way is to move R0_Low_%d and insert it between R0_%d and InvR0_%d. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Add varialbe csz to store cali_data->cali_dat_sz_per_dev, which is long enough to need two lines. - Add more comments on calibrated-data order --- sound/hda/codecs/side-codecs/tas2781_hda.c | 43 ++++++++++++++++++---- 1 file changed, 35 insertions(+), 8 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index f46d2e06c64f..d5913a727be4 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -33,6 +33,32 @@ const efi_guid_t tasdev_fct_efi_guid[] = { }; EXPORT_SYMBOL_NS_GPL(tasdev_fct_efi_guid, "SND_HDA_SCODEC_TAS2781"); +/* + * The order of calibrated-data writing is a bit different from the order + * in UEFI. Here is the conversion to match the order of calibrated-data + * writing. + */ +static void cali_cnv(unsigned char *data, unsigned int base, int offset) +{ + __be32 bedata[TASDEV_CALIB_N]; + int i; + + /* r0_reg */ + bedata[0] = cpu_to_be32(*(uint32_t *)&data[base]); + /* r0_low_reg */ + bedata[1] = cpu_to_be32(*(uint32_t *)&data[base + 8]); + /* invr0_reg */ + bedata[2] = cpu_to_be32(*(uint32_t *)&data[base + 4]); + /* pow_reg */ + bedata[3] = cpu_to_be32(*(uint32_t *)&data[base + 12]); + /* tlimit_reg */ + bedata[4] = cpu_to_be32(*(uint32_t *)&data[base + 16]); + + for (i = 0; i < TASDEV_CALIB_N; i++) + memcpy(&data[offset + i * 4 + 1], &bedata[i], + sizeof(bedata[i])); +} + static void tas2781_apply_calib(struct tasdevice_priv *p) { struct calidata *cali_data = &p->cali_data; @@ -46,6 +72,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) TASDEVICE_REG(0, 0x13, 0x70), TASDEVICE_REG(0, 0x18, 0x7c), }; + unsigned int csz = cali_data->cali_dat_sz_per_dev; unsigned int crc, oft, node_num; unsigned char *buf; int i, j, k, l; @@ -86,6 +113,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) for (j = 0, k = 0; j < node_num; j++) { oft = j * 6 + 3; + /* Calibration registers address */ if (tmp_val[oft] == TASDEV_UEFI_CALI_REG_ADDR_FLG) { for (i = 0; i < TASDEV_CALIB_N; i++) { buf = &data[(oft + i + 1) * 4]; @@ -93,7 +121,8 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) buf[2], buf[3]); } } else { - l = j * (cali_data->cali_dat_sz_per_dev + 1); + /* Calibrated data */ + l = j * (csz + 1); if (k >= p->ndev || l > oft * 4) { dev_err(p->dev, "%s: dev sum error\n", __func__); @@ -103,8 +132,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) data[l] = k; oft++; - for (i = 0; i < TASDEV_CALIB_N * 4; i++) - data[l + i + 1] = data[4 * oft + i]; + cali_cnv(data, 4 * oft, l); k++; } } @@ -127,12 +155,11 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) dev_err(p->dev, "%s: V1 CRC error\n", __func__); return; } - + /* reverse rearrangement in case of overlap */ for (j = p->ndev - 1; j >= 0; j--) { - l = j * (cali_data->cali_dat_sz_per_dev + 1); - for (i = TASDEV_CALIB_N * 4; i > 0 ; i--) - data[l + i] = data[p->index * 5 + i]; - data[l+i] = j; + l = j * (csz + 1); + cali_cnv(data, csz * j, l); + data[l] = j; } } -- 2.43.0

2 1

tlv: ignore xdir when type is DB_MINMAX_MUTE
by GitHub pull_request - opened 02 Sep '25

02 Sep '25

alsa-project/alsa-lib pull request #474 was opened from qaqland: Muting caused by rounding is meaningless, and it is also inconvenient for upper-layer applications to handling muting issues caused by DB_MINMAX_MUTE or xdir. Thus, the restriction on xdir is removed. Request URL : https://github.com/alsa-project/alsa-lib/pull/474 Patch URL : https://github.com/alsa-project/alsa-lib/pull/474.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] ALSA: hda/tas2781: Fix the order of TAS2563 calibrated-data
by Shenghao Ding 01 Sep '25

01 Sep '25

A bug reported by one of my customers that the order of TAS2563 calibrated-data is incorrect, the correct way is to move R0_Low_%d and insert it between R0_%d and InvR0_%d. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index dbd71e173119..853ff5d87966 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -313,7 +313,7 @@ static int tas2563_save_calibration(struct tas2781_hda *h) { efi_guid_t efi_guid = tasdev_fct_efi_guid[LENOVO]; char *vars[TASDEV_CALIB_N] = { - "R0_%d", "InvR0_%d", "R0_Low_%d", "Power_%d", "TLim_%d" + "R0_%d", "R0_Low_%d", "InvR0_%d", "Power_%d", "TLim_%d" }; efi_char16_t efi_name[TAS2563_CAL_VAR_NAME_MAX]; unsigned long max_size = TAS2563_CAL_DATA_SIZE; -- 2.43.0

3 2

[PATCH 1/2] ALSA: hda: tas2781: fix tas2563 EFI data endianness
by Gergo Koteles 30 Aug '25

30 Aug '25

Before conversion to unify the calibration data management, the tas2563_apply_calib() function performed the big endian conversion and wrote the calibration data to the device. The writing is now done by the common tasdev_load_calibrated_data() function, but without conversion. Put the values into the calibration data buffer with the expected endianness. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Cc: <stable(a)vger.kernel.org> Signed-off-by: Gergo Koteles <soyer(a)irl.hu> --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index e34b17f0c9b9..1eac751ab2a6 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -310,6 +310,7 @@ static int tas2563_save_calibration(struct tas2781_hda *h) struct cali_reg *r = &cd->cali_reg_array; unsigned int offset = 0; unsigned char *data; + __be32 bedata; efi_status_t status; unsigned int attr; int ret, i, j, k; @@ -351,6 +352,8 @@ static int tas2563_save_calibration(struct tas2781_hda *h) i, j, status); return -EINVAL; } + bedata = cpu_to_be32(*(uint32_t *)&data[offset]); + memcpy(&data[offset], &bedata, sizeof(bedata)); offset += TAS2563_CAL_DATA_SIZE; } } -- 2.51.0

2 2

[BUG] ALSA FireWire: poll timeout causes JACK to stop (6.10 – 6.17-rc2)
by Илья Кузнецов 30 Aug '25

30 Aug '25

------------------------------ Hello ALSA developers, I would like to report a reproducible issue with *snd-fireface* on Linux kernels 6.10 through 6.17-rc2. ------------------------------ System Information - *Kernel versions tested:* 6.10 → 6.17-rc2 - *Audio interface:* RME Fireface 800 - *Driver in use:* snd-fireface - *Audio server:* JACK (tested with jackd and QJackCtl) - *Distribution:* Debian 12 - *Kernel config:* ALSA FireWire drivers enabled, no FFADO in use - *Chip 1394: *Texas Instrumets ------------------------------ Steps to Reproduce 1. Boot kernel 6.10 … 6.17-rc2 with snd-fireface loaded. 2. Start JACK with ALSA backend, for example: jackd -d alsa -d hw:Fireface -p 64 -n 3 3. Use the system normally — during playback, recording, or even when idle. The issue may occur at any time without apparent external cause. Actual Behavior After some idle time, ALSA stops responding and JACK prints: ERROR: ALSA: poll time out, polled for ... usecs ERROR: JackAudioDriver::ProcessAsync: read error, stopping... JACK immediately terminates. The Fireface 800 does not recover until JACK is restarted. ------------------------------ Expected Behavior The ALSA FireWire driver should maintain the stream and allow JACK to continue running without manual restart, even when idle. ------------------------------ Notes 4. With *FFADO* backend (jackd -d firewire), the same hardware works reliably and does not exhibit this issue. 5. This suggests that the problem lies in the ALSA FireWire streaming engine rather than the hardware itself. 6. The problem has been reproducible across all tested kernels (vanila, rt, liquorix) from *6.10 through 6.17-rc2*. ------------------------------ Request Could you please confirm if this issue is known? Is there any ongoing work on the ALSA FireWire streaming engine ( snd-fireface) to address these poll timeout conditions? I can provide additional logs (dmesg, journalctl) or test patches if needed. Thank you, Ilya Kuznetsov.

1 0

URGENT: ALSAfirewire Drivers Introduce 90ms delay - Linux is Unusable for Pro Audio
by M. Armsby 29 Aug '25

29 Aug '25

Dear maintainers, Linux is being ridiculed in the professional audio community — not because it can't do low-latency audio, but because the ALSA FireWire stack actively breaks it. Professional gear costing thousands of dollars like Motu RME Focusrite etc. etc. Since kernel 5.16+, FireWire audio devices are correctly detected and functional — but they are burdened with an artificial +90ms latency offset due to missing timing metadata (e.g., cycle-match delay, transmission latency) in the driver's interface to userspace. PipeWire and WirePlumber rely on accurate latency reporting. When it's missing, they fall back to worst-case assumptions — adding ~90ms of delay. This is NOT a PipeWire bug. It is a FAILURE of the ALSA FireWire drivers to expose timing data that the hardware and firmware already provide. This bug: - Renders high-end interfaces (RME, Focusrite, Tascam, Motu, Terratec) unusable for real-time monitoring - Forces users to hack WirePlumber with hardcoded quantum rules - Damages Linux's credibility in music production - Makes Linux a JOKE in studios worldwide There is NO excuse for this in 2025. We have workarounds (see: https://forum.cockos.com/showthread.php?t=274978), but they are fragile, device-specific, and do not scale. We demand: 1. Immediate exposure of accurate device latency via ALSA control or hwdep interface 2. Proper reporting of transmission/cycle delay to userspace 3. A timeline for fixing this in mainline This is not a "nice-to-have". It is a **critical defect** in a core multimedia subsystem now using pipewire as standard. Linux can and must do better. Many are waiting to move to Linux, finally leaving windows. — Martin Armsby (Martin A - maa) Professional Audio Producer / Linux Advocate Please see: Imagine if Linux Audio just worked - Firewire <https://forum.cockos.com/showthread.php?t=302305> and <https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4785> Thank you for reading. -- Martin Armsby -- Martin Armsby

1 1

[GIT PULL] ASoC fixes for v6.17-rc3
by Mark Brown 29 Aug '25

29 Aug '25

The following changes since commit 1b237f190eb3d36f52dffe07a40b5eb210280e00: Linux 6.17-rc3 (2025-08-24 12:04:12 -0400) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.17-rc3 for you to fetch changes up to 3e7fd1febc3156d3d98fba229399a13b12d69707: ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op (2025-08-28 21:15:49 +0200) ---------------------------------------------------------------- ASoC: Fixes for v6.17 The main fixes here are for some of the cleanups done in the core in this release, we had broken component lookup in the case with a single bus and DMA controller. Otherwise it's driver specific changes, the shortlogs for the Intel WCL and rsnd drivers look like minor cleanups but are actually bugfixes (adding an op needed for correct functionality and reverting an inappropriate helper usage). ---------------------------------------------------------------- Ajye Huang (1): ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op Cryolitia PukNgae (1): ASoC: codecs: idt821034: fix wrong log in idt821034_chip_direction_output() Kuninori Morimoto (3): ASoC: soc-core: care NULL dirver name on snd_soc_lookup_component_nolocked() ASoC: soc-core: tidyup snd_soc_lookup_component_nolocked() ASoC: rsnd: tidyup direction name on rsnd_dai_connect() Mark Brown (1): ASoC: fixup snd_soc_lookup_component_nolocked() sound/soc/codecs/idt821034.c | 2 +- sound/soc/renesas/rcar/core.c | 2 +- sound/soc/soc-core.c | 25 +++++++++++++++---------- sound/soc/sof/intel/ptl.c | 1 + 4 files changed, 18 insertions(+), 12 deletions(-)

2 1

[PATCH v3 1/2] ASoC: tas2781: Add tas2118, tas2x20, tas5825 support
by Baojun Xu 29 Aug '25

29 Aug '25

Add tas2020, tas2118, tas2120, tas2320, tas5825 support in tas2781 driver. Tas2118, tas2x20 have no on-chip DSP, tas5825 have on-chip DSP but have no calibration required stereo smart amplifier. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v3: - No update. v2: - Update the mail list for maintainers of yaml file --- include/sound/tas2781.h | 11 +- include/sound/tas2x20-tlv.h | 259 ++++++++++++++++++++++++++ sound/soc/codecs/tas2781-comlib-i2c.c | 2 + sound/soc/codecs/tas2781-i2c.c | 156 +++++++++++----- 4 files changed, 381 insertions(+), 47 deletions(-) create mode 100644 include/sound/tas2x20-tlv.h diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index f0aefc04a957..73ac447b9327 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -51,7 +51,9 @@ /* Software Reset, compatble with new device (TAS5825). */ #define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) -#define TASDEVICE_REG_SWRESET_RESET (BIT(0) | BIT(4)) +#define TASDEVICE_REG_SWRESET_RESET BIT(0) + +#define TAS5825_REG_SWRESET_RESET (BIT(0) | BIT(4)) /* Checksum */ #define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) @@ -110,8 +112,14 @@ #define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) enum audio_device { + TAS2020, + TAS2118, + TAS2120, + TAS2320, TAS2563, TAS2781, + TAS5825, + TAS_OTHERS, }; enum dspbin_type { @@ -194,6 +202,7 @@ struct tasdevice_priv { unsigned char coef_binaryname[64]; unsigned char rca_binaryname[64]; unsigned char dev_name[32]; + const unsigned char (*dvc_tlv_table)[4]; const char *name_prefix; unsigned char ndev; unsigned int dspbin_typ; diff --git a/include/sound/tas2x20-tlv.h b/include/sound/tas2x20-tlv.h new file mode 100644 index 000000000000..6e6bcec4a0a1 --- /dev/null +++ b/include/sound/tas2x20-tlv.h @@ -0,0 +1,259 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2x20/TAS2118 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2x20/TAS2118 hda driver implements for one, two, or even multiple +// TAS2x20/TAS2118 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2X20_TLV_H__ +#define __TAS2X20_TLV_H__ + +#define TAS2X20_DVC_LEVEL TASDEVICE_REG(0x0, 0x2, 0x0c) +#define TAS2X20_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x07) + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_amp_tlv, 2100, 50, 0); + +/* pow(10, db/20) * pow(2,22) */ +static const __maybe_unused unsigned char tas2x20_dvc_table[][4] = { + { 0X00, 0X00, 0X0D, 0X00 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0X00 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0X00 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0X00 }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0X00 }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0X00 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X00 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X00 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X00 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0X00 }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X00 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X00 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0X00 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X00 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X00 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X00 }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X00 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X00 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0X00 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X00 }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X00 }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0X00 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0X00 }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0X00 }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X00 }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0X00 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X00 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X00 }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X00 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X00 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0X00 }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0X00 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0X00 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X00 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X00 }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X00 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X00 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X00 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0X00 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X00 }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0X00 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0X00 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0X00 }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0X00 }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X00 }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X00 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X00 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0X00 }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0X00 }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0X00 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0X00 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X00 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0X00 }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X00 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X00 }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0X00 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0X00 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0X00 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X00 }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X00 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X00 }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X00 }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X00 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X00 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X00 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X00 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0X00 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X00 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0X00 }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X00 }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0X00 }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X00 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0X00 }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0X00 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X00 }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0X00 }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X00 }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X00 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X00 }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0X00 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X00 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X00 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0X00 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0X00 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X00 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X00 }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X00 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0X00 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0X00 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X00 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X00 }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X00 }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X00 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X00 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0X00 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X00 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X00 }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0X00 }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X00 }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0X00 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X00 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0X00 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X00 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X00 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X00 }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0X00 }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X00 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X00 }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0X00 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X00 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0X00 }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X00 }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0X00 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X00 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X00 }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0X00 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X00 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X00 }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X00 }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X00 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0X00 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0X00 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0X00 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X00 }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X00 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X00 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X00 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0X00 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X00 }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0X00 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X00 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X00 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X00 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0X00 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X00 }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0X00 }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0X00 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0X00 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X00 }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X00 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0X00 }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X00 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X00 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0X00 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X00 }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0X00 }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X00 }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X00 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X00 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0X00 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X00 }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X00 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0X00 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X00 }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X00 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0X00 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X00 }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X00 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X00 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0X00 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X00 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0X00 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X00 }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X00 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0X00 }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X00 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X00 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X00 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X00 }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0X00 }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0X00 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0X00 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0X00 }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X00 }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X00 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0X00 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0X00 }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0X00 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X00 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X00 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0X00 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0X00 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X00 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X00 }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X00 }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0X00 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X00 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X00 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X00 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0X00 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0X00 }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X00 }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0X00 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X00 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X00 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0X00 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X00 }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X00 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X00 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0X00 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X00 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X00 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X00 }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X00 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0X00 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X00 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X00 }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0X00 }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X00 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X00 }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0X00 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X00 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0X00 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X00 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X00 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X00 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X00 }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X00 }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X00 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X00 }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0X00 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0X00 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X00 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0X00 }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X00 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0X00 }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X00 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X00 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0X00 }, /* 5.5db */ + { 0X7F, 0XB2, 0X61, 0X00 }, /* 6.0db */ +}; +#endif diff --git a/sound/soc/codecs/tas2781-comlib-i2c.c b/sound/soc/codecs/tas2781-comlib-i2c.c index c078bb0a8437..764304604e2c 100644 --- a/sound/soc/codecs/tas2781-comlib-i2c.c +++ b/sound/soc/codecs/tas2781-comlib-i2c.c @@ -320,6 +320,8 @@ void tasdevice_reset(struct tasdevice_priv *tas_dev) for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_write(tas_dev, i, TASDEVICE_REG_SWRESET, + tas_dev->chip_id == TAS5825 ? + TAS5825_REG_SWRESET_RESET : TASDEVICE_REG_SWRESET_RESET); if (ret < 0) dev_err(tas_dev->dev, diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index ea3cdb8553de..cf6bd4d63f31 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,8 +30,10 @@ #include <sound/tas2781.h> #include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> +#include <sound/tas2x20-tlv.h> #include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> +#include <sound/tas5825-tlv.h> #include <linux/unaligned.h> #define X2563_CL_STT_VAL(xreg, xval) \ @@ -98,16 +100,26 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { }; static const struct i2c_device_id tasdevice_id[] = { + { "tas2020", TAS2020 }, + { "tas2118", TAS2118 }, + { "tas2120", TAS2120 }, + { "tas2320", TAS2320 }, { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, + { "tas5825", TAS5825 }, {} }; MODULE_DEVICE_TABLE(i2c, tasdevice_id); #ifdef CONFIG_OF static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2020" }, + { .compatible = "ti,tas2118" }, + { .compatible = "ti,tas2120" }, + { .compatible = "ti,tas2320" }, { .compatible = "ti,tas2563" }, { .compatible = "ti,tas2781" }, + { .compatible = "ti,tas5825" }, {}, }; MODULE_DEVICE_TABLE(of, tasdevice_of_match); @@ -797,7 +809,7 @@ static int tasdev_nop_get( return 0; } -static int tas2563_digital_gain_get( +static int tasdevice_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -823,15 +835,15 @@ static int tas2563_digital_gain_get( while (r > 1 + l) { mid = (l + r) / 2; - ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + ar_mid = get_unaligned_be32(tas_dev->dvc_tlv_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]); + ar_l = get_unaligned_be32(tas_dev->dvc_tlv_table[l]); + ar_r = get_unaligned_be32(tas_dev->dvc_tlv_table[r]); /* find out the member same as or closer to the current volume */ ucontrol->value.integer.value[0] = @@ -841,7 +853,7 @@ static int tas2563_digital_gain_get( return 0; } -static int tas2563_digital_gain_put( +static int tasdevice_digital_gain_put( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -867,7 +879,7 @@ static int tas2563_digital_gain_put( } volrd = get_unaligned_be32(&data[0]); - volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + volwr = get_unaligned_be32(tas_dev->dvc_tlv_table[vol]); if (volrd == volwr) { rc = 0; @@ -876,7 +888,7 @@ static int tas2563_digital_gain_put( for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_bulk_write(tas_dev, i, reg, - (unsigned char *)tas2563_dvc_table[vol], 4); + (unsigned char *)tas_dev->dvc_tlv_table[vol], 4); if (ret) { dev_err(tas_dev->dev, "%s, set digital vol error in dev %d\n", @@ -892,11 +904,6 @@ static int tas2563_digital_gain_put( return rc; } -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), @@ -907,6 +914,16 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { SND_SOC_BYTES_EXT("Amp XMA2 Data", 6, tasdev_XMA2_data_get, NULL), }; +static const struct snd_kcontrol_new tas2x20_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2X20_AMP_LEVEL, + 0, 0, 42, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas2x20_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2X20_DVC_LEVEL, + 0, 0, ARRAY_SIZE(tas2x20_dvc_table) - 1, 0, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, + tas2x20_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -916,6 +933,15 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, tas2781_dvc_tlv), }; +static const struct snd_kcontrol_new tas5825_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS5825_AMP_LEVEL, + 0, 0, 31, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS5825_DVC_LEVEL, + 0, 0, 254, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_cali_controls[] = { SND_SOC_BYTES_EXT("Amp Latch Data", 3, tas2781_latch_reg_get, NULL), }; @@ -923,7 +949,7 @@ static const struct snd_kcontrol_new tas2781_cali_controls[] = { 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, - tas2563_digital_gain_get, tas2563_digital_gain_put, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, tas2563_dvc_tlv), }; @@ -968,7 +994,7 @@ static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; - uinfo->value.integer.min = TAS2563; + uinfo->value.integer.min = TAS2020; uinfo->value.integer.max = TAS2781; return 0; @@ -1168,9 +1194,9 @@ static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *active_dev_num, *chip_id; + char *active_dev_num, *chip_id, *fw_load; char *conf_name, *prog_name; - int nr_controls = 4; + int nr_controls = 5; int mix_index = 0; /* Alloc kcontrol via devm_kzalloc, which don't manually @@ -1228,6 +1254,19 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].get = tasdevice_get_chip_id; mix_index++; + fw_load = devm_kstrdup(tas_priv->dev, "Speaker Force Firmware Load", + GFP_KERNEL); + if (!fw_load) + return -ENOMEM; + + dsp_ctrls[mix_index].name = fw_load; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_info_bool_ext; + dsp_ctrls[mix_index].put = tasdev_force_fwload_put; + dsp_ctrls[mix_index].get = tasdev_force_fwload_get; + dsp_ctrls[mix_index].private_value = 0UL; + mix_index++; + return snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); } @@ -1587,6 +1626,9 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * failing to load DSP firmware is NOT an error. */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; + /* There is no DSP firmware required for TAS2118/2X20. */ + if (tas_priv->chip_id < TAS2563) + goto out; if (tas_priv->name_prefix) scnprintf(tas_priv->coef_binaryname, 64, "%s-%s_coef.bin", tas_priv->name_prefix, tas_priv->dev_name); @@ -1608,34 +1650,37 @@ static void tasdevice_fw_ready(const struct firmware *fmw, 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; - /* If calibrated data occurs error, dsp will still works with default - * calibrated data inside algo. - */ - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->name_prefix) - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s-%s_cal_0x%02x.bin", tas_priv->name_prefix, - tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - else - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s_cal_0x%02x.bin", tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - ret = tas2781_load_calibration(tas_priv, - tas_priv->cal_binaryname[i], i); - if (ret != 0) - dev_err(tas_priv->dev, - "%s: load %s error, default will effect\n", - __func__, tas_priv->cal_binaryname[i]); + /* There is no calibration required for TAS5825. */ + if (tas_priv->chip_id < TAS5825) { + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + /* If calibrated data occurs error, dsp will still works + * with default calibrated data inside algo. + */ + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->name_prefix) + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s-%s_cal_0x%02x.bin", + tas_priv->name_prefix, + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + else + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s_cal_0x%02x.bin", + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_priv, + tas_priv->cal_binaryname[i], i); + if (ret != 0) + dev_err(tas_priv->dev, + "%s: load %s error, keep default.\n", + __func__, tas_priv->cal_binaryname[i]); + } } tasdevice_prmg_load(tas_priv, 0); @@ -1658,7 +1703,8 @@ static void tasdevice_fw_ready(const struct firmware *fmw, comp, &acoustic_ctl_fops); #endif out: - if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { + if ((tas_priv->fw_state == TASDEVICE_RCA_FW_OK) && + (tas_priv->chip_id >= TAS2563)) { /* If DSP FW fail, DSP kcontrol won't be created. */ tasdevice_dsp_remove(tas_priv); } @@ -1804,13 +1850,27 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) int rc; switch (tas_priv->chip_id) { + case TAS2020: + case TAS2118: + case TAS2120: + case TAS2320: + p = (struct snd_kcontrol_new *)tas2x20_snd_controls; + size = ARRAY_SIZE(tas2x20_snd_controls); + tas_priv->dvc_tlv_table = tas2x20_dvc_table; + break; case TAS2781: p = (struct snd_kcontrol_new *)tas2781_snd_controls; size = ARRAY_SIZE(tas2781_snd_controls); break; + case TAS5825: + p = (struct snd_kcontrol_new *)tas5825_snd_controls; + size = ARRAY_SIZE(tas5825_snd_controls); + break; default: p = (struct snd_kcontrol_new *)tas2563_snd_controls; size = ARRAY_SIZE(tas2563_snd_controls); + tas_priv->dvc_tlv_table = tas2563_dvc_table; + break; } rc = snd_soc_add_component_controls(codec, p, size); @@ -1850,8 +1910,6 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .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, @@ -1967,7 +2025,13 @@ static void tasdevice_i2c_remove(struct i2c_client *client) #ifdef CONFIG_ACPI static const struct acpi_device_id tasdevice_acpi_match[] = { - { "TAS2781", TAS2781 }, + { "TXNW2020", TAS2020 }, + { "TXNW2118", TAS2118 }, + { "TXNW2120", TAS2120 }, + { "TXNW2320", TAS2320 }, + { "TXNW2563", TAS2563 }, + { "TXNW2781", TAS2781 }, + { "TXNW5825", TAS5825 }, {}, }; -- 2.43.0

1 1

when i click apply HDAjackretask is not respoding
by GitHub issues - opened 29 Aug '25

29 Aug '25

alsa-project/alsa-tools issue #40 was opened from m33rxj: I'm using a hp envy x360 and when i try change and override the pins to allow the front speakers and click apply, it says on the gui 'its not responding' it happened to me on ubuntu and now on cachyos. I'm not sure if my codec is unsupported? its RealTek ALC245 I'm trying to work on but even on the intel codec I get the same error. I forgot to mention I have pipewire but I'm unsure what I used for ubuntu at the time Any chance I can get some help incase I'm doing something wrong. <img width="656" height="869" alt="Image" src="https://github.com/user-attachments/assets/dfdd4d42-3979-4e89-9234-b49726d2…" /> <img width="808" height="295" alt="Image" src="https://github.com/user-attachments/assets/42f65770-e19d-40d3-8ec7-0521e345…" /> Issue URL : https://github.com/alsa-project/alsa-tools/issues/40 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH v3] ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op
by Ajye Huang 28 Aug '25

28 Aug '25

Add the missing op in the device description to avoid issues with jack detection. Fixes: 6b04629ae97a ("ASoC: SOF: Intel: add initial support for WCL") Acked-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> --- v2: Add Fixes: 6b04629ae97a ("ASoC: SOF: Intel: add initial support for WCL") v3: Add Acked-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> sound/soc/sof/intel/ptl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/ptl.c b/sound/soc/sof/intel/ptl.c index 1bc1f54c470d..4633cd01e7dd 100644 --- a/sound/soc/sof/intel/ptl.c +++ b/sound/soc/sof/intel/ptl.c @@ -143,6 +143,7 @@ const struct sof_intel_dsp_desc wcl_chip_info = { .read_sdw_lcount = hda_sdw_check_lcount_ext, .check_sdw_irq = lnl_dsp_check_sdw_irq, .check_sdw_wakeen_irq = lnl_sdw_check_wakeen_irq, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = mtl_dsp_check_ipc_irq, .cl_init = mtl_dsp_cl_init, .power_down_dsp = mtl_power_down_dsp, -- 2.25.1

2 1

ucm2: MediaTek: mt8391-evk: Add alsa-ucm support
by GitHub pull_request - opened 28 Aug '25

28 Aug '25

alsa-project/alsa-ucm-conf pull request #607 was opened from zoran055: Add alsa-ucm support for the MediaTek mt8391-evk platform. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/607 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/607.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ALSA: hda/tas2781: Fix EFI name for calibration beginning with 1 instead of 0
by Shenghao Ding 27 Aug '25

27 Aug '25

A bug reported by one of my customers that EFI name beginning with 0 instead of 1. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - remove unrelated change v1: - Fix EFI name beginning with 1 instead of 0 - Add extra comments on EFI name for calibration - Remove an extra space --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index ed7771ab9475..635cbd8820ac 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -340,7 +340,8 @@ static int tas2563_save_calibration(struct tas2781_hda *h) data[offset] = i; offset++; for (j = 0; j < TASDEV_CALIB_N; ++j) { - ret = snprintf(var8, sizeof(var8), vars[j], i); + /* EFI name for calibration started with 1, not 0 */ + ret = snprintf(var8, sizeof(var8), vars[j], i + 1); if (ret < 0 || ret >= sizeof(var8) - 1) { dev_err(p->dev, "%s: Read %s failed\n", @@ -349,7 +350,7 @@ static int tas2563_save_calibration(struct tas2781_hda *h) } /* * Our variable names are ASCII by construction, but - * EFI names are wide chars. Convert and zero-pad. + * EFI names are wide chars. Convert and zero-pad. */ memset(efi_name, 0, sizeof(efi_name)); for (k = 0; k < sizeof(var8) && var8[k]; k++) -- 2.43.0

4 5

Add UCM for RME Fireface UCX
by GitHub pull_request - opened 27 Aug '25

27 Aug '25

alsa-project/alsa-ucm-conf pull request #606 was opened from vascosantos: Adds UCM for RME Fireface UCX (heavily based on RME Fireface UCX II config). Device is assumed in CC mode. #605 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/606 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/606.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Feature request: Add support for RME Fireface UCX
by GitHub issues - opened 27 Aug '25

27 Aug '25

alsa-project/alsa-ucm-conf issue #605 was opened from vascosantos: I see there is already support for the RME Fireface UCX II. Could you please add support for the previous version, RME Fireface UCX? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/605 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v3] ALSA: hda/tas2781: Fix EFI name for calibration beginning with 1 instead of 0
by Shenghao Ding 27 Aug '25

27 Aug '25

A bug reported by one of my customers that EFI name beginning with 0 instead of 1. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - remove unrelated change v1: - Fix EFI name beginning with 1 instead of 0 - Add extra comments on EFI name for calibration --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index ed7771ab9475..dbd71e173119 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -340,8 +340,8 @@ static int tas2563_save_calibration(struct tas2781_hda *h) data[offset] = i; offset++; for (j = 0; j < TASDEV_CALIB_N; ++j) { - ret = snprintf(var8, sizeof(var8), vars[j], i); - + /* EFI name for calibration started with 1, not 0 */ + ret = snprintf(var8, sizeof(var8), vars[j], i + 1); if (ret < 0 || ret >= sizeof(var8) - 1) { dev_err(p->dev, "%s: Read %s failed\n", __func__, var8); -- 2.43.0

2 1

ADC Soundcard hack : Increase by 2 the number of channels for any adc soundcard
by Sylvain Grenier 27 Aug '25

27 Aug '25

Hello alsa team, To begin, thank you all the team for what you do, the alsa-project is a wonderfull open-source project! My goal is to increase the number of channels of my adc soundcard by separate once again a left-right for one channel. Using a small electronic scheme like this (it's a first simulation try, I have to test and improve), we can multiply the number of channels by 2. [image.png] Of course, the sampling frequency will be decreased by 2 using this method. Where can I begin with alsa to increase the number of channels and adjust the sampling frequency ? Something like a virtual soundcard maybe ? I have skills in dev but don't know where I have to start with alsa fot this hack... Best regards, Sylvain Grenier

1 0

Audio garbled when using Multi plugin to record from 2+ USB soundcards on Raspberry Pi
by GitHub issues - edited 26 Aug '25

26 Aug '25

alsa-project/alsa-lib issue #472 was edited from TommyKaneko: I have been trying to use the multi plugin to record multiple channels from multiple soundcards on the Raspberry Pi. The results are a garbled mess. ==== EDIT 26/08 ==== Having probed the issue some more, I would like to add more detail: The devices and drivers I try to use are: ```**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Device_1 [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` ...Curiously, I have actually had success using the multi plugin to record using a pro audio interface AND ONE `USB PnP Sound Device`, in this set up ```**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: USB [Scarlett 2i2 USB], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` HOWEVER, if I try to add another `USB PnP Sound Device` to the multi setup, then the result is a garbled mess. My conclusion is that the multi plugin works generally, but it does NOT work for two or more `USB PnP Sound Device`. Something tells me that the plugin cannot differentiate the addresses of the audio frames held in two or more of these cheap USB microphones. I have a more verbose post of my issue here: https://www.reddit.com/r/linuxaudio/comments/1m24po8/recording_on_multiple_… ===EDIT 26/08 The following diagnosis was wrong: I have diagnosed the reasons as thus: - current raspberry Pi OS uses alsa v1.2.8+rpt1 - the "fix" for the multi plugin was implemented in v1.2.9 : https://github.com/alsa-project/alsa-lib/commit/369df322577034ea872978d6470… I am aware that the multi plugin was not meant to be used on multiple soundcards that are not in sync, however if the clock drifts over time, this is ok for my application. I am here to seek advice on how I should go about creating a workaround. The limitations are that I need to use cheap USB microphones (it is a weight issue). Without going along the microcontroller route, I thought one way was to make an [external plugin](https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins…, based on the newly patched pcm_multi plugin. I am working on this, while climbing the steep learning curve. Perhaps I can contribute an external plugin for the project... In the meantime, does a developer have a better suggestion that I might have success with? Or any tips if you are familiar with the pcm_multi plugin? @perexg ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/472 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Audio garbled when using Multi plugin to record from 2+ USB soundcards on Raspberry Pi
by GitHub issues - edited 26 Aug '25

26 Aug '25

alsa-project/alsa-lib issue #472 was edited from TommyKaneko: I have been trying to use the multi plugin to record multiple channels from multiple soundcards on the Raspberry Pi. The results are a garbled mess. ==== EDIT 26/08 ==== Having probed the issue some more, I would like to add more detail: The devices and drivers I try to use are: ```**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Device_1 [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` ...Curiously, I have actually had success using the multi plugin to record using a pro audio interface AND ONE `USB PnP Sound Device`, in this set up ```**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: USB [Scarlett 2i2 USB], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0``` HOWEVER, if I try to add another `USB PnP Sound Device` to the multi setup, then the result is a garbled mess. My conclusion is that the multi plugin works generally, but it does NOT work for two or more `USB PnP Sound Device`. Something tells me that the plugin cannot differentiate the addresses of the audio frames held in two or more of these cheap USB microphones. I have a more verbose post of my issue here: https://www.reddit.com/r/linuxaudio/comments/1m24po8/recording_on_multiple_… ===EDIT 26/08 The following diagnosis was wrong: I have diagnosed the reasons as thus: - current raspberry Pi OS uses alsa v1.2.8+rpt1 - the "fix" for the multi plugin was implemented in v1.2.9 : https://github.com/alsa-project/alsa-lib/commit/369df322577034ea872978d6470… I am aware that the multi plugin was not meant to be used on multiple soundcards that are not in sync, however if the clock drifts over time, this is ok for my application. I am here to seek advice on how I should go about creating a workaround. The limitations are that I need to use cheap USB microphones (it is a weight issue). Without going along the microcontroller route, I thought one way was to make an [external plugin](https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins…, based on the newly patched pcm_multi plugin. I am working on this, while climbing the steep learning curve. Perhaps I can contribute an external plugin for the project... In the meantime, does a developer have a better suggestion that I might have success with? Or any tips if you are familiar with the pcm_multi plugin? @perexg ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/472 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Audio garbled when using Multi plugin to record from 2+ USB soundcards on Raspberry Pi
by GitHub issues - edited 26 Aug '25

26 Aug '25

alsa-project/alsa-lib issue #472 was edited from TommyKaneko: I have been trying to use the multi plugin to record multiple channels from multiple soundcards on the Raspberry Pi. The results are a garbled mess. ==== EDIT 26/08 ==== Having probed the issue some more, I would like to add more detail: The devices and drivers I try to use are: `**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Device_1 [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ` ...Curiously, I have actually had success using the multi plugin to record using a pro audio interface AND ONE `USB PnP Sound Device`, in this set up `**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: USB [Scarlett 2i2 USB], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0` HOWEVER, if I try to add another `USB PnP Sound Device` to the multi setup, then the result is a garbled mess. My conclusion is that the multi plugin works generally, but it does NOT work for two or more `USB PnP Sound Device`. Something tells me that the plugin cannot differentiate the addresses of the audio frames held in two or more of these cheap USB microphones. I have a more verbose post of my issue here: https://www.reddit.com/r/linuxaudio/comments/1m24po8/recording_on_multiple_… ===EDIT 26/08 The following diagnosis was wrong: I have diagnosed the reasons as thus: - current raspberry Pi OS uses alsa v1.2.8+rpt1 - the "fix" for the multi plugin was implemented in v1.2.9 : https://github.com/alsa-project/alsa-lib/commit/369df322577034ea872978d6470… I am aware that the multi plugin was not meant to be used on multiple soundcards that are not in sync, however if the clock drifts over time, this is ok for my application. I am here to seek advice on how I should go about creating a workaround. The limitations are that I need to use cheap USB microphones (it is a weight issue). Without going along the microcontroller route, I thought one way was to make an [external plugin](https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins…, based on the newly patched pcm_multi plugin. I am working on this, while climbing the steep learning curve. Perhaps I can contribute an external plugin for the project... In the meantime, does a developer have a better suggestion that I might have success with? Or any tips if you are familiar with the pcm_multi plugin? @perexg ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/472 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op
by Ajye Huang 26 Aug '25

26 Aug '25

Add the missing op in the device description to avoid issues with jack detection. Fixes: 6b04629ae97a ("ASoC: SOF: Intel: add initial support for WCL") Signed-off-by: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> --- v2: Add Fixes: 6b04629ae97a ("ASoC: SOF: Intel: add initial support for WCL") sound/soc/sof/intel/ptl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/ptl.c b/sound/soc/sof/intel/ptl.c index 1bc1f54c470d..4633cd01e7dd 100644 --- a/sound/soc/sof/intel/ptl.c +++ b/sound/soc/sof/intel/ptl.c @@ -143,6 +143,7 @@ const struct sof_intel_dsp_desc wcl_chip_info = { .read_sdw_lcount = hda_sdw_check_lcount_ext, .check_sdw_irq = lnl_dsp_check_sdw_irq, .check_sdw_wakeen_irq = lnl_sdw_check_wakeen_irq, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = mtl_dsp_check_ipc_irq, .cl_init = mtl_dsp_cl_init, .power_down_dsp = mtl_power_down_dsp, -- 2.25.1

1 1

[PATCH v1] ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op
by Ajye Huang 26 Aug '25

26 Aug '25

Add the missing op in the device description to avoid issues with jack detection. Signed-off-by: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> --- sound/soc/sof/intel/ptl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/ptl.c b/sound/soc/sof/intel/ptl.c index 1bc1f54c470d..4633cd01e7dd 100644 --- a/sound/soc/sof/intel/ptl.c +++ b/sound/soc/sof/intel/ptl.c @@ -143,6 +143,7 @@ const struct sof_intel_dsp_desc wcl_chip_info = { .read_sdw_lcount = hda_sdw_check_lcount_ext, .check_sdw_irq = lnl_dsp_check_sdw_irq, .check_sdw_wakeen_irq = lnl_sdw_check_wakeen_irq, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = mtl_dsp_check_ipc_irq, .cl_init = mtl_dsp_cl_init, .power_down_dsp = mtl_power_down_dsp, -- 2.25.1

2 2

[PATCH v1] ALSA: hda/tas2781: Fix EFI name for calibration beginning with 1 instead of 0
by Shenghao Ding 26 Aug '25

26 Aug '25

A bug reported by one of my customers that EFI name beginning with 0 instead of 1, and code clean for the string checking. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Fix EFI name beginning with 1 instead of 0 - Code clean for the string checking - Add extra comments on EFI name for calibration - Remove an extra space --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index ed7771ab9475..fecd5eac739b 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -340,16 +340,17 @@ static int tas2563_save_calibration(struct tas2781_hda *h) data[offset] = i; offset++; for (j = 0; j < TASDEV_CALIB_N; ++j) { - ret = snprintf(var8, sizeof(var8), vars[j], i); + /* EFI name for calibration started with 1, not 0 */ + ret = snprintf(var8, sizeof(var8), vars[j], i + 1); - if (ret < 0 || ret >= sizeof(var8) - 1) { + if (ret != strlen(var8)) { dev_err(p->dev, "%s: Read %s failed\n", __func__, var8); return -EINVAL; } /* * Our variable names are ASCII by construction, but - * EFI names are wide chars. Convert and zero-pad. + * EFI names are wide chars. Convert and zero-pad. */ memset(efi_name, 0, sizeof(efi_name)); for (k = 0; k < sizeof(var8) && var8[k]; k++) -- 2.43.0

2 1

my duo cast volume is too low
by GitHub issues - opened 25 Aug '25

25 Aug '25

alsa-project/alsa-ucm-conf issue #604 was opened from reflexionn: hello i bought the hyperx duocast mic for recording reasons i have a small youtube channel that goes by the name vironio i was dissapointed with the mic performance because it has very low volume and if i wanna record i have to be uncomfortably close to the microphone i thought thats just how the mic is and i lived with it i set the gain to around 70% because anything more would lead to weird noise that comes from the mic itself and i just record and then put the recording in adobe audition add a lot of effects and noise cancelation and audio gate and then i amplify the audio by around 9db so its hearable but not so long ago i saw a tiktoker with the same mic but he has a much higher volume and can use the microphone from around 2 to 3 inches away with the quality being really good so i thought its not actually the mics fault and i just didnt set it up right can anyone help me? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/604 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: MediaTek: mt8395-evk: Add support for SOF
by GitHub pull_request - edited 25 Aug '25

25 Aug '25

alsa-project/alsa-ucm-conf pull request #549 was edited from julien-massot: Add support for sof-mt8395-evk. Currently, enable only the minimum: earphone output and HDMI output. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/549 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/549.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 1/2] ASoC: tas2781: Add tas2118, tas2x20, tas5825 support
by Baojun Xu 22 Aug '25

22 Aug '25

Add tas2020, tas2118, tas2120, tas2320, tas5825 support in tas2781 driver. Tas2118, tas2x20 have no on-chip DSP, tas5825 have on-chip DSP but have no calibration required stereo smart amplifier. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v2: - Update the mail list for maintainers of yaml file --- include/sound/tas2781.h | 11 +- include/sound/tas2x20-tlv.h | 259 ++++++++++++++++++++++++++ sound/soc/codecs/tas2781-comlib-i2c.c | 2 + sound/soc/codecs/tas2781-i2c.c | 156 +++++++++++----- 4 files changed, 381 insertions(+), 47 deletions(-) create mode 100644 include/sound/tas2x20-tlv.h diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index f0aefc04a957..73ac447b9327 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -51,7 +51,9 @@ /* Software Reset, compatble with new device (TAS5825). */ #define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) -#define TASDEVICE_REG_SWRESET_RESET (BIT(0) | BIT(4)) +#define TASDEVICE_REG_SWRESET_RESET BIT(0) + +#define TAS5825_REG_SWRESET_RESET (BIT(0) | BIT(4)) /* Checksum */ #define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) @@ -110,8 +112,14 @@ #define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) enum audio_device { + TAS2020, + TAS2118, + TAS2120, + TAS2320, TAS2563, TAS2781, + TAS5825, + TAS_OTHERS, }; enum dspbin_type { @@ -194,6 +202,7 @@ struct tasdevice_priv { unsigned char coef_binaryname[64]; unsigned char rca_binaryname[64]; unsigned char dev_name[32]; + const unsigned char (*dvc_tlv_table)[4]; const char *name_prefix; unsigned char ndev; unsigned int dspbin_typ; diff --git a/include/sound/tas2x20-tlv.h b/include/sound/tas2x20-tlv.h new file mode 100644 index 000000000000..6e6bcec4a0a1 --- /dev/null +++ b/include/sound/tas2x20-tlv.h @@ -0,0 +1,259 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2x20/TAS2118 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2x20/TAS2118 hda driver implements for one, two, or even multiple +// TAS2x20/TAS2118 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2X20_TLV_H__ +#define __TAS2X20_TLV_H__ + +#define TAS2X20_DVC_LEVEL TASDEVICE_REG(0x0, 0x2, 0x0c) +#define TAS2X20_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x07) + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_amp_tlv, 2100, 50, 0); + +/* pow(10, db/20) * pow(2,22) */ +static const __maybe_unused unsigned char tas2x20_dvc_table[][4] = { + { 0X00, 0X00, 0X0D, 0X00 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0X00 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0X00 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0X00 }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0X00 }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0X00 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X00 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X00 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X00 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0X00 }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X00 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X00 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0X00 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X00 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X00 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X00 }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X00 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X00 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0X00 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X00 }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X00 }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0X00 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0X00 }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0X00 }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X00 }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0X00 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X00 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X00 }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X00 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X00 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0X00 }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0X00 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0X00 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X00 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X00 }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X00 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X00 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X00 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0X00 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X00 }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0X00 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0X00 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0X00 }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0X00 }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X00 }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X00 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X00 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0X00 }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0X00 }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0X00 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0X00 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X00 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0X00 }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X00 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X00 }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0X00 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0X00 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0X00 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X00 }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X00 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X00 }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X00 }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X00 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X00 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X00 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X00 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0X00 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X00 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0X00 }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X00 }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0X00 }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X00 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0X00 }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0X00 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X00 }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0X00 }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X00 }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X00 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X00 }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0X00 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X00 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X00 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0X00 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0X00 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X00 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X00 }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X00 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0X00 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0X00 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X00 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X00 }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X00 }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X00 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X00 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0X00 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X00 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X00 }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0X00 }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X00 }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0X00 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X00 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0X00 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X00 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X00 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X00 }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0X00 }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X00 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X00 }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0X00 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X00 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0X00 }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X00 }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0X00 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X00 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X00 }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0X00 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X00 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X00 }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X00 }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X00 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0X00 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0X00 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0X00 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X00 }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X00 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X00 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X00 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0X00 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X00 }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0X00 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X00 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X00 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X00 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0X00 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X00 }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0X00 }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0X00 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0X00 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X00 }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X00 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0X00 }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X00 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X00 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0X00 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X00 }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0X00 }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X00 }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X00 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X00 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0X00 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X00 }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X00 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0X00 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X00 }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X00 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0X00 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X00 }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X00 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X00 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0X00 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X00 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0X00 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X00 }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X00 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0X00 }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X00 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X00 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X00 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X00 }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0X00 }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0X00 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0X00 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0X00 }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X00 }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X00 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0X00 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0X00 }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0X00 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X00 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X00 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0X00 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0X00 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X00 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X00 }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X00 }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0X00 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X00 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X00 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X00 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0X00 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0X00 }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X00 }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0X00 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X00 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X00 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0X00 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X00 }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X00 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X00 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0X00 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X00 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X00 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X00 }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X00 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0X00 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X00 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X00 }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0X00 }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X00 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X00 }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0X00 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X00 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0X00 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X00 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X00 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X00 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X00 }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X00 }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X00 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X00 }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0X00 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0X00 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X00 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0X00 }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X00 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0X00 }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X00 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X00 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0X00 }, /* 5.5db */ + { 0X7F, 0XB2, 0X61, 0X00 }, /* 6.0db */ +}; +#endif diff --git a/sound/soc/codecs/tas2781-comlib-i2c.c b/sound/soc/codecs/tas2781-comlib-i2c.c index c078bb0a8437..764304604e2c 100644 --- a/sound/soc/codecs/tas2781-comlib-i2c.c +++ b/sound/soc/codecs/tas2781-comlib-i2c.c @@ -320,6 +320,8 @@ void tasdevice_reset(struct tasdevice_priv *tas_dev) for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_write(tas_dev, i, TASDEVICE_REG_SWRESET, + tas_dev->chip_id == TAS5825 ? + TAS5825_REG_SWRESET_RESET : TASDEVICE_REG_SWRESET_RESET); if (ret < 0) dev_err(tas_dev->dev, diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index ea3cdb8553de..cf6bd4d63f31 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,8 +30,10 @@ #include <sound/tas2781.h> #include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> +#include <sound/tas2x20-tlv.h> #include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> +#include <sound/tas5825-tlv.h> #include <linux/unaligned.h> #define X2563_CL_STT_VAL(xreg, xval) \ @@ -98,16 +100,26 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { }; static const struct i2c_device_id tasdevice_id[] = { + { "tas2020", TAS2020 }, + { "tas2118", TAS2118 }, + { "tas2120", TAS2120 }, + { "tas2320", TAS2320 }, { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, + { "tas5825", TAS5825 }, {} }; MODULE_DEVICE_TABLE(i2c, tasdevice_id); #ifdef CONFIG_OF static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2020" }, + { .compatible = "ti,tas2118" }, + { .compatible = "ti,tas2120" }, + { .compatible = "ti,tas2320" }, { .compatible = "ti,tas2563" }, { .compatible = "ti,tas2781" }, + { .compatible = "ti,tas5825" }, {}, }; MODULE_DEVICE_TABLE(of, tasdevice_of_match); @@ -797,7 +809,7 @@ static int tasdev_nop_get( return 0; } -static int tas2563_digital_gain_get( +static int tasdevice_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -823,15 +835,15 @@ static int tas2563_digital_gain_get( while (r > 1 + l) { mid = (l + r) / 2; - ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + ar_mid = get_unaligned_be32(tas_dev->dvc_tlv_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]); + ar_l = get_unaligned_be32(tas_dev->dvc_tlv_table[l]); + ar_r = get_unaligned_be32(tas_dev->dvc_tlv_table[r]); /* find out the member same as or closer to the current volume */ ucontrol->value.integer.value[0] = @@ -841,7 +853,7 @@ static int tas2563_digital_gain_get( return 0; } -static int tas2563_digital_gain_put( +static int tasdevice_digital_gain_put( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -867,7 +879,7 @@ static int tas2563_digital_gain_put( } volrd = get_unaligned_be32(&data[0]); - volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + volwr = get_unaligned_be32(tas_dev->dvc_tlv_table[vol]); if (volrd == volwr) { rc = 0; @@ -876,7 +888,7 @@ static int tas2563_digital_gain_put( for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_bulk_write(tas_dev, i, reg, - (unsigned char *)tas2563_dvc_table[vol], 4); + (unsigned char *)tas_dev->dvc_tlv_table[vol], 4); if (ret) { dev_err(tas_dev->dev, "%s, set digital vol error in dev %d\n", @@ -892,11 +904,6 @@ static int tas2563_digital_gain_put( return rc; } -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), @@ -907,6 +914,16 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { SND_SOC_BYTES_EXT("Amp XMA2 Data", 6, tasdev_XMA2_data_get, NULL), }; +static const struct snd_kcontrol_new tas2x20_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2X20_AMP_LEVEL, + 0, 0, 42, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas2x20_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2X20_DVC_LEVEL, + 0, 0, ARRAY_SIZE(tas2x20_dvc_table) - 1, 0, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, + tas2x20_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -916,6 +933,15 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, tas2781_dvc_tlv), }; +static const struct snd_kcontrol_new tas5825_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS5825_AMP_LEVEL, + 0, 0, 31, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS5825_DVC_LEVEL, + 0, 0, 254, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_cali_controls[] = { SND_SOC_BYTES_EXT("Amp Latch Data", 3, tas2781_latch_reg_get, NULL), }; @@ -923,7 +949,7 @@ static const struct snd_kcontrol_new tas2781_cali_controls[] = { 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, - tas2563_digital_gain_get, tas2563_digital_gain_put, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, tas2563_dvc_tlv), }; @@ -968,7 +994,7 @@ static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; - uinfo->value.integer.min = TAS2563; + uinfo->value.integer.min = TAS2020; uinfo->value.integer.max = TAS2781; return 0; @@ -1168,9 +1194,9 @@ static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *active_dev_num, *chip_id; + char *active_dev_num, *chip_id, *fw_load; char *conf_name, *prog_name; - int nr_controls = 4; + int nr_controls = 5; int mix_index = 0; /* Alloc kcontrol via devm_kzalloc, which don't manually @@ -1228,6 +1254,19 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].get = tasdevice_get_chip_id; mix_index++; + fw_load = devm_kstrdup(tas_priv->dev, "Speaker Force Firmware Load", + GFP_KERNEL); + if (!fw_load) + return -ENOMEM; + + dsp_ctrls[mix_index].name = fw_load; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_info_bool_ext; + dsp_ctrls[mix_index].put = tasdev_force_fwload_put; + dsp_ctrls[mix_index].get = tasdev_force_fwload_get; + dsp_ctrls[mix_index].private_value = 0UL; + mix_index++; + return snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); } @@ -1587,6 +1626,9 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * failing to load DSP firmware is NOT an error. */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; + /* There is no DSP firmware required for TAS2118/2X20. */ + if (tas_priv->chip_id < TAS2563) + goto out; if (tas_priv->name_prefix) scnprintf(tas_priv->coef_binaryname, 64, "%s-%s_coef.bin", tas_priv->name_prefix, tas_priv->dev_name); @@ -1608,34 +1650,37 @@ static void tasdevice_fw_ready(const struct firmware *fmw, 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; - /* If calibrated data occurs error, dsp will still works with default - * calibrated data inside algo. - */ - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->name_prefix) - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s-%s_cal_0x%02x.bin", tas_priv->name_prefix, - tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - else - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s_cal_0x%02x.bin", tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - ret = tas2781_load_calibration(tas_priv, - tas_priv->cal_binaryname[i], i); - if (ret != 0) - dev_err(tas_priv->dev, - "%s: load %s error, default will effect\n", - __func__, tas_priv->cal_binaryname[i]); + /* There is no calibration required for TAS5825. */ + if (tas_priv->chip_id < TAS5825) { + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + /* If calibrated data occurs error, dsp will still works + * with default calibrated data inside algo. + */ + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->name_prefix) + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s-%s_cal_0x%02x.bin", + tas_priv->name_prefix, + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + else + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s_cal_0x%02x.bin", + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_priv, + tas_priv->cal_binaryname[i], i); + if (ret != 0) + dev_err(tas_priv->dev, + "%s: load %s error, keep default.\n", + __func__, tas_priv->cal_binaryname[i]); + } } tasdevice_prmg_load(tas_priv, 0); @@ -1658,7 +1703,8 @@ static void tasdevice_fw_ready(const struct firmware *fmw, comp, &acoustic_ctl_fops); #endif out: - if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { + if ((tas_priv->fw_state == TASDEVICE_RCA_FW_OK) && + (tas_priv->chip_id >= TAS2563)) { /* If DSP FW fail, DSP kcontrol won't be created. */ tasdevice_dsp_remove(tas_priv); } @@ -1804,13 +1850,27 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) int rc; switch (tas_priv->chip_id) { + case TAS2020: + case TAS2118: + case TAS2120: + case TAS2320: + p = (struct snd_kcontrol_new *)tas2x20_snd_controls; + size = ARRAY_SIZE(tas2x20_snd_controls); + tas_priv->dvc_tlv_table = tas2x20_dvc_table; + break; case TAS2781: p = (struct snd_kcontrol_new *)tas2781_snd_controls; size = ARRAY_SIZE(tas2781_snd_controls); break; + case TAS5825: + p = (struct snd_kcontrol_new *)tas5825_snd_controls; + size = ARRAY_SIZE(tas5825_snd_controls); + break; default: p = (struct snd_kcontrol_new *)tas2563_snd_controls; size = ARRAY_SIZE(tas2563_snd_controls); + tas_priv->dvc_tlv_table = tas2563_dvc_table; + break; } rc = snd_soc_add_component_controls(codec, p, size); @@ -1850,8 +1910,6 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .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, @@ -1967,7 +2025,13 @@ static void tasdevice_i2c_remove(struct i2c_client *client) #ifdef CONFIG_ACPI static const struct acpi_device_id tasdevice_acpi_match[] = { - { "TAS2781", TAS2781 }, + { "TXNW2020", TAS2020 }, + { "TXNW2118", TAS2118 }, + { "TXNW2120", TAS2120 }, + { "TXNW2320", TAS2320 }, + { "TXNW2563", TAS2563 }, + { "TXNW2781", TAS2781 }, + { "TXNW5825", TAS5825 }, {}, }; -- 2.43.0

1 0

[PATCH] ALSA: hda: intel-dsp-config: Select SOF driver on MTL Chromebooks
by Brady Norander 22 Aug '25

22 Aug '25

The SOF driver is required for functional audio on MTL Chromebooks Signed-off-by: Brady Norander <bradynorander(a)gmail.com> --- sound/hda/core/intel-dsp-config.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/hda/core/intel-dsp-config.c b/sound/hda/core/intel-dsp-config.c index 3cb1e7fc3b3b..00f184917623 100644 --- a/sound/hda/core/intel-dsp-config.c +++ b/sound/hda/core/intel-dsp-config.c @@ -167,9 +167,9 @@ static const struct config_entry config_table[] = { /* * CoffeeLake, CannonLake, CometLake, IceLake, TigerLake, AlderLake, - * RaptorLake use legacy HDAudio driver except for Google Chromebooks - * and when DMICs are present. Two cases are required since Coreboot - * does not expose NHLT tables. + * RaptorLake, MeteorLake use legacy HDAudio driver except for Google + * Chromebooks and when DMICs are present. Two cases are required since + * Coreboot does not expose NHLT tables. * * When the Chromebook quirk is not present, it's based on information * that no such device exists. When the quirk is present, it could be @@ -516,6 +516,19 @@ static const struct config_entry config_table[] = { /* Meteor Lake */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_METEORLAKE) /* Meteorlake-P */ + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_MTL, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_MTL, -- 2.50.1

1 0

Mute LED not working.
by GitHub issues - opened 21 Aug '25

21 Aug '25

alsa-project/alsa-lib issue #473 was opened from Gropinou: https://alsa-project.org/db/?f=a68feee728b669f547491855254add15fddad73e Issue URL : https://github.com/alsa-project/alsa-lib/issues/473 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v6.17-rc2
by Mark Brown 21 Aug '25

21 Aug '25

The following changes since commit c17b750b3ad9f45f2b6f7e6f7f4679844244f0b9: Linux 6.17-rc2 (2025-08-17 15:22:10 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.17-rc2 for you to fetch changes up to 7c15e4cabfa96ed3cd717a2ed7a9961268ab21f7: ASoC: cs35l56: Fixes for CS35L63 for production (2025-08-20 19:56:20 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.17 A few fixes that came in during the past week, there's some updates for the CS35L56 which adjust the driver for production silicon and a fix for buggy resume of the ES9389. ---------------------------------------------------------------- Mark Brown (1): ASoC: cs35l56: Fixes for CS35L63 for production Richard Fitzgerald (1): ASoC: cs35l56: Handle new algorithms IDs for CS35L63 Stefan Binding (2): ASoC: cs35l56: Update Firmware Addresses for CS35L63 for production silicon ASoC: cs35l56: Remove SoundWire Clock Divider workaround for CS35L63 Zhang Yi (1): ASoC: codecs: ES9389: Modify the standby configuration include/sound/cs35l56.h | 5 +-- sound/soc/codecs/cs35l56-sdw.c | 69 --------------------------------------- sound/soc/codecs/cs35l56-shared.c | 29 ++++++++++++++-- sound/soc/codecs/cs35l56.c | 2 +- sound/soc/codecs/cs35l56.h | 3 -- sound/soc/codecs/es8389.c | 2 +- 6 files changed, 31 insertions(+), 79 deletions(-)

1 1

Cybersecurity Risk Assessment Request from Emerson for Advanced Linux Sound Architecture (ALSA)
by KATARE, SAURABH [EMR/MSOL/PUNE] 21 Aug '25

21 Aug '25

Hello, I hope this message finds you well. As part of our ongoing efforts to comply with the EU Cyber Resilience Act (CRA), we are currently conducting a cybersecurity risk assessment of third-party software vendors whose products or components are integrated into our systems. To support this initiative, we kindly request your input on the following questions related to your software product "Advanced Linux Sound Architecture (ALSA)" with version v1.2.1.2. Please provide your responses directly in the table below and do reply to all added in this email, Additional Information: * Purpose: This security assessment is part of our due diligence and regulatory compliance obligations under the EU CRA. * Confidentiality: All information shared will be treated as confidential and used solely for the purpose of this assessment. * Contact: Should you have any questions or need further clarification, please feel free to reach out by replying directly to this email. We kindly request your response by Monday, August 25, 2025, to ensure timely completion of our assessment process. Thank you for your cooperation and continued partnership in maintaining a secure and resilient digital environment. Sr. No. Queries to Vendor Response from Vendor (Yes/No) Additional Remarks from Vendor 1 Is Secure Software Development Lifecycle followed for developing this component? 2 Do you provide regular security updates for "ALSA" ? 3 Is there any discontinuation/End of life for the latest version of "ALSA" in near future? 4 Do you have Long Term support for "ALSA"? If yes please mention the version in Remark column 5 Is appropriate cybersecurity testing followed? If yes, is any specific standard for testing used? 6 Are there any vulnerabilities in the latest version which are not disclosed publicly? If yes, when will it be fixed and released? please mention in Remark column. 7 Is the vulnerability handing procedure available for "ALSA"? if yes mention the procedure in the Remark column. 8 Do you comply with EU-CRA requirements? 9 Do you provide proof of conformity regarding adherence to EU-CRA? If yes please mention details in Remark column Best regards, Saurabh. Saurabh Katare | Engineer, Software Development Emerson | Plot no 23, Rajiv Gandhi InfoTech Park | Phase II , Hinjawadi | Pune | Maharashtra | 411057 | India saurabh.katare(a)emerson.com

2 1

[PATCH v2] ASoC: amd: acp: Adjust pdm gain value
by Venkata Prasad Potturu 21 Aug '25

21 Aug '25

Set pdm gain value by setting PDM_MISC_CTRL_MASK value. To avoid low pdm gain value. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- Changes since v1: - remove pdm gain adjustment using module parameter. sound/soc/amd/acp/amd.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index cb8d97122f95..73a028e67246 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -130,7 +130,7 @@ #define PDM_DMA_INTR_MASK 0x10000 #define PDM_DEC_64 0x2 #define PDM_CLK_FREQ_MASK 0x07 -#define PDM_MISC_CTRL_MASK 0x10 +#define PDM_MISC_CTRL_MASK 0x18 #define PDM_ENABLE 0x01 #define PDM_DISABLE 0x00 #define DMA_EN_MASK 0x02 -- 2.43.0

3 2

Low maximum volume with Plantronics Blackwire 315.1
by Paul Menzel 21 Aug '25

21 Aug '25

Dear Linux folks, On the Intel Kaby Lake laptop Dell XPS 13 9360, using Debian sid/unstable with Debian sid/unstable and Linux 6.17.0-rc2-00053-gb19a97d57c15, connecting the USB headset Plantronics Blackwire 315.1, even turning the volume on the headset control (or GNOME, both are linked) to maximum, the volume is still very low. `pavucontrol` shows everything on 100 %. Is that a driver problem, or somehow a hardware property? I uploaded the files done by `/sbin/alsa-info` [1]. Kind regards, Paul [1]: https://alsa-project.org/db/?f=ccc9945a112e88c7b6bf29ecc072f05e0cb72f3f

1 0

[PATCH v1] ALSA: hda/tas2781: Enable init_profile_id for device initialization
by Shenghao Ding 21 Aug '25

21 Aug '25

Optimize the time consumption of profile switching, init_profile saves the common settings of different profiles, such as the dsp coefficients, etc, which can greatly reduce the profile switching time comsumption and remove the repetitive settings. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 318f8c58ae61..97c33fee9660 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -477,6 +477,12 @@ static void tasdevice_dspfw_init(void *context) if (tas_priv->fmw->nr_configurations > 0) tas_priv->cur_conf = 0; + /* Init common setting for different audio profiles */ + if (tas_priv->rcabin.init_profile_id >= 0) + tasdevice_select_cfg_blk(tas_priv, + tas_priv->rcabin.init_profile_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + /* If calibrated data occurs error, dsp will still works with default * calibrated data inside algo. */ @@ -779,6 +785,12 @@ static int tas2781_system_resume(struct device *dev) tasdevice_reset(tas_hda->priv); tasdevice_prmg_load(tas_hda->priv, tas_hda->priv->cur_prog); + /* Init common setting for different audio profiles */ + if (tas_hda->priv->rcabin.init_profile_id >= 0) + tasdevice_select_cfg_blk(tas_hda->priv, + tas_hda->priv->rcabin.init_profile_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + if (tas_hda->priv->playback_started) tasdevice_tuning_switch(tas_hda->priv, 0); -- 2.43.0

3 3

[GIT PULL] ASoC fixes for v6.17-rc2
by Mark Brown 21 Aug '25

21 Aug '25

The following changes since commit c17b750b3ad9f45f2b6f7e6f7f4679844244f0b9: Linux 6.17-rc2 (2025-08-17 15:22:10 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.17-rc2 for you to fetch changes up to 7c15e4cabfa96ed3cd717a2ed7a9961268ab21f7: ASoC: cs35l56: Fixes for CS35L63 for production (2025-08-20 19:56:20 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.17 A few fixes that came in during the past week, there's some updates for the CS35L56 which adjust the driver for production silicon and a fix for buggy resume of the ES9389. ---------------------------------------------------------------- Mark Brown (1): ASoC: cs35l56: Fixes for CS35L63 for production Richard Fitzgerald (1): ASoC: cs35l56: Handle new algorithms IDs for CS35L63 Stefan Binding (2): ASoC: cs35l56: Update Firmware Addresses for CS35L63 for production silicon ASoC: cs35l56: Remove SoundWire Clock Divider workaround for CS35L63 Zhang Yi (1): ASoC: codecs: ES9389: Modify the standby configuration include/sound/cs35l56.h | 5 +-- sound/soc/codecs/cs35l56-sdw.c | 69 --------------------------------------- sound/soc/codecs/cs35l56-shared.c | 29 ++++++++++++++-- sound/soc/codecs/cs35l56.c | 2 +- sound/soc/codecs/cs35l56.h | 3 -- sound/soc/codecs/es8389.c | 2 +- 6 files changed, 31 insertions(+), 79 deletions(-)

2 1

Audio garbled when using Multi plugin to record from 2+ USB soundcards on Raspberry Pi
by GitHub issues - opened 21 Aug '25

21 Aug '25

alsa-project/alsa-lib issue #472 was opened from TommyKaneko: I have been trying to use the multi plugin to record multiple channels from mutliple soundcards on the Raspberry Pi. The results are a garbled mess. I have diagnosed the reasons as thus: - current raspberry Pi OS uses alsa v1.2.8+rpt1 - the "fix" for the multi plugin was implemented in v1.2.9 : https://github.com/alsa-project/alsa-lib/commit/369df322577034ea872978d6470… I am aware that the multi plugin was not meant to be used on multiple soundcards that are not in sync, however if the clock drifts over time, this is ok for my application. I am here to seek advice on how I should go about creating a workaround. The limitations are that I need to use cheap USB microphones (it is a weight issue). Without going along the microcontroller route, I thought one way was to make an [external plugin](https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins…, based on the newly patched pcm_multi plugin. I am working on this, while climbing the steep learning curve. Perhaps I can contribute an external plugin for the project... In the meantime, does a developer have a better suggestion that I might have success with? Or any tips if you are familiar with the pcm_multi plugin? @perexg ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/472 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1 1/2] ASoC: tas2781: Add tas2118, tas2x20, tas5825 support
by Baojun Xu 20 Aug '25

20 Aug '25

Add tas2020, tas2118, tas2120, tas2320, tas5825 support in tas2781 driver. Tas2118, tas2x20 have no on-chip DSP, tas5825 have on-chip DSP but have no calibration required stereo smart amplifier. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781.h | 11 +- include/sound/tas2x20-tlv.h | 259 ++++++++++++++++++++++++++ sound/soc/codecs/tas2781-comlib-i2c.c | 2 + sound/soc/codecs/tas2781-i2c.c | 156 +++++++++++----- 4 files changed, 381 insertions(+), 47 deletions(-) create mode 100644 include/sound/tas2x20-tlv.h diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index f0aefc04a957..73ac447b9327 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -51,7 +51,9 @@ /* Software Reset, compatble with new device (TAS5825). */ #define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) -#define TASDEVICE_REG_SWRESET_RESET (BIT(0) | BIT(4)) +#define TASDEVICE_REG_SWRESET_RESET BIT(0) + +#define TAS5825_REG_SWRESET_RESET (BIT(0) | BIT(4)) /* Checksum */ #define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) @@ -110,8 +112,14 @@ #define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) enum audio_device { + TAS2020, + TAS2118, + TAS2120, + TAS2320, TAS2563, TAS2781, + TAS5825, + TAS_OTHERS, }; enum dspbin_type { @@ -194,6 +202,7 @@ struct tasdevice_priv { unsigned char coef_binaryname[64]; unsigned char rca_binaryname[64]; unsigned char dev_name[32]; + const unsigned char (*dvc_tlv_table)[4]; const char *name_prefix; unsigned char ndev; unsigned int dspbin_typ; diff --git a/include/sound/tas2x20-tlv.h b/include/sound/tas2x20-tlv.h new file mode 100644 index 000000000000..6e6bcec4a0a1 --- /dev/null +++ b/include/sound/tas2x20-tlv.h @@ -0,0 +1,259 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2x20/TAS2118 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2x20/TAS2118 hda driver implements for one, two, or even multiple +// TAS2x20/TAS2118 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2X20_TLV_H__ +#define __TAS2X20_TLV_H__ + +#define TAS2X20_DVC_LEVEL TASDEVICE_REG(0x0, 0x2, 0x0c) +#define TAS2X20_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x07) + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_amp_tlv, 2100, 50, 0); + +/* pow(10, db/20) * pow(2,22) */ +static const __maybe_unused unsigned char tas2x20_dvc_table[][4] = { + { 0X00, 0X00, 0X0D, 0X00 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0X00 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0X00 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0X00 }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0X00 }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0X00 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X00 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X00 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X00 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0X00 }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X00 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X00 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0X00 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X00 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X00 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X00 }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X00 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X00 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0X00 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X00 }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X00 }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0X00 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0X00 }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0X00 }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X00 }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0X00 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X00 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X00 }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X00 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X00 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0X00 }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0X00 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0X00 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X00 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X00 }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X00 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X00 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X00 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0X00 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X00 }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0X00 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0X00 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0X00 }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0X00 }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X00 }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X00 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X00 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0X00 }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0X00 }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0X00 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0X00 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X00 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0X00 }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X00 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X00 }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0X00 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0X00 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0X00 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X00 }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X00 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X00 }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X00 }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X00 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X00 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X00 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X00 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0X00 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X00 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0X00 }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X00 }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0X00 }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X00 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0X00 }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0X00 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X00 }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0X00 }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X00 }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X00 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X00 }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0X00 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X00 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X00 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0X00 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0X00 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X00 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X00 }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X00 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0X00 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0X00 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X00 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X00 }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X00 }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X00 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X00 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0X00 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X00 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X00 }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0X00 }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X00 }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0X00 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X00 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0X00 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X00 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X00 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X00 }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0X00 }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X00 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X00 }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0X00 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X00 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0X00 }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X00 }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0X00 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X00 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X00 }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0X00 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X00 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X00 }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X00 }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X00 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0X00 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0X00 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0X00 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X00 }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X00 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X00 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X00 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0X00 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X00 }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0X00 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X00 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X00 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X00 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0X00 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X00 }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0X00 }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0X00 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0X00 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X00 }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X00 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0X00 }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X00 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X00 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0X00 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X00 }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0X00 }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X00 }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X00 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X00 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0X00 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X00 }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X00 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0X00 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X00 }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X00 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0X00 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X00 }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X00 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X00 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0X00 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X00 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0X00 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X00 }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X00 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0X00 }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X00 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X00 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X00 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X00 }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0X00 }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0X00 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0X00 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0X00 }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X00 }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X00 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0X00 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0X00 }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0X00 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X00 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X00 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0X00 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0X00 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X00 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X00 }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X00 }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0X00 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X00 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X00 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X00 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0X00 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0X00 }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X00 }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0X00 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X00 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X00 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0X00 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X00 }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X00 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X00 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0X00 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X00 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X00 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X00 }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X00 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0X00 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X00 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X00 }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0X00 }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X00 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X00 }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0X00 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X00 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0X00 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X00 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X00 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X00 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X00 }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X00 }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X00 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X00 }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0X00 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0X00 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X00 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0X00 }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X00 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0X00 }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X00 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X00 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0X00 }, /* 5.5db */ + { 0X7F, 0XB2, 0X61, 0X00 }, /* 6.0db */ +}; +#endif diff --git a/sound/soc/codecs/tas2781-comlib-i2c.c b/sound/soc/codecs/tas2781-comlib-i2c.c index c078bb0a8437..764304604e2c 100644 --- a/sound/soc/codecs/tas2781-comlib-i2c.c +++ b/sound/soc/codecs/tas2781-comlib-i2c.c @@ -320,6 +320,8 @@ void tasdevice_reset(struct tasdevice_priv *tas_dev) for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_write(tas_dev, i, TASDEVICE_REG_SWRESET, + tas_dev->chip_id == TAS5825 ? + TAS5825_REG_SWRESET_RESET : TASDEVICE_REG_SWRESET_RESET); if (ret < 0) dev_err(tas_dev->dev, diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index ea3cdb8553de..cf6bd4d63f31 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,8 +30,10 @@ #include <sound/tas2781.h> #include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> +#include <sound/tas2x20-tlv.h> #include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> +#include <sound/tas5825-tlv.h> #include <linux/unaligned.h> #define X2563_CL_STT_VAL(xreg, xval) \ @@ -98,16 +100,26 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { }; static const struct i2c_device_id tasdevice_id[] = { + { "tas2020", TAS2020 }, + { "tas2118", TAS2118 }, + { "tas2120", TAS2120 }, + { "tas2320", TAS2320 }, { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, + { "tas5825", TAS5825 }, {} }; MODULE_DEVICE_TABLE(i2c, tasdevice_id); #ifdef CONFIG_OF static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2020" }, + { .compatible = "ti,tas2118" }, + { .compatible = "ti,tas2120" }, + { .compatible = "ti,tas2320" }, { .compatible = "ti,tas2563" }, { .compatible = "ti,tas2781" }, + { .compatible = "ti,tas5825" }, {}, }; MODULE_DEVICE_TABLE(of, tasdevice_of_match); @@ -797,7 +809,7 @@ static int tasdev_nop_get( return 0; } -static int tas2563_digital_gain_get( +static int tasdevice_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -823,15 +835,15 @@ static int tas2563_digital_gain_get( while (r > 1 + l) { mid = (l + r) / 2; - ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + ar_mid = get_unaligned_be32(tas_dev->dvc_tlv_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]); + ar_l = get_unaligned_be32(tas_dev->dvc_tlv_table[l]); + ar_r = get_unaligned_be32(tas_dev->dvc_tlv_table[r]); /* find out the member same as or closer to the current volume */ ucontrol->value.integer.value[0] = @@ -841,7 +853,7 @@ static int tas2563_digital_gain_get( return 0; } -static int tas2563_digital_gain_put( +static int tasdevice_digital_gain_put( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -867,7 +879,7 @@ static int tas2563_digital_gain_put( } volrd = get_unaligned_be32(&data[0]); - volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + volwr = get_unaligned_be32(tas_dev->dvc_tlv_table[vol]); if (volrd == volwr) { rc = 0; @@ -876,7 +888,7 @@ static int tas2563_digital_gain_put( for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_bulk_write(tas_dev, i, reg, - (unsigned char *)tas2563_dvc_table[vol], 4); + (unsigned char *)tas_dev->dvc_tlv_table[vol], 4); if (ret) { dev_err(tas_dev->dev, "%s, set digital vol error in dev %d\n", @@ -892,11 +904,6 @@ static int tas2563_digital_gain_put( return rc; } -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), @@ -907,6 +914,16 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { SND_SOC_BYTES_EXT("Amp XMA2 Data", 6, tasdev_XMA2_data_get, NULL), }; +static const struct snd_kcontrol_new tas2x20_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2X20_AMP_LEVEL, + 0, 0, 42, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas2x20_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2X20_DVC_LEVEL, + 0, 0, ARRAY_SIZE(tas2x20_dvc_table) - 1, 0, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, + tas2x20_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -916,6 +933,15 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, tas2781_dvc_tlv), }; +static const struct snd_kcontrol_new tas5825_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS5825_AMP_LEVEL, + 0, 0, 31, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS5825_DVC_LEVEL, + 0, 0, 254, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_cali_controls[] = { SND_SOC_BYTES_EXT("Amp Latch Data", 3, tas2781_latch_reg_get, NULL), }; @@ -923,7 +949,7 @@ static const struct snd_kcontrol_new tas2781_cali_controls[] = { 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, - tas2563_digital_gain_get, tas2563_digital_gain_put, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, tas2563_dvc_tlv), }; @@ -968,7 +994,7 @@ static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; - uinfo->value.integer.min = TAS2563; + uinfo->value.integer.min = TAS2020; uinfo->value.integer.max = TAS2781; return 0; @@ -1168,9 +1194,9 @@ static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *active_dev_num, *chip_id; + char *active_dev_num, *chip_id, *fw_load; char *conf_name, *prog_name; - int nr_controls = 4; + int nr_controls = 5; int mix_index = 0; /* Alloc kcontrol via devm_kzalloc, which don't manually @@ -1228,6 +1254,19 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].get = tasdevice_get_chip_id; mix_index++; + fw_load = devm_kstrdup(tas_priv->dev, "Speaker Force Firmware Load", + GFP_KERNEL); + if (!fw_load) + return -ENOMEM; + + dsp_ctrls[mix_index].name = fw_load; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_info_bool_ext; + dsp_ctrls[mix_index].put = tasdev_force_fwload_put; + dsp_ctrls[mix_index].get = tasdev_force_fwload_get; + dsp_ctrls[mix_index].private_value = 0UL; + mix_index++; + return snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); } @@ -1587,6 +1626,9 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * failing to load DSP firmware is NOT an error. */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; + /* There is no DSP firmware required for TAS2118/2X20. */ + if (tas_priv->chip_id < TAS2563) + goto out; if (tas_priv->name_prefix) scnprintf(tas_priv->coef_binaryname, 64, "%s-%s_coef.bin", tas_priv->name_prefix, tas_priv->dev_name); @@ -1608,34 +1650,37 @@ static void tasdevice_fw_ready(const struct firmware *fmw, 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; - /* If calibrated data occurs error, dsp will still works with default - * calibrated data inside algo. - */ - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->name_prefix) - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s-%s_cal_0x%02x.bin", tas_priv->name_prefix, - tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - else - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s_cal_0x%02x.bin", tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - ret = tas2781_load_calibration(tas_priv, - tas_priv->cal_binaryname[i], i); - if (ret != 0) - dev_err(tas_priv->dev, - "%s: load %s error, default will effect\n", - __func__, tas_priv->cal_binaryname[i]); + /* There is no calibration required for TAS5825. */ + if (tas_priv->chip_id < TAS5825) { + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + /* If calibrated data occurs error, dsp will still works + * with default calibrated data inside algo. + */ + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->name_prefix) + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s-%s_cal_0x%02x.bin", + tas_priv->name_prefix, + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + else + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s_cal_0x%02x.bin", + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_priv, + tas_priv->cal_binaryname[i], i); + if (ret != 0) + dev_err(tas_priv->dev, + "%s: load %s error, keep default.\n", + __func__, tas_priv->cal_binaryname[i]); + } } tasdevice_prmg_load(tas_priv, 0); @@ -1658,7 +1703,8 @@ static void tasdevice_fw_ready(const struct firmware *fmw, comp, &acoustic_ctl_fops); #endif out: - if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { + if ((tas_priv->fw_state == TASDEVICE_RCA_FW_OK) && + (tas_priv->chip_id >= TAS2563)) { /* If DSP FW fail, DSP kcontrol won't be created. */ tasdevice_dsp_remove(tas_priv); } @@ -1804,13 +1850,27 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) int rc; switch (tas_priv->chip_id) { + case TAS2020: + case TAS2118: + case TAS2120: + case TAS2320: + p = (struct snd_kcontrol_new *)tas2x20_snd_controls; + size = ARRAY_SIZE(tas2x20_snd_controls); + tas_priv->dvc_tlv_table = tas2x20_dvc_table; + break; case TAS2781: p = (struct snd_kcontrol_new *)tas2781_snd_controls; size = ARRAY_SIZE(tas2781_snd_controls); break; + case TAS5825: + p = (struct snd_kcontrol_new *)tas5825_snd_controls; + size = ARRAY_SIZE(tas5825_snd_controls); + break; default: p = (struct snd_kcontrol_new *)tas2563_snd_controls; size = ARRAY_SIZE(tas2563_snd_controls); + tas_priv->dvc_tlv_table = tas2563_dvc_table; + break; } rc = snd_soc_add_component_controls(codec, p, size); @@ -1850,8 +1910,6 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .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, @@ -1967,7 +2025,13 @@ static void tasdevice_i2c_remove(struct i2c_client *client) #ifdef CONFIG_ACPI static const struct acpi_device_id tasdevice_acpi_match[] = { - { "TAS2781", TAS2781 }, + { "TXNW2020", TAS2020 }, + { "TXNW2118", TAS2118 }, + { "TXNW2120", TAS2120 }, + { "TXNW2320", TAS2320 }, + { "TXNW2563", TAS2563 }, + { "TXNW2781", TAS2781 }, + { "TXNW5825", TAS5825 }, {}, }; -- 2.43.0

1 0

[tiwai-sound:for-next 17/18] sound/usb/usx2y/us144mkii_midi.c:65 tascam_midi_in_urb_complete() error: we previously assumed 'tascam' could be null (see line 54)
by Dan Carpenter 19 Aug '25

19 Aug '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: d9f0633867711cff87c5604a4f90daa8ab986e4f commit: 0ec417aa85467c6fad0563621b74868be0faf00d [17/18] ALSA: usb-audio: Add infrastructure for TASCAM US-144MKII config: powerpc64-randconfig-r073-20250819 (https://download.01.org/0day-ci/archive/20250819/202508192109.lcMrINK1-lkp@…) compiler: clang version 22.0.0git (https://github.com/llvm/llvm-project 93d24b6b7b148c47a2fa228a4ef31524fa1d9f3f) 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> | Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> | Closes: https://lore.kernel.org/r/202508192109.lcMrINK1-lkp@intel.com/ smatch warnings: sound/usb/usx2y/us144mkii_midi.c:65 tascam_midi_in_urb_complete() error: we previously assumed 'tascam' could be null (see line 54) vim +/tascam +65 sound/usb/usx2y/us144mkii_midi.c 67afec157fe63d Šerif Rami 2025-08-14 38 67afec157fe63d Šerif Rami 2025-08-14 39 void tascam_midi_in_urb_complete(struct urb *urb) 67afec157fe63d Šerif Rami 2025-08-14 40 { 67afec157fe63d Šerif Rami 2025-08-14 41 struct tascam_card *tascam = urb->context; 67afec157fe63d Šerif Rami 2025-08-14 42 int ret; 67afec157fe63d Šerif Rami 2025-08-14 43 67afec157fe63d Šerif Rami 2025-08-14 44 if (urb->status) { 67afec157fe63d Šerif Rami 2025-08-14 45 if (urb->status != -ENOENT && urb->status != -ECONNRESET && 67afec157fe63d Šerif Rami 2025-08-14 46 urb->status != -ESHUTDOWN && urb->status != -EPROTO) { 67afec157fe63d Šerif Rami 2025-08-14 47 dev_err_ratelimited(tascam->card->dev, ^^^^^^^^^^^^ Unchecked dereference. 67afec157fe63d Šerif Rami 2025-08-14 48 "MIDI IN URB failed: status %d\n", 67afec157fe63d Šerif Rami 2025-08-14 49 urb->status); 67afec157fe63d Šerif Rami 2025-08-14 50 } 67afec157fe63d Šerif Rami 2025-08-14 51 goto out; 67afec157fe63d Šerif Rami 2025-08-14 52 } 67afec157fe63d Šerif Rami 2025-08-14 53 67afec157fe63d Šerif Rami 2025-08-14 @54 if (tascam && atomic_read(&tascam->midi_in_active) && ^^^^^^ Is this required? 67afec157fe63d Šerif Rami 2025-08-14 55 urb->actual_length > 0) { 67afec157fe63d Šerif Rami 2025-08-14 56 kfifo_in_spinlocked(&tascam->midi_in_fifo, urb->transfer_buffer, 67afec157fe63d Šerif Rami 2025-08-14 57 urb->actual_length, &tascam->midi_in_lock); 67afec157fe63d Šerif Rami 2025-08-14 58 schedule_work(&tascam->midi_in_work); 67afec157fe63d Šerif Rami 2025-08-14 59 } 67afec157fe63d Šerif Rami 2025-08-14 60 67afec157fe63d Šerif Rami 2025-08-14 61 usb_get_urb(urb); 67afec157fe63d Šerif Rami 2025-08-14 62 usb_anchor_urb(urb, &tascam->midi_in_anchor); ^^^^^^^^ Unchecked dereference. 67afec157fe63d Šerif Rami 2025-08-14 63 ret = usb_submit_urb(urb, GFP_ATOMIC); 67afec157fe63d Šerif Rami 2025-08-14 64 if (ret < 0) { 67afec157fe63d Šerif Rami 2025-08-14 @65 dev_err(tascam->card->dev, 67afec157fe63d Šerif Rami 2025-08-14 66 "Failed to resubmit MIDI IN URB: error %d\n", ret); 67afec157fe63d Šerif Rami 2025-08-14 67 usb_unanchor_urb(urb); 67afec157fe63d Šerif Rami 2025-08-14 68 usb_put_urb(urb); 67afec157fe63d Šerif Rami 2025-08-14 69 } 67afec157fe63d Šerif Rami 2025-08-14 70 out: 67afec157fe63d Šerif Rami 2025-08-14 71 usb_put_urb(urb); 67afec157fe63d Šerif Rami 2025-08-14 72 } -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

Bug 220109 - Audio disappears on HP 15-fc000 after warm boot again
by Kailang 19 Aug '25

19 Aug '25

Hi Takashi, Attach patch will solve issue. BR, Kailang

2 1

[PATCH v2] ALSA: hda/tas2781: Add name prefix tas2781 for tas2781's dvc_tlv and amp_vol_tlv
by Shenghao Ding 18 Aug '25

18 Aug '25

With some new devices adding into the driver, dvc_tlv and amp_vol_tlv will cause confusion for customers on which devices they support. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Put Fixes tag. --- include/sound/tas2781-tlv.h | 6 +++--- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- sound/hda/codecs/side-codecs/tas2781_hda_spi.c | 6 ++++-- sound/soc/codecs/tas2781-i2c.c | 4 ++-- 4 files changed, 10 insertions(+), 8 deletions(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index ef9b9f19d212..273224df9282 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 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -15,7 +15,7 @@ #ifndef __TAS2781_TLV_H__ #define __TAS2781_TLV_H__ -static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 50, 0); -static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_dvc_tlv, -10000, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_amp_tlv, 1100, 50, 0); #endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 3471b9ab52cb..318f8c58ae61 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -269,7 +269,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { static const struct snd_kcontrol_new tas2781_snd_controls[] = { ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_putvol, tas2781_amp_tlv), ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c index 09a5d0f131b2..b9a55672bf15 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c @@ -494,9 +494,11 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new tas2781_snd_ctls[] = { ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_AMP_LEVEL, 1, 0, 20, 0, - tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_getvol, tas2781_amp_putvol, + tas2781_amp_tlv), ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_DVC_LVL, 0, 0, 200, 1, - tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), + tas2781_digital_getvol, tas2781_digital_putvol, + tas2781_dvc_tlv), ACARD_SINGLE_BOOL_EXT(NULL, 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e4ca43e006db..ea3cdb8553de 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -910,10 +910,10 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_putvol, tas2781_amp_tlv), SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, - tas2781_digital_putvol, dvc_tlv), + tas2781_digital_putvol, tas2781_dvc_tlv), }; static const struct snd_kcontrol_new tas2781_cali_controls[] = { -- 2.43.0

2 1

[PATCH v1] ALSA: hda/tas2781: Add name prefix tas2781 for tas2781's dvc_tlv and amp_vol_tlv
by Shenghao Ding 15 Aug '25

15 Aug '25

With some new devices adding into the driver, dvc_tlv and amp_vol_tlv will cause confusion for customers on which devices they support. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2781-tlv.h | 6 +++--- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- sound/hda/codecs/side-codecs/tas2781_hda_spi.c | 6 ++++-- sound/soc/codecs/tas2781-i2c.c | 4 ++-- 4 files changed, 10 insertions(+), 8 deletions(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index ef9b9f19d212..273224df9282 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 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -15,7 +15,7 @@ #ifndef __TAS2781_TLV_H__ #define __TAS2781_TLV_H__ -static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 50, 0); -static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_dvc_tlv, -10000, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_amp_tlv, 1100, 50, 0); #endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 3471b9ab52cb..318f8c58ae61 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -269,7 +269,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { static const struct snd_kcontrol_new tas2781_snd_controls[] = { ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_putvol, tas2781_amp_tlv), ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c index 09a5d0f131b2..b9a55672bf15 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c @@ -494,9 +494,11 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new tas2781_snd_ctls[] = { ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_AMP_LEVEL, 1, 0, 20, 0, - tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_getvol, tas2781_amp_putvol, + tas2781_amp_tlv), ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_DVC_LVL, 0, 0, 200, 1, - tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), + tas2781_digital_getvol, tas2781_digital_putvol, + tas2781_dvc_tlv), ACARD_SINGLE_BOOL_EXT(NULL, 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e4ca43e006db..ea3cdb8553de 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -910,10 +910,10 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_putvol, tas2781_amp_tlv), SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, - tas2781_digital_putvol, dvc_tlv), + tas2781_digital_putvol, tas2781_dvc_tlv), }; static const struct snd_kcontrol_new tas2781_cali_controls[] = { -- 2.43.0

2 1

[PATCH v1] ALSA: hda/tas2781: Normalize the volume kcontrol name
by Baojun Xu 14 Aug '25

14 Aug '25

Change the name of the kcontrol from "Gain" to "Volume". Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 92aae19cfc8f..e4bc3bc756b0 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -256,7 +256,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { }; static const struct snd_kcontrol_new tas2781_snd_controls[] = { - ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv), ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, -- 2.43.0

3 4

[GIT PULL] ASoC fixes for v6.17-rc1
by Mark Brown 14 Aug '25

14 Aug '25

The following changes since commit 8f5ae30d69d7543eee0d70083daf4de8fe15d585: Linux 6.17-rc1 (2025-08-10 19:41:16 +0300) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.17-rc1 for you to fetch changes up to 47ed64db8c17eb16541098add865178fb7e68744: ASoC: tas2781: Normalize the volume kcontrol name (2025-08-13 15:52:53 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.17 A reasonably small collection of fixes that came in since the merge window, mostly small and driver specific plus a cleanup of the menu reorganisation to address some user confusion with the way the generic drivers had been handled. ---------------------------------------------------------------- Alexey Klimov (1): ASoC: codecs: tx-macro: correct tx_macro_component_drv name Baojun Xu (1): ASoC: tas2781: Normalize the volume kcontrol name Colin Ian King (2): ASoC: codec: sma1307: replace spelling mistake with new error message ASoC: tas2781: Fix spelling mistake "dismatch" -> "mismatch" Kuninori Morimoto (1): ASoC: generic: tidyup standardized ASoC menu for generic Sergio Perez Gonzalez (1): ASoC: stm: stm32_i2s: Fix calc_clk_div() error handling in determine_rate() Shengjiu Wang (1): ASoC: fsl_sai: replace regmap_write with regmap_update_bits Shuming Fan (2): ASoC: rt721: fix FU33 Boost Volume control not working ASoC: rt1320: fix random cycle mute issue Thorsten Blum (1): ASoC: codecs: Call strscpy() with correct size argument sound/soc/Kconfig | 4 +++- sound/soc/codecs/aw87390.c | 8 +++----- sound/soc/codecs/aw88081.c | 5 ++--- sound/soc/codecs/aw88166.c | 8 +++----- sound/soc/codecs/aw88261.c | 8 +++----- sound/soc/codecs/aw88395/aw88395.c | 8 +++----- sound/soc/codecs/aw88399.c | 8 +++----- sound/soc/codecs/lpass-tx-macro.c | 2 +- sound/soc/codecs/rt1320-sdw.c | 3 ++- sound/soc/codecs/rt721-sdca.c | 2 ++ sound/soc/codecs/rt721-sdca.h | 4 ++++ sound/soc/codecs/sma1307.c | 2 +- sound/soc/codecs/tas2781-i2c.c | 6 +++--- sound/soc/fsl/fsl_sai.c | 20 ++++++++++---------- sound/soc/stm/stm32_i2s.c | 7 ++----- 15 files changed, 45 insertions(+), 50 deletions(-)

2 1

[PATCH v1] ASoC: tas2781: Normalize the volume kcontrol name
by Baojun Xu 13 Aug '25

13 Aug '25

Change the name of the kcontrol from "Gain" to "Volume". Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index aa5f56615dad..e4ca43e006db 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -908,10 +908,10 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { }; static const struct snd_kcontrol_new tas2781_snd_controls[] = { - SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv), - SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), }; -- 2.43.0

1 1

[PATCH v1] ASoC: tas2781: Normalize the volume kcontrol name
by Baojun Xu 13 Aug '25

13 Aug '25

As more devices become supported, add "tas2781" as a suffix before the TLV name, correct the DVC step, and normalize the volume kcontrol name. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781-tlv.h | 6 ++++-- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 4 ++-- sound/hda/codecs/side-codecs/tas2781_hda_spi.c | 5 +++-- sound/soc/codecs/tas2781-i2c.c | 8 ++++---- 4 files changed, 13 insertions(+), 10 deletions(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index d87263e43fdb..38b8f828a353 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -15,7 +15,9 @@ #ifndef __TAS2781_TLV_H__ #define __TAS2781_TLV_H__ -static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); -static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_dvc_tlv, + -10000, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_amp_tlv, + 1100, 50, 0); #endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index a0b132681804..6ec93b27aa7f 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -265,9 +265,9 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { }; static const struct snd_kcontrol_new tas2781_snd_controls[] = { - ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_putvol, tas2781_amp_tlv), ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c index 09a5d0f131b2..b4664a696bad 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c @@ -494,9 +494,10 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new tas2781_snd_ctls[] = { ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_AMP_LEVEL, 1, 0, 20, 0, - tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_getvol, tas2781_amp_putvol, tas2781_amp_tlv), ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_DVC_LVL, 0, 0, 200, 1, - tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), + tas2781_digital_getvol, tas2781_digital_putvol, + tas2781_dvc_tlv), ACARD_SINGLE_BOOL_EXT(NULL, 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9f4d965a1335..a51f20cd869a 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -908,12 +908,12 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { }; static const struct snd_kcontrol_new tas2781_snd_controls[] = { - SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), - SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, + tas2781_amp_putvol, tas2781_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, - tas2781_digital_putvol, dvc_tlv), + tas2781_digital_putvol, tas2781_dvc_tlv), }; static const struct snd_kcontrol_new tas2781_cali_controls[] = { -- 2.43.0

2 3

ucm2: Qualcomm: x1e80100: Add X1E001DE-DEVKIT configuration
by GitHub pull_request - opened 13 Aug '25

13 Aug '25

alsa-project/alsa-ucm-conf pull request #602 was opened from stephan-gh: The X1E001DE Devkit needs basically the same audio configuration as the X1E80100 CRD, so just include the HiFi.conf intended for the CRD. The only difference is that there are no speakers connected to the pin header on the mainboard by default, so remove the speakers from the available outputs. If someone wants to connect speakers there they could drop the RemoveDevice line after verifying that it works correctly for their speakers. At the moment we do not have any way to automatically detect if there are speakers connected or not. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/602 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/602.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RESEND PATCH v1] ASoC: tas2781: Add keyword "init" in profile section
by Shenghao Ding 12 Aug '25

12 Aug '25

Since version 0x105, the keyword 'init' was introduced into the profile, which is used for chip initialization, particularly to store common settings for other non-initialization profiles. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2781-dsp.h | 8 ++++++++ sound/soc/codecs/tas2781-fmwlib.c | 12 ++++++++++++ sound/soc/codecs/tas2781-i2c.c | 6 ++++++ 3 files changed, 26 insertions(+) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index c3a9efa73d5d..a21f34c0266e 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -198,6 +198,14 @@ struct tasdevice_rca { int ncfgs; struct tasdevice_config_info **cfg_info; int profile_cfg_id; + /* + * Since version 0x105, the keyword 'init' was introduced into the + * profile, which is used for chip initialization, particularly to + * store common settings for other non-initialization profiles. + * if (init_profile_id < 0) + * No init profile inside the RCA firmware. + */ + int init_profile_id; }; void tasdevice_select_cfg_blk(void *context, int conf_no, diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index c9c1e608ddb7..8baf56237624 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -180,6 +180,16 @@ static struct tasdevice_config_info *tasdevice_add_config( dev_err(tas_priv->dev, "add conf: Out of boundary\n"); goto out; } + /* If in the RCA bin file are several profiles with the + * keyword "init", init_profile_id only store the last + * init profile id. + */ + if (strnstr(&config_data[config_offset], "init", 64)) { + tas_priv->rcabin.init_profile_id = + tas_priv->rcabin.ncfgs - 1; + dev_dbg(tas_priv->dev, "%s: init profile id = %d\n", + __func__, tas_priv->rcabin.init_profile_id); + } config_offset += 64; } @@ -283,6 +293,8 @@ int tasdevice_rca_parser(void *context, const struct firmware *fmw) int i; rca = &(tas_priv->rcabin); + /* Initialize to none */ + rca->init_profile_id = -1; fw_hdr = &(rca->fw_hdr); if (!fmw || !fmw->data) { dev_err(tas_priv->dev, "Failed to read %s\n", diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9f4d965a1335..f8faeffd983d 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -1641,6 +1641,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; + /* Init common setting for different audio profiles */ + if (tas_priv->rcabin.init_profile_id >= 0) + tasdevice_select_cfg_blk(tas_priv, + tas_priv->rcabin.init_profile_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + #ifdef CONFIG_SND_SOC_TAS2781_ACOUST_I2C if (tas_priv->name_prefix) acoustic_debugfs_node = devm_kasprintf(tas_priv->dev, -- 2.43.0

2 1

[PATCH v1] ASoC: tas2781: Add keyword "init" in profile section
by Shenghao Ding 12 Aug '25

12 Aug '25

Since version 0x105, the keyword 'init' was introduced into the profile, which is used for chip initialization, particularly to store common settings for other non-initialization profiles. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2781-dsp.h | 8 ++++++++ sound/soc/codecs/tas2781-fmwlib.c | 12 ++++++++++++ sound/soc/codecs/tas2781-i2c.c | 6 ++++++ 3 files changed, 26 insertions(+) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index c3a9efa73d5d..a21f34c0266e 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -198,6 +198,14 @@ struct tasdevice_rca { int ncfgs; struct tasdevice_config_info **cfg_info; int profile_cfg_id; + /* + * Since version 0x105, the keyword 'init' was introduced into the + * profile, which is used for chip initialization, particularly to + * store common settings for other non-initialization profiles. + * if (init_profile_id < 0) + * No init profile inside the RCA firmware. + */ + int init_profile_id; }; void tasdevice_select_cfg_blk(void *context, int conf_no, diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index c9c1e608ddb7..8baf56237624 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -180,6 +180,16 @@ static struct tasdevice_config_info *tasdevice_add_config( dev_err(tas_priv->dev, "add conf: Out of boundary\n"); goto out; } + /* If in the RCA bin file are several profiles with the + * keyword "init", init_profile_id only store the last + * init profile id. + */ + if (strnstr(&config_data[config_offset], "init", 64)) { + tas_priv->rcabin.init_profile_id = + tas_priv->rcabin.ncfgs - 1; + dev_dbg(tas_priv->dev, "%s: init profile id = %d\n", + __func__, tas_priv->rcabin.init_profile_id); + } config_offset += 64; } @@ -283,6 +293,8 @@ int tasdevice_rca_parser(void *context, const struct firmware *fmw) int i; rca = &(tas_priv->rcabin); + /* Initialize to none */ + rca->init_profile_id = -1; fw_hdr = &(rca->fw_hdr); if (!fmw || !fmw->data) { dev_err(tas_priv->dev, "Failed to read %s\n", diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9f4d965a1335..f8faeffd983d 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -1641,6 +1641,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; + /* Init common setting for different audio profiles */ + if (tas_priv->rcabin.init_profile_id >= 0) + tasdevice_select_cfg_blk(tas_priv, + tas_priv->rcabin.init_profile_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + #ifdef CONFIG_SND_SOC_TAS2781_ACOUST_I2C if (tas_priv->name_prefix) acoustic_debugfs_node = devm_kasprintf(tas_priv->dev, -- 2.43.0

2 1

[PATCH v1] ALSA: hda: Add TAS5825 support
by Baojun Xu 11 Aug '25

11 Aug '25

Add TAS5825 support in TI's HDA driver. TAS5825 is an on-chip DSP, but no calibration is required, and no global address support smart amplifier devices. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781-dsp.h | 3 + include/sound/tas2781.h | 4 +- include/sound/tas5825-tlv.h | 24 +++++++ .../hda/codecs/side-codecs/tas2781_hda_i2c.c | 28 ++++++++ sound/soc/codecs/tas2781-fmwlib.c | 67 ++++++++++++++++++- 5 files changed, 121 insertions(+), 5 deletions(-) create mode 100644 include/sound/tas5825-tlv.h diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index c3a9efa73d5d..49bbf24d6559 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -34,6 +34,7 @@ #define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 #define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 #define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 +#define PPC3_VERSION_TAS5825_BASE 0x114200 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -53,6 +54,8 @@ enum tasdevice_dsp_dev_idx { TASDEVICE_DSP_TAS_2781_DUAL_MONO, TASDEVICE_DSP_TAS_2781_21, TASDEVICE_DSP_TAS_2781_QUAD, + TASDEVICE_DSP_TAS_5825_MONO, + TASDEVICE_DSP_TAS_5825_DUAL, TASDEVICE_DSP_TAS_MAX_DEVICE }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 3875e92f1ec5..f0aefc04a957 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -49,9 +49,9 @@ #define TASDEVICE_REG(book, page, reg) (((book * 256 * 128) + \ (page * 128)) + reg) -/* Software Reset */ +/* Software Reset, compatble with new device (TAS5825). */ #define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) -#define TASDEVICE_REG_SWRESET_RESET BIT(0) +#define TASDEVICE_REG_SWRESET_RESET (BIT(0) | BIT(4)) /* Checksum */ #define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) diff --git a/include/sound/tas5825-tlv.h b/include/sound/tas5825-tlv.h new file mode 100644 index 000000000000..95f2d3fad120 --- /dev/null +++ b/include/sound/tas5825-tlv.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS5825 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS5825 hda driver implements for one or two TAS5825 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS5825_TLV_H__ +#define __TAS5825_TLV_H__ + +#define TAS5825_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x4c) +#define TAS5825_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x54) + +static const __maybe_unused DECLARE_TLV_DB_SCALE( + tas5825_dvc_tlv, -10300, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE( + tas5825_amp_tlv, -1550, 50, 0); + +#endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 45ac5e41bd4f..71ee946eb748 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -26,6 +26,7 @@ #include <sound/tlv.h> #include <sound/tas2770-tlv.h> #include <sound/tas2781-tlv.h> +#include <sound/tas5825-tlv.h> #include "hda_local.h" #include "hda_auto_parser.h" @@ -50,6 +51,7 @@ enum device_chip_id { HDA_TAS2563, HDA_TAS2770, HDA_TAS2781, + HDA_TAS5825, HDA_OTHERS }; @@ -272,6 +274,17 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_force_fwload_get, tas2781_force_fwload_put), }; +static const struct snd_kcontrol_new tas5825_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS5825_AMP_LEVEL, + 0, 0, 31, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_amp_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS5825_DVC_LEVEL, + 0, 0, 254, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + static const struct snd_kcontrol_new tasdevice_prof_ctrl = { .name = "Speaker Profile Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, @@ -501,6 +514,12 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) ARRAY_SIZE(tas2781_snd_controls)); tasdevice_dspfw_init(context); break; + case HDA_TAS5825: + tasdev_add_kcontrols(tas_priv, hda_priv->snd_ctls, codec, + &tas5825_snd_controls[0], + ARRAY_SIZE(tas5825_snd_controls)); + tasdevice_dspfw_init(context); + break; case HDA_TAS2563: tasdevice_dspfw_init(context); break; @@ -628,6 +647,7 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW2781:00-tas2781-hda.0")) { device_name = "TXNW2781"; + hda_priv->hda_chip_id = HDA_TAS2781; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; } else if (strstr(dev_name(&clt->dev), "INT8866")) { @@ -639,6 +659,13 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) hda_priv->hda_chip_id = HDA_TAS2563; hda_priv->save_calibration = tas2563_save_calibration; tas_hda->priv->global_addr = TAS2563_GLOBAL_ADDR; + } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW5825")) { + /* + * TAS5825, integrated on-chip DSP without + * global I2C address and calibration supported. + */ + device_name = "TXNW5825"; + hda_priv->hda_chip_id = HDA_TAS5825; } else { return -ENODEV; } @@ -775,6 +802,7 @@ static const struct acpi_device_id tas2781_acpi_hda_match[] = { {"TIAS2781", 0 }, {"TXNW2770", 0 }, {"TXNW2781", 0 }, + {"TXNW5825", 0 }, {} }; MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index c9c1e608ddb7..d69faef8a4d7 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -91,7 +91,7 @@ struct blktyp_devidx_map { }; static const char deviceNumber[TASDEVICE_DSP_TAS_MAX_DEVICE] = { - 1, 2, 1, 2, 1, 1, 0, 2, 4, 3, 1, 2, 3, 4 + 1, 2, 1, 2, 1, 1, 0, 2, 4, 3, 1, 2, 3, 4, 1, 2 }; /* fixed m68k compiling issue: mapping table can save code field */ @@ -509,6 +509,56 @@ static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, return offset; } +static int fw_parse_tas5825_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 65 unused byts*/ + offset += 65; + offset = fw_parse_data_kernel(tas_fmw, &(program->dev_data), + fmw, offset); + if (offset < 0) + return offset; + } + + return offset; +} + +static int fw_parse_tas5825_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 8 bytes*/ + offset += 72; + offset = fw_parse_data_kernel(tas_fmw, &(config->dev_data), + fmw, offset); + if (offset < 0) + return offset; + } + + return offset; +} + static int fw_parse_program_data_kernel( struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) @@ -1826,7 +1876,8 @@ static void dspbin_type_check(struct tasdevice_priv *tas_priv, else tas_priv->dspbin_typ = TASDEV_ALPHA; } - if (tas_priv->dspbin_typ != TASDEV_BASIC) + if ((tas_priv->dspbin_typ != TASDEV_BASIC) && + (ppcver < PPC3_VERSION_TAS5825_BASE)) tas_priv->fw_parse_fct_param_address = fw_parse_fct_param_address; } @@ -1837,7 +1888,17 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION_BASE) { + if (ppcver >= PPC3_VERSION_TAS5825_BASE) { + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_tas5825_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_tas5825_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + dspbin_type_check(tas_priv, ppcver); + } else if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = -- 2.43.0

1 0

ucm2: Qualcomm: add Radxa Dragon Q6A
by GitHub pull_request - opened 10 Aug '25

10 Aug '25

alsa-project/alsa-ucm-conf pull request #601 was opened from strongtz: This is an upcoming SBC (single board computer) based on QCS6490. Full specs: - Headphone jack using WCD9380 codec - Onboard DisplayPort to HDMI bridge using Radxa RA620 - MI2S0 available via the 40-Pin GPIO header (unused by default) - AMICs using WCD9380 codec via a dedicated MIC header (unused by default) The kernel device tree part is not yet but going to be upstreamed soon. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/601 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/601.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[HDA][HP EliteDesk 800 G5][8086:a348] – HDMI audio missing after BIOS update
by Hermanns Cyrille 09 Aug '25

09 Aug '25

Hello ALSA team, On an HP EliteDesk 800 G5 Desktop Mini with BIOS R21 Ver. 02.22.00, audio controller Intel Cannon Lake PCH cAVS [8086:a348], HDMI audio outputs are no longer detected after this BIOS update. System: - Linux Mint 22.1 Cinnamon 6.4.8 - Kernel: 6.8.0-71-generic - Codecs: Conexant CX20632 (analog), Intel Kabylake HDMI (digital) Issue: After boot, HDMI outputs appear as “Not connected”. HDMI audio works only if I use hdajackretask (Intel Kabylake HDMI → Show unconnected pins → set to HDMI/DisplayPort) and select “Install boot override”. ALSA report: http://alsa-project.org/db/?f=a1ee220062b23ad5c13c6d38cde3f1d046ce6260 Request: Would it be possible to add a quirk for this model so HDMI pins are enabled automatically at boot? Thank you for your help.

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Problems with "snd_pcm_hw_params_set_rate_near" and "snd_pcm_hw_params_set_period_size_near" in x86_64 assembly.
by GitHub issues - opened 08 Aug '25

08 Aug '25

alsa-project/alsa-lib issue #471 was opened from sc1ntilla: ### **PREAMBLE:** I have never worked with any sound system before, so I'm almost sure that it's just me doing something wrong, but I can't see what, so I'm asking for help. I'm doing this project for fun, and to learn a lot of stuff, so feel free to give your opinions / advice if you want to. ### **WHAT HAPPENS:** When calling `snd_pcm_hw_params_set_rate_near` and `snd_pcm_hw_params_set_period_size_near` memory protection violation is issued. I find that strange, because `snd_pcm_hw_params_set_buffer_size_near` works perfectly fine, and it's similiar to these two. ### **WHAT I TRIED:** - Making sure that stack after call is on address divisible by 96 (by adding 8 to stack pointer), - Making sure that `*val` argument is on address divisible by 96, - Misaligning stack by 8, 16, 24 and 32 before calling these functions (by subtracting from stack pointer), - Providing 0 to `*val` argument, to rule out that writing to output crashes, - Providing valid output address to `*dir` argument to rule out that 0 is causing crashes, - Calling these functions in different order, - Trying to call `snd_pcm_set_params` instead, but it says _"Channels count (0) not available for PLAYBACK: Invalid argument"_ however, I'm sure that I provided a valid number (2). I even hardcoded it. But to no avail. Is it my misunderstanding of how alsa-lib works? Very likely. Is my assembly code faulty? Quite possible. I just can't see where the problem lies. If you see error somewhere, please let me know. ### **ADDITIONAL INFORMATION:** I have multiple audio outputs: monitors with built-in speakers, standalone speakers, and analog jack headphones. I hardcoded `hw:2,0` as device name to open. Sometimes it results in `snd_pcm_open` returning _"File descriptor in bad state"_. When I tried using `sysdefault` it often returns _"ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.14/work/alsa-lib-1.2.14/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave"_ Below is .data section in which globals are stored, and source of function that will be used to play audio buffers. .data section: ``` .section .data .align 32 test2: .8byte 0 test: .8byte 0 SND_PCM_NONBLOCK: .8byte 0x1 SND_PCM_ASYNC: .8byte 0x2 SND_PCM_STREAM_PLAYBACK: .8byte 0 SND_PCM_FORMAT_U8: .8byte 1 SND_PCM_FORMAT_U16_LE: .8byte 4 SND_PCM_FORMAT_U24_LE: .8byte 8 SND_PCM_FORMAT_U32_LE: .8byte 12 SND_PCM_FORMAT_S8: .8byte 0 SND_PCM_FORMAT_S16_LE: .8byte 2 SND_PCM_FORMAT_S24_LE: .8byte 6 SND_PCM_FORMAT_S32_LE: .8byte 10 SND_PCM_ACCESS_MMAP_INTERLEAVED: .8byte 0 SND_PCM_STATE_PREPARED: .8byte 2 AUDIO_BUFFER_STATE_IDLE: .8byte 0 AUDIO_BUFFER_STATE_WRITING: .8byte 1 AUDIO_BUFFER_STATE_PLAYING: .8byte 2 t1: .8byte 0 t2: .8byte 0 alsa: alsa.pcm_handle: .8byte 0 alsa.hw_params: .8byte 0 alsa.last_error: .8byte 0 alsa.initialized: .8byte 0 alsa.device_name: .ascii "hw:2,0\0" audio_buffers.count_max: .8byte 16 amd64_audio_last_error: .8byte 0 ``` function: ``` amd64_audio_play: /* rdi: (input) address to audio data: * (+0/2) audio format (1: PCM, 2: IEEE754) * (+2/2) amount of channels * (+4/4) rate in hertz * (+8/4) bytes per sec (frequency * bytes per block) * (+12/2) bytes per block (number of channels * bits per sample / 8) * (+14/2) bits per sample * (+16/8) total size of samples data * (+24/8) address to samples data */ /* Align stack on 96byte boundary: */ push %r15 mov %rsp, %r15 and $-32, %rsp mov %rsp, %rax mov $96, %r14 xor %rdx, %rdx div %r14 sub %rdx, %rsp sub $96, %rsp mov %r15, (%rsp) /* Store registers: */ sub $192, %rsp call amd64_util_store_gpr_regs /* Create stack space: */ sub $192, %rsp mov %rdi, (%rsp) /* stack layout: * (+0/8) rdi (address to audio data) * (+8/8) buffer size (altered by *buffer_size_near) * (+16/8) period size (set when setting period size) * (+24/8) rate in hertz * (+32/8) temp */ xor %r15, %r15 mov 16(%rdi), %r15 mov %r15, 8(%rsp) xor %r15, %r15 mov 4(%rdi), %r15d mov %r15d, 24(%rsp) movq $0, 32(%rsp) /* Open pcm if needed: */ cmpq $0, (alsa.pcm_handle) jne .amd64_audio_play.pcm_is_open lea alsa.pcm_handle, %rdi mov %rdi, (%rdi) lea alsa.device_name, %rsi mov (SND_PCM_ACCESS_MMAP_INTERLEAVED), %rdx mov (SND_PCM_NONBLOCK), %r10 call snd_pcm_open cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_open .amd64_audio_play.pcm_is_open: /* Allocate hw params: */ cmpq $0, (alsa.hw_params) jne .amd64_audio_play.hw_params_are_allocated lea alsa.hw_params, %rdi mov %rdi, (%rdi) call snd_pcm_hw_params_malloc cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_malloc .amd64_audio_play.hw_params_are_allocated: /* Get all possible combinations into hw params: */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi call snd_pcm_hw_params_any cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_any /* Set access: */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov (SND_PCM_ACCESS_MMAP_INTERLEAVED), %rdx call snd_pcm_hw_params_set_access cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_access /* Set format: */ xor %rdi, %rdi mov (%rsp), %rdi mov 14(%rdi), %r15w mov (%rdi), %r14w cmp $1, %r14w jne .amd64_audio_play.error.format_unsupported cmpw $32, %r15w je .amd64_audio_play.format32 cmpw $24, %r15w je .amd64_audio_play.format24 cmpw $16, %r15w je .amd64_audio_play.format16 cmpw $8, %r15w je .amd64_audio_play.format8 jmp .amd64_audio_play.error.format_unsupported .amd64_audio_play.format32: mov (SND_PCM_FORMAT_S32_LE), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format24: mov (SND_PCM_FORMAT_S24_LE), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format16: mov (SND_PCM_FORMAT_S16_LE), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format8: mov (SND_PCM_FORMAT_S8), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format_end: mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi call snd_pcm_hw_params_set_format cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_format /* Set channels: */ xor %rdx, %rdx mov (%rsp), %r15 mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov 2(%r15), %dx call snd_pcm_hw_params_set_channels cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_channels /* Set buffer size near: */ /* Returns 0, meaning it succeeded. No crashes generated. */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov %rsp, %rdx add $8, %rdx mov $0, %r10 call snd_pcm_hw_params_set_buffer_size_near cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_buffer_size_near /* Set rate near: */ /* CRASHES */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov %rsp, %rdx add $24, %rdx mov $0, %r10 call snd_pcm_hw_params_set_rate_near cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_rate_near /* Set period size near: */ /* CRASHES */ mov 8(%rsp), %rax mov $4, %r15 xor %rdx, %rdx div %r15 mov %rax, 16(%rsp) mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov %rsp, %rdx add $16, %rdx mov $0, %r10 call snd_pcm_hw_params_set_period_size_near cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_period_size_near /* Free hw params: */ cmpq $0, (alsa.hw_params) je .amd64_audio_play.hw_params_not_allocated mov (alsa.hw_params), %rdi call snd_pcm_hw_params_free movq $0, (alsa.hw_params) .amd64_audio_play.hw_params_not_allocated: /* Register errors: */ xor %rax, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_open: mov %rax, (amd64_audio_last_error) mov $-1, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_malloc: mov %rax, (amd64_audio_last_error) mov $-2, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_any: mov %rax, (amd64_audio_last_error) mov $-3, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_access: mov %rax, (amd64_audio_last_error) mov $-4, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.format_unsupported: mov $-5, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_format: mov %rax, (amd64_audio_last_error) mov $-6, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_channels: mov %rax, (amd64_audio_last_error) mov $-7, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_buffer_size_near: mov %rax, (amd64_audio_last_error) mov $-8, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_period_size_near: mov %rax, (amd64_audio_last_error) mov $-9, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_rate_near: mov %rax, (amd64_audio_last_error) mov $-10, %rax jmp .amd64_audio_play.exit /* Exit: */ .amd64_audio_play.exit: /* Print error to stdout: */ mov %rax, %rdi call amd64_util_print_dec call amd64_util_print_newline cmpq $0, (amd64_audio_last_error) je .amd64_audio_play.no_error mov (amd64_audio_last_error), %rdi call snd_strerror mov %rax, %rdi call amd64_util_print2 call amd64_util_print_newline .amd64_audio_play.no_error: /* Restore stack: */ add $192, %rsp /* Restore registers without rax: */ mov %rax, 128(%rsp) call amd64_util_restore_gpr_regs mov 128(%rsp), %rax add $192, %rsp /* Restore stack alignment: */ pop %rsp pop %r15 ret ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/471 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Update hdajackretask.desktop
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-tools pull request #39 was edited from trebmuh: - adds a FR comment - change the "GenericName" to "Comment" - adds a Icon= field (closes #38 ) Request URL : https://github.com/alsa-project/alsa-tools/pull/39 Patch URL : https://github.com/alsa-project/alsa-tools/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

Update hdajackretask.desktop
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-tools pull request #39 was edited from trebmuh: - adds a FR comment - change the "genericname" to "Comment" - adds a Icon= field (closes #38 ) Request URL : https://github.com/alsa-project/alsa-tools/pull/39 Patch URL : https://github.com/alsa-project/alsa-tools/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

Update hdajackretask.desktop
by GitHub pull_request - opened 08 Aug '25

08 Aug '25

alsa-project/alsa-tools pull request #39 was opened from trebmuh: - adds a FR comment - change the "genericname" to "Comment" - adds a Icon= field (close #38 ) Request URL : https://github.com/alsa-project/alsa-tools/pull/39 Patch URL : https://github.com/alsa-project/alsa-tools/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-tools

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ucm2: Qualcomm: Add MONACO-EVK HiFi config
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #597 was edited from mohsRafi: Add UCM2 configs for the Qualcomm MONACO-EVK Board to handle: - I2S Speaker Amplifier - I2S Mic Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Qualcomm: Add MONACO-EVK HiFi config
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #597 was edited from mohsRafi: Add UCM2 configs for the Qualcomm MONACO-EVK Board to handle: - I2S Speaker Amplifier - I2S Mic Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Qualcomm: Add MONACO-EVK HiFi config
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #597 was edited from mohsRafi: Add UCM2 configs for the Qualcomm QCS9075-IQ8-EVK Board to handle: - I2S Speaker Amplifier - I2S Mic Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Rename qcs6490-rb3gen2 and qcs9075-iq-evk ucm2 conf
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #596 was edited from mohsRafi: Rename the ucm2 conf for Qualcomm qcs6490-rb3gen2 and qcs9075-iq-evk qcs6490-rb3gen2-snd-card.conf -> QCS6490-RB3Gen2.conf qcs9075-iq-evk-snd-card.conf -> LEMANS-EVK.conf Removed snd-card tags from conf files Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/596 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/596.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: sof-soundwire: add rt721 ucm support
by GitHub pull_request - opened 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #600 was opened from jack-cy-yu: Add support for rt721's UCM. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/600 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/600.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC fixes for v6.17-merge-window
by Mark Brown 06 Aug '25

06 Aug '25

The following changes since commit e95122a32e777309412e30dc638dbc88b9036811: ASoC: codecs: Add acpi_match_table for aw88399 driver (2025-07-25 13:44:29 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.17-merge-window for you to fetch changes up to 614d416dd8aee2675fb591c598308a901a660db8: ASoC: SOF: Intel: hda-sdw-bpt: fix SND_SOF_SOF_HDA_SDW_BPT dependencies (2025-08-06 12:02:32 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.17 This is a relatively small set of fixes and device quirks that came in during the merge window, the AMD changes adding support for ACP 7.2 systems are all just adding IDs for the devices rather than any substantial code - the actual code is the same as for prior versions of the platform. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: SOF: Intel: hda-sdw-bpt: fix SND_SOF_SOF_HDA_SDW_BPT dependencies Baojun Xu (1): ASoC: tas2781: Fix the wrong step for TLV on tas2781 Cezary Rojewski (1): ASoC: Intel: avs: Fix uninitialized pointer error in probe() Mark Brown (1): Add audio support for acp7.2 platform Muhammad Usama Anjum (1): ASoC: SOF: amd: acp-loader: Use GFP_KERNEL for DMA allocations in resume context Peter Jakubek (1): ASoC: Intel: sof_sdw: Add quirk for Alienware Area 51 (2025) 0CCC SKU Shengjiu Wang (1): ASoC: wm8962: Clear master mode when enter runtime suspend Venkata Prasad Potturu (4): ASoC: SOF: amd: Add sof audio support for acp7.2 platform ASoC: amd: ps: Add SoundWire pci and dma driver support for acp7.2 platform ASoC: amd: acp: Add SoundWire legacy machine driver support for acp7.2 platform ASoC: amd: acp: Add SoundWire SOF machine driver support for acp7.2 platform include/sound/tas2781-tlv.h | 2 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 3 +++ sound/soc/amd/acp/acp-sdw-sof-mach.c | 1 + sound/soc/amd/acp/soc_amd_sdw_common.h | 2 ++ sound/soc/amd/ps/acp63.h | 1 + sound/soc/amd/ps/pci-ps.c | 4 ++++ sound/soc/amd/ps/ps-sdw-dma.c | 5 +++++ sound/soc/codecs/wm8962.c | 11 +++++++++++ sound/soc/intel/avs/core.c | 3 ++- sound/soc/intel/boards/sof_sdw.c | 8 ++++++++ sound/soc/sof/amd/acp-loader.c | 6 +++--- sound/soc/sof/amd/acp.c | 8 ++++++++ sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-acp70.c | 1 + sound/soc/sof/intel/Kconfig | 3 ++- 15 files changed, 53 insertions(+), 6 deletions(-)

2 1

[PATCH 0/4] Add audio support for acp7.2 platform
by Venkata Prasad Potturu 06 Aug '25

06 Aug '25

This patch series is to add legacy and sof audio support for acp7.2 platform. Venkata Prasad Potturu (4): ASoC: SOF: amd: Add sof audio support for acp7.2 platform ASoC: amd: ps: Add SoundWire pci and dma driver support for acp7.2 platform ASoC: amd: acp: Add SoundWire legacy machine driver support for acp7.2 platform ASoC: amd: acp: Add SoundWire SOF machine driver support for acp7.2 platform sound/soc/amd/acp/acp-sdw-legacy-mach.c | 3 +++ sound/soc/amd/acp/acp-sdw-sof-mach.c | 1 + sound/soc/amd/acp/soc_amd_sdw_common.h | 2 ++ sound/soc/amd/ps/acp63.h | 1 + sound/soc/amd/ps/pci-ps.c | 4 ++++ sound/soc/amd/ps/ps-sdw-dma.c | 5 +++++ sound/soc/sof/amd/acp.c | 8 ++++++++ sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-acp70.c | 1 + 9 files changed, 26 insertions(+) -- 2.43.0

2 5

[PATCH v1] ASoC: tas2781: Fix the wrong step for TLV on tas2781
by Baojun Xu 06 Aug '25

06 Aug '25

The step for TLV on tas2781, should be 50 (-0.5dB). Fixes: 678f38eba1f2 ("ASoC: tas2781: Add Header file for tas2781 driver") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781-tlv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index d87263e43fdb..ef9b9f19d212 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -15,7 +15,7 @@ #ifndef __TAS2781_TLV_H__ #define __TAS2781_TLV_H__ -static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 50, 0); static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); #endif -- 2.43.0

2 1

Re: ALSAfirewire broken / Pipewire 90ms delay
by Goran Kovac 06 Aug '25

06 Aug '25

While this is mostly a non issue for playback only, recording in studio environment is unfortunately unusable. Would be really nice if this can be solved. My RME 800 can go really low with buffer size but its always +90ms.

1 0

Re: ASoC HDMI errors with sof-hda-dsp on Intel platform (no PCM for intel-hdmi-hifiX)
by Damian Tometzki 06 Aug '25

06 Aug '25

Hi all, I'm experiencing ASoC errors on an Intel-based system using the `sof-hda-dsp` driver. After boot, the kernel logs repeated errors related to HDMI outputs: [ 90.933290] snd_hda_codec_generic ehdaudio0D2: snd_soc_find_pcm_from_dai: didn't find PCM for DAI intel-hdmi-hifi1 [ 90.933295] snd_hda_codec_generic ehdaudio0D2: ASoC error (-22): at snd_soc_dai_startup() on intel-hdmi-hifi1 ... [ 91.016259] HDMI3: ASoC error (-22): at dpcm_fe_dai_startup() on HDMI3 These errors are repeated for HDMI1, HDMI2, and HDMI3. It appears that `snd_soc_find_pcm_from_dai()` fails to resolve the DAI to a PCM, and this breaks HDMI audio initialization. **System info:** - Kernel: current Linus build 2938ae2fa7cd - Distro: Fedora 42 - HDMI monitor with audio capability isn´t connected Is there a known fix or workaround? Thanks in advance, Damian On Sat, Aug 2, 2025 at 12:46 PM Damian Tometzki <damian(a)riscv-rocks.de> wrote: > > Hi all, > > I'm experiencing ASoC errors on an Intel-based system using the > `sof-hda-dsp` driver. After boot, the kernel logs repeated errors > related to HDMI outputs: > > [ 90.933290] snd_hda_codec_generic ehdaudio0D2: > snd_soc_find_pcm_from_dai: didn't find PCM for DAI intel-hdmi-hifi1 > [ 90.933295] snd_hda_codec_generic ehdaudio0D2: ASoC error (-22): at > snd_soc_dai_startup() on intel-hdmi-hifi1 > ... > [ 91.016259] HDMI3: ASoC error (-22): at dpcm_fe_dai_startup() on HDMI3 > > These errors are repeated for HDMI1, HDMI2, and HDMI3. It appears that > `snd_soc_find_pcm_from_dai()` fails to resolve the DAI to a PCM, and > this breaks HDMI audio initialization. > > **System info:** > - Kernel: current Linus build 2938ae2fa7cd > - Distro: Fedora 42 > - HDMI monitor with audio capability isn´t connected > > Is there a known fix or workaround? > Thanks in advance, > Damian

1 0

amixer: always use line buffering for (s)events
by GitHub pull_request - opened 06 Aug '25

06 Aug '25

alsa-project/alsa-utils pull request #306 was opened from elenril: Line buffering is the obviously correct mode for (s)events, but block buffering would typically be used when piping its output to another program. Request URL : https://github.com/alsa-project/alsa-utils/pull/306 Patch URL : https://github.com/alsa-project/alsa-utils/pull/306.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v1] ALSA: hda/tas2781: Support L"SmartAmpCalibrationData" to save calibrated data
by Shenghao Ding 06 Aug '25

06 Aug '25

Some devices save the calibrated data into L"CALI_DATA", and others into L"SmartAmpCalibrationData". Driver code will support both. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/hda/codecs/side-codecs/tas2781_hda.c | 47 +++++++++++++++------- sound/hda/codecs/side-codecs/tas2781_hda.h | 2 +- 2 files changed, 33 insertions(+), 16 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index 34217ce9f28e..f46d2e06c64f 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -18,6 +18,8 @@ #include "tas2781_hda.h" +#define CALIBRATION_DATA_AREA_NUM 2 + const efi_guid_t tasdev_fct_efi_guid[] = { /* DELL */ EFI_GUID(0xcc92382d, 0x6337, 0x41cb, 0xa8, 0x8b, 0x8e, 0xce, 0x74, @@ -160,36 +162,51 @@ int tas2781_save_calibration(struct tas2781_hda *hda) * manufactory. */ efi_guid_t efi_guid = tasdev_fct_efi_guid[LENOVO]; - static efi_char16_t efi_name[] = TASDEVICE_CALIBRATION_DATA_NAME; + /* + * Some devices save the calibrated data into L"CALI_DATA", + * and others into L"SmartAmpCalibrationData". + */ + static efi_char16_t *efi_name[CALIBRATION_DATA_AREA_NUM] = { + L"CALI_DATA", + L"SmartAmpCalibrationData", + }; struct tasdevice_priv *p = hda->priv; struct calidata *cali_data = &p->cali_data; unsigned long total_sz = 0; unsigned int attr, size; unsigned char *data; efi_status_t status; + int i; if (hda->catlog_id < LENOVO) efi_guid = tasdev_fct_efi_guid[hda->catlog_id]; cali_data->cali_dat_sz_per_dev = 20; size = p->ndev * (cali_data->cali_dat_sz_per_dev + 1); - /* Get real size of UEFI variable */ - status = efi.get_variable(efi_name, &efi_guid, &attr, &total_sz, NULL); - cali_data->total_sz = total_sz > size ? total_sz : size; - if (status == EFI_BUFFER_TOO_SMALL) { - /* Allocate data buffer of data_size bytes */ - data = p->cali_data.data = devm_kzalloc(p->dev, - p->cali_data.total_sz, GFP_KERNEL); - if (!data) { - p->cali_data.total_sz = 0; - return -ENOMEM; + for (i = 0; i < CALIBRATION_DATA_AREA_NUM; i++) { + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name[i], &efi_guid, &attr, + &total_sz, NULL); + cali_data->total_sz = total_sz > size ? total_sz : size; + if (status == EFI_BUFFER_TOO_SMALL) { + /* Allocate data buffer of data_size bytes */ + data = cali_data->data = devm_kzalloc(p->dev, + cali_data->total_sz, GFP_KERNEL); + if (!data) { + status = -ENOMEM; + continue; + } + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name[i], &efi_guid, + &attr, &cali_data->total_sz, data); } - /* Get variable contents into buffer */ - status = efi.get_variable(efi_name, &efi_guid, &attr, - &p->cali_data.total_sz, data); + /* Check whether get the calibrated data */ + if (status == EFI_SUCCESS) + break; } + if (status != EFI_SUCCESS) { - p->cali_data.total_sz = 0; + cali_data->total_sz = 0; return status; } diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.h b/sound/hda/codecs/side-codecs/tas2781_hda.h index 575a701c8dfb..66188909a0bb 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.h +++ b/sound/hda/codecs/side-codecs/tas2781_hda.h @@ -11,7 +11,7 @@ /* Flag of calibration registers address. */ #define TASDEV_UEFI_CALI_REG_ADDR_FLG BIT(7) -#define TASDEVICE_CALIBRATION_DATA_NAME L"CALI_DATA" + #define TASDEV_CALIB_N 5 /* -- 2.43.0

2 1

ALC4082: capturing S/PDIF output causes volume to jump to max - asus rog crosshair x670e hero
by GitHub issues - edited 06 Aug '25

06 Aug '25

alsa-project/alsa-ucm-conf issue #598 was edited from LuanVSO: hi, i had a problem where opening the kde plasma volume applet would make the volume on my headphone jump to the maximum. after muting the "PCM 2" channel on alsamixer i notice that the volume on the headphone would go muted instead. so i edited the ALC4080-HiFi.conf file and remove every mention of S/PDIF and rebooted my machine. after the change the problem didn't happen anymore. what do you need to fix the mapping problem? i think this happens because the "PCM 2" volume controls is redirecting to headphone output, but it doenst have a slider, so it is either max or muted: <img width="1259" height="967" alt="Image" src="https://github.com/user-attachments/assets/3b266bb7-7e1c-4a7a-b448-affabf06…" /> (also the S/PDIF aways appeared as connected even though i did not have anything connected to it, don't know if it is expected) [alsa-info.txt](https://github.com/user-attachments/files/21583734/alsa-info… <details> <summary>edited ALC4080-HiFi.conf</summary> ``` Define { SpeakerMixer "PCM" SpeakerMindex "0" SpeakerJack "Speaker - Output Jack" SpeakerPCM "hw:${CardId}" HeadphonesName "Front Headphones" HeadphonesMixer "PCM" HeadphonesMindex "1" HeadphonesJack "Headphone - Output Jack" HeadphonesPCM "hw:${CardId},1" Mic1Name "Microphone" Mic1Mixer "Mic" Mic1Mindex "0" Mic1Jack "Mic - Input Jack" Mic1PCM "hw:${CardId},2" Mic2Name "" Mic2Mixer "" Mic2Mindex "0" Mic2Jack "" Mic2PCM "" Line1Name "Line Input" Line1Mixer "Line" Line1Mindex "0" Line1Jack "Line - Input Jack" Line1PCM "hw:${CardId},1" # SpdifName "S/PDIF Output" # SpdifPCM "hw:${CardId},3" # SpdifMixer "PCM" # SpdifMindex "2" } If.speaker_ctl { Condition { Type ControlExists Control "name='Speaker Playback Switch'" } True.Define.SpeakerMixer "Speaker" } If.front_hp_ctl { Condition { Type ControlExists Control "name='Front Headphone Playback Switch'" } True.Define { HeadphonesMixer "Front Headphone" HeadphonesMindex "0" } } #If.spdif_ctl { # Condition { # Type ControlExists # Control "name='IEC958 Playback Switch'" # } # True.Define { # SpdifMixer "IEC958" # SpdifMindex "0" # } #} #If.spdif_nodev { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0db0:36e7)" # } # True.Define.SpdifName "" #} #If.spdif_dev2 { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0b05:(1996|1a5[23c])|0db0:1feb)" # } # True.Define.SpdifPCM "hw:${CardId},2" #} If.gigabyte-aorus-ultra { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0414:a014)" } True.Define { Mic1Name "Front Microphone" Mic1PCM "hw:${CardId},0" Mic2Name "Rear Microphone" Mic2Mixer "Mic" Mic2Mindex "1" Mic2Jack "name='Mic - Input Jack',index=1" Mic2PCM "hw:${CardId},1" SpdifName "" Line1Name "" } } If.asus-rog-usb { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1996)" } True.Define { Mic1Name "Front Microphone" Mic2Name "Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.asrock-taichi { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(26ce:0a0[68b])" } True.Define { Line1Name "" Mic1Name "Rear Input" Mic1PCM "hw:${CardId},0" Mic2Name "Front Microphone" Mic2Mixer "Mic" Mic2Jack "Front Mic - Input Jack" Mic2PCM "hw:${CardId},1" } } If.wrx80-sage { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1984)" } True.Define { Mic2Name "Front Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.msi-meg-unify { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0db0:82c7)" } True.Define { HeadphonesName "" SpdifName "" } } If.asus-rog-strix { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1999)" } True.Define { Line1PCM "hw:${CardId},3" HeadphonesName "" SpdifName "" } } SectionVerb { EnableSequence [ disdevall "" ] Value.TQ "HiFi" } SectionDevice."Speaker" { Comment "Speakers" EnableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} on" ] DisableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} off" ] Value { PlaybackChannels 2 PlaybackPriority 200 PlaybackPCM "${var:SpeakerPCM}" JackControl "${var:SpeakerJack}" PlaybackMixerElem "${var:SpeakerMixer},${var:SpeakerMindex}" } Variant."HiFi 5+1".Value.PlaybackChannels 6 Variant."HiFi 7+1".Value.PlaybackChannels 8 } If.headphones { Condition { Type String Empty "${var:HeadphonesName}" } False.SectionDevice."Headphones" { Comment "${var:HeadphonesName}" EnableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} on" ] DisableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} off" ] Value { PlaybackPriority 300 PlaybackPCM "${var:HeadphonesPCM}" JackControl "${var:HeadphonesJack}" PlaybackMixerElem "${var:HeadphonesMixer},${var:HeadphonesMindex}" } } } #If.spdif { # Condition { # Type String # Empty "${var:SpdifName}" # } # False.SectionDevice."SPDIF" { # Comment "${var:SpdifName}" # # EnableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} on" # ] # # DisableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} off" # ] # # Value { # PlaybackPriority 100 # PlaybackPCM "${var:SpdifPCM}" # PlaybackMixerElem "${var:SpdifMixer},${var:SpdifMindex}" # } # } #} If.line1 { Condition { Type String Empty "${var:Line1Name}" } False.SectionDevice."Line1" { Comment "${var:Line1Name}" EnableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} on" ] DisableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} off" ] Value { CapturePriority 100 CapturePCM "${var:Line1PCM}" JackControl "${var:Line1Jack}" CaptureMixerElem "${var:Line1Mixer},${var:Line1Mindex}" } } } If.mic1 { Condition { Type String Empty "${var:Mic1Name}" } False.SectionDevice."Mic1" { Comment "${var:Mic1Name}" EnableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} on" ] DisableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} off" ] Value { CapturePriority 300 CapturePCM "${var:Mic1PCM}" JackControl "${var:Mic1Jack}" CaptureMixerElem "${var:Mic1Mixer},${var:Mic1Mindex}" } } } If.mic2 { Condition { Type String Empty "${var:Mic2Name}" } False.SectionDevice."Mic2" { Comment "${var:Mic2Name}" EnableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} on" ] DisableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} off" ] Value { CapturePriority 400 CapturePCM "${var:Mic2PCM}" JackControl "${var:Mic2Jack}" CaptureMixerElem "${var:Mic2Mixer},${var:Mic2Mindex}" } } } ``` </details> Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/598 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v1] ALSA: hda: Fix the wrong register was used for DVC of TAS2770
by Baojun Xu 05 Aug '25

05 Aug '25

The wrong register was used for digital volume control of TAS2770, The definition was changed, and usage was also updated. Fixes: ab29b3460c5c ("ALSA: hda: Add TAS2770 support") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2770-tlv.h | 4 ++-- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/tas2770-tlv.h b/include/sound/tas2770-tlv.h index c0bd495b4a07..5518b34abdde 100644 --- a/include/sound/tas2770-tlv.h +++ b/include/sound/tas2770-tlv.h @@ -14,10 +14,10 @@ #ifndef __TAS2770_TLV_H__ #define __TAS2770_TLV_H__ -#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x17) +#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x05) #define TAS2770_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) -static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, -1000, 50, 0); static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_amp_tlv, 1100, 50, 0); #endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index a0b132681804..45ac5e41bd4f 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -260,7 +260,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { 0, 0, 20, 0, tas2781_amp_getvol, tas2781_amp_putvol, tas2770_amp_tlv), ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2770_DVC_LEVEL, - 0, 0, 31, 0, tas2781_amp_getvol, + 0, 0, 200, 1, tas2781_amp_getvol, tas2781_amp_putvol, tas2770_dvc_tlv), }; -- 2.43.0

2 1

hdajackretask.desktop missing icon
by GitHub issues - opened 05 Aug '25

05 Aug '25

alsa-project/alsa-tools issue #38 was opened from Dejweed: The hdajackretask.desktop file doesn't have the `Icon=hdajackretask`. Issue URL : https://github.com/alsa-project/alsa-tools/issues/38 Repository URL: https://github.com/alsa-project/alsa-tools

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Subject: [BUG] snd_hda_intel runtime power management unsupported on MSI MPG Z390 GAMING PLUS with Realtek codec (kernel 6.8.0) causing excessive audio power draw
by Thomas van Houten 05 Aug '25

05 Aug '25

Dear Linux kernel and ALSA developers, I am reporting an issue related to runtime power management support for the onboard Realtek audio codec on my desktop motherboard, which results in continuously high audio hardware power usage and increased idle system power draw. *System Information:* * Kernel: 6.8.0-64-generic x86_64 * Distribution: Linux Mint 21.3 Virginia * Motherboard: MSI MPG Z390 GAMING PLUS (MS-7B51) * CPU: Intel Core i5-9400F * Audio Driver: snd_hda_intel * Audio Codec: Realtek (likely ALC1220 or similar) *Issue Summary:* The onboard Realtek audio codec does not enter runtime power management low-power states, causing it to remain at 100% power usage even when idle. This significantly contributes to my system's high idle power consumption (~160W total), despite low CPU and GPU draw. *Detailed Diagnostics:* 1. *PCI Audio Devices:* text |00:1f.3 Audio device [0403]: Intel Corporation Cannon Lake PCH cAVS [8086:a348] (rev 10) 03:00.1 Audio device [0403]: Advanced Micro Devices, Inc. [AMD/ATI] Navi 31 HDMI/DP Audio [1002:ab30] 07:00.1 Audio device [0403]: Advanced Micro Devices, Inc. [AMD/ATI] Navi 21/23 HDMI/DP Audio Controller [1002:ab28] | 2. *Kernel Messages related to snd_hda_intel or audio:* No logs found via|dmesg | grep -i snd_hda_intel|or|dmesg | grep -i audio|. 3. *Power management sysfs settings:* text |cat /sys/module/snd_hda_intel/parameters/power_save 10 cat /sys/class/sound/hwC*/power/runtime_status unsupported unsupported unsupported cat /sys/class/sound/hwC*/power/control auto auto auto | The|runtime_status|reports "unsupported", indicating no runtime PM support is available for these audio devices. 4. *PulseAudio and user-space processes inspection:* Stopped PulseAudio and all audio clients, with no change in power usage or device ownership of audio controls, confirming this is not caused by user-space audio applications. 5. *Powertop Reports:* text |100.0% Device Audio codec hwC2D0: Realtek | Consistently 100% usage from the Realtek audio codec, confirmed by powertop. 6. *Driver parameter set:* |snd_hda_intel|module|power_save|parameter successfully set to 10 seconds (value 10), but no effective power saving observed. *Impact:* The lack of runtime power management support for this audio codec causes continuous high power draw, increasing system idle power consumption significantly, reducing energy efficiency. *Request:* Please investigate whether this is a driver limitation, missing firmware quirk, or configuration issue with|snd_hda_intel|on the MSI MPG Z390 GAMING PLUS motherboard's Realtek codec. Guidance, workarounds, or patches to enable runtime power management would be appreciated. *Additional notes:* * Tested power management settings according to current best practices. * BIOS is up to date as of the latest MSI release. * System behavior consistent on Live USB of the same Linux Mint version—problem not due to local configuration. Thank you for your attention to this matter. I’m happy to provide additional logs or assist with testing patches. Best regards, Thomas van Houten

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ucm2: Qualcomm: add Lenovo Ideapad 5 (Slim 5x / 2in1) support
by GitHub pull_request - opened 04 Aug '25

04 Aug '25

alsa-project/alsa-ucm-conf pull request #599 was opened from jglathe: Same layout as T14s, 2 speakers, headphone jack, DMIC Tested on Ideapad 5 2in1 14Q8X9. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/599 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/599.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082: S/PDIF volume redirecting to headphone volume - asus rog crosshair x670e hero
by GitHub issues - edited 04 Aug '25

04 Aug '25

alsa-project/alsa-ucm-conf issue #598 was edited from LuanVSO: hi, i had a problem where opening the kde plasma volume applet would make the volume on my headphone jump to the maximum. after muting the "PCM 2" channel on alsamixer i notice that the volume on the headphone would go muted instead. so i edited the ALC4080-HiFi.conf file and remove every mention of S/PDIF and rebooted my machine. after the change the problem didn't happen anymore. what do you need to fix the mapping problem? i think this happens because the "PCM 2" volume controls is redirecting to headphone output, but it doenst have a slider, so it is either max or muted: <img width="1259" height="967" alt="Image" src="https://github.com/user-attachments/assets/3b266bb7-7e1c-4a7a-b448-affabf06…" /> (also the S/PDIF aways appeared as connected even though i did not have anything connected to it, don't know if it is expected) [alsa-info.txt](https://github.com/user-attachments/files/21583734/alsa-info… <details> <summary>edited ALC4080-HiFi.conf</summary> ``` Define { SpeakerMixer "PCM" SpeakerMindex "0" SpeakerJack "Speaker - Output Jack" SpeakerPCM "hw:${CardId}" HeadphonesName "Front Headphones" HeadphonesMixer "PCM" HeadphonesMindex "1" HeadphonesJack "Headphone - Output Jack" HeadphonesPCM "hw:${CardId},1" Mic1Name "Microphone" Mic1Mixer "Mic" Mic1Mindex "0" Mic1Jack "Mic - Input Jack" Mic1PCM "hw:${CardId},2" Mic2Name "" Mic2Mixer "" Mic2Mindex "0" Mic2Jack "" Mic2PCM "" Line1Name "Line Input" Line1Mixer "Line" Line1Mindex "0" Line1Jack "Line - Input Jack" Line1PCM "hw:${CardId},1" # SpdifName "S/PDIF Output" # SpdifPCM "hw:${CardId},3" # SpdifMixer "PCM" # SpdifMindex "2" } If.speaker_ctl { Condition { Type ControlExists Control "name='Speaker Playback Switch'" } True.Define.SpeakerMixer "Speaker" } If.front_hp_ctl { Condition { Type ControlExists Control "name='Front Headphone Playback Switch'" } True.Define { HeadphonesMixer "Front Headphone" HeadphonesMindex "0" } } #If.spdif_ctl { # Condition { # Type ControlExists # Control "name='IEC958 Playback Switch'" # } # True.Define { # SpdifMixer "IEC958" # SpdifMindex "0" # } #} #If.spdif_nodev { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0db0:36e7)" # } # True.Define.SpdifName "" #} #If.spdif_dev2 { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0b05:(1996|1a5[23c])|0db0:1feb)" # } # True.Define.SpdifPCM "hw:${CardId},2" #} If.gigabyte-aorus-ultra { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0414:a014)" } True.Define { Mic1Name "Front Microphone" Mic1PCM "hw:${CardId},0" Mic2Name "Rear Microphone" Mic2Mixer "Mic" Mic2Mindex "1" Mic2Jack "name='Mic - Input Jack',index=1" Mic2PCM "hw:${CardId},1" SpdifName "" Line1Name "" } } If.asus-rog-usb { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1996)" } True.Define { Mic1Name "Front Microphone" Mic2Name "Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.asrock-taichi { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(26ce:0a0[68b])" } True.Define { Line1Name "" Mic1Name "Rear Input" Mic1PCM "hw:${CardId},0" Mic2Name "Front Microphone" Mic2Mixer "Mic" Mic2Jack "Front Mic - Input Jack" Mic2PCM "hw:${CardId},1" } } If.wrx80-sage { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1984)" } True.Define { Mic2Name "Front Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.msi-meg-unify { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0db0:82c7)" } True.Define { HeadphonesName "" SpdifName "" } } If.asus-rog-strix { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1999)" } True.Define { Line1PCM "hw:${CardId},3" HeadphonesName "" SpdifName "" } } SectionVerb { EnableSequence [ disdevall "" ] Value.TQ "HiFi" } SectionDevice."Speaker" { Comment "Speakers" EnableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} on" ] DisableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} off" ] Value { PlaybackChannels 2 PlaybackPriority 200 PlaybackPCM "${var:SpeakerPCM}" JackControl "${var:SpeakerJack}" PlaybackMixerElem "${var:SpeakerMixer},${var:SpeakerMindex}" } Variant."HiFi 5+1".Value.PlaybackChannels 6 Variant."HiFi 7+1".Value.PlaybackChannels 8 } If.headphones { Condition { Type String Empty "${var:HeadphonesName}" } False.SectionDevice."Headphones" { Comment "${var:HeadphonesName}" EnableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} on" ] DisableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} off" ] Value { PlaybackPriority 300 PlaybackPCM "${var:HeadphonesPCM}" JackControl "${var:HeadphonesJack}" PlaybackMixerElem "${var:HeadphonesMixer},${var:HeadphonesMindex}" } } } #If.spdif { # Condition { # Type String # Empty "${var:SpdifName}" # } # False.SectionDevice."SPDIF" { # Comment "${var:SpdifName}" # # EnableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} on" # ] # # DisableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} off" # ] # # Value { # PlaybackPriority 100 # PlaybackPCM "${var:SpdifPCM}" # PlaybackMixerElem "${var:SpdifMixer},${var:SpdifMindex}" # } # } #} If.line1 { Condition { Type String Empty "${var:Line1Name}" } False.SectionDevice."Line1" { Comment "${var:Line1Name}" EnableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} on" ] DisableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} off" ] Value { CapturePriority 100 CapturePCM "${var:Line1PCM}" JackControl "${var:Line1Jack}" CaptureMixerElem "${var:Line1Mixer},${var:Line1Mindex}" } } } If.mic1 { Condition { Type String Empty "${var:Mic1Name}" } False.SectionDevice."Mic1" { Comment "${var:Mic1Name}" EnableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} on" ] DisableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} off" ] Value { CapturePriority 300 CapturePCM "${var:Mic1PCM}" JackControl "${var:Mic1Jack}" CaptureMixerElem "${var:Mic1Mixer},${var:Mic1Mindex}" } } } If.mic2 { Condition { Type String Empty "${var:Mic2Name}" } False.SectionDevice."Mic2" { Comment "${var:Mic2Name}" EnableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} on" ] DisableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} off" ] Value { CapturePriority 400 CapturePCM "${var:Mic2PCM}" JackControl "${var:Mic2Jack}" CaptureMixerElem "${var:Mic2Mixer},${var:Mic2Mindex}" } } } ``` </details> Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/598 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082: S/PDIF volume redirecting to headphone volume - asus rog crosshair x670e hero
by GitHub issues - opened 04 Aug '25

04 Aug '25

alsa-project/alsa-ucm-conf issue #598 was opened from LuanVSO: hi, i had a problem where opening the kde plasma volume applet would make the volume on my headphone jump to the maximum. after muting the "PCM 2" channel on alsamixer i notice that the volume on the headphone would go muted instead. so i edited the ALC4080-HiFi.conf file and remove every mention of S/PDIF and rebooted my machine. after the change the problem didn't happen anymore. what do you need to fix the mapping problem? i think this happens because the "PCM 2" volume controls is redirecting to headphone output, but it doenst have a slider, so it is either max or muted: <img width="1259" height="967" alt="Image" src="https://github.com/user-attachments/assets/3b266bb7-7e1c-4a7a-b448-affabf06…" /> (also the S/PDIF aways appeared as connected even though i did not have anything connected to it, don't know if it is expected) [alsa-info.sh](https://paste.centos.org/view/c6545d44) <details> <summary>edited ALC4080-HiFi.conf</summary> ``` Define { SpeakerMixer "PCM" SpeakerMindex "0" SpeakerJack "Speaker - Output Jack" SpeakerPCM "hw:${CardId}" HeadphonesName "Front Headphones" HeadphonesMixer "PCM" HeadphonesMindex "1" HeadphonesJack "Headphone - Output Jack" HeadphonesPCM "hw:${CardId},1" Mic1Name "Microphone" Mic1Mixer "Mic" Mic1Mindex "0" Mic1Jack "Mic - Input Jack" Mic1PCM "hw:${CardId},2" Mic2Name "" Mic2Mixer "" Mic2Mindex "0" Mic2Jack "" Mic2PCM "" Line1Name "Line Input" Line1Mixer "Line" Line1Mindex "0" Line1Jack "Line - Input Jack" Line1PCM "hw:${CardId},1" # SpdifName "S/PDIF Output" # SpdifPCM "hw:${CardId},3" # SpdifMixer "PCM" # SpdifMindex "2" } If.speaker_ctl { Condition { Type ControlExists Control "name='Speaker Playback Switch'" } True.Define.SpeakerMixer "Speaker" } If.front_hp_ctl { Condition { Type ControlExists Control "name='Front Headphone Playback Switch'" } True.Define { HeadphonesMixer "Front Headphone" HeadphonesMindex "0" } } #If.spdif_ctl { # Condition { # Type ControlExists # Control "name='IEC958 Playback Switch'" # } # True.Define { # SpdifMixer "IEC958" # SpdifMindex "0" # } #} #If.spdif_nodev { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0db0:36e7)" # } # True.Define.SpdifName "" #} #If.spdif_dev2 { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0b05:(1996|1a5[23c])|0db0:1feb)" # } # True.Define.SpdifPCM "hw:${CardId},2" #} If.gigabyte-aorus-ultra { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0414:a014)" } True.Define { Mic1Name "Front Microphone" Mic1PCM "hw:${CardId},0" Mic2Name "Rear Microphone" Mic2Mixer "Mic" Mic2Mindex "1" Mic2Jack "name='Mic - Input Jack',index=1" Mic2PCM "hw:${CardId},1" SpdifName "" Line1Name "" } } If.asus-rog-usb { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1996)" } True.Define { Mic1Name "Front Microphone" Mic2Name "Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.asrock-taichi { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(26ce:0a0[68b])" } True.Define { Line1Name "" Mic1Name "Rear Input" Mic1PCM "hw:${CardId},0" Mic2Name "Front Microphone" Mic2Mixer "Mic" Mic2Jack "Front Mic - Input Jack" Mic2PCM "hw:${CardId},1" } } If.wrx80-sage { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1984)" } True.Define { Mic2Name "Front Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.msi-meg-unify { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0db0:82c7)" } True.Define { HeadphonesName "" SpdifName "" } } If.asus-rog-strix { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1999)" } True.Define { Line1PCM "hw:${CardId},3" HeadphonesName "" SpdifName "" } } SectionVerb { EnableSequence [ disdevall "" ] Value.TQ "HiFi" } SectionDevice."Speaker" { Comment "Speakers" EnableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} on" ] DisableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} off" ] Value { PlaybackChannels 2 PlaybackPriority 200 PlaybackPCM "${var:SpeakerPCM}" JackControl "${var:SpeakerJack}" PlaybackMixerElem "${var:SpeakerMixer},${var:SpeakerMindex}" } Variant."HiFi 5+1".Value.PlaybackChannels 6 Variant."HiFi 7+1".Value.PlaybackChannels 8 } If.headphones { Condition { Type String Empty "${var:HeadphonesName}" } False.SectionDevice."Headphones" { Comment "${var:HeadphonesName}" EnableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} on" ] DisableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} off" ] Value { PlaybackPriority 300 PlaybackPCM "${var:HeadphonesPCM}" JackControl "${var:HeadphonesJack}" PlaybackMixerElem "${var:HeadphonesMixer},${var:HeadphonesMindex}" } } } #If.spdif { # Condition { # Type String # Empty "${var:SpdifName}" # } # False.SectionDevice."SPDIF" { # Comment "${var:SpdifName}" # # EnableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} on" # ] # # DisableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} off" # ] # # Value { # PlaybackPriority 100 # PlaybackPCM "${var:SpdifPCM}" # PlaybackMixerElem "${var:SpdifMixer},${var:SpdifMindex}" # } # } #} If.line1 { Condition { Type String Empty "${var:Line1Name}" } False.SectionDevice."Line1" { Comment "${var:Line1Name}" EnableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} on" ] DisableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} off" ] Value { CapturePriority 100 CapturePCM "${var:Line1PCM}" JackControl "${var:Line1Jack}" CaptureMixerElem "${var:Line1Mixer},${var:Line1Mindex}" } } } If.mic1 { Condition { Type String Empty "${var:Mic1Name}" } False.SectionDevice."Mic1" { Comment "${var:Mic1Name}" EnableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} on" ] DisableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} off" ] Value { CapturePriority 300 CapturePCM "${var:Mic1PCM}" JackControl "${var:Mic1Jack}" CaptureMixerElem "${var:Mic1Mixer},${var:Mic1Mindex}" } } } If.mic2 { Condition { Type String Empty "${var:Mic2Name}" } False.SectionDevice."Mic2" { Comment "${var:Mic2Name}" EnableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} on" ] DisableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} off" ] Value { CapturePriority 400 CapturePCM "${var:Mic2PCM}" JackControl "${var:Mic2Jack}" CaptureMixerElem "${var:Mic2Mixer},${var:Mic2Mindex}" } } } ``` </details> Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/598 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Regression: Microphone broken on Intel Ice Lake with alsa-ucm-conf 1.2.10-150600.3.5.1 (openSUSE Leap 15.6)
by Pablo Scatizza 04 Aug '25

04 Aug '25

Dear ALSA developers, I would like to report a regression affecting the microphone on my laptop using Intel's Ice Lake-LP audio controller with the `sof-hda-dsp` driver. After upgrading `alsa-ucm-conf` to version 1.2.10-150600.3.5.1 (as distributed in openSUSE Leap 15.6), the internal microphone stops working. No capture is possible via `arecord` or any application. The device still shows up in `arecord -l`, but it does not receive audio input. Rolling back to the previous version (1.2.10-150600.1.2) using: sudo zypper install --oldpackage alsa-ucm-conf-1.2.10-150600.1.2.noarch restores full microphone functionality. **System information:** - Distribution: openSUSE Leap 15.6 - Kernel: 6.4.0-150600.23.53-default - Audio device: Intel Corporation Ice Lake-LP Smart Sound Technology Audio Controller (rev 30) - ALSA driver: sof-hda-dsp **Working `arecord -l` output with the old version:** card 0: sofhdadsp [sof-hda-dsp], device 0: HDA Analog ( *) [] card 0: sofhdadsp [sof-hda-dsp], device 6: DMIC (*) [] card 0: sofhdadsp [sof-hda-dsp], device 7: DMIC16kHz (*) [] Let me know if I can provide logs, diffs between UCM files, or further testing. I'd be happy to help. Best regards, Pablo Scatizza Neuquén, Argentina

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Qualcomm: Add QCS8275-IQ8-EVK HiFi config
by GitHub pull_request - opened 04 Aug '25

04 Aug '25

alsa-project/alsa-ucm-conf pull request #597 was opened from mohsRafi: Add UCM2 configs for the Qualcomm QCS9075-IQ8-EVK Board to handle: - I2S Speaker Amplifier - I2S Mic Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 1/2] ALSA: hda/realtek: Accelerate the front MIC charging and discharging of the 897 codec
by wangdich9700＠163.com 04 Aug '25

04 Aug '25

From: wangdicheng <wangdicheng(a)kylinos.cn> The front MIC capacitance of the Inspur board is too large, resulting in a longer charging and discharging time required, which in turn prevents recording for 5 seconds before startup. Therefore, software is used to increase the delay by 2 seconds to avoid this issue. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 877137cb09ac..88675eae9447 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1001,6 +1001,14 @@ static int alc_resume(struct hda_codec *codec) if (!spec->no_depop_delay) msleep(150); /* to avoid pop noise */ codec->patch_ops.init(codec); + + if (codec->core.subsystem_id == 0x10ec1304) { + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x04); + msleep(50); + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44); + msleep(2000); + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80); + } snd_hda_regmap_sync(codec); hda_call_check_power_status(codec, 0x01); return 0; -- 2.25.1

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[PATCH REQUEST] ALSA: hda/tas2781: Add support for ASUS VivoBook 16 Flip TP3607SA
by Ludd ite 02 Aug '25

02 Aug '25

Summary ======= The ASUS VivoBook 16 Flip TP3607SA laptop contains TI TAS2781 smart amplifiers but they are not supported by the current snd_hda_scodec_tas2781_i2c driver. The driver fails to initialize with "Platform not supported" error, preventing internal speakers from working. Hardware Information ==================== Laptop Model: ASUS VivoBook 16 Flip TP3607SA_TP3607SA BIOS Version: 1.0 PCI Audio Device: Intel Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem Vendor: 0x1043 (ASUS) Subsystem Device: 0x3e30 Audio Codec: Realtek ALC294 (HDA:10ec0294,10433e30,00100004) Smart Amplifier: TI TAS2781 on I2C bus (ACPI ID: TIAS2781:00) System Information ================== OS: Ubuntu 25.04 (development) Kernel: 6.14.0-1007-realtime Audio System: PipeWire with SOF (Sound Open Firmware) SOF Firmware: intel/sof-ipc4/lnl/sof-lnl.ri (version 2.12.0.1) Topology: intel/sof-ipc4-tplg/sof-hda-generic-2ch.tplg Current Behavior ================ 1. Audio Detection: SOF firmware loads successfully and creates speaker sink in PipeWire 2. Audio Pipeline: Complete audio pipeline works (volume controls, routing, mixing) 3. TAS2781 Driver: Fails to bind with the following errors: tas2781-hda i2c-TIAS2781:00: Failed to get Speaker id. tas2781-hda i2c-TIAS2781:00: read acpi error, ret: -2 tas2781-hda i2c-TIAS2781:00: error -ENOENT: Platform not supported tas2781-hda i2c-TIAS2781:00: probe with driver tas2781-hda failed with error -2 4. Result: No sound from internal speakers (Bluetooth and HDMI audio work fine) Technical Analysis ================== - I2C Device Present: i2c-TIAS2781:00 device exists and is detected - ACPI Information: Device has ACPI ID TIAS2781: but driver cannot read required configuration - Volume Control: Audio shows active volume bars in pavucontrol, indicating software pipeline works - Driver Status: TAS2781 driver loads but doesn't bind to the device due to platform whitelist Evidence of TAS2781 Hardware ============================= I2C device exists: $ ls /sys/bus/i2c/devices/i2c-TIAS2781:00/ firmware_node modalias name power subsystem Device modalias: $ cat /sys/bus/i2c/devices/i2c-TIAS2781:00/modalias acpi:TIAS2781: PipeWire detects speaker sink: $ pactl list sinks short | grep Speaker 60 alsa_output.pci-0000_00_1f.3-platform-skl_hda_dsp_generic.HiFi__Speaker__sink Expected Behavior ================= The TAS2781 driver should recognize the ASUS VivoBook 16 Flip TP3607SA (subsystem ID 1043:3e30) as a supported platform and successfully initialize the smart amplifiers, enabling internal speaker audio output. Workarounds Attempted ===================== 1. Manual I2C device creation: Created tas2781 devices at addresses 0x38-0x3b, but they fail with communication timeouts (error -121) 2. Driver parameters: Tried various modprobe options, but the driver doesn't expose platform override parameters 3. GPIO manipulation: Attempted to enable GPIO pins that might control amplifier power, without success 4. Kernel versions: Tested multiple 6.14.x kernels including recent versions with November 26, 2024 ASUS TAS2781 improvements Request ======= Please add support for ASUS VivoBook 16 Flip TP3607SA (subsystem ID 1043:3e30) to the TAS2781 driver's supported platform list. This appears to be a matter of adding the device ID to the driver's whitelist and possibly providing the appropriate ACPI configuration parsing. Additional Notes ================ - This is a recent laptop model (2024) with Intel Lunar Lake platform - Other ASUS laptops may have similar TAS2781 configurations - The November 26, 2024 commit "ALSA: hda/tas2781: Add speaker id check for ASUS projects" suggests ongoing work on ASUS TAS2781 support - Hardware detection is working perfectly through SOF firmware Contact Information =================== Available for testing patches or providing additional debugging information. Attachments =========== - Full dmesg output available upon request - ACPI tables dump available if needed - Detailed hardware information via sudo lshw available

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conf/pistachio: fix syntax
by GitHub pull_request - opened 01 Aug '25

01 Aug '25

alsa-project/alsa-lib pull request #470 was opened from blino: It was missing closing brackets since its introduction. Fixes: 4dfa8f08fb83 ("conf/cards: add support for pistachio-card.") Request URL : https://github.com/alsa-project/alsa-lib/pull/470 Patch URL : https://github.com/alsa-project/alsa-lib/pull/470.patch Repository URL: https://github.com/alsa-project/alsa-lib

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retire_capture_urb: Corrected urb data len
by Sean Anderson 01 Aug '25

01 Aug '25

Hi all, I have a "FiiO DigiHug USB Audio" sound card (1852:7022) [3]. I have had no problems with the audio, but I did notice a large number of message like retire_capture_urb: 4992 callbacks suppressed in my dmesg [1]. This is caused by the "Corrected urb data len." warning. The patch adding this warning [2] makes it seem like this warning should be an uncommon occurance. However, based on the number of suppressed callbacks, this seems to be happening at a rate of around 500 Hz. Is this buggy hardware? Or is this a bug in the driver? Does there need to be a quirk? Or perhaps the warning above should be a debug instead? I'm using 5.15.0-46-generic. I know, I know; this is a vendor kernel. However, I went through the commits for sound/usb/pcm.c and didn't see any major differences. I also previously saw this issue on 5.4, so this probably isn't a regression. --Sean [1] I would have liked to post the actual message, but I was unable to figure out how to disable ratelimiting. I tried setting /proc/sys/kernel/printk_ratelimit to 0, /proc/sys/kernel/printk_ratelimit_burst to 5000, and /proc/sys/kernel/printk to 8. However, nothing seemed to have any effect (except that the callbacks suppressed messages were disabled). I find this very strange, given that the callbacks suppressed message uses KERN_WARN, and so does dev_warn_ratelimited. [2] https://lore.kernel.org/all/4B28A659.6070303@ladisch.de/T/ [3] Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x1852 GYROCOM C&C Co., LTD idProduct 0x7022 bcdDevice 0.01 iManufacturer 1 FiiO iProduct 2 DigiHug USB Audio iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0182 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 58 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0012 1x 18 bytes bInterval 32 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 3 FiiO USB DAC-E10 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x003e bInCollection 2 baInterfaceNr(0) 2 baInterfaceNr(1) 3 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 9 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bSourceID 16 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 16 bSourceID 9 bControlSize 1 bmaControls(0) 0x01 Mute Control bmaControls(1) 0x02 Volume Control bmaControls(2) 0x02 Volume Control iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 6 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 32000 tSamFreq[ 3] 44100 tSamFreq[ 4] 48000 tSamFreq[ 5] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0184 1x 388 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 6 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 32000 tSamFreq[ 3] 44100 tSamFreq[ 4] 48000 tSamFreq[ 5] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0246 1x 582 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 32000 tSamFreq[ 1] 44100 tSamFreq[ 2] 48000 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0184 1x 388 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 4 Discrete tSamFreq[ 0] 32000 tSamFreq[ 1] 44100 tSamFreq[ 2] 48000 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0246 1x 582 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x2001 IEC1937_AC-3 AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 3 (FORMAT_TYPE_III) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 can't get debug descriptor: Resource temporarily unavailable Device Status: 0x0000 (Bus Powered)

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pcm: add a loop to snd_pcm_avail_delay() to avoid bogus delay values
by GitHub pull_request - opened 31 Jul '25

31 Jul '25

alsa-project/alsa-lib pull request #469 was opened from kv2019i: snd_pcm_avail_delay() is expected to report avail and delay values in atomic fashion. However the function does two separate syscalls and it cannot guarantee the avail value is the same as was used to calculate the delay. This is a problem as the reported delay is always relative to avail frames value. If application (like e.g. alsa_conformance_test) uses snd_pcm_avail_delay() to estimate the effective play position, it can observe bogus delay values (and effective play position going backwards) if snd_pcm_avail_delay() is called during a DMA burst where hw_ptr moves quickly. This commit adds a loop similar to that used in snd_pcm_hw_htimestamp() to wait until we get a stable avail reading, and only then extract the delay. This will avoid bogus values if function is called during DMA bursts. Closes: https://github.com/alsa-project/alsa-lib/issues/468 Request URL : https://github.com/alsa-project/alsa-lib/pull/469 Patch URL : https://github.com/alsa-project/alsa-lib/pull/469.patch Repository URL: https://github.com/alsa-project/alsa-lib

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snd_pcm_avail_delay() reporting inaccurate delay values
by GitHub issues - opened 31 Jul '25

31 Jul '25

alsa-project/alsa-lib issue #468 was opened from kv2019i: The alsa_conformance_test tool (https://chromium.googlesource.com/chromiumos/platform/audiotest/+/HEAD/alsa…) has recently added support to use snd_pcm_avail_delay() to estimate the actual playback position. This has proven a very useful tool to test ALSA drivers on hardware (like that support by https://github.com/thesofproject/ ) where there is significant buffering in hardware (and hw_ptr can move in bursts). When testing with high sampling rates, an issue was found where the estimated playback position seemed to go backwards in time, as observed by alsa_conformance_test. This was observed on multiple Intel platform using SOF firmware for DSP. Debugging the kernel driver delay() implementation showed, that driver never reported incorrect delay values, but somehow alsa_conformance_test still observed these. Example log: ``` alsa_conformance_test -P hw:sofnocodec,2 -d 1 -r 96000 -B 480 -c 2 -f S32_LE -p 480 --avail-delay --debug ... TIME_DIFF(s) HW_LEVEL PLAYED DELAY DIFF RATE [.. snip ] 0.000999950 960 3324 1476 96 96004.800240 0.000995762 864 3420 1380 96 96408.579560 0.001024820 672 3516 1284 96 93674.986827 0.000987638 621 3613 1187 97 98214.122988 [Notice] frame_diff -1 >= 4096. 0.000049322 576 3612 1188 -1 -20274.928024 # KV: the delay value of 1187 is never reported by driver and it causes the playback position to go backwards in time despite driver reporting the accurate delay correctly 0.000992228 480 3708 1092 96 96751.956204 0.000972403 864 3804 1476 96 98724.500027 0.000998287 768 3900 1380 96 96164.730183 0.000979150 768 3996 1284 96 98044.222029 0.000999349 672 4092 1188 96 96062.536711 0.001018172 480 4188 1092 96 94286.623478 0.000977025 960 4284 1476 96 98257.465264 0.001022131 768 4380 1380 96 93921.424945 0.000977795 768 4476 1284 96 98180.088873 0.001019073 601 4573 1187 97 95184.545170 [Notice] frame_diff -1 >= 4096. 0.000049433 576 4572 1188 -1 -20229.401412 0.000953494 480 4668 1092 96 100682.332558 0.000981225 960 4764 1476 96 97836.887564 0.001025294 768 4860 1380 96 93631.680279 ``` When instrumenting alsa-lib snd_pcm_avail_delay() it became apparent the function is calling in middle of a DMA burst (when hw_ptr is moving fast ahead and the "avail" number is changing fast). alsa-lib does two syscalls to read avail and delay, the two values are not in sync and we end up reporting a bogus value. The error is not a big one, but this is enough to fail alsa_conformance_test test plans and push the calculcated standard rate error above acceptance criteria. This could also confuse media players who use this for video lip-sync, although in theory applications should handle this (a 0...3 frame error in delay reporting is not going to make a difference unless negative value breaks the application logic like happens here with alsa_conformance_test). We can observe the error especially with higher rates (96, 176.4 and 192kHz). This is harder to hit with the more typical 48khz use-cases, and thus maybe has not been hit before. Versions tested: 1.12.24, 1.12.11 Issue URL : https://github.com/alsa-project/alsa-lib/issues/468 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ucm: use close_range on _GNU_SOURCE
by Jochen Sprickerhof 30 Jul '25

30 Jul '25

--- src/ucm/ucm_exec.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/src/ucm/ucm_exec.c b/src/ucm/ucm_exec.c index 276cf592..b79a84ae 100644 --- a/src/ucm/ucm_exec.c +++ b/src/ucm/ucm_exec.c @@ -254,8 +254,12 @@ int uc_mgr_exec(const char *prog) close(f); +#if defined(_GNU_SOURCE) + close_range(3, maxfd, 0); +#else for (f = 3; f < maxfd; f++) close(f); +#endif /* install default handlers for the forked process */ signal(SIGINT, SIG_DFL); -- 2.50.0

1 0

[PATCH v2] ALSA: hda: Fix the wrong register was used for DVC of TAS2770
by Baojun Xu 29 Jul '25

29 Jul '25

The wrong register was used for digital volume control of TAS2770, The definition was changed, and usage was also updated. Fixes: ab29b3460c5c ("ALSA: hda: Add TAS2770 support") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v2 - Correct number from -1000 to -10000 for DVC. --- include/sound/tas2770-tlv.h | 4 ++-- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/tas2770-tlv.h b/include/sound/tas2770-tlv.h index c0bd495b4a07..c7380925417a 100644 --- a/include/sound/tas2770-tlv.h +++ b/include/sound/tas2770-tlv.h @@ -14,10 +14,10 @@ #ifndef __TAS2770_TLV_H__ #define __TAS2770_TLV_H__ -#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x17) +#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x05) #define TAS2770_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) -static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, -10000, 50, 0); static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_amp_tlv, 1100, 50, 0); #endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index a0b132681804..45ac5e41bd4f 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -260,7 +260,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { 0, 0, 20, 0, tas2781_amp_getvol, tas2781_amp_putvol, tas2770_amp_tlv), ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2770_DVC_LEVEL, - 0, 0, 31, 0, tas2781_amp_getvol, + 0, 0, 200, 1, tas2781_amp_getvol, tas2781_amp_putvol, tas2770_dvc_tlv), }; -- 2.43.0

1 0

[PATCH] ASoC: amd: acp: Adjust pdm dmic gain using module parameter
by Venkata Prasad Potturu 29 Jul '25

29 Jul '25

Adjust pdm dimc gain value using module param. In case of regressions for any users that the new pdm_gain value is too high and for additional debugging, introduce a module parameter that would let them configure it. This parameter should be removed in the future: * If it's determined that the parameter is not needed, just hardcode the correct value as before * If users do end up using it to debug and report different values we should introduce a config knob that can have policy set by ucm. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/acp/acp-legacy-common.c | 3 ++- sound/soc/amd/acp/acp-pdm.c | 3 ++- sound/soc/amd/acp/amd.h | 6 +++++- 3 files changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index 57982d057c3a..dd804fb95790 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -173,7 +173,8 @@ static void set_acp_pdm_clk(struct snd_pcm_substream *substream, /* Enable default ACP PDM clk */ writel(PDM_CLK_FREQ_MASK, chip->base + ACP_WOV_CLK_CTRL); pdm_ctrl = readl(chip->base + ACP_WOV_MISC_CTRL); - pdm_ctrl |= PDM_MISC_CTRL_MASK; + pdm_ctrl &= ~ACP_WOV_GAIN_CONTROL; + pdm_ctrl |= FIELD_PREP(ACP_WOV_GAIN_CONTROL, clamp(pdm_gain, 0, 3)); writel(pdm_ctrl, chip->base + ACP_WOV_MISC_CTRL); set_acp_pdm_ring_buffer(substream, dai); } diff --git a/sound/soc/amd/acp/acp-pdm.c b/sound/soc/amd/acp/acp-pdm.c index 1bfc34c2aa53..ffb622a7a69a 100644 --- a/sound/soc/amd/acp/acp-pdm.c +++ b/sound/soc/amd/acp/acp-pdm.c @@ -38,7 +38,8 @@ static int acp_dmic_prepare(struct snd_pcm_substream *substream, /* Enable default DMIC clk */ writel(PDM_CLK_FREQ_MASK, chip->base + ACP_WOV_CLK_CTRL); dmic_ctrl = readl(chip->base + ACP_WOV_MISC_CTRL); - dmic_ctrl |= PDM_MISC_CTRL_MASK; + dmic_ctrl &= ~ACP_WOV_GAIN_CONTROL; + dmic_ctrl |= FIELD_PREP(ACP_WOV_GAIN_CONTROL, clamp(pdm_gain, 0, 3)); writel(dmic_ctrl, chip->base + ACP_WOV_MISC_CTRL); period_bytes = frames_to_bytes(substream->runtime, diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index cb8d97122f95..f2567e06ccd3 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -130,7 +130,7 @@ #define PDM_DMA_INTR_MASK 0x10000 #define PDM_DEC_64 0x2 #define PDM_CLK_FREQ_MASK 0x07 -#define PDM_MISC_CTRL_MASK 0x10 +#define ACP_WOV_GAIN_CONTROL GENMASK(4, 3) #define PDM_ENABLE 0x01 #define PDM_DISABLE 0x00 #define DMA_EN_MASK 0x02 @@ -138,6 +138,10 @@ #define PDM_TIMEOUT 1000 #define ACP_REGION2_OFFSET 0x02000000 +static int pdm_gain = 3; +module_param(pdm_gain, int, 0644); +MODULE_PARM_DESC(pdm_gain, "Gain control (0-3)"); + struct acp_chip_info { char *name; /* Platform name */ struct resource *res; -- 2.43.0

4 4

Support ohos build
by GitHub pull_request - opened 29 Jul '25

29 Jul '25

alsa-project/alsa-lib pull request #467 was opened from wyjstrong: https://github.com/alsa-project/alsa-lib/issues/466 Request URL : https://github.com/alsa-project/alsa-lib/pull/467 Patch URL : https://github.com/alsa-project/alsa-lib/pull/467.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Support ohos build
by GitHub issues - opened 29 Jul '25

29 Jul '25

alsa-project/alsa-lib issue #466 was opened from wyjstrong: Hello, ALSA is an important audio architecture in the Linux system. With it, developers can call Linux system interfaces in their programs to control and manage the underlying operating system. Thanks to its combination of high efficiency and ease of use, ALSA is also suitable for embedded devices, which highly aligns with OpenHarmony's philosophy of supporting diverse devices — currently, it has been widely used in the field of mobile application development for OpenHarmony. We will submit the adaptation code via a Pull Request (PR) as soon as possible. This issue is created to conduct in-depth discussions with the community on the integration of the adaptation code. We sincerely welcome your valuable suggestions. Thank you. Issue URL : https://github.com/alsa-project/alsa-lib/issues/466 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC updates for v6.17-2
by Mark Brown 28 Jul '25

28 Jul '25

The following changes since commit 347e9f5043c89695b01e66b3ed111755afcf1911: Linux 6.16-rc6 (2025-07-13 14:25:58 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.17-2 for you to fetch changes up to e95122a32e777309412e30dc638dbc88b9036811: ASoC: codecs: Add acpi_match_table for aw88399 driver (2025-07-25 13:44:29 +0100) ---------------------------------------------------------------- ASoC: More updates for v6.17 A few more updates, mostly fixes and device IDs plus some small enhancements for the FSL xcvr driver. ---------------------------------------------------------------- Ai Chao (3): ASoC: renesas: Use helper function for_each_child_of_node_scoped() ASoC: meson: Use helper function for_each_child_of_node_scoped() ASoC: imx-card: Use helper function for_each_child_of_node_scoped() Alexander Stein (1): ASoC: codec: tlv320aic32x4: Fix reset GPIO check Arnd Bergmann (2): ASoC: ops: dynamically allocate struct snd_ctl_elem_value ASoC: SDCA: fix HID dependency Balamurugan C (3): ASoC: Intel: soc-acpi: Add entry for sof_es8336 in PTL match table. ASoC: Intel: soc-acpi: Add entry for HDMI_In capture support in PTL match table ASoC: Intel: sof_rt5682: Add HDMI-In capture with rt5682 support for PTL. Bard Liao (2): ASoC: sdw_utils: add component_name string to dai_info ASoC: sdw_utils: generate combined spk components string Bram Vlerick (2): ASoC: tas571x: add support for tas5753 ASoC: dt-bindings: tas57xx: add tas5753 compatibility Brian Masney (6): ASoC: codecs: da7219: convert from round_rate() to determine_rate() ASoC: codecs: rt5682: convert from round_rate() to determine_rate() ASoC: codecs: rt5682s: convert from round_rate() to determine_rate() ASoC: qcom: qdsp6: q6dsp-lpass-clocks: convert from round_rate() to determine_rate() ASoC: stm: stm32_i2s: convert from round_rate() to determine_rate() ASoC: stm: stm32_sai_sub: convert from round_rate() to determine_rate() Cezary Rojewski (2): ASoC: codecs: rt5640: Drop dummy register names ASoC: Intel: avs: Add rt5640 machine board Chancel Liu (1): ASoC: imx-card: Add WM8524 support Charles Keepax (23): HID: core: Add bus define for SoundWire bus MAINTAINERS: Add SDCA maintainers entry ASoC: SDCA: Add missing default in switch in entity_pde_event() ASoC: SDCA: Fixup some kernel doc errors ASoC: SDCA: Minor selected/detected mode control fixups ASoC: SDCA: Add flag for unused IRQs ASoC: SDCA: Add some initial IRQ handlers ASoC: SDCA: Kconfig/Makefile fixups ASoC: SDCA: Pull HID and IRQ into the primary SDCA module ASoC: SDCA: Fix off by one error in IRQ bound check ASoC: SDCA: Avoid use of uninitialised local name variable ASoC: SDCA: Update memory allocations to zero initialise ASoC: SDCA: Allow read-only controls to be deferrable ASoC: SDCA: Remove overly chatty input pin list warning ASoC: SDCA: Move SDCA search functions and export ASoC: soc-dai: Add private data to snd_soc_dai ASoC: SDCA: Add helper to add DAI constraints ASoC: SDCA: Add a helper to get the SoundWire port number ASoC: SDCA: Add hw_params() helper function ASoC: SDCA: Shrink detected_mode_handler() stack frame ASoC: SDCA: Fix implicit cast from le16 ASoC: SDCA: Fix some holes in the regmap readable/writeable helpers ASoC: SDCA: Add support for -cn- value properties Chen Ni (1): ASoC: tegra: AHUB: Remove unneeded semicolon Chen-Yu Tsai (9): ASoC: dt-bindings: Convert MT8173 AFE binding to dt-schema ASoC: dt-bindings: mt8173-afe-pcm: Add power domain ASoC: dt-bindings: mt8173-afe-pcm: Allow specifying reserved memory region ASoC: dt-bindings: mt8186-afe-pcm: Allow specifying reserved memory region ASoC: dt-bindings: mt8192-afe-pcm: Allow specifying reserved memory region ASoC: mediatek: use reserved memory or enable buffer pre-allocation ASoC: mediatek: mt8183-afe-pcm: Support >32 bit DMA addresses ASoC: mediatek: mt8173-afe-pcm: use local `dev` pointer in driver callbacks ASoC: mediatek: mt8183-afe-pcm: use local `dev` pointer in driver callbacks ChiYuan Huang (2): ASoC: dt-bindings: rt9123: Append RTQ9124 description ASoC: codecs: Add support for Richtek RTQ9124 Frank Li (1): ASoC: dt-bindings: covert mxs-audio-sgtl5000.txt to yaml format Geert Uytterhoeven (2): ASoC: renesas: msiof: Convert to <linux/spi/sh_msiof.h> ASoC: img: Imagination Technologies sound should depend on MIPS Jyri Sarha (3): ASoC: SOF: ipc4: Add sof_ipc4_pipeline_state_str() for debugging ASoC: SOF: ipc4-pcm: Pipe instances to dev_dbg in multi_pipeline_state() ASoC: SOF: ipc4-pcm: Harmonize sof_ipc4_set_pipeline_state() dbg print Krzysztof Kozlowski (20): ASoC: codecs: Constify regmap configuration static variables ASoC: fsl: Constify reg_default array ASoC: codecs: wcd9335: Drop unused sido_input_src field ASoC: codecs: wcd934x: Drop unused num_rx_port/num_tx_port fields ASoC: codecs: wcd937x: Drop unused 'struct wcd937x_priv' fields ASoC: codecs: wcd938x: Drop unused 'struct wcd938x_priv' fields ASoC: codecs: wcd938x: Drop unused variant field ASoC: codecs: wcd939x: Drop unused 'struct wcd939x_priv' fields ASoC: codecs: wcd937x: Simplify with devm_regulator_bulk_get_enable() ASoC: codecs: wcd938x: Simplify with devm_regulator_bulk_get_enable() ASoC: codecs: wcd939x: Simplify with devm_regulator_bulk_get_enable() ASoC: codecs: wcd939x: Simplify return from devm_gpiod_get() error ASoC: dt-bindings: qcom,wcd939x: Document missing VDD_PX supply ASoC: codecs: wcd939x: Add VDD_PX supply ASoC: codecs: lpass: Drop unused AIF_INVALID first DAI identifier ASoC: codecs: wcd937x: Use simple defines for chipid register value ASoC: codecs: wcd938x: Use simple defines for chipid register value ASoC: codecs: wcd939x: Use simple defines for chipid register value ASoC: codecs: wcd939x: Add defines for major/minor version decoding ASoC: dt-bindings: qcom,lpass-va-macro: Define clock-names in top-level Kuninori Morimoto (62): ASoC: soc-core: save ID if param was set in fmt_single_name() ASoC: remove component->id ASoC: hdac_hdmi: remove hdac_hdmi_jack[_port]_init() ASoC: soc-dai: tidyup return value of snd_soc_xlate_tdm_slot_mask() ASoC: soc-ops-test: dynamically allocate struct snd_ctl_elem_value ASoC: ti: Standardize ASoC menu ASoC: adi: Standardize ASoC menu ASoC: amd: Standardize ASoC menu ASoC: bcm: Standardize ASoC menu ASoC: dwc: Standardize ASoC menu ASoC: fsl: Standardize ASoC menu ASoC: img: Standardize ASoC menu ASoC: mxs: Standardize ASoC menu ASoC: pxa: Standardize ASoC menu ASoC: sti: Standardize ASoC menu ASoC: stm: Standardize ASoC menu ASoC: sof: Standardize ASoC menu ASoC: au1x: Standardize ASoC menu ASoC: sdca: Standardize ASoC menu ASoC: sprd: Standardize ASoC menu ASoC: qcom: Standardize ASoC menu ASoC: meson: Standardize ASoC menu ASoC: apple: Standardize ASoC menu ASoC: atmel: Standardize ASoC menu ASoC: intel: Standardize ASoC menu ASoC: spear: Standardize ASoC menu ASoC: sunxi: Standardize ASoC menu ASoC: tegra: Standardize ASoC menu ASoC: ux500: Standardize ASoC menu ASoC: cirrus: Standardize ASoC menu ASoC: google: Standardize ASoC menu ASoC: jz4740: Standardize ASoC menu ASoC: xilinx: Standardize ASoC menu ASoC: xtensa: Standardize ASoC menu ASoC: renesas: Standardize ASoC menu ASoC: generic: Standardize ASoC menu ASoC: samsung: Standardize ASoC menu ASoC: kirkwood: Standardize ASoC menu ASoC: loongson: Standardize ASoC menu ASoC: mediatek: Standardize ASoC menu ASoC: rockchip: Standardize ASoC menu ASoC: starfive: Standardize ASoC menu ASoC: uniphier: Standardize ASoC menu ASoC: hisilicon: Standardize ASoC menu ASoC: soc-dapm: remove unnecessary definition ASoC: soc-dapm: remove EXPORT_SYMBOL_GPL() for snd_soc_dapm_free() ASoC: soc-dapm: remove snd_soc_dapm_nc_pin[_unlocked]() ASoC: soc-dapm: remove snd_soc_dapm_weak_routes() ASoC: soc-dapm: reordering function definitions ASoC: soc-dapm: reordering header definitions ASoC: soc-dapm: use component instead of cmpnt ASoC: soc-dapm: use common name for dapm ASoC: soc-dapm: add prefix on dapm_mark_endpoints_dirty() ASoC: soc-dapm: add prefix on dapm_xxx_event() ASoC: soc-dapm: add prefix on soc_dapm_dev_attrs ASoC: rt700: don't set dapm->bias_level ASoC: rt715: don't set dapm->bias_level ASoC: soc-dapm: set bias_level if snd_soc_dapm_set_bias_level() was successed ASoC: samsung: bell: don't set dapm->bias_level directly ASoC: samsung: speyside: don't set dapm->bias_level directly ASoC: samsung: tobermory: don't set dapm->bias_level directly ASoC: samsung: littlemill: don't set dapm->bias_level directly Laurentiu Mihalcea (1): ASoC: dt-bindings: cirrus,cs42xx8: add 'port' property Luca Weiss (6): ASoC: qcom: sm8250: set card driver name from match data ASoC: qcom: sm8250: add DisplayPort Jack support ASoC: qcom: sm8250: Add Fairphone 5 soundcard compatible ASoC: dt-bindings: qcom,q6afe: Document q6usb subnode ASoC: dt-bindings: qcom,sm8250: Add Fairphone 4 sound card ASoC: qcom: sm8250: Add Fairphone 4 soundcard compatible Mac Chiang (2): ASoC: Intel: soc-acpi-intel-ptl-match: add support ptl-rt721-l0 ASoC: Intel: soc-acpi-intel-lnl-match: add rt1320_l12_rt714_l0 support Maciej Strozek (1): ASoC: SDCA: Generic interrupt support Mark Brown (32): ASoC: codecs: wcd93xx: Few simplifications of code and ASoC: Intel: avs: Add rt5640 board support Use helper function for_each_child_of_node_scoped() ASoC: tas571x: add support for tas5753 ASoC: remove component->id Cleanup in rockchip_sai.c ASoC: Add Richtek RTQ9124 support Add DisplayPort sound support for Fairphone 5 ASoC: codecs: More const and unused member cleanups ASoC: mediatek: use reserved memory or enable ASoC: core/topology/Intel: ASoC: SOF: ipc4/Intel: Support for partial context ASoC: SOF: ipc4-pcm: Harmonize set pipeline state dbg ASoC: hdac_hdmi: Rate limit logging on connection and disconnection ASoC: codecs: wcd93xx: Drop enums for chipid values Add SoundWire machines for ACP7.0/ACP7.1 sof stack ASoC: Standardize ASoC menu ASoC: sdw_utils: generate combined spk components ASoC: Add SDCA IRQ support and some misc fixups Add sound card support for QCS8275 treewide: Remove redundant ASoC: soc-dapm: cleanups ASoC: Intel: add support for new devices ASoC: don't set dapm->bias_level directly ASoC: set bias_level at if Update SDCA Kconfig ASoC: codec: Convert to GPIO descriptors for ASoC: convert from clk round_rate() to SDCA Bug Fixes Add SDCA DAI ops helpers More minor SDCA changes ASoC: fsl_xcvr: get channel status data in two cases Pei Xiao (2): ASOC: rockchip: fix capture stream handling in rockchip_sai_xfer_stop ASOC: rockchip: Use helper function devm_clk_get_enabled() Peng Fan (4): ASoC: codec: rockchip_sai: Remove including of_gpio.h ASoC: codec: tlv320aic32x4: Drop aic32x4_pdata usage ASoC: codec: tlv320aic32x4: Sort headers alphabetically ASoC: codec: tlv320aic32x4: Convert to GPIO descriptors Peter Ujfalusi (13): ASoC: topology: Do not call snd_soc_remove_pcm_runtime() for ignored links ASoC: core: Check for rtd == NULL in snd_soc_remove_pcm_runtime() ASoC: Intel: skl_hda_dsp_generic: Implement add_dai_link to filter HDMI PCMs ASoC: Intel: sof_sdw: Implement add_dai_link to filter HDMI PCMs ASoC: SOF: ipc4-priv: Add kernel doc for fw_context_save of sof_ipc4_fw_data ASoC: SOF: ipc4/Intel: Add support for library restore firmware functionality ASoC: SOF: ipc4-pcm: Look for best matching hw_config for SSP ASoC: SOF: pcm: Remove local create_page_table() wrapper function ASoC: SOF: topology: Parse the dapm_widget_tokens in case of DSPless mode ASoC: SOF: pcm: Reverse check for prepared stream in sof_pcm_hw_params() ASoC: SOF: Intel: hda: Do not probe Soundwire in nocodec mode ASoC: SOF: ipc4-pcm: Enable delay reporting for ChainDMA streams ASoC: sdw_utils: Add missed component_name strings for speaker amps Prasad Kumpatla (2): ASoC: dt-bindings: qcom,sm8250: Add QCS8275 sound card ASoC: qcom: sc8280xp: Add support for QCS8275 Richard Fitzgerald (1): firmware: cs_dsp: Remove unused struct list_head from cs_dsp_coeff_ctl Rob Herring (Arm) (1): ASoC: Use of_reserved_mem_region_to_resource() for "memory-region" Sachin Mokashi (1): ASoC: Intel: Replace deprecated strcpy() with strscpy() Sakari Ailus (11): PM: runtime: Document return values of suspend-related API functions PM: runtime: Mark last busy stamp in pm_runtime_put_autosuspend() PM: runtime: Mark last busy stamp in pm_runtime_put_sync_autosuspend() PM: runtime: Mark last busy stamp in pm_runtime_autosuspend() PM: runtime: Mark last busy stamp in pm_request_autosuspend() Documentation: PM: *_autosuspend() functions update last busy time ASoC: atmel: Remove redundant pm_runtime_mark_last_busy() calls ASoC: codecs: Remove redundant pm_runtime_mark_last_busy() calls ASoC: Intel: Remove redundant pm_runtime_mark_last_busy() calls ASoC: component: Remove redundant pm_runtime_mark_last_busy() calls ASoC: SOF: Remove redundant pm_runtime_mark_last_busy() calls Seppo Ingalsuo (1): ASoC: SOF: ipc4-topology: Add load of ASRC component Shenghao Ding (2): ASoC: tas2781: Drop the unnecessary symbol imply ASoC: tas2781: Move the "include linux/debugfs.h" into tas2781.h Shengjiu Wang (6): ASoC: wm8524: enable constraints when sysclk is configured. ASoC: fsl_mqs: Distinguish different modules by system manager indices ASoC: fsl_mqs: rename system manager indices for i.MX95 ASoC: fsl-asoc-card: add sysclk_ratio for calculate sysclk frequency ASoC: fsl_xcvr: get channel status data when PHY is not exists ASoC: fsl_xcvr: get channel status data with firmware exists Shuming Fan (3): ASoC: SDCA: add support for HIDE entity properties and HID descriptor/report ASoC: SDCA: add a HID device for HIDE entity ASoC: SDCA: add route by the number of input pins in MU entity Srinivas Kandagatla (1): ASoC: qcom: use drvdata instead of component to keep id Stephen Rothwell (1): ASoC: SDCA: Check devm_mutex_init() return value Terry Cheong (1): ASoC: hdmi-codec: use SND_JACK_AVOUT as jack status Varshini Rajendran (1): ASoC: dt-bindings: atmel,at91-ssc: add microchip,sam9x7-ssc Venkata Prasad Potturu (4): ASoC: amd: acp: Fix pointer assignments for snd_soc_acpi_mach structures ASoC: amd: acp: Add legacy driver support acp7.2 based platforms ASoC: amd: acp: Enable I2S support for acp7.2 based platforms ASoC: amd: acp: Enable acp7.2 platform based DMIC support in machine driver Vijendar Mukunda (2): ASoC: amd: acp: add soundwire machine for ACP7.0 and ACP7.1 sof stack ASoC: SOF: amd: add alternate machines for ACP7.0 and ACP7.1 platforms Weidong Wang (2): ASoC: codecs: Add calibration function to aw88399 chip ASoC: codecs: Add acpi_match_table for aw88399 driver Xinxin Wan (1): ASoC: codecs: rt5640: Retry DEVICE_ID verification .../devicetree/bindings/sound/atmel,at91-ssc.yaml | 11 +- .../devicetree/bindings/sound/cirrus,cs42xx8.yaml | 4 + .../bindings/sound/fsl,mxs-audio-sgtl5000.yaml | 81 ++ .../bindings/sound/mediatek,mt8173-afe-pcm.yaml | 98 ++ .../devicetree/bindings/sound/mt8186-afe-pcm.yaml | 5 + .../devicetree/bindings/sound/mt8192-afe-pcm.yaml | 5 + .../devicetree/bindings/sound/mtk-afe-pcm.txt | 45 - .../bindings/sound/mxs-audio-sgtl5000.txt | 42 - .../bindings/sound/qcom,lpass-va-macro.yaml | 28 +- .../devicetree/bindings/sound/qcom,q6afe.yaml | 13 + .../devicetree/bindings/sound/qcom,sm8250.yaml | 2 + .../devicetree/bindings/sound/qcom,wcd939x.yaml | 3 + .../devicetree/bindings/sound/richtek,rt9123.yaml | 9 +- .../devicetree/bindings/sound/ti,tas57xx.yaml | 2 + Documentation/power/runtime_pm.rst | 50 +- MAINTAINERS | 11 + arch/mips/configs/generic/board-marduk.config | 1 - drivers/firmware/cirrus/cs_dsp.c | 45 +- drivers/hid/hid-core.c | 3 + include/linux/firmware/cirrus/cs_dsp.h | 2 - include/linux/firmware/imx/sm.h | 20 +- include/linux/pm_runtime.h | 187 +++- include/sound/sdca_asoc.h | 19 + include/sound/sdca_function.h | 85 +- include/sound/sdca_hid.h | 25 + include/sound/sdca_interrupts.h | 78 ++ include/sound/soc-component.h | 1 - include/sound/soc-dai.h | 3 + include/sound/soc-dapm.h | 236 +++-- include/sound/soc_sdw_utils.h | 1 + include/sound/sof/ipc4/header.h | 2 + include/sound/tas2781.h | 4 + include/sound/tlv320aic32x4.h | 9 - include/uapi/linux/input.h | 1 + sound/soc/Kconfig | 8 +- sound/soc/adi/Kconfig | 9 +- sound/soc/amd/Kconfig | 3 + sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-i2s.c | 3 + sound/soc/amd/acp/acp-legacy-common.c | 2 + sound/soc/amd/acp/acp-mach-common.c | 1 + sound/soc/amd/acp/acp-pci.c | 9 +- sound/soc/amd/acp/acp-platform.c | 2 + sound/soc/amd/acp/acp70.c | 1 + sound/soc/amd/acp/acp_common.h | 1 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 12 + sound/soc/amd/acp/amd-acpi-mach.c | 4 +- sound/soc/amd/acp/amd.h | 8 +- sound/soc/amd/mach-config.h | 1 + sound/soc/apple/Kconfig | 4 + sound/soc/atmel/Kconfig | 11 +- sound/soc/atmel/mchp-spdifrx.c | 6 - sound/soc/au1x/Kconfig | 4 + sound/soc/bcm/Kconfig | 4 + sound/soc/cirrus/Kconfig | 3 + sound/soc/codecs/Kconfig | 12 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/arizona-jack.c | 2 - sound/soc/codecs/aw88395/aw88395_device.h | 21 + sound/soc/codecs/aw88399.c | 449 ++++++++- sound/soc/codecs/aw88399.h | 34 + sound/soc/codecs/cros_ec_codec.c | 30 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs35l41.c | 2 - sound/soc/codecs/cs35l45.c | 1 - sound/soc/codecs/cs35l56-sdw.c | 1 - sound/soc/codecs/cs35l56.c | 1 - sound/soc/codecs/cs42l42.c | 1 - sound/soc/codecs/cs42l43-jack.c | 5 - sound/soc/codecs/cs42l43.c | 2 - sound/soc/codecs/cs48l32.c | 1 - sound/soc/codecs/da7218.c | 2 +- sound/soc/codecs/da7219.c | 68 +- sound/soc/codecs/es8375.c | 2 +- sound/soc/codecs/hda.c | 3 - sound/soc/codecs/hdac_hdmi.c | 307 +----- sound/soc/codecs/hdac_hdmi.h | 10 - sound/soc/codecs/hdmi-codec.c | 6 +- sound/soc/codecs/lpass-rx-macro.c | 1 - sound/soc/codecs/lpass-tx-macro.c | 1 - sound/soc/codecs/lpass-va-macro.c | 1 - sound/soc/codecs/lpass-wsa-macro.c | 1 - sound/soc/codecs/max98363.c | 3 +- sound/soc/codecs/max98373-i2c.c | 2 +- sound/soc/codecs/max98373-sdw.c | 3 +- sound/soc/codecs/max98388.c | 2 +- sound/soc/codecs/max98390.c | 2 +- sound/soc/codecs/max98396.c | 4 +- sound/soc/codecs/max98504.c | 2 +- sound/soc/codecs/max98520.c | 2 +- sound/soc/codecs/max98927.c | 2 +- sound/soc/codecs/rt1017-sdca-sdw.c | 1 - sound/soc/codecs/rt1308-sdw.c | 1 - sound/soc/codecs/rt1316-sdw.c | 1 - sound/soc/codecs/rt1318-sdw.c | 1 - sound/soc/codecs/rt1320-sdw.c | 1 - sound/soc/codecs/rt5640.c | 43 +- sound/soc/codecs/rt5640.h | 6 +- sound/soc/codecs/rt5682-sdw.c | 1 - sound/soc/codecs/rt5682.c | 26 +- sound/soc/codecs/rt5682s.c | 26 +- sound/soc/codecs/rt700.c | 4 +- sound/soc/codecs/rt711-sdca.c | 2 - sound/soc/codecs/rt711.c | 2 - sound/soc/codecs/rt712-sdca-dmic.c | 1 - sound/soc/codecs/rt712-sdca.c | 2 - sound/soc/codecs/rt715-sdca.c | 1 - sound/soc/codecs/rt715.c | 3 +- sound/soc/codecs/rt721-sdca.c | 2 - sound/soc/codecs/rt722-sdca-sdw.c | 2 +- sound/soc/codecs/rt722-sdca.c | 2 - sound/soc/codecs/rt9123.c | 3 - sound/soc/codecs/rtq9124.c | 543 +++++++++++ sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2781-i2c.c | 3 - sound/soc/codecs/tas571x.c | 52 + sound/soc/codecs/tlv320aic32x4.c | 53 +- sound/soc/codecs/wcd-mbhc-v2.c | 2 - sound/soc/codecs/wcd9335.c | 5 - sound/soc/codecs/wcd934x.c | 4 - sound/soc/codecs/wcd937x.c | 38 +- sound/soc/codecs/wcd937x.h | 1 - sound/soc/codecs/wcd938x.c | 64 +- sound/soc/codecs/wcd939x.c | 69 +- sound/soc/codecs/wm8524.c | 55 +- sound/soc/codecs/wsa881x.c | 5 +- sound/soc/codecs/wsa883x.c | 3 +- sound/soc/codecs/wsa884x.c | 3 +- sound/soc/dwc/Kconfig | 3 + sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl-asoc-card.c | 13 +- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_mqs.c | 11 +- sound/soc/fsl/fsl_sai.c | 4 +- sound/soc/fsl/fsl_xcvr.c | 25 +- sound/soc/fsl/imx-card.c | 53 +- sound/soc/generic/Kconfig | 4 + sound/soc/google/Kconfig | 3 + sound/soc/hisilicon/Kconfig | 4 + sound/soc/img/Kconfig | 14 +- sound/soc/intel/Kconfig | 4 + sound/soc/intel/atom/sst/sst_pvt.c | 1 - sound/soc/intel/avs/board_selection.c | 27 + sound/soc/intel/avs/boards/Kconfig | 12 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/rt5640.c | 270 ++++++ sound/soc/intel/avs/core.c | 1 - sound/soc/intel/avs/debugfs.c | 2 - sound/soc/intel/avs/ipc.c | 1 - sound/soc/intel/avs/pcm.c | 1 - sound/soc/intel/boards/cht_bsw_rt5672.c | 3 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 13 + sound/soc/intel/boards/sof_es8336.c | 10 + sound/soc/intel/boards/sof_rt5682.c | 7 + sound/soc/intel/boards/sof_sdw.c | 14 + sound/soc/intel/catpt/pcm.c | 6 - sound/soc/intel/catpt/sysfs.c | 1 - sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 34 + sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 57 ++ sound/soc/jz4740/Kconfig | 4 + sound/soc/kirkwood/Kconfig | 3 + sound/soc/loongson/Kconfig | 2 +- sound/soc/mediatek/Kconfig | 4 + .../soc/mediatek/common/mtk-afe-platform-driver.c | 4 +- sound/soc/mediatek/common/mtk-base-afe.h | 1 + sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 66 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 56 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 7 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 4 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 7 + sound/soc/meson/Kconfig | 2 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 16 +- sound/soc/mxs/Kconfig | 2 +- sound/soc/pxa/Kconfig | 4 + sound/soc/qcom/Kconfig | 2 +- sound/soc/qcom/lpass-platform.c | 27 +- sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 8 +- sound/soc/qcom/sc8280xp.c | 1 + sound/soc/qcom/sm8250.c | 28 +- sound/soc/renesas/Kconfig | 2 +- sound/soc/renesas/rcar/core.c | 35 +- sound/soc/renesas/rcar/ctu.c | 8 +- sound/soc/renesas/rcar/dma.c | 4 +- sound/soc/renesas/rcar/dvc.c | 8 +- sound/soc/renesas/rcar/mix.c | 8 +- sound/soc/renesas/rcar/msiof.c | 94 +- sound/soc/renesas/rcar/src.c | 10 +- sound/soc/renesas/rcar/ssi.c | 18 +- sound/soc/renesas/rcar/ssiu.c | 7 +- sound/soc/rockchip/Kconfig | 23 +- sound/soc/rockchip/rockchip_sai.c | 52 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/bells.c | 2 - sound/soc/samsung/littlemill.c | 2 - sound/soc/samsung/speyside.c | 2 - sound/soc/samsung/tobermory.c | 2 - sound/soc/sdca/Kconfig | 20 + sound/soc/sdca/Makefile | 6 +- sound/soc/sdca/sdca_asoc.c | 469 +++++++-- sound/soc/sdca/sdca_functions.c | 264 ++++- sound/soc/sdca/sdca_hid.c | 127 +++ sound/soc/sdca/sdca_interrupts.c | 444 +++++++++ sound/soc/sdca/sdca_regmap.c | 29 +- sound/soc/sdw_utils/soc_sdw_cs_amp.c | 10 - sound/soc/sdw_utils/soc_sdw_maxim.c | 9 - sound/soc/sdw_utils/soc_sdw_rt_amp.c | 6 - sound/soc/sdw_utils/soc_sdw_rt_mf_sdca.c | 7 - sound/soc/sdw_utils/soc_sdw_utils.c | 38 + sound/soc/soc-component.c | 1 - sound/soc/soc-core.c | 25 +- sound/soc/soc-dai.c | 16 +- sound/soc/soc-dapm.c | 1018 +++++++++----------- sound/soc/soc-ops-test.c | 29 +- sound/soc/soc-ops.c | 26 +- sound/soc/soc-topology.c | 7 +- sound/soc/sof/Kconfig | 2 +- sound/soc/sof/amd/pci-acp70.c | 1 + sound/soc/sof/control.c | 1 - sound/soc/sof/debug.c | 1 - sound/soc/sof/imx/imx-common.c | 36 +- sound/soc/sof/intel/hda-loader.c | 7 +- sound/soc/sof/intel/hda.c | 5 + sound/soc/sof/ipc3-dtrace.c | 1 - sound/soc/sof/ipc4-loader.c | 7 +- sound/soc/sof/ipc4-pcm.c | 137 ++- sound/soc/sof/ipc4-priv.h | 6 + sound/soc/sof/ipc4-topology.c | 94 +- sound/soc/sof/ipc4-topology.h | 25 + sound/soc/sof/ipc4.c | 39 +- sound/soc/sof/mediatek/mt8186/mt8186.c | 11 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 11 +- sound/soc/sof/pcm.c | 27 +- sound/soc/sof/sof-client-ipc-flood-test.c | 1 - sound/soc/sof/sof-client-ipc-kernel-injector.c | 1 - sound/soc/sof/sof-client-ipc-msg-injector.c | 1 - sound/soc/sof/sof-client-probes.c | 3 - sound/soc/sof/topology.c | 15 +- sound/soc/spear/Kconfig | 4 + sound/soc/sprd/Kconfig | 4 + sound/soc/starfive/Kconfig | 12 +- sound/soc/sti/Kconfig | 2 +- sound/soc/stm/Kconfig | 2 +- sound/soc/stm/stm32_i2s.c | 21 +- sound/soc/stm/stm32_sai_sub.c | 14 +- sound/soc/sunxi/Kconfig | 2 +- sound/soc/tegra/Kconfig | 4 + sound/soc/tegra/tegra210_ahub.c | 2 +- sound/soc/ti/Kconfig | 2 +- sound/soc/uniphier/Kconfig | 14 +- sound/soc/ux500/Kconfig | 2 +- sound/soc/xilinx/Kconfig | 4 + sound/soc/xtensa/Kconfig | 4 + 253 files changed, 5171 insertions(+), 2291 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,mxs-audio-sgtl5000.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8173-afe-pcm.yaml delete mode 100644 Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt delete mode 100644 Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt create mode 100644 include/sound/sdca_hid.h create mode 100644 include/sound/sdca_interrupts.h delete mode 100644 sound/soc/codecs/hdac_hdmi.h create mode 100644 sound/soc/codecs/rtq9124.c create mode 100644 sound/soc/intel/avs/boards/rt5640.c create mode 100644 sound/soc/sdca/sdca_hid.c create mode 100644 sound/soc/sdca/sdca_interrupts.c

2 1

Re: [PATCH v3 2/4] dt-bindings: sound: bindings for tac5x1x family of codecs
by Holalu Yogendra, Niranjan 28 Jul '25

28 Jul '25

Hi Krzysztof, Thanks for the detailed review comments. Request you to take a look at comments below. On Fri, Jun 27, 2025 at 1:18 PM +0530, Krzysztof Kozlowski wrote: > On Thu, Jun 26, 2025 at 11:43:31PM +0530, Niranjan H Y wrote: > > + ti,vref: > > + description: VREF required voltage. If node is omitted then VREF is > Voltage is in microvolt. Use proper suffix/values. Is this convention ? Is there any document on these I can refer ? > > > + oneOf: > > + - const: 0 > > + description: VREF output is powered to 2.75V. > > + - const: 1 > > + description: VREF output is powered to 2.5V. > > + - const: 2 > > + description: VREF output is powered to 1.375V. > > + ti,gpios-func: > > + description: | > > + Array indicating the GPIO1, GPIO2, GPO1 Functionality in the same order. > > + Each integer elemnent in the array represent the following > > + - 0 TAC5X1X_GPIO_DISABLE - GPIO is Disabled > > + - 1 TAC5X1X_GPIO_GPI - General Purpose Input > > + - 2 ADC3XXX_GPIO_GPO - General Purpose Output > > + - 3 TAC5X1X_GPIO_IRQ - Chip Interrupt > > + - 4 TAC5X1X_GPIO_PDMCLK - PDM CLK Output > > + - 5 TAC5X1X_GPIO_P_DOUT - Primary ASI DOUT > > + - 6 TAC5X1X_GPIO_P_DOUT2 - Primary ASI DOUT2 > > + - 7 TAC5X1X_GPIO_S_DOUT - Secondary ASI DOUT > > + - 8 TAC5X1X_GPIO_S_DOUT2 - Secondary ASI DOUT2 > > + - 9 TAC5X1X_GPIO_S_BCLK - Secondary BCLK Output > > + - 10 TAC5X1X_GPIO_S_FSYNC - Secondary FSYNC Output > > + - 11 TAC5X1X_GPIO_CLKOUT - General Purpose Output > > + - 12 TAC5X1X_GPIO_DOUT_MUX > > + - 13 TAC5X1X_GPIO_DAISY_OUT > > + $ref: /schemas/types.yaml#/definitions/uint32-array > > + minItems: 3 > > + maxItems: 3 > > Don't encode gpios and pins as some custom properties. Could you please suggest what is the alternative. Please suggest some example which I can refer ? Regards Niranjan

1 0

[PATCH v2] ALSA: hda: Add TAS2770 support
by Baojun Xu 26 Jul '25

26 Jul '25

Add TAS2770 support in TI's HDA driver. And add hda_chip_id for more products. Distinguish DSP and non-DSP in firmware loading function. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v2: - Rewrite the patch description - Move registers define for TAS2770 from tas2781.h to tas2770-tlv.h - Add new chip ID define for HDA - Add digital volume control in tas2770_snd_controls - Change public kcontrol name from tas27xx_ to tasdevice_ - Create new function for kcontrol adding - Create new function for DSP binary loading - Rename chip_id to hda_chip_id in i2c_probe for the supported devices --- include/sound/tas2770-tlv.h | 23 +++ .../hda/codecs/side-codecs/tas2781_hda_i2c.c | 161 +++++++++++------- 2 files changed, 127 insertions(+), 57 deletions(-) create mode 100644 include/sound/tas2770-tlv.h diff --git a/include/sound/tas2770-tlv.h b/include/sound/tas2770-tlv.h new file mode 100644 index 000000000000..c0bd495b4a07 --- /dev/null +++ b/include/sound/tas2770-tlv.h @@ -0,0 +1,23 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2770 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2770 hda driver implements for one, two, or even multiple +// TAS2770 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2770_TLV_H__ +#define __TAS2770_TLV_H__ + +#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x17) +#define TAS2770_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_amp_tlv, 1100, 50, 0); + +#endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index bacc3f6ed4bd..65a48326f5b9 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -24,6 +24,7 @@ #include <sound/tas2781.h> #include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> +#include <sound/tas2770-tlv.h> #include <sound/tas2781-tlv.h> #include "hda_local.h" @@ -45,9 +46,18 @@ #define TAS2563_CAL_TLIM TASDEVICE_REG(0, 0x10, 0x14) #define TAS2563_CAL_R0 TASDEVICE_REG(0, 0x0f, 0x34) +enum device_chip_id { + HDA_TAS2563, + HDA_TAS2770, + HDA_TAS2781, + HDA_OTHERS +}; + struct tas2781_hda_i2c_priv { struct snd_kcontrol *snd_ctls[2]; int (*save_calibration)(struct tas2781_hda *h); + + int hda_chip_id; }; static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) @@ -245,6 +255,15 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static const struct snd_kcontrol_new tas2770_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2770_AMP_LEVEL, + 0, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, tas2770_amp_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2770_DVC_LEVEL, + 0, 0, 31, 0, tas2781_amp_getvol, + tas2781_amp_putvol, tas2770_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -253,7 +272,7 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_force_fwload_get, tas2781_force_fwload_put), }; -static const struct snd_kcontrol_new tas2781_prof_ctrl = { +static const struct snd_kcontrol_new tasdevice_prof_ctrl = { .name = "Speaker Profile Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = tasdevice_info_profile, @@ -261,7 +280,7 @@ static const struct snd_kcontrol_new tas2781_prof_ctrl = { .put = tasdevice_set_profile_id, }; -static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl = { +static const struct snd_kcontrol_new tasdevice_dsp_prog_ctrl = { .name = "Speaker Program Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = tasdevice_info_programs, @@ -269,7 +288,7 @@ static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl = { .put = tasdevice_program_put, }; -static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl = { +static const struct snd_kcontrol_new tasdevice_dsp_conf_ctrl = { .name = "Speaker Config Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = tasdevice_info_config, @@ -378,44 +397,34 @@ static void tas2781_hda_remove_controls(struct tas2781_hda *tas_hda) snd_ctl_remove(codec->card, tas_hda->prof_ctl); } -static void tasdev_fw_ready(const struct firmware *fmw, void *context) +static void tasdev_add_kcontrols(struct tasdevice_priv *tas_priv, + struct snd_kcontrol **ctls, struct hda_codec *codec, + const struct snd_kcontrol_new *tas_snd_ctrls, int num_ctls) { - struct tasdevice_priv *tas_priv = context; - struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); - struct tas2781_hda_i2c_priv *hda_priv = tas_hda->hda_priv; - struct hda_codec *codec = tas_priv->codec; - int i, ret, spk_id; - - pm_runtime_get_sync(tas_priv->dev); - mutex_lock(&tas_priv->codec_lock); + int i, ret; - ret = tasdevice_rca_parser(tas_priv, fmw); - if (ret) - goto out; - - tas_hda->prof_ctl = snd_ctl_new1(&tas2781_prof_ctrl, 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.name, ret); - goto out; - } - - for (i = 0; i < ARRAY_SIZE(tas2781_snd_controls); i++) { - hda_priv->snd_ctls[i] = snd_ctl_new1(&tas2781_snd_controls[i], - tas_priv); - ret = snd_ctl_add(codec->card, hda_priv->snd_ctls[i]); + for (i = 0; i < num_ctls; i++) { + ctls[i] = snd_ctl_new1( + &tas_snd_ctrls[i], tas_priv); + ret = snd_ctl_add(codec->card, ctls[i]); if (ret) { dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", - tas2781_snd_controls[i].name, ret); - goto out; + tas_snd_ctrls[i].name, ret); + break; } } +} - tasdevice_dsp_remove(tas_priv); +static void tasdevice_dspfw_init(void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); + struct tas2781_hda_i2c_priv *hda_priv = tas_hda->hda_priv; + struct hda_codec *codec = tas_priv->codec; + int ret, spk_id; + tasdevice_dsp_remove(tas_priv); tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; if (tas_priv->speaker_id != NULL) { // Speaker id need to be checked for ASUS only. @@ -441,28 +450,12 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) dev_err(tas_priv->dev, "dspfw load %s error\n", tas_priv->coef_binaryname); tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - goto out; - } - - tas_hda->dsp_prog_ctl = snd_ctl_new1(&tas2781_dsp_prog_ctrl, - 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.name, ret); - goto out; - } - - tas_hda->dsp_conf_ctl = snd_ctl_new1(&tas2781_dsp_conf_ctrl, - 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.name, ret); - goto out; + return; } + tasdev_add_kcontrols(tas_priv, &tas_hda->dsp_prog_ctl, codec, + &tasdevice_dsp_prog_ctrl, 1); + tasdev_add_kcontrols(tas_priv, &tas_hda->dsp_conf_ctl, codec, + &tasdevice_dsp_conf_ctrl, 1); tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; tasdevice_prmg_load(tas_priv, 0); @@ -475,9 +468,45 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) * calibrated data inside algo. */ hda_priv->save_calibration(tas_hda); +} + +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 tas2781_hda_i2c_priv *hda_priv = tas_hda->hda_priv; + struct hda_codec *codec = tas_priv->codec; + int ret; + + pm_runtime_get_sync(tas_priv->dev); + mutex_lock(&tas_priv->codec_lock); + + ret = tasdevice_rca_parser(tas_priv, fmw); + if (ret) + goto out; - tasdevice_tuning_switch(tas_hda->priv, 0); - tas_hda->priv->playback_started = true; + tas_priv->fw_state = TASDEVICE_RCA_FW_OK; + tasdev_add_kcontrols(tas_priv, &tas_hda->prof_ctl, codec, + &tasdevice_prof_ctrl, 1); + + switch (hda_priv->hda_chip_id) { + case HDA_TAS2770: + tasdev_add_kcontrols(tas_priv, hda_priv->snd_ctls, codec, + &tas2770_snd_controls[0], + ARRAY_SIZE(tas2770_snd_controls)); + break; + case HDA_TAS2781: + tasdev_add_kcontrols(tas_priv, hda_priv->snd_ctls, codec, + &tas2781_snd_controls[0], + ARRAY_SIZE(tas2781_snd_controls)); + tasdevice_dspfw_init(context); + break; + case HDA_TAS2563: + tasdevice_dspfw_init(context); + break; + default: + break; + } out: mutex_unlock(&tas_hda->priv->codec_lock); @@ -581,16 +610,33 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) return -ENOMEM; if (strstr(dev_name(&clt->dev), "TIAS2781")) { - device_name = "TIAS2781"; + /* + * TAS2781, integrated on-chip DSP with + * global I2C address supported. + */ + device_name = "TIAS2781"; + hda_priv->hda_chip_id = HDA_TAS2781; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; + } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW2770")) { + /* + * TAS2770, has no on-chip DSP, so no calibration data + * required; has no global I2C address supported. + */ + device_name = "TXNW2770"; + hda_priv->hda_chip_id = HDA_TAS2770; } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW2781:00-tas2781-hda.0")) { device_name = "TXNW2781"; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; } else if (strstr(dev_name(&clt->dev), "INT8866")) { - device_name = "INT8866"; + /* + * TAS2563, integrated on-chip DSP with + * global I2C address supported. + */ + device_name = "INT8866"; + hda_priv->hda_chip_id = HDA_TAS2563; hda_priv->save_calibration = tas2563_save_calibration; tas_hda->priv->global_addr = TAS2563_GLOBAL_ADDR; } else { @@ -727,6 +773,7 @@ static const struct i2c_device_id tas2781_hda_i2c_id[] = { static const struct acpi_device_id tas2781_acpi_hda_match[] = { {"INT8866", 0 }, {"TIAS2781", 0 }, + {"TXNW2770", 0 }, {"TXNW2781", 0 }, {} }; -- 2.43.0

3 2

tinycompress: add support for new SNDRV_COMPRESS_TSTAMP64 and _AVAIL64
by GitHub pull_request - opened 25 Jul '25

25 Jul '25

alsa-project/tinycompress pull request #29 was opened from jorisverhaegen: These 2 ioctl are under review to be added to kernel. See https://lore.kernel.org/linux-sound/20250725114249.2086974-1-verhaegen@goog… This pull request should be merged after this kernel is available. Request URL : https://github.com/alsa-project/tinycompress/pull/29 Patch URL : https://github.com/alsa-project/tinycompress/pull/29.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

HDMI card is marked "off" on system resume and stays so until reset
by GitHub issues - opened 24 Jul '25

24 Jul '25

alsa-project/alsa-lib issue #465 was opened from j39m: (Continued from [Pipewire issue 4809](https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4809).) # Basic Symptoms When resuming, my device often fails to continue playback over my HDMI output. Higher-level utils (`pavucontrol`, `pw-dump`, etc.) report that the corresponding "sink" is not present (until I reset the output). Pipewire dev `@pvir` read my logs thusly: <details> <summary>`alsactl monitor` and `amixer`</summary> Last state update after resume: ``` 2025-07-24 22:15:16,844: alsactl node hw:0, #7 (0,0,0,HDMI/DP,pcm=7 Jack,0) VALUE 2025-07-24 22:15:16,854: amixer numid=7,iface=CARD,name='HDMI/DP,pcm=7 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off ``` The next update then comes only when workaround is run: ``` 2025-07-24 22:16:02,772: alsactl node hw:0, #12 (3,7,0,ELD,0) VALUE INFO 2025-07-24 22:16:02,781: amixer numid=7,iface=CARD,name='HDMI/DP,pcm=7 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on ``` It appears there was no notification for the off->on transition, or the off->on only happens when doing the workaround cycling displays, so probably this is ALSA or HDMI bug. </details> I haven't tried waiting longer than this, but I assume that if ALSA thinks the card is still gone ~45s after resume, it's already a bit too far gone to be worried about exact timing. # System readout <details> ``` [j39m@flaglock6 ~/Downloads/tmp] $ rpm -q alsa-lib intel-audio-firmware alsa-lib-1.2.14-3.fc42.x86_64 intel-audio-firmware-20250708-1.fc42.noarch [j39m@flaglock6 ~/Downloads/tmp] $ inxi -A Audio: Device-1: Advanced Micro Devices [AMD/ATI] Raven/Raven2/Fenghuang HDMI/DP Audio driver: snd_hda_intel Device-2: Advanced Micro Devices [AMD] Audio Coprocessor driver: snd_pci_acp3x Device-3: Advanced Micro Devices [AMD] Family 17h/19h/1ah HD Audio driver: snd_hda_intel API: ALSA v: k6.15.6-200.fc42.x86_64 status: kernel-api Server-1: PipeWire v: 1.4.6 status: active ``` </details> Issue URL : https://github.com/alsa-project/alsa-lib/issues/465 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC updates for v6.17
by Mark Brown 24 Jul '25

24 Jul '25

The following changes since commit 347e9f5043c89695b01e66b3ed111755afcf1911: Linux 6.16-rc6 (2025-07-13 14:25:58 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.17 for you to fetch changes up to c58c35ef6ae62e36927f506a5afc66610b7261d9: ASoC: qcom: sm8250: Add Fairphone 4 soundcard compatible (2025-07-23 15:09:30 +0100) ---------------------------------------------------------------- ASoC: Updates for v6.17 There's a few new drivers here and quite a lot of cleanup work from Morimoto-san but generally this has been quite a quiet release, resulting in a fairly small diffstat. Highlights include: - Refactoring of the Kconfig menus to be hopefully more consistant and easier to navigate. - Refactoring of the DAPM code, mainly hiding functionality that doesn't need to be exposed to drivers. - Removal of the unused upstream weak paths DAPM functionality. - Further work on the generic handling for SoundWire SDCA devices. - Cleanups of our usage of the PM autosuspend functions, this pulls in some PM core changes on a shared tag. - Support for AMD ACP7.2 and SoundWire on ACP 7.1, Fairphone 4 & 5, various Intel systems, Qualcomm QCS8275, Richtek RTQ9124 and TI TAS5753. ---------------------------------------------------------------- Ai Chao (3): ASoC: renesas: Use helper function for_each_child_of_node_scoped() ASoC: meson: Use helper function for_each_child_of_node_scoped() ASoC: imx-card: Use helper function for_each_child_of_node_scoped() Alexander Stein (1): ASoC: codec: tlv320aic32x4: Fix reset GPIO check Arnd Bergmann (2): ASoC: ops: dynamically allocate struct snd_ctl_elem_value ASoC: SDCA: fix HID dependency Balamurugan C (3): ASoC: Intel: soc-acpi: Add entry for sof_es8336 in PTL match table. ASoC: Intel: soc-acpi: Add entry for HDMI_In capture support in PTL match table ASoC: Intel: sof_rt5682: Add HDMI-In capture with rt5682 support for PTL. Bard Liao (2): ASoC: sdw_utils: add component_name string to dai_info ASoC: sdw_utils: generate combined spk components string Bram Vlerick (2): ASoC: tas571x: add support for tas5753 ASoC: dt-bindings: tas57xx: add tas5753 compatibility Brian Masney (6): ASoC: codecs: da7219: convert from round_rate() to determine_rate() ASoC: codecs: rt5682: convert from round_rate() to determine_rate() ASoC: codecs: rt5682s: convert from round_rate() to determine_rate() ASoC: qcom: qdsp6: q6dsp-lpass-clocks: convert from round_rate() to determine_rate() ASoC: stm: stm32_i2s: convert from round_rate() to determine_rate() ASoC: stm: stm32_sai_sub: convert from round_rate() to determine_rate() Cezary Rojewski (2): ASoC: codecs: rt5640: Drop dummy register names ASoC: Intel: avs: Add rt5640 machine board Charles Keepax (21): HID: core: Add bus define for SoundWire bus MAINTAINERS: Add SDCA maintainers entry ASoC: SDCA: Add missing default in switch in entity_pde_event() ASoC: SDCA: Fixup some kernel doc errors ASoC: SDCA: Minor selected/detected mode control fixups ASoC: SDCA: Add flag for unused IRQs ASoC: SDCA: Add some initial IRQ handlers ASoC: SDCA: Kconfig/Makefile fixups ASoC: SDCA: Pull HID and IRQ into the primary SDCA module ASoC: SDCA: Fix off by one error in IRQ bound check ASoC: SDCA: Avoid use of uninitialised local name variable ASoC: SDCA: Update memory allocations to zero initialise ASoC: SDCA: Allow read-only controls to be deferrable ASoC: SDCA: Remove overly chatty input pin list warning ASoC: SDCA: Move SDCA search functions and export ASoC: soc-dai: Add private data to snd_soc_dai ASoC: SDCA: Add helper to add DAI constraints ASoC: SDCA: Add a helper to get the SoundWire port number ASoC: SDCA: Add hw_params() helper function ASoC: SDCA: Shrink detected_mode_handler() stack frame ASoC: SDCA: Fix implicit cast from le16 Chen Ni (1): ASoC: tegra: AHUB: Remove unneeded semicolon Chen-Yu Tsai (9): ASoC: dt-bindings: Convert MT8173 AFE binding to dt-schema ASoC: dt-bindings: mt8173-afe-pcm: Add power domain ASoC: dt-bindings: mt8173-afe-pcm: Allow specifying reserved memory region ASoC: dt-bindings: mt8186-afe-pcm: Allow specifying reserved memory region ASoC: dt-bindings: mt8192-afe-pcm: Allow specifying reserved memory region ASoC: mediatek: use reserved memory or enable buffer pre-allocation ASoC: mediatek: mt8183-afe-pcm: Support >32 bit DMA addresses ASoC: mediatek: mt8173-afe-pcm: use local `dev` pointer in driver callbacks ASoC: mediatek: mt8183-afe-pcm: use local `dev` pointer in driver callbacks ChiYuan Huang (2): ASoC: dt-bindings: rt9123: Append RTQ9124 description ASoC: codecs: Add support for Richtek RTQ9124 Frank Li (1): ASoC: dt-bindings: covert mxs-audio-sgtl5000.txt to yaml format Geert Uytterhoeven (2): ASoC: renesas: msiof: Convert to <linux/spi/sh_msiof.h> ASoC: img: Imagination Technologies sound should depend on MIPS Jyri Sarha (3): ASoC: SOF: ipc4: Add sof_ipc4_pipeline_state_str() for debugging ASoC: SOF: ipc4-pcm: Pipe instances to dev_dbg in multi_pipeline_state() ASoC: SOF: ipc4-pcm: Harmonize sof_ipc4_set_pipeline_state() dbg print Krzysztof Kozlowski (20): ASoC: codecs: Constify regmap configuration static variables ASoC: fsl: Constify reg_default array ASoC: codecs: wcd9335: Drop unused sido_input_src field ASoC: codecs: wcd934x: Drop unused num_rx_port/num_tx_port fields ASoC: codecs: wcd937x: Drop unused 'struct wcd937x_priv' fields ASoC: codecs: wcd938x: Drop unused 'struct wcd938x_priv' fields ASoC: codecs: wcd938x: Drop unused variant field ASoC: codecs: wcd939x: Drop unused 'struct wcd939x_priv' fields ASoC: codecs: wcd937x: Simplify with devm_regulator_bulk_get_enable() ASoC: codecs: wcd938x: Simplify with devm_regulator_bulk_get_enable() ASoC: codecs: wcd939x: Simplify with devm_regulator_bulk_get_enable() ASoC: codecs: wcd939x: Simplify return from devm_gpiod_get() error ASoC: dt-bindings: qcom,wcd939x: Document missing VDD_PX supply ASoC: codecs: wcd939x: Add VDD_PX supply ASoC: codecs: lpass: Drop unused AIF_INVALID first DAI identifier ASoC: codecs: wcd937x: Use simple defines for chipid register value ASoC: codecs: wcd938x: Use simple defines for chipid register value ASoC: codecs: wcd939x: Use simple defines for chipid register value ASoC: codecs: wcd939x: Add defines for major/minor version decoding ASoC: dt-bindings: qcom,lpass-va-macro: Define clock-names in top-level Kuninori Morimoto (62): ASoC: soc-core: save ID if param was set in fmt_single_name() ASoC: remove component->id ASoC: hdac_hdmi: remove hdac_hdmi_jack[_port]_init() ASoC: soc-dai: tidyup return value of snd_soc_xlate_tdm_slot_mask() ASoC: soc-ops-test: dynamically allocate struct snd_ctl_elem_value ASoC: ti: Standardize ASoC menu ASoC: adi: Standardize ASoC menu ASoC: amd: Standardize ASoC menu ASoC: bcm: Standardize ASoC menu ASoC: dwc: Standardize ASoC menu ASoC: fsl: Standardize ASoC menu ASoC: img: Standardize ASoC menu ASoC: mxs: Standardize ASoC menu ASoC: pxa: Standardize ASoC menu ASoC: sti: Standardize ASoC menu ASoC: stm: Standardize ASoC menu ASoC: sof: Standardize ASoC menu ASoC: au1x: Standardize ASoC menu ASoC: sdca: Standardize ASoC menu ASoC: sprd: Standardize ASoC menu ASoC: qcom: Standardize ASoC menu ASoC: meson: Standardize ASoC menu ASoC: apple: Standardize ASoC menu ASoC: atmel: Standardize ASoC menu ASoC: intel: Standardize ASoC menu ASoC: spear: Standardize ASoC menu ASoC: sunxi: Standardize ASoC menu ASoC: tegra: Standardize ASoC menu ASoC: ux500: Standardize ASoC menu ASoC: cirrus: Standardize ASoC menu ASoC: google: Standardize ASoC menu ASoC: jz4740: Standardize ASoC menu ASoC: xilinx: Standardize ASoC menu ASoC: xtensa: Standardize ASoC menu ASoC: renesas: Standardize ASoC menu ASoC: generic: Standardize ASoC menu ASoC: samsung: Standardize ASoC menu ASoC: kirkwood: Standardize ASoC menu ASoC: loongson: Standardize ASoC menu ASoC: mediatek: Standardize ASoC menu ASoC: rockchip: Standardize ASoC menu ASoC: starfive: Standardize ASoC menu ASoC: uniphier: Standardize ASoC menu ASoC: hisilicon: Standardize ASoC menu ASoC: soc-dapm: remove unnecessary definition ASoC: soc-dapm: remove EXPORT_SYMBOL_GPL() for snd_soc_dapm_free() ASoC: soc-dapm: remove snd_soc_dapm_nc_pin[_unlocked]() ASoC: soc-dapm: remove snd_soc_dapm_weak_routes() ASoC: soc-dapm: reordering function definitions ASoC: soc-dapm: reordering header definitions ASoC: soc-dapm: use component instead of cmpnt ASoC: soc-dapm: use common name for dapm ASoC: soc-dapm: add prefix on dapm_mark_endpoints_dirty() ASoC: soc-dapm: add prefix on dapm_xxx_event() ASoC: soc-dapm: add prefix on soc_dapm_dev_attrs ASoC: rt700: don't set dapm->bias_level ASoC: rt715: don't set dapm->bias_level ASoC: soc-dapm: set bias_level if snd_soc_dapm_set_bias_level() was successed ASoC: samsung: bell: don't set dapm->bias_level directly ASoC: samsung: speyside: don't set dapm->bias_level directly ASoC: samsung: tobermory: don't set dapm->bias_level directly ASoC: samsung: littlemill: don't set dapm->bias_level directly Laurentiu Mihalcea (1): ASoC: dt-bindings: cirrus,cs42xx8: add 'port' property Luca Weiss (6): ASoC: qcom: sm8250: set card driver name from match data ASoC: qcom: sm8250: add DisplayPort Jack support ASoC: qcom: sm8250: Add Fairphone 5 soundcard compatible ASoC: dt-bindings: qcom,q6afe: Document q6usb subnode ASoC: dt-bindings: qcom,sm8250: Add Fairphone 4 sound card ASoC: qcom: sm8250: Add Fairphone 4 soundcard compatible Mac Chiang (2): ASoC: Intel: soc-acpi-intel-ptl-match: add support ptl-rt721-l0 ASoC: Intel: soc-acpi-intel-lnl-match: add rt1320_l12_rt714_l0 support Maciej Strozek (1): ASoC: SDCA: Generic interrupt support Mark Brown (30): ASoC: codecs: wcd93xx: Few simplifications of code and ASoC: Intel: avs: Add rt5640 board support Use helper function for_each_child_of_node_scoped() ASoC: tas571x: add support for tas5753 ASoC: remove component->id Cleanup in rockchip_sai.c ASoC: Add Richtek RTQ9124 support Add DisplayPort sound support for Fairphone 5 ASoC: codecs: More const and unused member cleanups ASoC: mediatek: use reserved memory or enable ASoC: core/topology/Intel: ASoC: SOF: ipc4/Intel: Support for partial context ASoC: SOF: ipc4-pcm: Harmonize set pipeline state dbg ASoC: hdac_hdmi: Rate limit logging on connection and disconnection ASoC: codecs: wcd93xx: Drop enums for chipid values Add SoundWire machines for ACP7.0/ACP7.1 sof stack ASoC: Standardize ASoC menu ASoC: sdw_utils: generate combined spk components ASoC: Add SDCA IRQ support and some misc fixups Add sound card support for QCS8275 treewide: Remove redundant ASoC: soc-dapm: cleanups ASoC: Intel: add support for new devices ASoC: don't set dapm->bias_level directly ASoC: set bias_level at if Update SDCA Kconfig ASoC: codec: Convert to GPIO descriptors for ASoC: convert from clk round_rate() to SDCA Bug Fixes Add SDCA DAI ops helpers Pei Xiao (2): ASOC: rockchip: fix capture stream handling in rockchip_sai_xfer_stop ASOC: rockchip: Use helper function devm_clk_get_enabled() Peng Fan (4): ASoC: codec: rockchip_sai: Remove including of_gpio.h ASoC: codec: tlv320aic32x4: Drop aic32x4_pdata usage ASoC: codec: tlv320aic32x4: Sort headers alphabetically ASoC: codec: tlv320aic32x4: Convert to GPIO descriptors Peter Ujfalusi (13): ASoC: topology: Do not call snd_soc_remove_pcm_runtime() for ignored links ASoC: core: Check for rtd == NULL in snd_soc_remove_pcm_runtime() ASoC: Intel: skl_hda_dsp_generic: Implement add_dai_link to filter HDMI PCMs ASoC: Intel: sof_sdw: Implement add_dai_link to filter HDMI PCMs ASoC: SOF: ipc4-priv: Add kernel doc for fw_context_save of sof_ipc4_fw_data ASoC: SOF: ipc4/Intel: Add support for library restore firmware functionality ASoC: SOF: ipc4-pcm: Look for best matching hw_config for SSP ASoC: SOF: pcm: Remove local create_page_table() wrapper function ASoC: SOF: topology: Parse the dapm_widget_tokens in case of DSPless mode ASoC: SOF: pcm: Reverse check for prepared stream in sof_pcm_hw_params() ASoC: SOF: Intel: hda: Do not probe Soundwire in nocodec mode ASoC: SOF: ipc4-pcm: Enable delay reporting for ChainDMA streams ASoC: sdw_utils: Add missed component_name strings for speaker amps Prasad Kumpatla (2): ASoC: dt-bindings: qcom,sm8250: Add QCS8275 sound card ASoC: qcom: sc8280xp: Add support for QCS8275 Richard Fitzgerald (1): firmware: cs_dsp: Remove unused struct list_head from cs_dsp_coeff_ctl Rob Herring (Arm) (1): ASoC: Use of_reserved_mem_region_to_resource() for "memory-region" Sachin Mokashi (1): ASoC: Intel: Replace deprecated strcpy() with strscpy() Sakari Ailus (11): PM: runtime: Document return values of suspend-related API functions PM: runtime: Mark last busy stamp in pm_runtime_put_autosuspend() PM: runtime: Mark last busy stamp in pm_runtime_put_sync_autosuspend() PM: runtime: Mark last busy stamp in pm_runtime_autosuspend() PM: runtime: Mark last busy stamp in pm_request_autosuspend() Documentation: PM: *_autosuspend() functions update last busy time ASoC: atmel: Remove redundant pm_runtime_mark_last_busy() calls ASoC: codecs: Remove redundant pm_runtime_mark_last_busy() calls ASoC: Intel: Remove redundant pm_runtime_mark_last_busy() calls ASoC: component: Remove redundant pm_runtime_mark_last_busy() calls ASoC: SOF: Remove redundant pm_runtime_mark_last_busy() calls Seppo Ingalsuo (1): ASoC: SOF: ipc4-topology: Add load of ASRC component Shenghao Ding (2): ASoC: tas2781: Drop the unnecessary symbol imply ASoC: tas2781: Move the "include linux/debugfs.h" into tas2781.h Shengjiu Wang (4): ASoC: wm8524: enable constraints when sysclk is configured. ASoC: fsl_mqs: Distinguish different modules by system manager indices ASoC: fsl_mqs: rename system manager indices for i.MX95 ASoC: fsl-asoc-card: add sysclk_ratio for calculate sysclk frequency Shuming Fan (3): ASoC: SDCA: add support for HIDE entity properties and HID descriptor/report ASoC: SDCA: add a HID device for HIDE entity ASoC: SDCA: add route by the number of input pins in MU entity Srinivas Kandagatla (1): ASoC: qcom: use drvdata instead of component to keep id Stephen Rothwell (1): ASoC: SDCA: Check devm_mutex_init() return value Terry Cheong (1): ASoC: hdmi-codec: use SND_JACK_AVOUT as jack status Venkata Prasad Potturu (4): ASoC: amd: acp: Fix pointer assignments for snd_soc_acpi_mach structures ASoC: amd: acp: Add legacy driver support acp7.2 based platforms ASoC: amd: acp: Enable I2S support for acp7.2 based platforms ASoC: amd: acp: Enable acp7.2 platform based DMIC support in machine driver Vijendar Mukunda (2): ASoC: amd: acp: add soundwire machine for ACP7.0 and ACP7.1 sof stack ASoC: SOF: amd: add alternate machines for ACP7.0 and ACP7.1 platforms Weidong Wang (1): ASoC: codecs: Add calibration function to aw88399 chip Xinxin Wan (1): ASoC: codecs: rt5640: Retry DEVICE_ID verification .../devicetree/bindings/sound/cirrus,cs42xx8.yaml | 4 + .../bindings/sound/fsl,mxs-audio-sgtl5000.yaml | 81 ++ .../bindings/sound/mediatek,mt8173-afe-pcm.yaml | 98 ++ .../devicetree/bindings/sound/mt8186-afe-pcm.yaml | 5 + .../devicetree/bindings/sound/mt8192-afe-pcm.yaml | 5 + .../devicetree/bindings/sound/mtk-afe-pcm.txt | 45 - .../bindings/sound/mxs-audio-sgtl5000.txt | 42 - .../bindings/sound/qcom,lpass-va-macro.yaml | 28 +- .../devicetree/bindings/sound/qcom,q6afe.yaml | 13 + .../devicetree/bindings/sound/qcom,sm8250.yaml | 2 + .../devicetree/bindings/sound/qcom,wcd939x.yaml | 3 + .../devicetree/bindings/sound/richtek,rt9123.yaml | 9 +- .../devicetree/bindings/sound/ti,tas57xx.yaml | 2 + Documentation/power/runtime_pm.rst | 50 +- MAINTAINERS | 11 + arch/mips/configs/generic/board-marduk.config | 1 - drivers/firmware/cirrus/cs_dsp.c | 45 +- drivers/hid/hid-core.c | 3 + include/linux/firmware/cirrus/cs_dsp.h | 2 - include/linux/firmware/imx/sm.h | 20 +- include/linux/pm_runtime.h | 187 +++- include/sound/sdca_asoc.h | 19 + include/sound/sdca_function.h | 71 ++ include/sound/sdca_hid.h | 25 + include/sound/sdca_interrupts.h | 78 ++ include/sound/soc-component.h | 1 - include/sound/soc-dai.h | 3 + include/sound/soc-dapm.h | 236 +++-- include/sound/soc_sdw_utils.h | 1 + include/sound/sof/ipc4/header.h | 2 + include/sound/tas2781.h | 4 + include/sound/tlv320aic32x4.h | 9 - include/uapi/linux/input.h | 1 + sound/soc/Kconfig | 8 +- sound/soc/adi/Kconfig | 9 +- sound/soc/amd/Kconfig | 3 + sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-i2s.c | 3 + sound/soc/amd/acp/acp-legacy-common.c | 2 + sound/soc/amd/acp/acp-mach-common.c | 1 + sound/soc/amd/acp/acp-pci.c | 9 +- sound/soc/amd/acp/acp-platform.c | 2 + sound/soc/amd/acp/acp70.c | 1 + sound/soc/amd/acp/acp_common.h | 1 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 12 + sound/soc/amd/acp/amd-acpi-mach.c | 4 +- sound/soc/amd/acp/amd.h | 8 +- sound/soc/amd/mach-config.h | 1 + sound/soc/apple/Kconfig | 4 + sound/soc/atmel/Kconfig | 11 +- sound/soc/atmel/mchp-spdifrx.c | 6 - sound/soc/au1x/Kconfig | 4 + sound/soc/bcm/Kconfig | 4 + sound/soc/cirrus/Kconfig | 3 + sound/soc/codecs/Kconfig | 12 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/arizona-jack.c | 2 - sound/soc/codecs/aw88395/aw88395_device.h | 21 + sound/soc/codecs/aw88399.c | 440 ++++++++- sound/soc/codecs/aw88399.h | 34 + sound/soc/codecs/cros_ec_codec.c | 30 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs35l41.c | 2 - sound/soc/codecs/cs35l45.c | 1 - sound/soc/codecs/cs35l56-sdw.c | 1 - sound/soc/codecs/cs35l56.c | 1 - sound/soc/codecs/cs42l42.c | 1 - sound/soc/codecs/cs42l43-jack.c | 5 - sound/soc/codecs/cs42l43.c | 2 - sound/soc/codecs/cs48l32.c | 1 - sound/soc/codecs/da7218.c | 2 +- sound/soc/codecs/da7219.c | 68 +- sound/soc/codecs/es8375.c | 2 +- sound/soc/codecs/hda.c | 3 - sound/soc/codecs/hdac_hdmi.c | 307 +----- sound/soc/codecs/hdac_hdmi.h | 10 - sound/soc/codecs/hdmi-codec.c | 6 +- sound/soc/codecs/lpass-rx-macro.c | 1 - sound/soc/codecs/lpass-tx-macro.c | 1 - sound/soc/codecs/lpass-va-macro.c | 1 - sound/soc/codecs/lpass-wsa-macro.c | 1 - sound/soc/codecs/max98363.c | 3 +- sound/soc/codecs/max98373-i2c.c | 2 +- sound/soc/codecs/max98373-sdw.c | 3 +- sound/soc/codecs/max98388.c | 2 +- sound/soc/codecs/max98390.c | 2 +- sound/soc/codecs/max98396.c | 4 +- sound/soc/codecs/max98504.c | 2 +- sound/soc/codecs/max98520.c | 2 +- sound/soc/codecs/max98927.c | 2 +- sound/soc/codecs/rt1017-sdca-sdw.c | 1 - sound/soc/codecs/rt1308-sdw.c | 1 - sound/soc/codecs/rt1316-sdw.c | 1 - sound/soc/codecs/rt1318-sdw.c | 1 - sound/soc/codecs/rt1320-sdw.c | 1 - sound/soc/codecs/rt5640.c | 43 +- sound/soc/codecs/rt5640.h | 6 +- sound/soc/codecs/rt5682-sdw.c | 1 - sound/soc/codecs/rt5682.c | 26 +- sound/soc/codecs/rt5682s.c | 26 +- sound/soc/codecs/rt700.c | 4 +- sound/soc/codecs/rt711-sdca.c | 2 - sound/soc/codecs/rt711.c | 2 - sound/soc/codecs/rt712-sdca-dmic.c | 1 - sound/soc/codecs/rt712-sdca.c | 2 - sound/soc/codecs/rt715-sdca.c | 1 - sound/soc/codecs/rt715.c | 3 +- sound/soc/codecs/rt721-sdca.c | 2 - sound/soc/codecs/rt722-sdca-sdw.c | 2 +- sound/soc/codecs/rt722-sdca.c | 2 - sound/soc/codecs/rt9123.c | 3 - sound/soc/codecs/rtq9124.c | 543 +++++++++++ sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2781-i2c.c | 3 - sound/soc/codecs/tas571x.c | 52 + sound/soc/codecs/tlv320aic32x4.c | 53 +- sound/soc/codecs/wcd-mbhc-v2.c | 2 - sound/soc/codecs/wcd9335.c | 5 - sound/soc/codecs/wcd934x.c | 4 - sound/soc/codecs/wcd937x.c | 38 +- sound/soc/codecs/wcd937x.h | 1 - sound/soc/codecs/wcd938x.c | 64 +- sound/soc/codecs/wcd939x.c | 69 +- sound/soc/codecs/wm8524.c | 55 +- sound/soc/codecs/wsa881x.c | 5 +- sound/soc/codecs/wsa883x.c | 3 +- sound/soc/codecs/wsa884x.c | 3 +- sound/soc/dwc/Kconfig | 3 + sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl-asoc-card.c | 13 +- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_mqs.c | 11 +- sound/soc/fsl/fsl_sai.c | 4 +- sound/soc/fsl/imx-card.c | 13 +- sound/soc/generic/Kconfig | 4 + sound/soc/google/Kconfig | 3 + sound/soc/hisilicon/Kconfig | 4 + sound/soc/img/Kconfig | 14 +- sound/soc/intel/Kconfig | 4 + sound/soc/intel/atom/sst/sst_pvt.c | 1 - sound/soc/intel/avs/board_selection.c | 27 + sound/soc/intel/avs/boards/Kconfig | 12 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/rt5640.c | 270 ++++++ sound/soc/intel/avs/core.c | 1 - sound/soc/intel/avs/debugfs.c | 2 - sound/soc/intel/avs/ipc.c | 1 - sound/soc/intel/avs/pcm.c | 1 - sound/soc/intel/boards/cht_bsw_rt5672.c | 3 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 13 + sound/soc/intel/boards/sof_es8336.c | 10 + sound/soc/intel/boards/sof_rt5682.c | 7 + sound/soc/intel/boards/sof_sdw.c | 14 + sound/soc/intel/catpt/pcm.c | 6 - sound/soc/intel/catpt/sysfs.c | 1 - sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 34 + sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 57 ++ sound/soc/jz4740/Kconfig | 4 + sound/soc/kirkwood/Kconfig | 3 + sound/soc/loongson/Kconfig | 2 +- sound/soc/mediatek/Kconfig | 4 + .../soc/mediatek/common/mtk-afe-platform-driver.c | 4 +- sound/soc/mediatek/common/mtk-base-afe.h | 1 + sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 66 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 56 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 7 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 4 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 7 + sound/soc/meson/Kconfig | 2 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 16 +- sound/soc/mxs/Kconfig | 2 +- sound/soc/pxa/Kconfig | 4 + sound/soc/qcom/Kconfig | 2 +- sound/soc/qcom/lpass-platform.c | 27 +- sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 8 +- sound/soc/qcom/sc8280xp.c | 1 + sound/soc/qcom/sm8250.c | 28 +- sound/soc/renesas/Kconfig | 2 +- sound/soc/renesas/rcar/core.c | 35 +- sound/soc/renesas/rcar/ctu.c | 8 +- sound/soc/renesas/rcar/dma.c | 4 +- sound/soc/renesas/rcar/dvc.c | 8 +- sound/soc/renesas/rcar/mix.c | 8 +- sound/soc/renesas/rcar/msiof.c | 94 +- sound/soc/renesas/rcar/src.c | 10 +- sound/soc/renesas/rcar/ssi.c | 18 +- sound/soc/renesas/rcar/ssiu.c | 7 +- sound/soc/rockchip/Kconfig | 23 +- sound/soc/rockchip/rockchip_sai.c | 52 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/bells.c | 2 - sound/soc/samsung/littlemill.c | 2 - sound/soc/samsung/speyside.c | 2 - sound/soc/samsung/tobermory.c | 2 - sound/soc/sdca/Kconfig | 20 + sound/soc/sdca/Makefile | 6 +- sound/soc/sdca/sdca_asoc.c | 469 +++++++-- sound/soc/sdca/sdca_functions.c | 165 +++- sound/soc/sdca/sdca_hid.c | 127 +++ sound/soc/sdca/sdca_interrupts.c | 444 +++++++++ sound/soc/sdw_utils/soc_sdw_cs_amp.c | 10 - sound/soc/sdw_utils/soc_sdw_maxim.c | 9 - sound/soc/sdw_utils/soc_sdw_rt_amp.c | 6 - sound/soc/sdw_utils/soc_sdw_rt_mf_sdca.c | 7 - sound/soc/sdw_utils/soc_sdw_utils.c | 38 + sound/soc/soc-component.c | 1 - sound/soc/soc-core.c | 25 +- sound/soc/soc-dai.c | 16 +- sound/soc/soc-dapm.c | 1018 +++++++++----------- sound/soc/soc-ops-test.c | 29 +- sound/soc/soc-ops.c | 26 +- sound/soc/soc-topology.c | 7 +- sound/soc/sof/Kconfig | 2 +- sound/soc/sof/amd/pci-acp70.c | 1 + sound/soc/sof/control.c | 1 - sound/soc/sof/debug.c | 1 - sound/soc/sof/imx/imx-common.c | 36 +- sound/soc/sof/intel/hda-loader.c | 7 +- sound/soc/sof/intel/hda.c | 5 + sound/soc/sof/ipc3-dtrace.c | 1 - sound/soc/sof/ipc4-loader.c | 7 +- sound/soc/sof/ipc4-pcm.c | 137 ++- sound/soc/sof/ipc4-priv.h | 6 + sound/soc/sof/ipc4-topology.c | 94 +- sound/soc/sof/ipc4-topology.h | 25 + sound/soc/sof/ipc4.c | 39 +- sound/soc/sof/mediatek/mt8186/mt8186.c | 11 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 11 +- sound/soc/sof/pcm.c | 27 +- sound/soc/sof/sof-client-ipc-flood-test.c | 1 - sound/soc/sof/sof-client-ipc-kernel-injector.c | 1 - sound/soc/sof/sof-client-ipc-msg-injector.c | 1 - sound/soc/sof/sof-client-probes.c | 3 - sound/soc/sof/topology.c | 15 +- sound/soc/spear/Kconfig | 4 + sound/soc/sprd/Kconfig | 4 + sound/soc/starfive/Kconfig | 12 +- sound/soc/sti/Kconfig | 2 +- sound/soc/stm/Kconfig | 2 +- sound/soc/stm/stm32_i2s.c | 21 +- sound/soc/stm/stm32_sai_sub.c | 14 +- sound/soc/sunxi/Kconfig | 2 +- sound/soc/tegra/Kconfig | 4 + sound/soc/tegra/tegra210_ahub.c | 2 +- sound/soc/ti/Kconfig | 2 +- sound/soc/uniphier/Kconfig | 14 +- sound/soc/ux500/Kconfig | 2 +- sound/soc/xilinx/Kconfig | 4 + sound/soc/xtensa/Kconfig | 4 + 250 files changed, 4999 insertions(+), 2236 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,mxs-audio-sgtl5000.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8173-afe-pcm.yaml delete mode 100644 Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt delete mode 100644 Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt create mode 100644 include/sound/sdca_hid.h create mode 100644 include/sound/sdca_interrupts.h delete mode 100644 sound/soc/codecs/hdac_hdmi.h create mode 100644 sound/soc/codecs/rtq9124.c create mode 100644 sound/soc/intel/avs/boards/rt5640.c create mode 100644 sound/soc/sdca/sdca_hid.c create mode 100644 sound/soc/sdca/sdca_interrupts.c

2 1

[GIT PULL] ASoC updates for v6.18
by Mark Brown 24 Jul '25

24 Jul '25

The following changes since commit 347e9f5043c89695b01e66b3ed111755afcf1911: Linux 6.16-rc6 (2025-07-13 14:25:58 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.18 for you to fetch changes up to c58c35ef6ae62e36927f506a5afc66610b7261d9: ASoC: qcom: sm8250: Add Fairphone 4 soundcard compatible (2025-07-23 15:09:30 +0100) ---------------------------------------------------------------- ASoC: Updates for v6.18 There's a few new drivers here and quite a lot of cleanup work from Morimoto-san but generally this has been quite a quiet release, resulting in a fairly small diffstat. Highlights include: - Refactoring of the Kconfig menus to be hopefully more consistant and easier to navigate. - Refactoring of the DAPM code, mainly hiding functionality that doesn't need to be exposed to drivers. - Removal of the unused upstream weak paths DAPM functionality. - Further work on the generic handling for SoundWire SDCA devices. - Cleanups of our usage of the PM autosuspend functions, this pulls in some PM core changes on a shared tag. - Support for AMD ACP7.2 and SoundWire on ACP 7.1, Fairphone 4 & 5, various Intel systems, Qualcomm QCS8275, Richtek RTQ9124 and TI TAS5753. ---------------------------------------------------------------- Ai Chao (3): ASoC: renesas: Use helper function for_each_child_of_node_scoped() ASoC: meson: Use helper function for_each_child_of_node_scoped() ASoC: imx-card: Use helper function for_each_child_of_node_scoped() Alexander Stein (1): ASoC: codec: tlv320aic32x4: Fix reset GPIO check Arnd Bergmann (2): ASoC: ops: dynamically allocate struct snd_ctl_elem_value ASoC: SDCA: fix HID dependency Balamurugan C (3): ASoC: Intel: soc-acpi: Add entry for sof_es8336 in PTL match table. ASoC: Intel: soc-acpi: Add entry for HDMI_In capture support in PTL match table ASoC: Intel: sof_rt5682: Add HDMI-In capture with rt5682 support for PTL. Bard Liao (2): ASoC: sdw_utils: add component_name string to dai_info ASoC: sdw_utils: generate combined spk components string Bram Vlerick (2): ASoC: tas571x: add support for tas5753 ASoC: dt-bindings: tas57xx: add tas5753 compatibility Brian Masney (6): ASoC: codecs: da7219: convert from round_rate() to determine_rate() ASoC: codecs: rt5682: convert from round_rate() to determine_rate() ASoC: codecs: rt5682s: convert from round_rate() to determine_rate() ASoC: qcom: qdsp6: q6dsp-lpass-clocks: convert from round_rate() to determine_rate() ASoC: stm: stm32_i2s: convert from round_rate() to determine_rate() ASoC: stm: stm32_sai_sub: convert from round_rate() to determine_rate() Cezary Rojewski (2): ASoC: codecs: rt5640: Drop dummy register names ASoC: Intel: avs: Add rt5640 machine board Charles Keepax (21): HID: core: Add bus define for SoundWire bus MAINTAINERS: Add SDCA maintainers entry ASoC: SDCA: Add missing default in switch in entity_pde_event() ASoC: SDCA: Fixup some kernel doc errors ASoC: SDCA: Minor selected/detected mode control fixups ASoC: SDCA: Add flag for unused IRQs ASoC: SDCA: Add some initial IRQ handlers ASoC: SDCA: Kconfig/Makefile fixups ASoC: SDCA: Pull HID and IRQ into the primary SDCA module ASoC: SDCA: Fix off by one error in IRQ bound check ASoC: SDCA: Avoid use of uninitialised local name variable ASoC: SDCA: Update memory allocations to zero initialise ASoC: SDCA: Allow read-only controls to be deferrable ASoC: SDCA: Remove overly chatty input pin list warning ASoC: SDCA: Move SDCA search functions and export ASoC: soc-dai: Add private data to snd_soc_dai ASoC: SDCA: Add helper to add DAI constraints ASoC: SDCA: Add a helper to get the SoundWire port number ASoC: SDCA: Add hw_params() helper function ASoC: SDCA: Shrink detected_mode_handler() stack frame ASoC: SDCA: Fix implicit cast from le16 Chen Ni (1): ASoC: tegra: AHUB: Remove unneeded semicolon Chen-Yu Tsai (9): ASoC: dt-bindings: Convert MT8173 AFE binding to dt-schema ASoC: dt-bindings: mt8173-afe-pcm: Add power domain ASoC: dt-bindings: mt8173-afe-pcm: Allow specifying reserved memory region ASoC: dt-bindings: mt8186-afe-pcm: Allow specifying reserved memory region ASoC: dt-bindings: mt8192-afe-pcm: Allow specifying reserved memory region ASoC: mediatek: use reserved memory or enable buffer pre-allocation ASoC: mediatek: mt8183-afe-pcm: Support >32 bit DMA addresses ASoC: mediatek: mt8173-afe-pcm: use local `dev` pointer in driver callbacks ASoC: mediatek: mt8183-afe-pcm: use local `dev` pointer in driver callbacks ChiYuan Huang (2): ASoC: dt-bindings: rt9123: Append RTQ9124 description ASoC: codecs: Add support for Richtek RTQ9124 Frank Li (1): ASoC: dt-bindings: covert mxs-audio-sgtl5000.txt to yaml format Geert Uytterhoeven (2): ASoC: renesas: msiof: Convert to <linux/spi/sh_msiof.h> ASoC: img: Imagination Technologies sound should depend on MIPS Jyri Sarha (3): ASoC: SOF: ipc4: Add sof_ipc4_pipeline_state_str() for debugging ASoC: SOF: ipc4-pcm: Pipe instances to dev_dbg in multi_pipeline_state() ASoC: SOF: ipc4-pcm: Harmonize sof_ipc4_set_pipeline_state() dbg print Krzysztof Kozlowski (20): ASoC: codecs: Constify regmap configuration static variables ASoC: fsl: Constify reg_default array ASoC: codecs: wcd9335: Drop unused sido_input_src field ASoC: codecs: wcd934x: Drop unused num_rx_port/num_tx_port fields ASoC: codecs: wcd937x: Drop unused 'struct wcd937x_priv' fields ASoC: codecs: wcd938x: Drop unused 'struct wcd938x_priv' fields ASoC: codecs: wcd938x: Drop unused variant field ASoC: codecs: wcd939x: Drop unused 'struct wcd939x_priv' fields ASoC: codecs: wcd937x: Simplify with devm_regulator_bulk_get_enable() ASoC: codecs: wcd938x: Simplify with devm_regulator_bulk_get_enable() ASoC: codecs: wcd939x: Simplify with devm_regulator_bulk_get_enable() ASoC: codecs: wcd939x: Simplify return from devm_gpiod_get() error ASoC: dt-bindings: qcom,wcd939x: Document missing VDD_PX supply ASoC: codecs: wcd939x: Add VDD_PX supply ASoC: codecs: lpass: Drop unused AIF_INVALID first DAI identifier ASoC: codecs: wcd937x: Use simple defines for chipid register value ASoC: codecs: wcd938x: Use simple defines for chipid register value ASoC: codecs: wcd939x: Use simple defines for chipid register value ASoC: codecs: wcd939x: Add defines for major/minor version decoding ASoC: dt-bindings: qcom,lpass-va-macro: Define clock-names in top-level Kuninori Morimoto (62): ASoC: soc-core: save ID if param was set in fmt_single_name() ASoC: remove component->id ASoC: hdac_hdmi: remove hdac_hdmi_jack[_port]_init() ASoC: soc-dai: tidyup return value of snd_soc_xlate_tdm_slot_mask() ASoC: soc-ops-test: dynamically allocate struct snd_ctl_elem_value ASoC: ti: Standardize ASoC menu ASoC: adi: Standardize ASoC menu ASoC: amd: Standardize ASoC menu ASoC: bcm: Standardize ASoC menu ASoC: dwc: Standardize ASoC menu ASoC: fsl: Standardize ASoC menu ASoC: img: Standardize ASoC menu ASoC: mxs: Standardize ASoC menu ASoC: pxa: Standardize ASoC menu ASoC: sti: Standardize ASoC menu ASoC: stm: Standardize ASoC menu ASoC: sof: Standardize ASoC menu ASoC: au1x: Standardize ASoC menu ASoC: sdca: Standardize ASoC menu ASoC: sprd: Standardize ASoC menu ASoC: qcom: Standardize ASoC menu ASoC: meson: Standardize ASoC menu ASoC: apple: Standardize ASoC menu ASoC: atmel: Standardize ASoC menu ASoC: intel: Standardize ASoC menu ASoC: spear: Standardize ASoC menu ASoC: sunxi: Standardize ASoC menu ASoC: tegra: Standardize ASoC menu ASoC: ux500: Standardize ASoC menu ASoC: cirrus: Standardize ASoC menu ASoC: google: Standardize ASoC menu ASoC: jz4740: Standardize ASoC menu ASoC: xilinx: Standardize ASoC menu ASoC: xtensa: Standardize ASoC menu ASoC: renesas: Standardize ASoC menu ASoC: generic: Standardize ASoC menu ASoC: samsung: Standardize ASoC menu ASoC: kirkwood: Standardize ASoC menu ASoC: loongson: Standardize ASoC menu ASoC: mediatek: Standardize ASoC menu ASoC: rockchip: Standardize ASoC menu ASoC: starfive: Standardize ASoC menu ASoC: uniphier: Standardize ASoC menu ASoC: hisilicon: Standardize ASoC menu ASoC: soc-dapm: remove unnecessary definition ASoC: soc-dapm: remove EXPORT_SYMBOL_GPL() for snd_soc_dapm_free() ASoC: soc-dapm: remove snd_soc_dapm_nc_pin[_unlocked]() ASoC: soc-dapm: remove snd_soc_dapm_weak_routes() ASoC: soc-dapm: reordering function definitions ASoC: soc-dapm: reordering header definitions ASoC: soc-dapm: use component instead of cmpnt ASoC: soc-dapm: use common name for dapm ASoC: soc-dapm: add prefix on dapm_mark_endpoints_dirty() ASoC: soc-dapm: add prefix on dapm_xxx_event() ASoC: soc-dapm: add prefix on soc_dapm_dev_attrs ASoC: rt700: don't set dapm->bias_level ASoC: rt715: don't set dapm->bias_level ASoC: soc-dapm: set bias_level if snd_soc_dapm_set_bias_level() was successed ASoC: samsung: bell: don't set dapm->bias_level directly ASoC: samsung: speyside: don't set dapm->bias_level directly ASoC: samsung: tobermory: don't set dapm->bias_level directly ASoC: samsung: littlemill: don't set dapm->bias_level directly Laurentiu Mihalcea (1): ASoC: dt-bindings: cirrus,cs42xx8: add 'port' property Luca Weiss (6): ASoC: qcom: sm8250: set card driver name from match data ASoC: qcom: sm8250: add DisplayPort Jack support ASoC: qcom: sm8250: Add Fairphone 5 soundcard compatible ASoC: dt-bindings: qcom,q6afe: Document q6usb subnode ASoC: dt-bindings: qcom,sm8250: Add Fairphone 4 sound card ASoC: qcom: sm8250: Add Fairphone 4 soundcard compatible Mac Chiang (2): ASoC: Intel: soc-acpi-intel-ptl-match: add support ptl-rt721-l0 ASoC: Intel: soc-acpi-intel-lnl-match: add rt1320_l12_rt714_l0 support Maciej Strozek (1): ASoC: SDCA: Generic interrupt support Mark Brown (30): ASoC: codecs: wcd93xx: Few simplifications of code and ASoC: Intel: avs: Add rt5640 board support Use helper function for_each_child_of_node_scoped() ASoC: tas571x: add support for tas5753 ASoC: remove component->id Cleanup in rockchip_sai.c ASoC: Add Richtek RTQ9124 support Add DisplayPort sound support for Fairphone 5 ASoC: codecs: More const and unused member cleanups ASoC: mediatek: use reserved memory or enable ASoC: core/topology/Intel: ASoC: SOF: ipc4/Intel: Support for partial context ASoC: SOF: ipc4-pcm: Harmonize set pipeline state dbg ASoC: hdac_hdmi: Rate limit logging on connection and disconnection ASoC: codecs: wcd93xx: Drop enums for chipid values Add SoundWire machines for ACP7.0/ACP7.1 sof stack ASoC: Standardize ASoC menu ASoC: sdw_utils: generate combined spk components ASoC: Add SDCA IRQ support and some misc fixups Add sound card support for QCS8275 treewide: Remove redundant ASoC: soc-dapm: cleanups ASoC: Intel: add support for new devices ASoC: don't set dapm->bias_level directly ASoC: set bias_level at if Update SDCA Kconfig ASoC: codec: Convert to GPIO descriptors for ASoC: convert from clk round_rate() to SDCA Bug Fixes Add SDCA DAI ops helpers Pei Xiao (2): ASOC: rockchip: fix capture stream handling in rockchip_sai_xfer_stop ASOC: rockchip: Use helper function devm_clk_get_enabled() Peng Fan (4): ASoC: codec: rockchip_sai: Remove including of_gpio.h ASoC: codec: tlv320aic32x4: Drop aic32x4_pdata usage ASoC: codec: tlv320aic32x4: Sort headers alphabetically ASoC: codec: tlv320aic32x4: Convert to GPIO descriptors Peter Ujfalusi (13): ASoC: topology: Do not call snd_soc_remove_pcm_runtime() for ignored links ASoC: core: Check for rtd == NULL in snd_soc_remove_pcm_runtime() ASoC: Intel: skl_hda_dsp_generic: Implement add_dai_link to filter HDMI PCMs ASoC: Intel: sof_sdw: Implement add_dai_link to filter HDMI PCMs ASoC: SOF: ipc4-priv: Add kernel doc for fw_context_save of sof_ipc4_fw_data ASoC: SOF: ipc4/Intel: Add support for library restore firmware functionality ASoC: SOF: ipc4-pcm: Look for best matching hw_config for SSP ASoC: SOF: pcm: Remove local create_page_table() wrapper function ASoC: SOF: topology: Parse the dapm_widget_tokens in case of DSPless mode ASoC: SOF: pcm: Reverse check for prepared stream in sof_pcm_hw_params() ASoC: SOF: Intel: hda: Do not probe Soundwire in nocodec mode ASoC: SOF: ipc4-pcm: Enable delay reporting for ChainDMA streams ASoC: sdw_utils: Add missed component_name strings for speaker amps Prasad Kumpatla (2): ASoC: dt-bindings: qcom,sm8250: Add QCS8275 sound card ASoC: qcom: sc8280xp: Add support for QCS8275 Richard Fitzgerald (1): firmware: cs_dsp: Remove unused struct list_head from cs_dsp_coeff_ctl Rob Herring (Arm) (1): ASoC: Use of_reserved_mem_region_to_resource() for "memory-region" Sachin Mokashi (1): ASoC: Intel: Replace deprecated strcpy() with strscpy() Sakari Ailus (11): PM: runtime: Document return values of suspend-related API functions PM: runtime: Mark last busy stamp in pm_runtime_put_autosuspend() PM: runtime: Mark last busy stamp in pm_runtime_put_sync_autosuspend() PM: runtime: Mark last busy stamp in pm_runtime_autosuspend() PM: runtime: Mark last busy stamp in pm_request_autosuspend() Documentation: PM: *_autosuspend() functions update last busy time ASoC: atmel: Remove redundant pm_runtime_mark_last_busy() calls ASoC: codecs: Remove redundant pm_runtime_mark_last_busy() calls ASoC: Intel: Remove redundant pm_runtime_mark_last_busy() calls ASoC: component: Remove redundant pm_runtime_mark_last_busy() calls ASoC: SOF: Remove redundant pm_runtime_mark_last_busy() calls Seppo Ingalsuo (1): ASoC: SOF: ipc4-topology: Add load of ASRC component Shenghao Ding (2): ASoC: tas2781: Drop the unnecessary symbol imply ASoC: tas2781: Move the "include linux/debugfs.h" into tas2781.h Shengjiu Wang (4): ASoC: wm8524: enable constraints when sysclk is configured. ASoC: fsl_mqs: Distinguish different modules by system manager indices ASoC: fsl_mqs: rename system manager indices for i.MX95 ASoC: fsl-asoc-card: add sysclk_ratio for calculate sysclk frequency Shuming Fan (3): ASoC: SDCA: add support for HIDE entity properties and HID descriptor/report ASoC: SDCA: add a HID device for HIDE entity ASoC: SDCA: add route by the number of input pins in MU entity Srinivas Kandagatla (1): ASoC: qcom: use drvdata instead of component to keep id Stephen Rothwell (1): ASoC: SDCA: Check devm_mutex_init() return value Terry Cheong (1): ASoC: hdmi-codec: use SND_JACK_AVOUT as jack status Venkata Prasad Potturu (4): ASoC: amd: acp: Fix pointer assignments for snd_soc_acpi_mach structures ASoC: amd: acp: Add legacy driver support acp7.2 based platforms ASoC: amd: acp: Enable I2S support for acp7.2 based platforms ASoC: amd: acp: Enable acp7.2 platform based DMIC support in machine driver Vijendar Mukunda (2): ASoC: amd: acp: add soundwire machine for ACP7.0 and ACP7.1 sof stack ASoC: SOF: amd: add alternate machines for ACP7.0 and ACP7.1 platforms Weidong Wang (1): ASoC: codecs: Add calibration function to aw88399 chip Xinxin Wan (1): ASoC: codecs: rt5640: Retry DEVICE_ID verification .../devicetree/bindings/sound/cirrus,cs42xx8.yaml | 4 + .../bindings/sound/fsl,mxs-audio-sgtl5000.yaml | 81 ++ .../bindings/sound/mediatek,mt8173-afe-pcm.yaml | 98 ++ .../devicetree/bindings/sound/mt8186-afe-pcm.yaml | 5 + .../devicetree/bindings/sound/mt8192-afe-pcm.yaml | 5 + .../devicetree/bindings/sound/mtk-afe-pcm.txt | 45 - .../bindings/sound/mxs-audio-sgtl5000.txt | 42 - .../bindings/sound/qcom,lpass-va-macro.yaml | 28 +- .../devicetree/bindings/sound/qcom,q6afe.yaml | 13 + .../devicetree/bindings/sound/qcom,sm8250.yaml | 2 + .../devicetree/bindings/sound/qcom,wcd939x.yaml | 3 + .../devicetree/bindings/sound/richtek,rt9123.yaml | 9 +- .../devicetree/bindings/sound/ti,tas57xx.yaml | 2 + Documentation/power/runtime_pm.rst | 50 +- MAINTAINERS | 11 + arch/mips/configs/generic/board-marduk.config | 1 - drivers/firmware/cirrus/cs_dsp.c | 45 +- drivers/hid/hid-core.c | 3 + include/linux/firmware/cirrus/cs_dsp.h | 2 - include/linux/firmware/imx/sm.h | 20 +- include/linux/pm_runtime.h | 187 +++- include/sound/sdca_asoc.h | 19 + include/sound/sdca_function.h | 71 ++ include/sound/sdca_hid.h | 25 + include/sound/sdca_interrupts.h | 78 ++ include/sound/soc-component.h | 1 - include/sound/soc-dai.h | 3 + include/sound/soc-dapm.h | 236 +++-- include/sound/soc_sdw_utils.h | 1 + include/sound/sof/ipc4/header.h | 2 + include/sound/tas2781.h | 4 + include/sound/tlv320aic32x4.h | 9 - include/uapi/linux/input.h | 1 + sound/soc/Kconfig | 8 +- sound/soc/adi/Kconfig | 9 +- sound/soc/amd/Kconfig | 3 + sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-i2s.c | 3 + sound/soc/amd/acp/acp-legacy-common.c | 2 + sound/soc/amd/acp/acp-mach-common.c | 1 + sound/soc/amd/acp/acp-pci.c | 9 +- sound/soc/amd/acp/acp-platform.c | 2 + sound/soc/amd/acp/acp70.c | 1 + sound/soc/amd/acp/acp_common.h | 1 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 12 + sound/soc/amd/acp/amd-acpi-mach.c | 4 +- sound/soc/amd/acp/amd.h | 8 +- sound/soc/amd/mach-config.h | 1 + sound/soc/apple/Kconfig | 4 + sound/soc/atmel/Kconfig | 11 +- sound/soc/atmel/mchp-spdifrx.c | 6 - sound/soc/au1x/Kconfig | 4 + sound/soc/bcm/Kconfig | 4 + sound/soc/cirrus/Kconfig | 3 + sound/soc/codecs/Kconfig | 12 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/arizona-jack.c | 2 - sound/soc/codecs/aw88395/aw88395_device.h | 21 + sound/soc/codecs/aw88399.c | 440 ++++++++- sound/soc/codecs/aw88399.h | 34 + sound/soc/codecs/cros_ec_codec.c | 30 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs35l41.c | 2 - sound/soc/codecs/cs35l45.c | 1 - sound/soc/codecs/cs35l56-sdw.c | 1 - sound/soc/codecs/cs35l56.c | 1 - sound/soc/codecs/cs42l42.c | 1 - sound/soc/codecs/cs42l43-jack.c | 5 - sound/soc/codecs/cs42l43.c | 2 - sound/soc/codecs/cs48l32.c | 1 - sound/soc/codecs/da7218.c | 2 +- sound/soc/codecs/da7219.c | 68 +- sound/soc/codecs/es8375.c | 2 +- sound/soc/codecs/hda.c | 3 - sound/soc/codecs/hdac_hdmi.c | 307 +----- sound/soc/codecs/hdac_hdmi.h | 10 - sound/soc/codecs/hdmi-codec.c | 6 +- sound/soc/codecs/lpass-rx-macro.c | 1 - sound/soc/codecs/lpass-tx-macro.c | 1 - sound/soc/codecs/lpass-va-macro.c | 1 - sound/soc/codecs/lpass-wsa-macro.c | 1 - sound/soc/codecs/max98363.c | 3 +- sound/soc/codecs/max98373-i2c.c | 2 +- sound/soc/codecs/max98373-sdw.c | 3 +- sound/soc/codecs/max98388.c | 2 +- sound/soc/codecs/max98390.c | 2 +- sound/soc/codecs/max98396.c | 4 +- sound/soc/codecs/max98504.c | 2 +- sound/soc/codecs/max98520.c | 2 +- sound/soc/codecs/max98927.c | 2 +- sound/soc/codecs/rt1017-sdca-sdw.c | 1 - sound/soc/codecs/rt1308-sdw.c | 1 - sound/soc/codecs/rt1316-sdw.c | 1 - sound/soc/codecs/rt1318-sdw.c | 1 - sound/soc/codecs/rt1320-sdw.c | 1 - sound/soc/codecs/rt5640.c | 43 +- sound/soc/codecs/rt5640.h | 6 +- sound/soc/codecs/rt5682-sdw.c | 1 - sound/soc/codecs/rt5682.c | 26 +- sound/soc/codecs/rt5682s.c | 26 +- sound/soc/codecs/rt700.c | 4 +- sound/soc/codecs/rt711-sdca.c | 2 - sound/soc/codecs/rt711.c | 2 - sound/soc/codecs/rt712-sdca-dmic.c | 1 - sound/soc/codecs/rt712-sdca.c | 2 - sound/soc/codecs/rt715-sdca.c | 1 - sound/soc/codecs/rt715.c | 3 +- sound/soc/codecs/rt721-sdca.c | 2 - sound/soc/codecs/rt722-sdca-sdw.c | 2 +- sound/soc/codecs/rt722-sdca.c | 2 - sound/soc/codecs/rt9123.c | 3 - sound/soc/codecs/rtq9124.c | 543 +++++++++++ sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2781-i2c.c | 3 - sound/soc/codecs/tas571x.c | 52 + sound/soc/codecs/tlv320aic32x4.c | 53 +- sound/soc/codecs/wcd-mbhc-v2.c | 2 - sound/soc/codecs/wcd9335.c | 5 - sound/soc/codecs/wcd934x.c | 4 - sound/soc/codecs/wcd937x.c | 38 +- sound/soc/codecs/wcd937x.h | 1 - sound/soc/codecs/wcd938x.c | 64 +- sound/soc/codecs/wcd939x.c | 69 +- sound/soc/codecs/wm8524.c | 55 +- sound/soc/codecs/wsa881x.c | 5 +- sound/soc/codecs/wsa883x.c | 3 +- sound/soc/codecs/wsa884x.c | 3 +- sound/soc/dwc/Kconfig | 3 + sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl-asoc-card.c | 13 +- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_mqs.c | 11 +- sound/soc/fsl/fsl_sai.c | 4 +- sound/soc/fsl/imx-card.c | 13 +- sound/soc/generic/Kconfig | 4 + sound/soc/google/Kconfig | 3 + sound/soc/hisilicon/Kconfig | 4 + sound/soc/img/Kconfig | 14 +- sound/soc/intel/Kconfig | 4 + sound/soc/intel/atom/sst/sst_pvt.c | 1 - sound/soc/intel/avs/board_selection.c | 27 + sound/soc/intel/avs/boards/Kconfig | 12 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/rt5640.c | 270 ++++++ sound/soc/intel/avs/core.c | 1 - sound/soc/intel/avs/debugfs.c | 2 - sound/soc/intel/avs/ipc.c | 1 - sound/soc/intel/avs/pcm.c | 1 - sound/soc/intel/boards/cht_bsw_rt5672.c | 3 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 13 + sound/soc/intel/boards/sof_es8336.c | 10 + sound/soc/intel/boards/sof_rt5682.c | 7 + sound/soc/intel/boards/sof_sdw.c | 14 + sound/soc/intel/catpt/pcm.c | 6 - sound/soc/intel/catpt/sysfs.c | 1 - sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 34 + sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 57 ++ sound/soc/jz4740/Kconfig | 4 + sound/soc/kirkwood/Kconfig | 3 + sound/soc/loongson/Kconfig | 2 +- sound/soc/mediatek/Kconfig | 4 + .../soc/mediatek/common/mtk-afe-platform-driver.c | 4 +- sound/soc/mediatek/common/mtk-base-afe.h | 1 + sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 66 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 56 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 7 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 4 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 7 + sound/soc/meson/Kconfig | 2 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 16 +- sound/soc/mxs/Kconfig | 2 +- sound/soc/pxa/Kconfig | 4 + sound/soc/qcom/Kconfig | 2 +- sound/soc/qcom/lpass-platform.c | 27 +- sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 8 +- sound/soc/qcom/sc8280xp.c | 1 + sound/soc/qcom/sm8250.c | 28 +- sound/soc/renesas/Kconfig | 2 +- sound/soc/renesas/rcar/core.c | 35 +- sound/soc/renesas/rcar/ctu.c | 8 +- sound/soc/renesas/rcar/dma.c | 4 +- sound/soc/renesas/rcar/dvc.c | 8 +- sound/soc/renesas/rcar/mix.c | 8 +- sound/soc/renesas/rcar/msiof.c | 94 +- sound/soc/renesas/rcar/src.c | 10 +- sound/soc/renesas/rcar/ssi.c | 18 +- sound/soc/renesas/rcar/ssiu.c | 7 +- sound/soc/rockchip/Kconfig | 23 +- sound/soc/rockchip/rockchip_sai.c | 52 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/bells.c | 2 - sound/soc/samsung/littlemill.c | 2 - sound/soc/samsung/speyside.c | 2 - sound/soc/samsung/tobermory.c | 2 - sound/soc/sdca/Kconfig | 20 + sound/soc/sdca/Makefile | 6 +- sound/soc/sdca/sdca_asoc.c | 469 +++++++-- sound/soc/sdca/sdca_functions.c | 165 +++- sound/soc/sdca/sdca_hid.c | 127 +++ sound/soc/sdca/sdca_interrupts.c | 444 +++++++++ sound/soc/sdw_utils/soc_sdw_cs_amp.c | 10 - sound/soc/sdw_utils/soc_sdw_maxim.c | 9 - sound/soc/sdw_utils/soc_sdw_rt_amp.c | 6 - sound/soc/sdw_utils/soc_sdw_rt_mf_sdca.c | 7 - sound/soc/sdw_utils/soc_sdw_utils.c | 38 + sound/soc/soc-component.c | 1 - sound/soc/soc-core.c | 25 +- sound/soc/soc-dai.c | 16 +- sound/soc/soc-dapm.c | 1018 +++++++++----------- sound/soc/soc-ops-test.c | 29 +- sound/soc/soc-ops.c | 26 +- sound/soc/soc-topology.c | 7 +- sound/soc/sof/Kconfig | 2 +- sound/soc/sof/amd/pci-acp70.c | 1 + sound/soc/sof/control.c | 1 - sound/soc/sof/debug.c | 1 - sound/soc/sof/imx/imx-common.c | 36 +- sound/soc/sof/intel/hda-loader.c | 7 +- sound/soc/sof/intel/hda.c | 5 + sound/soc/sof/ipc3-dtrace.c | 1 - sound/soc/sof/ipc4-loader.c | 7 +- sound/soc/sof/ipc4-pcm.c | 137 ++- sound/soc/sof/ipc4-priv.h | 6 + sound/soc/sof/ipc4-topology.c | 94 +- sound/soc/sof/ipc4-topology.h | 25 + sound/soc/sof/ipc4.c | 39 +- sound/soc/sof/mediatek/mt8186/mt8186.c | 11 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 11 +- sound/soc/sof/pcm.c | 27 +- sound/soc/sof/sof-client-ipc-flood-test.c | 1 - sound/soc/sof/sof-client-ipc-kernel-injector.c | 1 - sound/soc/sof/sof-client-ipc-msg-injector.c | 1 - sound/soc/sof/sof-client-probes.c | 3 - sound/soc/sof/topology.c | 15 +- sound/soc/spear/Kconfig | 4 + sound/soc/sprd/Kconfig | 4 + sound/soc/starfive/Kconfig | 12 +- sound/soc/sti/Kconfig | 2 +- sound/soc/stm/Kconfig | 2 +- sound/soc/stm/stm32_i2s.c | 21 +- sound/soc/stm/stm32_sai_sub.c | 14 +- sound/soc/sunxi/Kconfig | 2 +- sound/soc/tegra/Kconfig | 4 + sound/soc/tegra/tegra210_ahub.c | 2 +- sound/soc/ti/Kconfig | 2 +- sound/soc/uniphier/Kconfig | 14 +- sound/soc/ux500/Kconfig | 2 +- sound/soc/xilinx/Kconfig | 4 + sound/soc/xtensa/Kconfig | 4 + 250 files changed, 4999 insertions(+), 2236 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,mxs-audio-sgtl5000.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8173-afe-pcm.yaml delete mode 100644 Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt delete mode 100644 Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt create mode 100644 include/sound/sdca_hid.h create mode 100644 include/sound/sdca_interrupts.h delete mode 100644 sound/soc/codecs/hdac_hdmi.h create mode 100644 sound/soc/codecs/rtq9124.c create mode 100644 sound/soc/intel/avs/boards/rt5640.c create mode 100644 sound/soc/sdca/sdca_hid.c create mode 100644 sound/soc/sdca/sdca_interrupts.c

2 1

[GIT PULL] ASoC fixes for v6.16-rc7
by Mark Brown 24 Jul '25

24 Jul '25

The following changes since commit 7bab1bd9fdf15b9fa7e6a4b0151deab93df3c80d: ASoC: amd: yc: Add DMI quirk for HP Laptop 17 cp-2033dx (2025-07-16 11:50:33 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.16-rc7 for you to fetch changes up to 696e123aa36bf0bc72bda98df96dd8f379a6e854: ASoC: mediatek: common: fix device and OF node leak (2025-07-23 13:03:57 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.16 A few device specific fixes, none especially remarkable though all useful. ---------------------------------------------------------------- Derek Fang (1): ASoC: rt5650: Eliminate the high frequency glitch Guoqing Jiang (1): ASoC: mediatek: mt8365-dai-i2s: pass correct size to mt8365_dai_set_priv Johan Hovold (1): ASoC: mediatek: common: fix device and OF node leak Ranjani Sridharan (1): ASoC: SOF: Intel: PTL: Add the sdw_process_wakeen op Shuming Fan (1): ASoC: SDCA: correct the calculation of the maximum init table size sound/soc/codecs/rt5645.c | 1 + sound/soc/mediatek/common/mtk-soundcard-driver.c | 4 ++++ sound/soc/mediatek/mt8365/mt8365-dai-i2s.c | 3 +-- sound/soc/sdca/sdca_functions.c | 2 +- sound/soc/sof/intel/ptl.c | 1 + 5 files changed, 8 insertions(+), 3 deletions(-)

2 1

New Card: ESI Juli@ eX
by GitHub issues - reopened 24 Jul '25

24 Jul '25

alsa-project/alsa-lib issue #462 was reopened from Carlisle96: I am not sure where to ask or how to report this: There was a new card released ESI-Audio Juli@ eX https://www.esi-audio.com/products/juliaex/ I am wondering if this is supported or planned to be supported. It is not yet listed here: https://www.alsa-project.org/wiki/Matrix:Vendor-ESI Issue URL : https://github.com/alsa-project/alsa-lib/issues/462 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

شهادة الأبوة في المغرب
by walid falcon 23 Jul '25

23 Jul '25

كل ما تحتاج معرفته حول شهادة الأبوة بالمغرب يمكنك تحميل الطلب من خلال الرابط المباشر https://www.targir.com/2025/04/blog-post_14.html معلومات شاملة لاستخراج شهادة الأبوة والإجراءات القانونية المرتبطة بها ما هي شهادة الأبوة في المغرب؟ شهادة الأبوة وثيقة قانونية تصدر بناءً على حكم قضائي، وتُثبت العلاقة البنوية بين الطفل ووالده البيولوجي، في حال عدم وجود عقد زواج رسمي أو تسجيل سابق. تُعَد هذه الشهادة جزءًا من الإجراءات القضائية التي تدخل ضمن قضايا الحالة المدنية. الحالات التي تستوجب استخراج شهادة الأبوة 1. الأطفال غير المسجلين في الحالة المدنية عندما يولد طفل خارج إطار الزواج أو لم يُسجَّل بصفة قانونية في دفتر الحالة المدنية، يحتاج الأب إلى رفع دعوى للحصول على شهادة الأبوة. 2. إثبات النسب في غياب عقد زواج في حالات عدم توفر وثيقة الزواج بين الأبوين، يُلجأ إلى المحكمة لتقديم التصريح بالأبوة. إجراءات الحصول على شهادة الأبوة أولا: رفع دعوى أمام المحكمة يتوجب على الأب أو ولي أمر الطفل تقديم طلب إلى المحكمة الابتدائية التي يقع فيها مقر سكن الطفل أو الأب، وذلك من أجل إصدار حكم قضائي بإثبات الأبوة. ثانياً: الوثائق المطلوبة نسخة من بطاقة التعريف الوطنية عقد الازدياد الخاص بالطفل (إن وُجد) شهادة طبية للولادة شهادة السكنى وثائق أو إثباتات تدعم علاقة الأب بالطفل نموذج شهادة الأبوة pdf تتيح بعض المواقع الإلكترونية تحميل نموذج شهادة الأبوة PDF قابل للطباعة، غير أن إصدارها الرسمي لا يتم إلا بعد حكم قضائي، ويتم توجيهها لمصلحة الحالة المدنية لتسجيل الطفل. الأسئلة الشائعة حول شهادة الأبوة هل يمكن تقديم طلب شهادة الأبوة إلكترونياً؟ لا، المسطرة تظل قضائية وتتطلب الحضور أمام المحكمة. ما هي مدة الحصول على شهادة الأبوة؟ قد تختلف المدة حسب عدد الجلسات، ولكن عادة ما تستغرق بين شهر إلى ثلاثة أشهر، حسب تعقيد الملف. صلة بين شهادة الأبوة وبعض التصاريح القانونية الأخرى تم الربط ضمن هذا الدليل بين شهادة الأبوة وبعض المفاهيم القانونية الأخرى التي يبحث عنها الكثير، مثل: 1. التصريح الأمني وتصريح السفر غالباً ما يُطلب في بعض الحالات القانونية توفير تصريح أمني عند استخراج أوراق ثبوتية، كما هو الحال في: الأوراق المطلوبة لاستخراج تصريح السفر من منطقة التجنيد تصريح أمني مستعجل التصريح الأمني الإلكتروني 2. استبدال التصريح برخصة قيادة في حالات مثل شروط إصدار رخصة قيادة بعد التصريح أو تجديد التصريح إلى رخصة، من المهم التأكد من إثبات الهوية القانونية، وهو ما يجعل شهادة الأبوة ضرورية في بعض الحالات القانونية الخاصة. شهادة الأبوة الغاية من هذه الوثيقة هي إثبات علاقة الأبوة بالابن سواء كان راشدا أم قاصرا وغالبا ما تطلب في بعض المساطر الإدارية. صاحب الطلب المغربي المقيم بالخارج الوثائق المطلوبة بالنسبة للراشد : · بطاقة التسجيل ؛ · الدفتر العائلي بالنسبة للقاصر : · بطاقة تسجيل الأب ؛ · الدفتر العائلي . المصلحة المكلفة باستلام الوثائق المطلوبة المصالح القنصلية التابعة لمحل إقامة صاحب الطلب المصلحة المكلفة بتقديم الخدمة المصالح القنصلية التابعة لمحل إقامة صاحب الطلب في حالة تعذر الحصول على الخدمة مؤسسة الوسيط لأجل الاتصال وزارة الشؤون الخارجية والتعاون مصدر المعلومات (الإدارة المعنية) وزارة الشؤون الخارجية والتعاون_أرشيف يمكنك تحميل الطلب من خلال الرابط المباشر نموذج شهادة الأبوة بالمغرب محكمة الأسرة بـ: [اسم المدينة] رقم الملف: [يُترك للمحكمة] رقم القضية: [يُترك للمحكمة] بيان تصريح بالأبوة أنا الموقع أسفله: الاسم الكامل: [الاسم الكامل للأب] رقم بطاقة التعريف الوطنية: [رقم CIN] تاريخ ومكان الازدياد: [تاريخ ومكان ولادة الأب] العنوان الكامل: [عنوان الإقامة] المهنة: [المهنة الحالية] أُصرّح بموجب هذا، وبكامل قواي العقلية والقانونية، بما يلي: أن الطفل (ة) المسمى (ة): [اسم الطفل/ة] المولود (ة) بتاريخ: [تاريخ الولادة] بـ: [مكان الولادة] هو(هي) ابني/ابنتي البيولوجي(ة)، وقد وُلد(ت) نتيجة علاقة أبوية قائمة بيني وبين السيدة: [اسم الأم]، المولودة بتاريخ: [تاريخ ميلاد الأم]، والحاملة لرقم بطاقة التعريف الوطنية: [رقم CIN الأم]. وأتحمل كامل المسؤولية القانونية والأبوية تجاه هذا الطفل/ة، وأطلب من المحكمة الموقرة إصدار حكم قضائي يثبت نسب الطفل لي، وتمكينه من التسجيل في الحالة المدنية. الإمضاء الإسم الكامل: _____________________ التوقيع: ___________________________ التاريخ: ____ / ____ / ________ ملاحظات : يُرفق هذا النموذج بالوثائق الداعمة كـ: نسخة البطاقة الوطنية، شهادة السكنى، شهادة الولادة، وغيرها. يُودَع لدى كتابة ضبط المحكمة الابتدائية (قسم قضاء الأسرة) التي يقع ضمن اختصاصها محل ولادة الطفل أو محل سكن الأب.

1 0

Cannot write control 'numid=11,iface=HWDEP,name='DDS''
by GitHub issues - opened 22 Jul '25

22 Jul '25

alsa-project/alsa-utils issue #305 was opened from rubin55: When running `alsactl restore` on an RME HDSPe AIO Pro (using the `snd-hdspe` kernel module), I get the following error: ``` # alsactl restore -U Found hardware: "HDSPe" "RME HDSPe" "" "0x0000" "0x0000" Hardware is initialized using a generic method alsactl: set_control:1494: Cannot write control 'numid=11,iface=HWDEP,name='DDS'' : Invalid argument ``` This seems to prevent systemd's `alsa-restore` oneshot from working, although I don't really understand why; running `alsactl store/restore -U` from command line shows the same error, but *does* store/restore volume levels. The same via systemd, logs the errors in `systemd status alsa-restore` but volume levels are never restored. Issue URL : https://github.com/alsa-project/alsa-utils/issues/305 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH RESEND] ASoC: rt5650: Eliminate the high frequency glitch
by derek.fang＠realtek.com 21 Jul '25

21 Jul '25

From: Derek Fang <derek.fang(a)realtek.com> The glitch was detected in the high frequency of the HP playback. This patch adjusts the DAC dither setting to avoid this situation for almost all cases. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 08df87238..29a403526 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -82,6 +82,7 @@ static const struct reg_sequence rt5650_init_list[] = { {0xf6, 0x0100}, {RT5645_PWR_ANLG1, 0x02}, {RT5645_IL_CMD3, 0x6728}, + {RT5645_PR_BASE + 0x3a, 0x0000}, }; static const struct reg_default rt5645_reg[] = { -- 2.34.1

2 1

Rename qcs6490-rb3gen2 and qcs9075-iq-evk ucm2 conf
by GitHub pull_request - opened 20 Jul '25

20 Jul '25

alsa-project/alsa-ucm-conf pull request #596 was opened from mohsRafi: Rename the ucm2 conf for Qualcomm qcs6490-rb3gen2 and qcs9075-iq-evk qcs6490-rb3gen2-snd-card.conf -> QCS6490-RB3Gen2.conf qcs9075-iq-evk-snd-card.conf -> QCS9075-IQ-EVK.conf Removed snd-card tags from conf files Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/596 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/596.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1] ALSA: hda: Add TAS2770 support
by Baojun Xu 19 Jul '25

19 Jul '25

Add TAS2770 support in TI's HDA driver. And add hda_chip_id for more product support in the future. Separated DSP and non-DSP in firmware load function. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2770-tlv.h | 23 +++++ sound/pci/hda/tas2781_hda_i2c.c | 157 ++++++++++++++++++++++---------- 2 files changed, 133 insertions(+), 47 deletions(-) create mode 100644 include/sound/tas2770-tlv.h diff --git a/include/sound/tas2770-tlv.h b/include/sound/tas2770-tlv.h new file mode 100644 index 000000000000..c0bd495b4a07 --- /dev/null +++ b/include/sound/tas2770-tlv.h @@ -0,0 +1,23 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2770 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2770 hda driver implements for one, two, or even multiple +// TAS2770 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2770_TLV_H__ +#define __TAS2770_TLV_H__ + +#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x17) +#define TAS2770_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_amp_tlv, 1100, 50, 0); + +#endif diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index d91eed9f7804..2fdff0f88af7 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -24,6 +24,7 @@ #include <sound/tas2781.h> #include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> +#include <sound/tas2770-tlv.h> #include <sound/tas2781-tlv.h> #include "hda_local.h" @@ -45,9 +46,18 @@ #define TAS2563_CAL_TLIM TASDEVICE_REG(0, 0x10, 0x14) #define TAS2563_CAL_R0 TASDEVICE_REG(0, 0x0f, 0x34) +enum device_chip_id { + HDA_TAS2563, + HDA_TAS2770, + HDA_TAS2781, + HDA_OTHERS +}; + struct tas2781_hda_i2c_priv { struct snd_kcontrol *snd_ctls[2]; int (*save_calibration)(struct tas2781_hda *h); + + int hda_chip_id; }; static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) @@ -246,6 +256,15 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static const struct snd_kcontrol_new tas2770_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2770_AMP_LEVEL, + 0, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, tas2770_amp_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2770_DVC_LEVEL, + 0, 0, 31, 0, tas2781_amp_getvol, + tas2781_amp_putvol, tas2770_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -254,7 +273,7 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_force_fwload_get, tas2781_force_fwload_put), }; -static const struct snd_kcontrol_new tas2781_prof_ctrl = { +static const struct snd_kcontrol_new tasdevice_prof_ctrl = { .name = "Speaker Profile Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = tasdevice_info_profile, @@ -262,7 +281,7 @@ static const struct snd_kcontrol_new tas2781_prof_ctrl = { .put = tasdevice_set_profile_id, }; -static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl = { +static const struct snd_kcontrol_new tasdevice_dsp_prog_ctrl = { .name = "Speaker Program Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = tasdevice_info_programs, @@ -270,7 +289,7 @@ static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl = { .put = tasdevice_program_put, }; -static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl = { +static const struct snd_kcontrol_new tasdevice_dsp_conf_ctrl = { .name = "Speaker Config Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = tasdevice_info_config, @@ -379,44 +398,15 @@ static void tas2781_hda_remove_controls(struct tas2781_hda *tas_hda) snd_ctl_remove(codec->card, tas_hda->prof_ctl); } -static void tasdev_fw_ready(const struct firmware *fmw, void *context) +static void tasdevice_dspfw_init(void *context) { struct tasdevice_priv *tas_priv = context; struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); struct tas2781_hda_i2c_priv *hda_priv = tas_hda->hda_priv; struct hda_codec *codec = tas_priv->codec; - int i, ret, spk_id; - - pm_runtime_get_sync(tas_priv->dev); - mutex_lock(&tas_priv->codec_lock); - - ret = tasdevice_rca_parser(tas_priv, fmw); - if (ret) - goto out; - - tas_hda->prof_ctl = snd_ctl_new1(&tas2781_prof_ctrl, 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.name, ret); - goto out; - } - - for (i = 0; i < ARRAY_SIZE(tas2781_snd_controls); i++) { - hda_priv->snd_ctls[i] = snd_ctl_new1(&tas2781_snd_controls[i], - tas_priv); - ret = snd_ctl_add(codec->card, hda_priv->snd_ctls[i]); - if (ret) { - dev_err(tas_priv->dev, - "Failed to add KControl %s = %d\n", - tas2781_snd_controls[i].name, ret); - goto out; - } - } + int ret, spk_id; tasdevice_dsp_remove(tas_priv); - tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; if (tas_priv->speaker_id != NULL) { // Speaker id need to be checked for ASUS only. @@ -442,48 +432,101 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) dev_err(tas_priv->dev, "dspfw load %s error\n", tas_priv->coef_binaryname); tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - goto out; + return; } - tas_hda->dsp_prog_ctl = snd_ctl_new1(&tas2781_dsp_prog_ctrl, + tas_hda->dsp_prog_ctl = snd_ctl_new1(&tasdevice_dsp_prog_ctrl, 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.name, ret); - goto out; + tasdevice_dsp_prog_ctrl.name, ret); + return; } - tas_hda->dsp_conf_ctl = snd_ctl_new1(&tas2781_dsp_conf_ctrl, + tas_hda->dsp_conf_ctl = snd_ctl_new1(&tasdevice_dsp_conf_ctrl, 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.name, ret); - goto out; + tasdevice_dsp_conf_ctrl.name, ret); + return; } - tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + tasdevice_prmg_load(tas_priv, 0); 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. */ hda_priv->save_calibration(tas_hda); +} + +static void tasdev_add_kcontrols(struct tasdevice_priv *tas_priv, + struct snd_kcontrol **ctls, struct hda_codec *codec, + const struct snd_kcontrol_new *tas_snd_ctrls, int num_ctls) +{ + int i, ret; + + for (i = 0; i < num_ctls; i++) { + ctls[i] = snd_ctl_new1( + &tas_snd_ctrls[i], tas_priv); + ret = snd_ctl_add(codec->card, ctls[i]); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas_snd_ctrls[i].name, ret); + break; + } + } +} + +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 tas2781_hda_i2c_priv *hda_priv = tas_hda->hda_priv; + struct hda_codec *codec = tas_priv->codec; + int ret; + + pm_runtime_get_sync(tas_priv->dev); + mutex_lock(&tas_priv->codec_lock); - tasdevice_tuning_switch(tas_hda->priv, 0); - tas_hda->priv->playback_started = true; + ret = tasdevice_rca_parser(tas_priv, fmw); + if (ret) + goto out; + + tas_priv->fw_state = TASDEVICE_RCA_FW_OK; + tasdev_add_kcontrols(tas_priv, &tas_hda->prof_ctl, codec, + &tasdevice_prof_ctrl, 1); + + switch (hda_priv->hda_chip_id) { + case HDA_TAS2770: + tasdev_add_kcontrols(tas_priv, hda_priv->snd_ctls, codec, + &tas2770_snd_controls[0], + ARRAY_SIZE(tas2770_snd_controls)); + break; + case HDA_TAS2781: + tasdev_add_kcontrols(tas_priv, hda_priv->snd_ctls, codec, + &tas2781_snd_controls[0], + ARRAY_SIZE(tas2781_snd_controls)); + tasdevice_dspfw_init(context); + break; + case HDA_TAS2563: + tasdevice_dspfw_init(context); + break; + default: + break; + } out: mutex_unlock(&tas_hda->priv->codec_lock); release_firmware(fmw); - pm_runtime_mark_last_busy(tas_hda->dev); pm_runtime_put_autosuspend(tas_hda->dev); } @@ -584,15 +627,34 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) return -ENOMEM; if (strstr(dev_name(&clt->dev), "TIAS2781")) { + /* + * TAS2781, integrated on-chip DSP with + * global I2C address supported. + */ device_name = "TIAS2781"; + hda_priv->hda_chip_id = HDA_TAS2781; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; + } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW2770")) { + /* + * TAS2770, has no on-chip DSP, so no calibration data + * required; has no global I2C address supported. + */ + device_name = "TXNW2770"; + hda_priv->hda_chip_id = HDA_TAS2770; } else if (strstr(dev_name(&clt->dev), "INT8866")) { + /* + * TAS2563, integrated on-chip DSP with + * global I2C address supported. + */ device_name = "INT8866"; + hda_priv->hda_chip_id = HDA_TAS2563; hda_priv->save_calibration = tas2563_save_calibration; tas_hda->priv->global_addr = TAS2563_GLOBAL_ADDR; - } else + } else { return -ENODEV; + } + tas_hda->priv->irq = clt->irq; ret = tas2781_read_acpi(tas_hda->priv, device_name); @@ -724,6 +786,7 @@ static const struct i2c_device_id tas2781_hda_i2c_id[] = { static const struct acpi_device_id tas2781_acpi_hda_match[] = { {"TIAS2781", 0 }, {"INT8866", 0 }, + {"TXNW2770", 0 }, {} }; MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); -- 2.34.1

3 2

Internal Microphone not working
by GitHub issues - opened 19 Jul '25

19 Jul '25

alsa-project/alsa-lib issue #464 was opened from crashthepose: After updating my computer my internal microphone is not working. Originally when I went to my Audio settings it would say under Input "headset unplugged". I was able to go into the HDAjackretask interface and override one of the pins where now the internal microphone is showing on the input list but now I only hear static noise. Info on my computer: OS: Pop!_OS 22.04 LTS x86_64 Host: ASUSLaptop_Q530VJ 1.0 Kernal: 6.12.10-76061203-generic Desktop Environment: GNOME 42.9 Terminal: gnome-terminal CPU: 13th Gen Intel i7-13620H (16) @ 4.700GHz GPU: Intel Device a7a8 GPU: NVIDIA 01:00.0 NVIDIA Corporation Device 25ac Memory: 7828MiB / 15610MiB Here is my alsa info: http://alsa-project.org/db/?f=ed8089460c52b3b91175d52dfd51c4bad1f7388f I've never submitted a bug before so apologies if i'm not doing this the correct way. Issue URL : https://github.com/alsa-project/alsa-lib/issues/464 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

speaker-test 1.2.13 dose not produce 20kHz sine wav
by GitHub issues - opened 17 Jul '25

17 Jul '25

alsa-project/alsa-utils issue #304 was opened from gharig: Hello Alsa-Dev, I hope that I’m writing to the proper place to report a problem, please advise if not… I’m not able to produce 20kHz sine wav with speaker-test 1.2.13 Running on Tinycore 16.0 . Below is what I submitted to Tinycore Team, and they suggested that I contact you with my findings. The hardware I’m using is Winsystems E3825 dual-core, 1.33GHz SATA 2.0 channel mSATA socket (shared with miniPCIe) 2 GB of Soldered Down DDR3 [https://winsystems.com/product/ppm-c407/](url) Tinycore Team; My problem today is that speaker-test does not produce a 20kHz sine wave on Tinycore 16.0 to the line out. speaker-test does produce something just not a 20kHz sine wave. Code: ` speaker-test -t sine -f 20000 -X -l 0` However, using the exact same hardware running Ubuntu 20.24 or 22.04 the following code works and a 20kHz sine wave is produced. The difference being that the alsa versions are different between the 3 Operating Systems. Tinycore 16.0 Alsa version is 1.2.13 - failure Ubuntu 20.04 Alsa version is 1.2.2 - success Ubuntu 22.04 Alsa version is 1.2.6 - success Through my testing, starting at 8000 Hz, and incrementing by 1000 Hz, speaker-test does produce a sine wave, until I went over 15000 Hz then no more sine wave. :) :) Code: ``` speaker-test -t sine -f 8000 -X -l 0 speaker-test -t sine -f 15000 -X -l 0 ``` :( Code: ` speaker-test -t sine -f 16000 -X -l 0` The work around for the time being is to configure speaker-test to play a 20kHz wav file, this dose work and produces a beautiful sine wav :) Code: ` speaker-test -t wav -f /opt/20kHz.wav -l 0` Now for the people who will say why on earth do you want to produce a sound that no one can hear, and my reply is "Exactly". I send the 20kHz sine wav across the audio line as a way to supervise that the audio line is fully functioning. The second the 20kHz tone stops we are notified and send someone out to fix the problem, whether it be hardware or someone accidentality cuts the wire, we want to know. Basically the 20 kHz sine wave does not interferer with paging in the system. My first thoughts on why the 20 kHz sine is not produced, is because of missing driver. But I made sure that every sound driver that was loaded on Ubuntu was loaded on Tinycore. Code: ` sudo modprobe snd_seq_midi snd_seq_midi_event snd_intel_dspcfg` I also loaded AlsaPlugins.tcz just incase, but that did not make a difference. Code: ` tceload -i alsa-plugins.tcz` The last thing I tried after was to set the format sample -F parameter to 96000 Code: `speaker-test -t sine -f 20000 -X -l 0 -F 96000` At first when I did this I thought I was so smart. A sine wave appeared, however I did not have a speaker hooked up, but when the speaker got plugged in I heard a high pitched noise. So the parameter affected the output but not the desired output that I need. Another thought that pass through my mind is could this be a configuration setting during the build of alsa that need to be turned on. After all 15000 Hz does work. This is slightly related to a previous post linked below. [https://forum.tinycorelinux.net/index.php/topic,27673.0.html](url) Thank you Any help is greatly appreciated gharig Issue URL : https://github.com/alsa-project/alsa-utils/issues/304 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[GIT PULL] ASoC fixes for v6.16-rc6
by Mark Brown 17 Jul '25

17 Jul '25

The following changes since commit 3b3312f28ee2d9c386602f8521e419cfc69f4823: ASoC: cs35l56: probe() should fail if the device ID is not recognized (2025-07-03 12:22:27 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.16-rc6 for you to fetch changes up to 7bab1bd9fdf15b9fa7e6a4b0151deab93df3c80d: ASoC: amd: yc: Add DMI quirk for HP Laptop 17 cp-2033dx (2025-07-16 11:50:33 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.16 A relatively large set of changes, but most of them are quirk information for various x86 systems. There is one more substantial fix for a NULL pointer dereference when removing the AVS driver, plus one for Kconfig dependencies. ---------------------------------------------------------------- Adam Queler (1): ASoC: amd: yc: Add DMI entries to support HP 15-fb1xxx Alexandru Andries (1): ASoC: amd: yc: add DMI quirk for ASUS M6501RM Amadeusz Sławiński (1): ASoC: Intel: avs: Fix NULL ptr deref on rmmod Arnd Bergmann (1): ASoC: Intel: fix SND_SOC_SOF dependencies Bard Liao (1): ASoC: Intel: soc-acpi: add support for HP Omen14 ARL Lane Odenbach (1): ASoC: amd: yc: Add DMI quirk for HP Laptop 17 cp-2033dx Oder Chiou (1): ASoC: rt5660: Fix the dmic data source from GPIO2 sound/soc/amd/yc/acp6x-mach.c | 21 ++++++++++++++++ sound/soc/codecs/rt5660.c | 7 ++++-- sound/soc/intel/avs/pcm.c | 4 +++- sound/soc/intel/boards/Kconfig | 2 +- sound/soc/intel/common/soc-acpi-intel-arl-match.c | 29 +++++++++++++++++++++++ 5 files changed, 59 insertions(+), 4 deletions(-)

2 1

[GIT PULL] ASoC fixes for v6.16-rc5
by Mark Brown 17 Jul '25

17 Jul '25

The following changes since commit 3b3312f28ee2d9c386602f8521e419cfc69f4823: ASoC: cs35l56: probe() should fail if the device ID is not recognized (2025-07-03 12:22:27 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.16-rc5 for you to fetch changes up to a609bd74b8680dba62c44f7e6d00d381ddb2d3c0: ASoC: Intel: avs: Fix NULL ptr deref on rmmod (2025-07-08 14:54:40 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.16 Two patches here, one quirk for an AMD system and a fix for an issue on remove of the AVS driver. ---------------------------------------------------------------- Alexandru Andries (1): ASoC: amd: yc: add DMI quirk for ASUS M6501RM Amadeusz Sławiński (1): ASoC: Intel: avs: Fix NULL ptr deref on rmmod sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/intel/avs/pcm.c | 4 +++- 2 files changed, 10 insertions(+), 1 deletion(-)

1 0

[PATCH] ASoC: qcom: x1e80100: Support boards with two speakers
by Krzysztof Kozlowski 17 Jul '25

17 Jul '25

Some Qualcomm X1E laptops have only two speakers. Regardless whether this sound card driver is suitable for them (we could re-use one for some older SoC), we should set reasonable channel map depending on the number of channels, not always 4-speaker setup. This change is necessary for bringing audio support on Lenovo Thinkpad T14s with Qualcomm X1E78100 and only two speakers. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/x1e80100.c | 40 ++++++++++++++++++++++++++++++++++----- 1 file changed, 35 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/x1e80100.c b/sound/soc/qcom/x1e80100.c index 898b5c26bf1e..8eb57fc12f0d 100644 --- a/sound/soc/qcom/x1e80100.c +++ b/sound/soc/qcom/x1e80100.c @@ -95,23 +95,53 @@ static int x1e80100_snd_hw_params(struct snd_pcm_substream *substream, return qcom_snd_sdw_hw_params(substream, params, &data->sruntime[cpu_dai->id]); } +static int x1e80100_snd_hw_map_channels(unsigned int *ch_map, int num) +{ + switch (num) { + case 1: + ch_map[0] = PCM_CHANNEL_FC; + break; + case 2: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + break; + case 3: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + ch_map[2] = PCM_CHANNEL_FC; + break; + case 4: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_LB; + ch_map[2] = PCM_CHANNEL_FR; + ch_map[3] = PCM_CHANNEL_RB; + break; + default: + return -EINVAL; + } + + return 0; +} + static int x1e80100_snd_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; - const unsigned int rx_slot[4] = { PCM_CHANNEL_FL, - PCM_CHANNEL_LB, - PCM_CHANNEL_FR, - PCM_CHANNEL_RB }; + unsigned int channels = substream->runtime->channels; + unsigned int rx_slot[4]; int ret; switch (cpu_dai->id) { case WSA_CODEC_DMA_RX_0: case WSA_CODEC_DMA_RX_1: + ret = x1e80100_snd_hw_map_channels(rx_slot, channels); + if (ret) + return ret; + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, - ARRAY_SIZE(rx_slot), rx_slot); + channels, rx_slot); if (ret) return ret; break; -- 2.43.0

3 2

qcom-lpass: RX HPH Mode is set to a ULP mode which causes distortion
by GitHub issues - opened 17 Jul '25

17 Jul '25

alsa-project/alsa-ucm-conf issue #595 was opened from valpackett: The headphone enable sequence for all Qualcomm devices sets `CLS_H_ULP` mode: https://github.com/alsa-project/alsa-ucm-conf/blob/1b69ade9b6d7ee37a87c08b1… I was trying to figure out why I was getting horrendous distortion on the headphone jack of my Dell Latitude 7455 laptop above a certain relatively-quiet volume level. After lots of fiddling and searching (thanks, [XDA forum audio modders](https://xdaforums.com/t/mod-audio-biquads-utilizing-qualcomms-audi… from several years ago!) I figured it out. Changing that to `CLS_AB_HIFI` has completely fixed the distortion! Should the default be changed to a HIFI mode for everyone, i.e. right in that file? ping: @Srinivas-Kandagatla Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/595 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Alsa speaker-test dose not generate 20kHz Sine Wav
by Harig, George William 16 Jul '25

16 Jul '25

Hello Alsa-Dev, I hope that I'm writing to the proper place to report a problem, please advise if not... I'm not able to produce 20kHz sine wav with speaker-test 1.2.13 Running on Tinycore 16.0 . Below is what I submitted to Tinycore Team, and they suggested that I contact you with my findings. The hardware I'm using is Winsystems E3825 dual-core, 1.33GHz SATA 2.0 channel mSATA socket (shared with miniPCIe) 2 GB of Soldered Down DDR3 Link to product: https://winsystems.com/product/ppm-c407/ Tinycore Team; My problem today is that speaker-test does not produce a 20kHz sine wave on Tinycore 16.0 to the line out. speaker-test does produce something just not a 20kHz sine wave. Code: speaker-test -t sine -f 20000 -X -l 0 However, using the exact same hardware running Ubuntu 20.24 or 22.04 the following code works and a 20kHz sine wave is produced. The difference being that the alsa versions are different between the 3 Operating Systems. Tinycore 16.0 Alsa version is 1.2.13 - failure Ubuntu 20.04 Alsa version is 1.2.2 - success Ubuntu 22.04 Alsa version is 1.2.6 - success Through my testing, starting at 8000 Hz, and incrementing by 1000 Hz, speaker-test does produce a sine wave, until I went over 15000 Hz then no more sine wave. :) :) Code: speaker-test -t sine -f 8000 -X -l 0 speaker-test -t sine -f 15000 -X -l 0 :( Code: speaker-test -t sine -f 16000 -X -l 0 The work around for the time being is to configure speaker-test to play a 20kHz wav file, this dose work and produces a beautiful sine wav :) Code: speaker-test -t wav -f /opt/20kHz.wav -l 0 Now for the people who will say why on earth do you want to produce a sound that no one can hear, and my reply is "Exactly". I send the 20kHz sine wav across the audio line as a way to supervise that the audio line is fully functioning. The second the 20kHz tone stops we are notified and send someone out to fix the problem, whether it be hardware or someone accidentality cuts the wire, we want to know. Basically the 20 kHz sine wave does not interferer with paging in the system. My first thoughts on why the 20 kHz sine is not produced, is because of missing driver. But I made sure that every sound driver that was loaded on Ubuntu was loaded on Tinycore. Code: sudo modprobe snd_seq_midi snd_seq_midi_event snd_intel_dspcfg I also loaded AlsaPlugins.tcz just incase, but that did not make a difference. Code: tceload -i alsa-plugins.tcz The last thing I tried after was to set the format sample -F parameter to 96000 Code: speaker-test -t sine -f 20000 -X -l 0 -F 96000 At first when I did this I thought I was so smart. A sine wave appeared, however I did not have a speaker hooked up, but when the speaker got plugged in I heard a high pitched noise. So the parameter affected the output but not the desired output that I need. Another thought that pass through my mind is could this be a configuration setting during the build of alsa that need to be turned on. After all 15000 Hz does work. This is slightly related to a previous post linked below. https://forum.tinycorelinux.net/index.php/topic,27673.0.html Thank you Any help is greatly appreciated gharig

1 0

Missing analog-stereo profile for Dell Latitude 3420 (ALC3204, SOF, Fedora 42)
by GitHub issues - opened 14 Jul '25

14 Jul '25

alsa-project/alsa-ucm-conf issue #594 was opened from KrzysztofWisiecki: **Summary:** On my Dell Latitude 3420 (Intel Tiger Lake, Realtek ALC3204, sof-hda-dsp), running Fedora 42 with PipeWire and WirePlumber (no PulseAudio), I am unable to select a standard "analog-stereo" profile for the internal sound card. Only "pro-audio" and "off" profiles are available. This makes it impossible to use speakers and the internal microphone together in a typical desktop setup. **System details:** Laptop: Dell Latitude 3420 Codec: Realtek ALC3204 Sound driver: sof-hda-dsp (SOF, not snd_hda_intel) OS: Fedora 42, fully updated PipeWire: Compiled with libpipewire 1.4.6 Linked with libpipewire 1.4.6 WirePlumber: Compiled with libwireplumber 0.5.8 Linked with libwireplumber 0.5.8 Kernel: 6.14.0-63.fc42.x86_64 **What is the issue?** No "analog-stereo" (or other typical analog) profiles are available. Only these profiles are listed: off pro-audio Profile: off: Off (sinks: 0, sources: 0, priority: 0, available: yes) pro-audio: Pro Audio (sinks: 5, sources: 3, priority: 1, available: yes) Active Profile: pro-audio ALSA devices are present: cat /proc/asound/cards 0 [sofhdadsp ]: sof-hda-dsp - sof-hda-dsp DellInc.-Latitude3420--03RT7P aplay -l card 0: sofhdadsp [sof-hda-dsp], device 0: HDA Analog (*) [] ... card 0: sofhdadsp [sof-hda-dsp], device 6: DMIC (*) [] ... The analog inputs/outputs exist at the ALSA layer. There is no file /usr/share/alsa-card-profile/mixer/profile-sets/sof-hda-dsp.conf by default. Attempting to copy or create a minimal sof-hda-dsp.conf with analog-stereo mapping does not enable the analog profile – only "pro-audio" appears. What have I tried? Created a custom /usr/share/alsa-card-profile/mixer/profile-sets/sof-hda-dsp.conf (based on default.conf/analog-only.conf/minimal analog-stereo mapping). Set api.acp.auto-profile and api.acp.auto-port to true for this device in /etc/wireplumber/wireplumber.conf.d/99-auto-profile.conf. Restarted PipeWire and WirePlumber. Reinstalled firmware (alsa-sof-firmware), alsa-card-profiles, alsa-ucm-conf. Checked for conflicting blacklist or modprobe options. Verified that everything works as expected with snd_hda_intel legacy driver (dsp_driver=1), but I want to use SOF Tried updating to the latest Fedora 42 and checked for updates in alsa-card-profiles/ucm-conf. Expected: "analog-stereo" (or similar) profile should be available for the internal sound card (sof-hda-dsp) so that both speakers and microphones can be used in a standard desktop configuration. Automatic profile switching should work (i.e., plugging in headphones/switching between speakers and headphones, using internal mic). Actual: Only "pro-audio" and "off" profiles are available. Manual switching is possible in "pro-audio", but lacks the convenience and user experience of standard profiles. No "analog-stereo", "analog-duplex" or similar options are listed. Request / What I'd like to see: Please add or enable standard analog profiles ("analog-stereo", "analog-duplex", etc.) for Dell Latitude 3420 with Realtek ALC3204 and sof-hda-dsp, if possible. Any guidance on manually enabling such profiles in the meantime would be appreciated. Let me know if you need full logs, system info, or additional tests! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/594 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: rt5660: Fix the dmic data source from GPIO2
by Oder Chiou 14 Jul '25

14 Jul '25

The patch fixes an issue with the dmic data pin connected to GPIO2. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5660.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index 82b92e83be4c..44c3a3b92f98 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1315,14 +1315,17 @@ static int rt5660_i2c_probe(struct i2c_client *i2c) regmap_update_bits(rt5660->regmap, RT5660_GPIO_CTRL1, RT5660_GP1_PIN_MASK, RT5660_GP1_PIN_DMIC1_SCL); - if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_GPIO2) + if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_GPIO2) { regmap_update_bits(rt5660->regmap, RT5660_DMIC_CTRL1, RT5660_SEL_DMIC_DATA_MASK, RT5660_SEL_DMIC_DATA_GPIO2); - else if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_IN1P) + regmap_update_bits(rt5660->regmap, RT5660_GPIO_CTRL1, + RT5660_GP2_PIN_MASK, RT5660_GP2_PIN_DMIC1_SDA); + } else if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_IN1P) { regmap_update_bits(rt5660->regmap, RT5660_DMIC_CTRL1, RT5660_SEL_DMIC_DATA_MASK, RT5660_SEL_DMIC_DATA_IN1P); + } } return devm_snd_soc_register_component(&i2c->dev, -- 2.34.1

2 1

I'm using Ubuntu 24.04.2 LTS with Wayland on an iMac 27" (2017) equipped with the Cirrus Logic CS8409 audio codec. After compiling the driver manually (https://github.com/egorenar/snd-hda-codec-cs8409), ALSA detects the audio device correctly and it works properly with PipeWire and PulseAudio. However, it breaks with each kernel update, requiring recompilation. **Request**: Please consider adding native support for CS8409 in `alsa-ucm-conf`. Modern systems like Ubuntu 24.04.2 handle the device well, and i
by GitHub issues - edited 14 Jul '25

14 Jul '25

alsa-project/alsa-ucm-conf issue #592 was edited from Marek-mahav: -w Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/592 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Feature request: Add native support for CS8409 codec
by GitHub issues - opened 14 Jul '25

14 Jul '25

alsa-project/alsa-ucm-conf issue #593 was opened from Marek-mahav: I've tested the CS8409 driver on Ubuntu 24.04.2 with Wayland… [celý text tu] Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/593 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Feature request: Add native support for Cirrus Logic CS8409 codec
by GitHub issues - opened 14 Jul '25

14 Jul '25

alsa-project/alsa-ucm-conf issue #592 was opened from Marek-mahav: -w Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/592 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Feature request: Add native support for Cirrus Logic CS8409 codec
by GitHub issues - opened 14 Jul '25

14 Jul '25

alsa-project/alsa-ucm-conf issue #591 was opened from Marek-mahav: -w Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/591 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

timer/seq: fix minor issues
by GitHub pull_request - opened 12 Jul '25

12 Jul '25

alsa-project/alsa-gobject pull request #85 was opened from takaswie: As long as building on Ubuntu 25.04, some codes takes warnings from gcc 14 and gi-docgen 2025.3-1. This patchset includes some fixes to solve the codes. Request URL : https://github.com/alsa-project/alsa-gobject/pull/85 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/85.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH v1] ALSA: hda/tas2781: Add TAS2770 support
by Baojun Xu 12 Jul '25

12 Jul '25

Add TAS2770 support in HDA. Create a header file include/sound/tas2770-tlv.h, Set chip_id in i2c probe, check it while sound control is created, and the DSP firmware binary file parser. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2770-tlv.h | 20 ++++++++++++ include/sound/tas2781.h | 4 +++ sound/pci/hda/tas2781_hda_i2c.c | 58 +++++++++++++++++++++++++-------- 3 files changed, 69 insertions(+), 13 deletions(-) create mode 100644 include/sound/tas2770-tlv.h diff --git a/include/sound/tas2770-tlv.h b/include/sound/tas2770-tlv.h new file mode 100644 index 000000000000..2a2a74372366 --- /dev/null +++ b/include/sound/tas2770-tlv.h @@ -0,0 +1,20 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2770 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2770 hda driver implements for one, two, or even multiple +// TAS2770 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2770_TLV_H__ +#define __TAS2770_TLV_H__ + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_amp_tlv, 1100, 50, 0); + +#endif diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 3875e92f1ec5..f3d9c80fad36 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -63,6 +63,9 @@ /* Volume control */ #define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0c) +#define TAS2770_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x17) +#define TAS2770_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) + #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) @@ -111,6 +114,7 @@ enum audio_device { TAS2563, + TAS2770, TAS2781, }; diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index b7ee22840d78..39f00e33e49d 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -24,6 +24,7 @@ #include <sound/tas2781.h> #include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> +#include <sound/tas2770-tlv.h> #include <sound/tas2781-tlv.h> #include "hda_local.h" @@ -245,6 +246,12 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static const struct snd_kcontrol_new tas2770_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2770_AMP_LEVEL, + 0, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, tas2770_amp_tlv), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -253,7 +260,7 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_force_fwload_get, tas2781_force_fwload_put), }; -static const struct snd_kcontrol_new tas2781_prof_ctrl = { +static const struct snd_kcontrol_new tas27xx_prof_ctrl = { .name = "Speaker Profile Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = tasdevice_info_profile, @@ -393,26 +400,45 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) if (ret) goto out; - tas_hda->prof_ctl = snd_ctl_new1(&tas2781_prof_ctrl, tas_priv); + tas_hda->prof_ctl = snd_ctl_new1(&tas27xx_prof_ctrl, 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.name, ret); + tas27xx_prof_ctrl.name, ret); goto out; } - for (i = 0; i < ARRAY_SIZE(tas2781_snd_controls); i++) { - hda_priv->snd_ctls[i] = snd_ctl_new1(&tas2781_snd_controls[i], - tas_priv); - ret = snd_ctl_add(codec->card, hda_priv->snd_ctls[i]); - if (ret) { - dev_err(tas_priv->dev, - "Failed to add KControl %s = %d\n", - tas2781_snd_controls[i].name, ret); - goto out; + if (tas_priv->chip_id == TAS2770) { + for (i = 0; i < ARRAY_SIZE(tas2770_snd_controls); i++) { + hda_priv->snd_ctls[i] = snd_ctl_new1( + &tas2770_snd_controls[i], tas_priv); + ret = snd_ctl_add(codec->card, hda_priv->snd_ctls[i]); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2770_snd_controls[i].name, ret); + goto out; + } + } + } else if (tas_priv->chip_id == TAS2781) { + for (i = 0; i < ARRAY_SIZE(tas2781_snd_controls); i++) { + hda_priv->snd_ctls[i] = snd_ctl_new1( + &tas2781_snd_controls[i], tas_priv); + ret = snd_ctl_add(codec->card, hda_priv->snd_ctls[i]); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_snd_controls[i].name, ret); + goto out; + } } } + // No dsp supported in TAS2770. + if (tas_priv->chip_id == TAS2770) { + tas_hda->priv->playback_started = true; + goto out; + } tasdevice_dsp_remove(tas_priv); @@ -453,7 +479,6 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) tas2781_dsp_prog_ctrl.name, ret); goto out; } - tas_hda->dsp_conf_ctl = snd_ctl_new1(&tas2781_dsp_conf_ctrl, tas_priv); ret = snd_ctl_add(codec->card, tas_hda->dsp_conf_ctl); @@ -582,15 +607,21 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) if (strstr(dev_name(&clt->dev), "TIAS2781")) { device_name = "TIAS2781"; + tas_hda->priv->chip_id = TAS2781; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; + } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW2770")) { + device_name = "TXNW2770"; + tas_hda->priv->chip_id = TAS2770; } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW2781:00-tas2781-hda.0")) { device_name = "TXNW2781"; + tas_hda->priv->chip_id = TAS2781; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; } else if (strstr(dev_name(&clt->dev), "INT8866")) { device_name = "INT8866"; + tas_hda->priv->chip_id = TAS2563; hda_priv->save_calibration = tas2563_save_calibration; tas_hda->priv->global_addr = TAS2563_GLOBAL_ADDR; } else { @@ -727,6 +758,7 @@ static const struct i2c_device_id tas2781_hda_i2c_id[] = { static const struct acpi_device_id tas2781_acpi_hda_match[] = { {"INT8866", 0 }, {"TIAS2781", 0 }, + {"TXNW2770", 0 }, {"TXNW2781", 0 }, {} }; -- 2.43.0

1 0

Realtek ALC897 - no 5.1, 7.1 surround sound
by GitHub issues - opened 10 Jul '25

10 Jul '25

alsa-project/alsa-ucm-conf issue #590 was opened from tomaszdrozdz: Hello all. In nutshell: - I can not set 5.1 sound on my Fedora. - My sound card is Realtek ALC897. First I described it here: https://discussion.fedoraproject.org/t/5-1-7-1-sound-speakers/156932/4 Then here: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4788 And so I got here. If you need more info or action to be taken from me please let me know. Help, assist - Please. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/590 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: rt5650: Eliminate the high frequency glitch
by derek.fang＠realtek.com 10 Jul '25

10 Jul '25

From: Derek Fang <derek.fang(a)realtek.com> The glitch was detected in the high frequency of the HP playback. This patch adjusts the DAC dither setting to avoid this situation for almost all cases. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 08df87238..29a403526 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -82,6 +82,7 @@ static const struct reg_sequence rt5650_init_list[] = { {0xf6, 0x0100}, {RT5645_PWR_ANLG1, 0x02}, {RT5645_IL_CMD3, 0x6728}, + {RT5645_PR_BASE + 0x3a, 0x0000}, }; static const struct reg_default rt5645_reg[] = { -- 2.34.1

1 0

[GIT PULL] ASoC fixes for v6.16-rc5
by Mark Brown 10 Jul '25

10 Jul '25

The following changes since commit 3b3312f28ee2d9c386602f8521e419cfc69f4823: ASoC: cs35l56: probe() should fail if the device ID is not recognized (2025-07-03 12:22:27 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.16-rc5 for you to fetch changes up to a609bd74b8680dba62c44f7e6d00d381ddb2d3c0: ASoC: Intel: avs: Fix NULL ptr deref on rmmod (2025-07-08 14:54:40 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.16 Two patches here, one quirk for an AMD system and a fix for an issue on remove of the AVS driver. ---------------------------------------------------------------- Alexandru Andries (1): ASoC: amd: yc: add DMI quirk for ASUS M6501RM Amadeusz Sławiński (1): ASoC: Intel: avs: Fix NULL ptr deref on rmmod sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/intel/avs/pcm.c | 4 +++- 2 files changed, 10 insertions(+), 1 deletion(-)

2 1

[PATCH] ASoC: samsung: Implement abox generic structure
by ew.kim＠samsung.com 10 Jul '25

10 Jul '25

From: ew kim <ew.kim(a)samsung.com> Implemet basic abox generic drivers. This driver is a management driver for the generic drivers used in Automotive Abox, connecting them to SOC drivers. It supports various Exynos Automotive SOCs. Signed-off-by: ew kim <ew.kim(a)samsung.com> --- sound/soc/samsung/Kconfig | 2 + sound/soc/samsung/Makefile | 1 + sound/soc/samsung/auto_abox/Kconfig | 14 + sound/soc/samsung/auto_abox/generic/Kbuild | 12 + .../samsung/auto_abox/generic/abox_generic.c | 568 ++++++++++++++++++ .../auto_abox/generic/include/abox_generic.h | 87 +++ 6 files changed, 684 insertions(+) create mode 100644 sound/soc/samsung/auto_abox/Kconfig create mode 100644 sound/soc/samsung/auto_abox/generic/Kbuild create mode 100644 sound/soc/samsung/auto_abox/generic/abox_generic.c create mode 100644 sound/soc/samsung/auto_abox/generic/include/abox_generic.h diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 60b4b7b75215..359aa67f49db 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -148,4 +148,6 @@ config SND_SOC_SAMSUNG_MIDAS_WM1811 help Say Y if you want to add support for SoC audio on the Midas boards. +source "sound/soc/samsung/auto_abox/Kconfig" + endif #SND_SOC_SAMSUNG diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 8d5f09147900..5d99cfbfa71c 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -42,3 +42,4 @@ obj-$(CONFIG_SND_SOC_ARNDALE) += snd-soc-arndale.o obj-$(CONFIG_SND_SOC_SAMSUNG_TM2_WM5110) += snd-soc-tm2-wm5110.o obj-$(CONFIG_SND_SOC_SAMSUNG_ARIES_WM8994) += snd-soc-aries-wm8994.o obj-$(CONFIG_SND_SOC_SAMSUNG_MIDAS_WM1811) += snd-soc-midas-wm1811.o +obj-$(CONFIG_SND_SOC_SAMSUNG_AUTO_ABOX) += auto_abox/generic/ \ No newline at end of file diff --git a/sound/soc/samsung/auto_abox/Kconfig b/sound/soc/samsung/auto_abox/Kconfig new file mode 100644 index 000000000000..d5f565d3d9c2 --- /dev/null +++ b/sound/soc/samsung/auto_abox/Kconfig @@ -0,0 +1,14 @@ +#: SPDX-License-Identifier: GPL-2.0-only + +menu "Exynosauto Automotive Abox Modules Options" + +config SND_SOC_SAMSUNG_AUTO_ABOX + tristate "ASoC support for Samsung Exynosauto Automotive ABOX Audio" + select SND_SOC_COMPRESS + help + Say Y or M if you want to add support for codecs attached to + the Samsung SoC Auto ABOX interface. You will also need to + select the audio interfaces to support below. + +endmenu + diff --git a/sound/soc/samsung/auto_abox/generic/Kbuild b/sound/soc/samsung/auto_abox/generic/Kbuild new file mode 100644 index 000000000000..fa6ba7091730 --- /dev/null +++ b/sound/soc/samsung/auto_abox/generic/Kbuild @@ -0,0 +1,12 @@ +# SPDX-License-Identifier: GPL-2.0 +# Exynosauto Automotive Abox Driver Support + +snd-soc-samsung-abox-generic-$(CONFIG_SND_SOC_SAMSUNG_AUTO_ABOX) := \ + abox_generic.o + +ccflags-y += -I./include + +obj-$(CONFIG_SND_SOC_SAMSUNG_AUTO_ABOX) += \ + snd-soc-samsung-abox-generic.o + + diff --git a/sound/soc/samsung/auto_abox/generic/abox_generic.c b/sound/soc/samsung/auto_abox/generic/abox_generic.c new file mode 100644 index 000000000000..c4588e0a6a60 --- /dev/null +++ b/sound/soc/samsung/auto_abox/generic/abox_generic.c @@ -0,0 +1,568 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2020 Samsung Electronics Co., Ltd. + * http://www.samsung.com/ + * + * EXYNOS - sound/soc/samsung/auto_abox/generic/abox_generic.c + */ + +//#define DEBUG +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_address.h> +#include <linux/of_platform.h> +#include <linux/delay.h> +#include <linux/suspend.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> +#include <linux/of_reserved_mem.h> +#include <linux/types.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/sched/clock.h> +#include <linux/ktime.h> +#include <linux/iommu.h> +#include <linux/clk-provider.h> +#include <linux/kmod.h> +#include <linux/umh.h> +#include <linux/string.h> + +#include "include/abox_generic.h" + +static struct abox_generic_data *g_abox_generic_data; + +/** + * @cnotice + * @prdcode + * @Sub_SW_Component{abox_generic} + * @ALM_Link {work item url} + * @purpose "get value from virtual address" + * @logic "return global abox_generic_data" + * \image html + * @params + * @param{in, -, -, -} + * @endparam + * @retval {-, struct *abox_generic_data, !NULL, NULL} + */ +struct abox_generic_data *abox_generic_get_abox_generic_data(void) +{ + return g_abox_generic_data; +} + +static struct abox_generic_data *abox_generic_get_generic_data_from_child(struct device *child_dev) +{ + struct device *generic_dev = child_dev->parent; + struct abox_generic_data *generic_data = NULL; + + if (!generic_dev) { + pr_err("%s Failed to get generic device\n", __func__); + return NULL; + } + generic_data = dev_get_drvdata(generic_dev); + if (!generic_data) { + dev_err(generic_dev, "%s Failed to get generic data\n", __func__); + return NULL; + } + return generic_data; +} + +int abox_generic_set_dma_buffer(struct device *pcm_dev) +{ + struct abox_generic_data *generic_data = abox_generic_get_generic_data_from_child(pcm_dev); + int ret = 0; + + if (!generic_data) { + dev_err(pcm_dev, "%s Failed to get generic data\n", __func__); + return 0; + } + + if (!generic_data->soc_ioctl) { + dev_err(pcm_dev, "%s Failed to get soc_ioctl\n", __func__); + return 0; + } + ret = generic_data->soc_ioctl(generic_data->soc_dev, ABOX_SOC_IOCTL_SET_DMA_BUFFER, pcm_dev); + + return ret; +} + +int abox_generic_set_pp_pointer(struct device *pcm_dev) +{ + struct abox_generic_data *generic_data = abox_generic_get_generic_data_from_child(pcm_dev); + int ret = 0; + + if (!generic_data) { + dev_err(pcm_dev, "%s Failed to get generic data\n", __func__); + return 0; + } + if (!generic_data->soc_ioctl) { + dev_err(pcm_dev, "%s Failed to get soc_ioctl\n", __func__); + return 0; + } + ret = generic_data->soc_ioctl(generic_data->soc_dev, ABOX_SOC_IOCTL_SET_PP_POINTER, pcm_dev); + + return ret; +} + +int abox_generic_attach_soc_callback(struct device *soc_dev, + SOC_IOCTL soc_ioctl) +{ + struct abox_generic_data *generic_data = ABOX_GENERIC_DATA; + + dev_info(soc_dev, "%s(%d) Attach SoC IOCTL\n", __func__, __LINE__); + if (!generic_data) { + dev_err(soc_dev, "%s Generic Drv is not ready\n", __func__); + return -ENODATA; + } + generic_data->soc_dev = soc_dev; + generic_data->soc_ioctl = soc_ioctl; + + generic_data->num_of_rdma = generic_data->soc_ioctl(generic_data->soc_dev, + ABOX_SOC_IOCTL_GET_NUM_OF_RDMA, NULL); + generic_data->num_of_wdma = generic_data->soc_ioctl(generic_data->soc_dev, + ABOX_SOC_IOCTL_GET_NUM_OF_WDMA, NULL); + generic_data->num_of_uaif = generic_data->soc_ioctl(generic_data->soc_dev, + ABOX_SOC_IOCTL_GET_NUM_OF_UAIF, NULL); + dev_info(soc_dev, "%s(%d) num_of_rdma:%d\n", __func__, __LINE__, generic_data->num_of_rdma); + dev_info(soc_dev, "%s(%d) num_of_wdma:%d\n", __func__, __LINE__, generic_data->num_of_wdma); + dev_info(soc_dev, "%s(%d) num_of_uaif:%d\n", __func__, __LINE__, generic_data->num_of_uaif); + + return 0; +} +EXPORT_SYMBOL(abox_generic_attach_soc_callback); + +struct platform_device *abox_generic_get_pcm_platform_dev(int pcm_id, int stream_type) +{ + struct abox_generic_data *generic_data = ABOX_GENERIC_DATA; + struct platform_device **pdev_pcm = NULL; + + if (stream_type == SNDRV_PCM_STREAM_PLAYBACK) + pdev_pcm = generic_data->pdev_pcm_playback; + else + pdev_pcm = generic_data->pdev_pcm_capture; + + return pdev_pcm[pcm_id]; +} +EXPORT_SYMBOL(abox_generic_get_pcm_platform_dev); + +int abox_generic_get_num_of_pcm(int stream_type) +{ + struct abox_generic_data *generic_data = ABOX_GENERIC_DATA; + + if (!generic_data) { + pr_err("%s Failed to get abox_generic_data\n", __func__); + return -ENODATA; + } + + return (stream_type == SNDRV_PCM_STREAM_PLAYBACK) ? generic_data->num_of_pcm_playback : + generic_data->num_of_pcm_capture; +} +EXPORT_SYMBOL(abox_generic_get_num_of_pcm); + +int abox_generic_get_num_of_i2s_dummy(void) +{ + struct abox_generic_data *generic_data = ABOX_GENERIC_DATA; + + if (!generic_data) { + pr_err("%s Failed to get abox_generic_data\n", __func__); + return -ENODATA; + } + + return generic_data->num_of_i2s_dummy; +} +EXPORT_SYMBOL(abox_generic_get_num_of_i2s_dummy); + +int abox_generic_get_num_of_dma(struct device *pcm_dev, int stream_type) +{ + struct abox_generic_data *generic_data = abox_generic_get_generic_data_from_child(pcm_dev); + + return (stream_type == SNDRV_PCM_STREAM_PLAYBACK) ? + generic_data->num_of_rdma : generic_data->num_of_wdma; +} + +/** + * @cnotice + * @prdcode + * @Sub_SW_Component{abox generic} + * @ALM_Link {work item url} + * @purpose "Registering the pcm_playback/pcm_capture pdev to abox driver" + * @logic "Get the pcm playback instance and update the pcm playback/capture id and + * increament the pcm playback/capture count" + * \image html + * @params + * @param{in, pdev_pcm_dev, struct platform_device *, !NULL} + * @param{in, id, unsigned int, 0 ~ 32} + * @param{in, stream_type, int, 0 ~ 1} + * @endparam + * @retval{ret, int, Undefined, 0, <0} + */ +int abox_generic_register_pcm_dev(struct platform_device *pdev_pcm, + unsigned int id, int stream_type) +{ + struct device *pcm_dev = &pdev_pcm->dev; + struct abox_generic_data *generic_data = abox_generic_get_generic_data_from_child(pcm_dev); + int num_of_pcm_dev = 0; + + dev_dbg(pcm_dev, "[%s] PCM%d Attached Stream_type:%d\n", __func__, id, stream_type); + num_of_pcm_dev = (stream_type == SNDRV_PCM_STREAM_PLAYBACK) ? + generic_data->num_of_pcm_playback : generic_data->num_of_pcm_capture; + if (id >= num_of_pcm_dev) { + dev_err(pcm_dev, "%s: invalid id(%u) : Stream Type:%d\n", __func__, id, stream_type); + return -EINVAL; + } + + if (stream_type == SNDRV_PCM_STREAM_PLAYBACK) + generic_data->pdev_pcm_playback[id] = pdev_pcm; + else + generic_data->pdev_pcm_capture[id] = pdev_pcm; + + return 0; +} + +/** + * @cnotice + * @prdcode + * @Sub_SW_Component{abox} + * @ALM_Link {work item url} + * @purpose "finding struct device of frontend from backend" + * @logic "get device of backend using component and alsa macro" + * \image html audio-interface_abox_abox_find_fe_dev_from_rtd.png + * @params + * @param{in, be, struct snd_soc_pcm_runtime *, -} + * @endparam + * @retval {-, struct device *, !NULL, NULL} + */ +struct device *abox_generic_find_fe_dev_from_rtd(struct snd_soc_pcm_runtime *be) +{ + struct abox_generic_data *generic_data = ABOX_GENERIC_DATA; + struct snd_soc_dpcm *dpcm = NULL; + struct snd_soc_pcm_runtime *fe = NULL; + int stream_type = 0; + + if (!generic_data) + return NULL; + + for (stream_type = 0; stream_type <= SNDRV_PCM_STREAM_LAST; stream_type++) { + int cmpnt_index = 0; + struct snd_soc_component *component = NULL; + + for_each_dpcm_fe(be, stream_type, dpcm) { + fe = dpcm->fe; + if (fe) + break; + } + if (!fe) + continue; + + for_each_rtd_components(fe, cmpnt_index, component) { + struct platform_device **pdev = NULL; + int num_of_pcm_dev = 0; + int i = 0; + + if (stream_type == SNDRV_PCM_STREAM_PLAYBACK) { + num_of_pcm_dev = generic_data->num_of_pcm_playback; + pdev = generic_data->pdev_pcm_playback; + } else { + num_of_pcm_dev = generic_data->num_of_pcm_capture; + pdev = generic_data->pdev_pcm_capture; + } + for (i = 0; i < num_of_pcm_dev; i++) + if (pdev[i] && component->dev == &pdev[i]->dev) + return component->dev; + } + } + + return NULL; +} +EXPORT_SYMBOL_GPL(abox_generic_find_fe_dev_from_rtd); + +int abox_generic_request_soc_ioctl(struct device *generic_dev, enum abox_soc_ioctl_cmd cmd, + void *data) +{ + struct abox_generic_data *generic_data = dev_get_drvdata(generic_dev); + struct device *soc_dev = generic_data->soc_dev; + + if (IS_ERR_OR_NULL(soc_dev)) { + dev_err(generic_dev, "%s SoC Device is not ready\n", __func__); + return -ENODATA; + } + return generic_data->soc_ioctl(soc_dev, cmd, data); +} + +/** + * @cnotice + * @prdcode + * @Sub_SW_Component{abox generic} + * @ALM_Link + * @purpose "suspend of the abox generic to be called by S2R" + * @logic + * \image html + * @params + * @param{in, dev, struct device *, !NULL} + * @endparam + * @retval{ret, int, Undefined, 0, <0} + */ +static int abox_generic_suspend(struct device *dev) +{ + struct abox_generic_data *data = dev_get_drvdata(dev); + int ret = 0; + + dev_info(dev, "%s start\n", __func__); + if (!data) { + dev_err(dev, "%s: Invalid abox generic data\n", __func__); + return -ENODATA; + } + + dev_info(dev, "%s end\n", __func__); + + return ret; +} + +/** + * @cnotice + * @prdcode + * @Sub_SW_Component{abox generic} + * @ALM_Link + * @purpose "Resume the abox generic during S2R operation" + * @logic + * \image html + * @params + * @param{in, dev, struct device *, !NULL} + * @endparam + * @retval{ret, int, Undefined, 0, <0} + */ +static int abox_generic_resume(struct device *dev) +{ + struct abox_generic_data *data = dev_get_drvdata(dev); + int ret = 0; + + dev_info(dev, "%s start\n", __func__); + if (!data) { + dev_err(dev, "%s: Invalid abox generic data\n", __func__); + return -ENODATA; + } + + dev_info(dev, "%s end\n", __func__); + return ret; +} + +static struct platform_driver *abox_generic_sub_drivers[] = { +}; + +/** + * @cnotice + * @prdcode + * @Sub_SW_Component{abox} + * @ALM_Link {work item url} + * @purpose "Read property from device tree node" + * @logic + * \image html + * @params + * @param{in, dev, struct:: device *, !NULL} + * @param{in, dev->of_node, struct:: device_node, !NULL} + * @param{out, data->num_of_pcm_playback, unsigned int, 0} + * @param{out, data->num_of_pcm_capture, unsigned int, 0} + * @param{out, data->num_of_i2s_dummy, unsigned int, 0} + * @param{out, abox_sfr_set, int, 0} + * @endparam + * @retval{ret, int, 0, 0, > 0} + */ +static int abox_generic_read_property_from_dt(struct device *dev, struct abox_generic_data *data) +{ + struct device_node *np = dev->of_node; + int ret = 0; + + ret = of_property_read_u32(np, "samsung,num-of-pcm_playback", &data->num_of_pcm_playback); + if (ret < 0) { + dev_err(dev, "%s property reading fail\n", "samsung,num-of-pcm_playback"); + return ret; + } + ret = of_property_read_u32(np, "samsung,num-of-pcm_capture", &data->num_of_pcm_capture); + if (ret < 0) { + dev_err(dev, "%s property reading fail\n", "samsung,num-of-pcm_capture"); + return ret; + } + ret = of_property_read_u32(np, "samsung,num-of-i2s-dummy-backend", &data->num_of_i2s_dummy); + if (ret < 0) { + dev_err(dev, "%s property reading fail\n", "samsung,num-of-i2s-dummy-backend"); + return ret; + } + + return ret; +} + +/** + * @cnotice + * @prdcode + * @Sub_SW_Component{abox generic} + * @ALM_Link {work item url} + * @purpose "Allocate memory for abox generic" + * @logic + * \image html + * @params + * @param{in, dev, struct:: device *, !NULL} + * @param{in, data, struct:: abox_gneric_data, !NULL} + * @param{out, data->pdev_pcm_playback, struct:: platform_device, !NULL} + * @param{out, data->pdev_pcm_capture, struct:: platform_device, !NULL} + * @endparam + * @retval{ret, int, 0, 0, > 0} + */ +static int abox_generic_allocate_memory(struct device *dev, struct abox_generic_data *data) +{ + int ret = 0; + + data->pdev_pcm_playback = devm_kzalloc(dev, + sizeof(struct platform_device *) * data->num_of_pcm_playback, GFP_KERNEL); + if (!data->pdev_pcm_playback) { + dev_err(dev, "%s Can't allocate memory for pdev_pcm_playback\n", __func__); + ret = -ENOMEM; + return ret; + } + data->pdev_pcm_capture = devm_kzalloc(dev, + sizeof(struct platform_device *) * data->num_of_pcm_capture, GFP_KERNEL); + if (!data->pdev_pcm_capture) { + dev_err(dev, "%s Can't allocate memory for pdev_pcm_capture\n", __func__); + ret = -ENOMEM; + return ret; + } + + return ret; +} + +/** + * @cnotice + * @prdcode + * @Sub_SW_Component{abox generic} + * @ALM_Link {work item url} + * @purpose "Probing the abox generic" + * @logic + * \image html + * @params + * @param{in, pdev, struct platform_device *, !NULL} + * @param{in, pdev->dev, struct:: device, !NULL} + * @endparam + * @retval{ret, int, 0, 0, > 0} + */ +static int samsung_abox_generic_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct device_node *np = dev->of_node; + struct abox_generic_data *data; + int ret = 0; + + dev_info(dev, "%s\n", __func__); + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); + if (!data) { + return -ENOMEM; + } + + data->pdev = pdev; + ret = abox_generic_read_property_from_dt(dev, data); + if (ret < 0) { + dev_err(dev, "%s Failed to read property. ret:%d\n", __func__, ret); + return ret; + } + ret = abox_generic_allocate_memory(dev, data); + if (ret < 0) { + dev_err(dev, "%s Failed to allocate memory. ret:%d\n", __func__, ret); + return ret; + } + g_abox_generic_data = data; + platform_set_drvdata(pdev, data); + + platform_register_drivers(abox_generic_sub_drivers, ARRAY_SIZE(abox_generic_sub_drivers)); + ret = of_platform_populate(np, NULL, NULL, dev); + if (ret < 0) { + dev_err(dev, "Failed to populate sub-platform_devices. ret:%d\n", ret); + return ret; + } + + return ret; +} + +/** + * @cnotice + * @prdcode + * @Sub_SW_Component{abox generic} + * @ALM_Link {work item url} + * @purpose "Disbaling the abox generic" + * @logic "Disbale the abox generic" + * \image html + * @params + * @param{in, pdev->dev, struct::device, !NULL} + * @endparam + * @noret + */ +static void samsung_abox_generic_remove(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct abox_generic_data *data = dev_get_drvdata(dev); + + dev_info(dev, "%s\n", __func__); + + if (!data) { + dev_err(dev, "%s: Invalid abox generic data\n", __func__); + return; + } + return; +} + +/** + * @cnotice + * @prdcode + * @Sub_SW_Component{abox generic} + * @ALM_Link {work item url} + * @purpose "shutdown of the abox generic" + * @logic "Disbale the abox hardware by calling the following function + * pm_runtime_disable(dev)" + * \image html + * @params + * @param{in, pdev->dev, struct:: device, !NULL} + * @endparam + * @noret + */ +static void samsung_abox_generic_shutdown(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct abox_generic_data *data = dev_get_drvdata(dev); + + if (!data) { + dev_err(dev, "%s: Invalid abox generic data\n", __func__); + return; + } + return; +} + +static const struct of_device_id samsung_abox_generic_match[] = { + { + .compatible = "samsung,abox_generic", + }, + {}, +}; +MODULE_DEVICE_TABLE(of, samsung_abox_generic_match); + +static const struct dev_pm_ops samsung_abox_generic_pm = { + SET_SYSTEM_SLEEP_PM_OPS(abox_generic_suspend, abox_generic_resume) +}; + +struct platform_driver samsung_abox_generic_driver = { + .probe = samsung_abox_generic_probe, + .remove = samsung_abox_generic_remove, + .shutdown = samsung_abox_generic_shutdown, + .driver = { + .name = "samsung-abox-generic", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(samsung_abox_generic_match), + .pm = &samsung_abox_generic_pm, + }, +}; + +module_platform_driver(samsung_abox_generic_driver); +/* Module information */ +MODULE_AUTHOR("Eunwoo Kim, <ew.kim(a)samsung.com>"); +MODULE_DESCRIPTION("Samsung ASoC A-Box Generic Driver"); +MODULE_ALIAS("platform:samsung-abox-generic"); +MODULE_LICENSE("GPL v2"); + diff --git a/sound/soc/samsung/auto_abox/generic/include/abox_generic.h b/sound/soc/samsung/auto_abox/generic/include/abox_generic.h new file mode 100644 index 000000000000..1c954272e2b5 --- /dev/null +++ b/sound/soc/samsung/auto_abox/generic/include/abox_generic.h @@ -0,0 +1,87 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * ALSA SoC - Samsung ABOX Share Function and Data structure + * for Exynos specific extensions + * + * Copyright (C) 2013-2020 Samsung Electronics Co., Ltd. + * + * EXYNOS - sound/soc/samsung/abox/include/abox_generic.h + */ + +#ifndef __SND_SOC_ABOX_GENERIC_BASE_H +#define __SND_SOC_ABOX_GENERIC_BASE_H + +#define ABOX_GENERIC_DATA abox_generic_get_abox_generic_data(); + +struct snd_soc_pcm_runtime; + +enum abox_soc_ioctl_cmd { + ABOX_SOC_IOCTL_GET_NUM_OF_RDMA, + ABOX_SOC_IOCTL_GET_NUM_OF_WDMA, + ABOX_SOC_IOCTL_GET_NUM_OF_UAIF, + ABOX_SOC_IOCTL_GET_SOC_TIMER, + ABOX_SOC_IOCTL_SET_DMA_BUFFER, + ABOX_SOC_IOCTL_SET_PP_POINTER, + ABOX_SOC_IOCTL_SET_PERF_PERIOD, + ABOX_SOC_IOCTL_CHECK_TIME_MUTEX, + ABOX_SOC_IOCTL_CHECK_TIME_NO_MUTEX, + ABOX_SOC_IOCTL_PCM_DUMP_INTR, + ABOX_SOC_IOCTL_PCM_DUMP_CLOSE, + ABOX_SOC_IOCTL_PCM_DUMP_ADD_CONTROL, + ABOX_SOC_IOCTL_MAX +}; + +typedef int (*SOC_IOCTL)(struct device *soc_dev, enum abox_soc_ioctl_cmd cmd, void *data); + +struct abox_generic_data { + struct platform_device *pdev; + struct platform_device **pdev_pcm_playback; + struct platform_device **pdev_pcm_capture; + unsigned int num_of_pcm_playback; + unsigned int num_of_pcm_capture; + unsigned int num_of_i2s_dummy; + unsigned int num_of_rdma; + unsigned int num_of_wdma; + unsigned int num_of_uaif; + struct device *soc_dev; + SOC_IOCTL soc_ioctl; +}; + + +/************ Internal API ************/ + +struct abox_generic_data *abox_generic_get_abox_generic_data(void); + +int abox_generic_set_dma_buffer(struct device *pcm_dev); + +int abox_generic_request_soc_ioctl(struct device *generic_dev, enum abox_soc_ioctl_cmd cmd, + void *data); + +int abox_generic_set_pp_pointer(struct device *pcm_dev); + + + + +/************ External API ************/ + +extern struct device *abox_generic_find_fe_dev_from_rtd(struct snd_soc_pcm_runtime *be); + +extern struct platform_device *abox_generic_get_pcm_platform_dev(int pcm_id, + int stream_type); + +extern int abox_generic_get_num_of_pcm(int stream_type); + +extern int abox_generic_get_num_of_i2s_dummy(void); + +extern int abox_generic_get_num_of_dma(struct device *pcm_dev, + int stream_type); + +extern int abox_generic_attach_soc_callback(struct device *soc_dev, + SOC_IOCTL soc_ioctl); + +extern int abox_generic_register_pcm_dev(struct platform_device *pdev_pcm_dev, + unsigned int id, int stream_type); + + +#endif //__SND_SOC_ABOX_GENERIC_BASE_H + -- 2.25.1

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[PATCH] ASoC: dt-bindings: Add samsung,abox-generic document
by ew kim 09 Jul '25

09 Jul '25

Add soundcard bindings for the abox generic of exynus automotive. Signed-off-by: ew kim <ew.kim(a)samsung.com> --- .../bindings/sound/samsung,abox-generic.yaml | 77 +++++++++++++++++++ 1 file changed, 77 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,abox-generic.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,abox-generic.yaml b/Documentation/devicetree/bindings/sound/samsung,abox-generic.yaml new file mode 100644 index 000000000000..bf641a197903 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,abox-generic.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/soc/samsung/abox-generic.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Samsung Exynos Automotive Abox Generic + +maintainers: + - name: Eunwoo Kim + email: ew.kim(a)samsung.com + +description: | + Samsung Exynos Automotive Abox Generic core node. + + This node represents the generic Abox controller which manages + various Abox sub-drivers and acts as a hub connecting them to + SoC-level audio drivers. It provides PCM playback and capture + configuration, as well as dummy I2S backend handling, enabling + integration between Abox IP blocks and the SoC audio subsystem. + + Typically, this node is used to initialize and configure playback, + capture, and routing between the Abox and other SoC audio components. + +properties: + compatible: + const: samsung,abox_generic + + samsung,num-of-pcm_playback: + description: Number of PCM playback instances. + $ref: /schemas/types.yaml#/definitions/uint32 + + samsung,num-of-pcm_capture: + description: Number of PCM capture instances. + $ref: /schemas/types.yaml#/definitions/uint32 + + samsung,num-of-i2s-dummy-backend: + description: Number of dummy I2S backend instances. + $ref: /schemas/types.yaml#/definitions/uint32 + + status: + description: Availability status. + enum: [ "okay", "disabled" ] + + '#address-cells': + description: Number of address cells in the node. + const: 2 + + '#size-cells': + description: Number of size cells in the node. + const: 1 + + ranges: + description: Standard `ranges` property for bus address mapping. + type: object + +required: + - compatible + - samsung,num-of-pcm_playback + - samsung,num-of-pcm_capture + - samsung,num-of-i2s-dummy-backend + +additionalProperties: false + +examples: + - | + abox_generic: abox_generic@generic { + compatible = "samsung,abox_generic"; + samsung,num-of-pcm_playback = <32>; + samsung,num-of-pcm_capture = <32>; + samsung,num-of-i2s-dummy-backend = <5>; + status = "disabled"; + #address-cells = <2>; + #size-cells = <1>; + ranges; + }; + -- 2.25.1

3 2

[tiwai-sound:test/hda-reorg 17/42] sound/soc/intel/atom/sst/sst_acpi.c:272: undefined reference to `snd_intel_acpi_dsp_driver_probe'
by kernel test robot 08 Jul '25

08 Jul '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/hda-reorg head: be6979113a769a90bab2a55a30ae83b76f5971e6 commit: d1bc64a913342041aaacdde0c0920eb0d5b96281 [17/42] ALSA: hda: Move HD-audio core stuff into sound/hda/core config: i386-randconfig-063-20250522 (https://download.01.org/0day-ci/archive/20250708/202507081351.iyDgGOUP-lkp@…) compiler: gcc-12 (Debian 12.2.0-14+deb12u1) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250708/202507081351.iyDgGOUP-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/202507081351.iyDgGOUP-lkp@intel.com/ All errors (new ones prefixed by >>): ld: sound/soc/intel/atom/sst/sst_acpi.o: in function `sst_acpi_probe': >> sound/soc/intel/atom/sst/sst_acpi.c:272: undefined reference to `snd_intel_acpi_dsp_driver_probe' vim +272 sound/soc/intel/atom/sst/sst_acpi.c 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 255 5129ad6e4e6999 sound/soc/intel/sst/sst_acpi.c Wei Yongjun 2014-12-09 256 static int sst_acpi_probe(struct platform_device *pdev) 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 257 { 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 258 struct device *dev = &pdev->dev; 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 259 int ret = 0; 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 260 struct intel_sst_drv *ctx; 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 261 const struct acpi_device_id *id; 7feb2f786a46d3 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2017-10-12 262 struct snd_soc_acpi_mach *mach; 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 263 struct platform_device *mdev; 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 264 struct platform_device *plat_dev; 12cc291b0b5850 sound/soc/intel/atom/sst/sst_acpi.c Vinod Koul 2015-11-05 265 struct sst_platform_info *pdata; 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 266 unsigned int dev_id; 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 267 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 268 id = acpi_match_device(dev->driver->acpi_match_table, dev); 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 269 if (!id) 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 270 return -ENODEV; df5f5edaef4b65 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2020-11-12 271 df5f5edaef4b65 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2020-11-12 @272 ret = snd_intel_acpi_dsp_driver_probe(dev, id->id); df5f5edaef4b65 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2020-11-12 273 if (ret != SND_INTEL_DSP_DRIVER_ANY && ret != SND_INTEL_DSP_DRIVER_SST) { df5f5edaef4b65 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2020-11-12 274 dev_dbg(dev, "SST ACPI driver not selected, aborting probe\n"); df5f5edaef4b65 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2020-11-12 275 return -ENODEV; df5f5edaef4b65 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2020-11-12 276 } df5f5edaef4b65 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2020-11-12 277 1635c694124fe6 sound/soc/intel/atom/sst/sst_acpi.c Colin Ian King 2016-09-16 278 dev_dbg(dev, "for %s\n", id->id); 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 279 7feb2f786a46d3 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2017-10-12 280 mach = (struct snd_soc_acpi_mach *)id->driver_data; 7feb2f786a46d3 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2017-10-12 281 mach = snd_soc_acpi_find_machine(mach); 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 282 if (mach == NULL) { 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 283 dev_err(dev, "No matching machine driver found\n"); 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 284 return -ENODEV; 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 285 } 73a33f6f6d44db sound/soc/intel/atom/sst/sst_acpi.c Vinod Koul 2016-07-08 286 536cfd2f375d36 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2019-05-30 287 if (soc_intel_is_byt()) 6004640638a2cb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2017-10-12 288 mach->pdata = &byt_rvp_platform_data; 6004640638a2cb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2017-10-12 289 else 6004640638a2cb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2017-10-12 290 mach->pdata = &chv_platform_data; 12cc291b0b5850 sound/soc/intel/atom/sst/sst_acpi.c Vinod Koul 2015-11-05 291 pdata = mach->pdata; 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 292 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 293 if (!strcmp(id->id, "LPE0F28")) { 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 294 struct resource *rsrc; 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 295 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 296 /* Use regular BYT SST PCI VID:PID */ 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 297 dev_id = 0x80860F28; 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 298 byt_rvp_platform_data.res_info = &lpe8086_res_info; 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 299 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 300 /* 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 301 * The "LPE0F28" ACPI device has separate IO-mem resources for: 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 302 * DDR, SHIM, MBOX, IRAM, DRAM, CFG 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 303 * None of which covers the entire LPE base address range. 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 304 * lpe8086_res_info.acpi_lpe_res_index points to the SHIM. 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 305 * Patch this to cover the entire base address range as expected 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 306 * by sst_platform_get_resources(). 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 307 */ 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 308 rsrc = platform_get_resource(pdev, IORESOURCE_MEM, 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 309 pdata->res_info->acpi_lpe_res_index); 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 310 if (!rsrc) { c1895ba181e560 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-26 311 dev_err(dev, "Invalid SHIM base\n"); 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 312 return -EIO; 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 313 } 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 314 rsrc->start -= pdata->res_info->shim_offset; 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 315 rsrc->end = rsrc->start + 0x200000 - 1; 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 316 } else { 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 317 ret = kstrtouint(id->id, 16, &dev_id); 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 318 if (ret < 0) { 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 319 dev_err(dev, "Unique device id conversion error: %d\n", ret); 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 320 return ret; 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 321 } 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 322 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 323 if (soc_intel_is_byt_cr(pdev)) 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 324 byt_rvp_platform_data.res_info = &bytcr_res_info; 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 325 } 6668610b4d8ce9 sound/soc/intel/atom/sst/sst_acpi.c Hans de Goede 2024-10-25 326 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 327 dev_dbg(dev, "ACPI device id: %x\n", dev_id); 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 328 a68bc0d43e1b96 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-08-12 329 ret = sst_alloc_drv_context(&ctx, dev, dev_id); a68bc0d43e1b96 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-08-12 330 if (ret < 0) a68bc0d43e1b96 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-08-12 331 return ret; a68bc0d43e1b96 sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-08-12 332 3ee1cd4f81e15f sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2018-11-01 333 /* update machine parameters */ 3ee1cd4f81e15f sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2018-11-01 334 mach->mach_params.acpi_ipc_irq_index = 3ee1cd4f81e15f sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2018-11-01 335 pdata->res_info->acpi_ipc_irq_index; 3ee1cd4f81e15f sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2018-11-01 336 caf94ed8629afb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-01-04 337 plat_dev = platform_device_register_data(dev, pdata->platform, -1, caf94ed8629afb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-01-04 338 NULL, 0); f5d40b400fe2de sound/soc/intel/sst/sst_acpi.c Wei Yongjun 2014-12-09 339 if (IS_ERR(plat_dev)) { caf94ed8629afb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-01-04 340 dev_err(dev, "Failed to create machine device: %s\n", caf94ed8629afb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-01-04 341 pdata->platform); f5d40b400fe2de sound/soc/intel/sst/sst_acpi.c Wei Yongjun 2014-12-09 342 return PTR_ERR(plat_dev); 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 343 } 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 344 caf94ed8629afb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-01-04 345 /* caf94ed8629afb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-01-04 346 * Create platform device for sst machine driver, caf94ed8629afb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-01-04 347 * pass machine info as pdata caf94ed8629afb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-01-04 348 */ caf94ed8629afb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-01-04 349 mdev = platform_device_register_data(dev, mach->drv_name, -1, caf94ed8629afb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-01-04 350 (const void *)mach, sizeof(*mach)); f5d40b400fe2de sound/soc/intel/sst/sst_acpi.c Wei Yongjun 2014-12-09 351 if (IS_ERR(mdev)) { caf94ed8629afb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-01-04 352 dev_err(dev, "Failed to create machine device: %s\n", caf94ed8629afb sound/soc/intel/atom/sst/sst_acpi.c Pierre-Louis Bossart 2016-01-04 353 mach->drv_name); f5d40b400fe2de sound/soc/intel/sst/sst_acpi.c Wei Yongjun 2014-12-09 354 return PTR_ERR(mdev); 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 355 } 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 356 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 357 /* Fill sst platform data */ 12cc291b0b5850 sound/soc/intel/atom/sst/sst_acpi.c Vinod Koul 2015-11-05 358 ctx->pdata = pdata; 12cc291b0b5850 sound/soc/intel/atom/sst/sst_acpi.c Vinod Koul 2015-11-05 359 strcpy(ctx->firmware_name, mach->fw_filename); 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 360 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 361 ret = sst_platform_get_resources(ctx); 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 362 if (ret) 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 363 return ret; 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 364 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 365 ret = sst_context_init(ctx); 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 366 if (ret < 0) 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 367 return ret; 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 368 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 369 sst_configure_runtime_pm(ctx); 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 370 platform_set_drvdata(pdev, ctx); 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 371 return ret; 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 372 } 336cfbb05edf7b sound/soc/intel/sst/sst_acpi.c Vinod Koul 2014-11-11 373 :::::: The code at line 272 was first introduced by commit :::::: df5f5edaef4b653fa731dcf3753e71766f95c2cd ASoC: Intel: Atom: add dynamic selection of DSP driver :::::: TO: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH v2] ALSA: hda/tas2781: Fix calibration data parser issue
by Baojun Xu 07 Jul '25

07 Jul '25

We will copy calibration data from position behind to front. We has created a variable (tmp_val) point on top of calibration data buffer, and tmp_val[1] is max of node number in original calibration data structure, it will be overwritten after first data copy, so can't be used as max node number check in for loop. So we create a new variable to save max of node number (tmp_val[1]), used to check if max node number was reached in for loop. And a point need to be increased to point at calibration data in node. Data saved position also need to be increased one byte. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- Change in v2: - Add more description about this fix. --- sound/pci/hda/tas2781_hda.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/tas2781_hda.c b/sound/pci/hda/tas2781_hda.c index 5f1d4b3e9688..34217ce9f28e 100644 --- a/sound/pci/hda/tas2781_hda.c +++ b/sound/pci/hda/tas2781_hda.c @@ -44,7 +44,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) TASDEVICE_REG(0, 0x13, 0x70), TASDEVICE_REG(0, 0x18, 0x7c), }; - unsigned int crc, oft; + unsigned int crc, oft, node_num; unsigned char *buf; int i, j, k, l; @@ -80,8 +80,9 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) dev_err(p->dev, "%s: CRC error\n", __func__); return; } + node_num = tmp_val[1]; - for (j = 0, k = 0; j < tmp_val[1]; j++) { + for (j = 0, k = 0; j < node_num; j++) { oft = j * 6 + 3; if (tmp_val[oft] == TASDEV_UEFI_CALI_REG_ADDR_FLG) { for (i = 0; i < TASDEV_CALIB_N; i++) { @@ -99,8 +100,9 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) } data[l] = k; + oft++; for (i = 0; i < TASDEV_CALIB_N * 4; i++) - data[l + i] = data[4 * oft + i]; + data[l + i + 1] = data[4 * oft + i]; k++; } } -- 2.43.0

1 0

[PATCH v1] ALSA: hda/tas2781: Add bus name in device name check
by Baojun Xu 06 Jul '25

06 Jul '25

Device name start from bus name, as we use strstarts() to do compare, need add it for TXNW2781 device. Fixes: b2904df0a347 ("ALSA: hda/tas2781: Add compatible for hardware id TIAS2781 and TXNW2781") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- sound/pci/hda/tas2781_hda_i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index b9cdbca951e4..530c2266ab3b 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -588,7 +588,7 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; } else if (strstarts(dev_name(&clt->dev), - "TXNW2781:00-tas2781-hda.0")) { + "i2c-TXNW2781:00-tas2781-hda.0")) { device_name = "TXNW2781"; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; -- 2.43.0

1 1

[PATCH v1] ALSA: hda/tas2781: Fix calibration data parser issue
by Baojun Xu 06 Jul '25

06 Jul '25

Calibration data was overwritten during parser, it cause issue. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- sound/pci/hda/tas2781_hda.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/tas2781_hda.c b/sound/pci/hda/tas2781_hda.c index 5f1d4b3e9688..34217ce9f28e 100644 --- a/sound/pci/hda/tas2781_hda.c +++ b/sound/pci/hda/tas2781_hda.c @@ -44,7 +44,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) TASDEVICE_REG(0, 0x13, 0x70), TASDEVICE_REG(0, 0x18, 0x7c), }; - unsigned int crc, oft; + unsigned int crc, oft, node_num; unsigned char *buf; int i, j, k, l; @@ -80,8 +80,9 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) dev_err(p->dev, "%s: CRC error\n", __func__); return; } + node_num = tmp_val[1]; - for (j = 0, k = 0; j < tmp_val[1]; j++) { + for (j = 0, k = 0; j < node_num; j++) { oft = j * 6 + 3; if (tmp_val[oft] == TASDEV_UEFI_CALI_REG_ADDR_FLG) { for (i = 0; i < TASDEV_CALIB_N; i++) { @@ -99,8 +100,9 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) } data[l] = k; + oft++; for (i = 0; i < TASDEV_CALIB_N * 4; i++) - data[l + i] = data[4 * oft + i]; + data[l + i + 1] = data[4 * oft + i]; k++; } } -- 2.43.0

1 0

test: update midifile library to ANSI C
by GitHub pull_request - opened 06 Jul '25

06 Jul '25

alsa-project/alsa-lib pull request #463 was opened from atsampson: The `midifile` library used by the `playmidi1` program dates from 1989, and used pre-ANSI function definitions and prototypes. GCC 15 now defaults to C23 where `()` means the same as `(void)` in prototypes, which causes type mismatch errors. Update the code to use ANSI function definitions and prototypes, so it'll compile happily as anything from ANSI C to C23. This revealed that `playmidi1`'s `do_tempo` had the wrong argument type, so correct that as well. Fixes #452. Request URL : https://github.com/alsa-project/alsa-lib/pull/463 Patch URL : https://github.com/alsa-project/alsa-lib/pull/463.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[tiwai-sound:test/hda-reorg 41/53] sound/hda/codecs/cirrus/cs421x.c:561:35: warning: 'cs421x_codec_ops' defined but not used
by kernel test robot 05 Jul '25

05 Jul '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/hda-reorg head: 6119728edf86cefcbf7b20e000414c85916e900d commit: 484072d3f99c402030fcee94e28ab3ea76d04247 [41/53] ALSA: hda/cirrus: Split to cs420x and cs421x drivers config: sh-allmodconfig (https://download.01.org/0day-ci/archive/20250705/202507051737.DMAkxayN-lkp@…) compiler: sh4-linux-gcc (GCC) 15.1.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250705/202507051737.DMAkxayN-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/202507051737.DMAkxayN-lkp@intel.com/ All warnings (new ones prefixed by >>): >> sound/hda/codecs/cirrus/cs421x.c:561:35: warning: 'cs421x_codec_ops' defined but not used [-Wunused-const-variable=] 561 | static const struct hda_codec_ops cs421x_codec_ops = { | ^~~~~~~~~~~~~~~~ vim +/cs421x_codec_ops +561 sound/hda/codecs/cirrus/cs421x.c 560 > 561 static const struct hda_codec_ops cs421x_codec_ops = { 562 .probe = cs421x_probe, 563 .remove = snd_hda_gen_remove, 564 .build_controls = snd_hda_gen_build_controls, 565 .build_pcms = snd_hda_gen_build_pcms, 566 .init = cs421x_init, 567 .unsol_event = snd_hda_jack_unsol_event, 568 .suspend = cs421x_suspend, 569 .stream_pm = snd_hda_gen_stream_pm, 570 }; 571 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[tiwai-sound:test/hda-reorg 29/53] sound/hda/common/hwdep.c:26:6: warning: label followed by a declaration is a C23 extension
by kernel test robot 05 Jul '25

05 Jul '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/hda-reorg head: 6119728edf86cefcbf7b20e000414c85916e900d commit: 5b965be899c8e3286f5111b7e90207afe5124b67 [29/53] ALSA: hda: Move common codec driver into sound/hda/common directory config: riscv-randconfig-001-20250705 (https://download.01.org/0day-ci/archive/20250705/202507051454.VintnWOF-lkp@…) compiler: clang version 21.0.0git (https://github.com/llvm/llvm-project 61529d9e36fa86782a2458e6bdeedf7f376ef4b5) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250705/202507051454.VintnWOF-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/202507051454.VintnWOF-lkp@intel.com/ All warnings (new ones prefixed by >>): In file included from sound/hda/common/hwdep.c:10: In file included from include/linux/compat.h:17: include/linux/fs.h:3979:15: warning: label followed by a declaration is a C23 extension [-Wc23-extensions] 3979 | if (unlikely(get_user(c, path))) | ^ arch/riscv/include/asm/uaccess.h:274:3: note: expanded from macro 'get_user' 274 | __get_user((x), __p) : \ | ^ arch/riscv/include/asm/uaccess.h:244:2: note: expanded from macro '__get_user' 244 | __get_user_error(__gu_val, __gu_ptr, __gu_err); \ | ^ arch/riscv/include/asm/uaccess.h:207:2: note: expanded from macro '__get_user_error' 207 | __get_user_nocheck(x, ptr, __gu_failed); \ | ^ arch/riscv/include/asm/uaccess.h:196:3: note: expanded from macro '__get_user_nocheck' 196 | __get_user_8((x), __gu_ptr, label); \ | ^ arch/riscv/include/asm/uaccess.h:130:2: note: expanded from macro '__get_user_8' 130 | u32 __user *__ptr = (u32 __user *)(ptr); \ | ^ >> sound/hda/common/hwdep.c:26:6: warning: label followed by a declaration is a C23 extension [-Wc23-extensions] 26 | if (get_user(verb, &arg->verb)) | ^ arch/riscv/include/asm/uaccess.h:274:3: note: expanded from macro 'get_user' 274 | __get_user((x), __p) : \ | ^ arch/riscv/include/asm/uaccess.h:244:2: note: expanded from macro '__get_user' 244 | __get_user_error(__gu_val, __gu_ptr, __gu_err); \ | ^ arch/riscv/include/asm/uaccess.h:207:2: note: expanded from macro '__get_user_error' 207 | __get_user_nocheck(x, ptr, __gu_failed); \ | ^ arch/riscv/include/asm/uaccess.h:196:3: note: expanded from macro '__get_user_nocheck' 196 | __get_user_8((x), __gu_ptr, label); \ | ^ arch/riscv/include/asm/uaccess.h:130:2: note: expanded from macro '__get_user_8' 130 | u32 __user *__ptr = (u32 __user *)(ptr); \ | ^ sound/hda/common/hwdep.c:40:6: warning: label followed by a declaration is a C23 extension [-Wc23-extensions] 40 | if (get_user(verb, &arg->verb)) | ^ arch/riscv/include/asm/uaccess.h:274:3: note: expanded from macro 'get_user' 274 | __get_user((x), __p) : \ | ^ arch/riscv/include/asm/uaccess.h:244:2: note: expanded from macro '__get_user' 244 | __get_user_error(__gu_val, __gu_ptr, __gu_err); \ | ^ arch/riscv/include/asm/uaccess.h:207:2: note: expanded from macro '__get_user_error' 207 | __get_user_nocheck(x, ptr, __gu_failed); \ | ^ arch/riscv/include/asm/uaccess.h:196:3: note: expanded from macro '__get_user_nocheck' 196 | __get_user_8((x), __gu_ptr, label); \ | ^ arch/riscv/include/asm/uaccess.h:130:2: note: expanded from macro '__get_user_8' 130 | u32 __user *__ptr = (u32 __user *)(ptr); \ | ^ 3 warnings generated. vim +26 sound/hda/common/hwdep.c 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 17 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 18 /* 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 19 * write/read an out-of-bound verb 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 20 */ 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 21 static int verb_write_ioctl(struct hda_codec *codec, 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 22 struct hda_verb_ioctl __user *arg) 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 23 { 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 24 u32 verb, res; 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 25 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 @26 if (get_user(verb, &arg->verb)) 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 27 return -EFAULT; 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 28 res = snd_hda_codec_read(codec, verb >> 24, 0, 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 29 (verb >> 8) & 0xffff, verb & 0xff); 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 30 if (put_user(res, &arg->res)) 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 31 return -EFAULT; 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 32 return 0; 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 33 } 2807314d467e7d sound/pci/hda/hda_hwdep.c Takashi Iwai 2007-07-27 34 :::::: The code at line 26 was first introduced by commit :::::: 2807314d467e7dd929c42050031aabbd28e78f0b [ALSA] hda-intel - Add hwdep interface :::::: TO: Takashi Iwai <tiwai(a)suse.de> :::::: CC: Jaroslav Kysela <perex(a)perex.cz> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[GIT PULL] ASoC fixes for v6.16-rc4
by Mark Brown 04 Jul '25

04 Jul '25

The following changes since commit d0b3b7b22dfa1f4b515fd3a295b3fd958f9e81af: Linux 6.16-rc4 (2025-06-29 13:09:04 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.16-rc4 for you to fetch changes up to 3b3312f28ee2d9c386602f8521e419cfc69f4823: ASoC: cs35l56: probe() should fail if the device ID is not recognized (2025-07-03 12:22:27 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.16 An update for the MAINTAINERS file, plus a number of small driver specific fixes and device quirks. ---------------------------------------------------------------- Arun Raghavan (1): ASoC: fsl_sai: Force a software reset when starting in consumer mode Bard Liao (1): ASoC: Intel: SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_ACPI_INTEL_MATCH Charles Keepax (1): ASoC: Intel: soc-acpi: arl: Correct order of cs42l43 matches Richard Fitzgerald (1): ASoC: cs35l56: probe() should fail if the device ID is not recognized Shengjiu Wang (1): ASoC: fsl_asrc: use internal measured ratio for non-ideal ratio mode Simon Trimmer (1): ASoC: Intel: sof_sdw: Add quirks for Lenovo P1 and P16 Srinivas Kandagatla (1): MAINTAINERS: update Qualcomm audio codec drivers list MAINTAINERS | 16 +++++----------- sound/soc/codecs/cs35l56-shared.c | 2 +- sound/soc/fsl/fsl_asrc.c | 3 ++- sound/soc/fsl/fsl_sai.c | 14 ++++++++------ sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_sdw.c | 3 +++ sound/soc/intel/common/soc-acpi-intel-arl-match.c | 12 ++++++------ 7 files changed, 26 insertions(+), 25 deletions(-)

2 1

[BUG] Mackie DLZ Creator XS – USB Audio: write error -77 after accepted hw_params
by Brian Knutsson 04 Jul '25

04 Jul '25

Hi ALSA devs, I'm reporting an issue with the Mackie DLZ Creator XS USB audio device on Linux (tested on Ubuntu 24.04 and 25.05 with kernel 6.8+). The device is detected correctly via ALSA (card 2: XS), but as soon as playback is initiated – whether via `aplay` or `speaker-test` – it disconnects with `write error: -77, File descriptor in bad state`. Relevant kernel logs show: usb 3-7: Product: DLZ Creator XS usb 3-7: Manufacturer: LOUD Audio, LLC usb 3-7: USB disconnect, device number X This occurs with and without PipeWire/udev running. I’ve created a full ALSA log with `alsa-info` and opened a Kernel Bugzilla entry: <https://bugzilla.kernel.org/show_bug.cgi?id=220143> Is there any known quirk or patch needed for this hardware? Or a way to assist debugging further (usbmon trace etc.)? Thanks in advance, Brian Knutsson

2 1

[PATCH v1] ALSA: hda/tas2781: Add bus name in device name check
by Baojun Xu 03 Jul '25

03 Jul '25

Device name start from bus name, as we use strstarts() to do compare, need add it for TXNW2781 device. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- sound/pci/hda/tas2781_hda_i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index b9cdbca95..530c2266a 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -588,7 +588,7 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; } else if (strstarts(dev_name(&clt->dev), - "TXNW2781:00-tas2781-hda.0")) { + "i2c-TXNW2781:00-tas2781-hda.0")) { device_name = "TXNW2781"; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; -- 2.43.0

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[PATCH 0/3] Add legacy driver support for ACP7.2 based
by Venkata Prasad Potturu 02 Jul '25

02 Jul '25

This patch series is to add I2S and PDM support for ACP7.2 based platforms. Venkata Prasad Potturu (3): ASoC: amd: acp: Add legacy driver support acp7.2 based platforms ASoC: amd: acp: Enable I2S support for acp7.2 based platforms ASoC: amd: acp: Enable acp7.2 platform based DMIC support in machine driver sound/soc/amd/acp/acp-i2s.c | 3 +++ sound/soc/amd/acp/acp-legacy-common.c | 2 ++ sound/soc/amd/acp/acp-mach-common.c | 1 + sound/soc/amd/acp/acp-pci.c | 1 + sound/soc/amd/acp/acp-platform.c | 2 ++ sound/soc/amd/acp/acp70.c | 1 + sound/soc/amd/acp/acp_common.h | 1 + 7 files changed, 11 insertions(+) -- 2.43.0

2 4

New Card: ESI Juli@ eX
by GitHub issues - opened 01 Jul '25

01 Jul '25

alsa-project/alsa-lib issue #462 was opened from Carlisle96: I am not sure where to ask or how to report this: There was a new card released ESI-Audio Juli@ eX https://www.esi-audio.com/products/juliaex/ I am wondering if this is supported or planned to be supported. It is not yet listed here: https://www.alsa-project.org/wiki/Matrix:Vendor-ESI Issue URL : https://github.com/alsa-project/alsa-lib/issues/462 Repository URL: https://github.com/alsa-project/alsa-lib

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Request for help: Outputting audio as an I2S stream
by Steve Broshar 01 Jul '25

01 Jul '25

Summary: My goal is to output audio to an ADAU1467 via I2S on Linux 6.6. I tried and then abandoned using a custom, device-specific driver. Now trying to use built-in simple-sound-card. But, driver does not initialize (probe). Don't know how to debug/fix the issue. I've been working with chapgpt extensively since I have found no other source of information on how to setup linux to output an I2S audio stream. In fact, I posted a question to a linux maillist about an issue a couple weeks ago and the reply from Lars Clausen showed output from chatgpt. In the old days folks would joke about googling something for someone. Today, I guess we can do the same WRT chatgpt. But, that question was about updating a driver. Chatgpt tells me that I shouldn't need a device-specific driver since the device is configured and controlled elsewhere. I just need to pump an audio stream into it. So, I started over with my solution; no longer working on a custom driver. Apparently, I can use the simple-sound-card and a dummy codec. But, it does not work. Maybe chatgpt is wrong and I need a device-specific driver. Or maybe my setup is wrong. I have worked with chatgpt to resolve my issues, but have hit a wall. Hoping to get help from humans now. 😊 The boot-time log (attached) shows nothing about my audio setup until the last line: platform sound-i2s: deferred probe pending I think that ‘deferred probe’ means that an initial probe failed. How can I tell what failed? What prevented it from succeeding? Would be nice if the driver told me what failed. But, there’s no error message, just an info msg that probe is pending. Here's the device tree setup that I added (based on input from chatgpt): / { // Compton audio output can be accomplished via a built-in driver -- simple-audio-card sound_i2s: sound-i2s { // specify the built-in simple sound card compatible = "simple-audio-card"; // seems to be a descriptive name that shows in userland; i.e. via 'aplay -l' simple-audio-card,name = "imx8mn-adauslave"; // might specify I2S (as opposed to descriptive text) simple-audio-card,format = "i2s"; // defines CPU as master for bit-clock simple-audio-card,bitclock-master = <&sound_i2s_cpu>; // defines CPU as master for frame-clock simple-audio-card,frame-master = <&sound_i2s_cpu>; sound_i2s_cpu: simple-audio-card,cpu { sound-dai = <&sai3>; dai-tdm-slot-num = <2>; /* stereo */ dai-tdm-slot-width = <32>; /* bits per slot */ }; simple-audio-card,codec { sound-dai = <&dummy_codec>; }; }; // Compton audio output does not require a device-specific codec; the built-in, dummy driver is sufficient dummy_codec: dummy-codec { compatible = "linux,snd-soc-dummy"; #sound-dai-cells = <0>; }; Here's the device tree config that comes with the NXP IMX8M build for nano: pinctrl_sai3: sai3grp { fsl,pins = < MX8MN_IOMUXC_SAI3_TXFS_SAI3_TX_SYNC 0xd6 MX8MN_IOMUXC_SAI3_TXC_SAI3_TX_BCLK 0xd6 MX8MN_IOMUXC_SAI3_MCLK_SAI3_MCLK 0xd6 MX8MN_IOMUXC_SAI3_TXD_SAI3_TX_DATA0 0xd6 >; }; sai3: sai@30030000 { compatible = "fsl,imx8mn-sai", "fsl,imx8mq-sai"; reg = <0x30030000 0x10000>; #sound-dai-cells = <0>; interrupts = <GIC_SPI 50 IRQ_TYPE_LEVEL_HIGH>; clocks = <&clk IMX8MN_CLK_SAI3_IPG>, <&clk IMX8MN_CLK_DUMMY>, <&clk IMX8MN_CLK_SAI3_ROOT>, <&clk IMX8MN_CLK_DUMMY>, <&clk IMX8MN_CLK_DUMMY>; clock-names = "bus", "mclk0", "mclk1", "mclk2", "mclk3"; dmas = <&sdma2 4 2 0>, <&sdma2 5 2 0>; dma-names = "rx", "tx"; status = "disabled"; }; &sai3 { #sound-dai-cells = <0>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sai3>; assigned-clocks = <&clk IMX8MN_CLK_SAI3>; assigned-clock-parents = <&clk IMX8MN_AUDIO_PLL1_OUT>; assigned-clock-rates = <24576000>; fsl,sai-mclk-direction-output; status = "okay"; }; In summary, how do I further debug this? In general, and if you know, how does one output an audio stream as I2S? Steve Broshar Palmer Wireless Medtech

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[PATCH v1] ASoC: Intel: soc-acpi-intel-ptl-match: Add rt721_l0 support
by Ajye Huang 30 Jun '25

30 Jun '25

Patch adds driver data & match table for rt721 multi-function codec on PTL at sdw link0. Signed-off-by: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> --- .../intel/common/soc-acpi-intel-ptl-match.c | 24 +++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c index eae75f3f0fa4..0257e12e8b6e 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c @@ -330,6 +330,15 @@ static const struct snd_soc_acpi_adr_device rt1320_3_group1_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt721_0_single_adr[] = { + { + .adr = 0x000030025d072101ull, + .num_endpoints = ARRAY_SIZE(rt_mf_endpoints), + .endpoints = rt_mf_endpoints, + .name_prefix = "rt721" + } +}; + static const struct snd_soc_acpi_adr_device rt721_3_single_adr[] = { { .adr = 0x000330025d072101ull, @@ -339,6 +348,15 @@ static const struct snd_soc_acpi_adr_device rt721_3_single_adr[] = { } }; +static const struct snd_soc_acpi_link_adr ptl_rt721_l0[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt721_0_single_adr), + .adr_d = rt721_0_single_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr ptl_rt721_l3[] = { { .mask = BIT(3), @@ -664,6 +682,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_sdw_machines[] = { .sof_tplg_filename = "sof-ptl-rt712-l3-rt1320-l3.tplg", .get_function_tplg_files = sof_sdw_get_tplg_files, }, + { + .link_mask = BIT(0), + .links = ptl_rt721_l0, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-ptl-rt721.tplg", + }, { .link_mask = BIT(3), .links = ptl_rt721_l3, -- 2.25.1

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ucm2: Qualcomm: Add Dell Latitude 7455 / Inspiron 14 Plus 7441 support
by GitHub pull_request - edited 29 Jun '25

29 Jun '25

alsa-project/alsa-ucm-conf pull request #589 was edited from valpackett: (Both SKUs are identical, the codename is Thena) This configuration is: - 4 speakers - 2 DMICs - headset with mic - DisplayPort (not yet) --- audioreach: https://github.com/linux-msm/audioreach-topology/pull/25 cc @0xB0D @alexVinarskis Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/589 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/589.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: Qualcomm: Add Dell Latitude 7455 / Inspiron 14 Plus 7441 support
by GitHub pull_request - opened 29 Jun '25

29 Jun '25

alsa-project/alsa-ucm-conf pull request #589 was opened from valpackett: (Both SKUs are identical, the codename is Thena) This configuration is: - 4 speakers - 2 DMICs - headset with mic - DisplayPort (not yet) --- cc @0xB0D @alexVinarskis Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/589 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/589.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Focusrite Scarlett 18i20 Gen 1: UCM configuration defines incorrect capture channel count
by GitHub issues - opened 29 Jun '25

29 Jun '25

alsa-project/alsa-ucm-conf issue #588 was opened from mkuhl: ## Summary The ALSA UCM configuration for Focusrite Scarlett 18i20 incorrectly defines 20 capture channels (`Define.DirectCaptureChannels 20`), but the 1st generation hardware only supports 18 channels. This causes the device to fail initialization with UCM profile errors. ## Device Information - **Device**: Focusrite Scarlett 18i20 (1st Generation) - **USB ID**: 1235:800c - **USB Name**: Focusrite Scarlett 18i20 USB - **Driver**: snd_usb_audio ## Channel Configuration (Gen 1) - 8 analog inputs - 8 ADAT inputs - 2 S/PDIF inputs - **Total**: 18 input channels (not 20) ## Error Message ``` spa.alsa: Error in ALSA UCM profile for _ucm0007.hw:USB,0 (HiFi: Direct 2: source): CaptureChannels=20 > avail 18 spa.alsa: Errors in ALSA UCM profile for card Scarlett 18i20 ``` ## Affected Files The channel definition is in the main configuration file: `/usr/share/alsa/ucm2/USB-Audio/Focusrite/Scarlett-18i20.conf` Current (incorrect): ``` Define.DirectCaptureChannels 20 ``` Should be: ``` Define.DirectCaptureChannels 18 ``` This file then includes `Scarlett-18i20-HiFi.conf` which uses the defined value. ## System Information - **Distribution**: Fedora 42 / Aurora-dx - **Kernel**: 6.15.3-200.fc42.x86_64 - **PipeWire**: 1.4.5-1.fc42 - **WirePlumber**: 0.5.8-1.fc42 - **ALSA UCM**: 1.2.14-2.fc42 - **ALSA lib**: 1.2.14-2.fc42 ## Steps to Reproduce 1. Connect Focusrite Scarlett 18i20 Gen 1 (USB ID 1235:800c) 2. Start PipeWire/WirePlumber 3. Check device status: `wpctl status | grep Scarlett` 4. Device shows "[ALSA UCM error]" 5. Journal shows channel count mismatch error ## Workaround Users can create a local override: ```bash mkdir -p ~/.local/share/alsa/ucm2/USB-Audio/Focusrite/ cp /usr/share/alsa/ucm2/USB-Audio/Focusrite/Scarlett-18i20.conf \ ~/.local/share/alsa/ucm2/USB-Audio/Focusrite/ cp /usr/share/alsa/ucm2/USB-Audio/Focusrite/Scarlett-18i20-HiFi.conf \ ~/.local/share/alsa/ucm2/USB-Audio/Focusrite/ sed -i 's/Define.DirectCaptureChannels 20/Define.DirectCaptureChannels 18/g' \ ~/.local/share/alsa/ucm2/USB-Audio/Focusrite/Scarlett-18i20.conf ``` ## Root Cause Analysis The UCM configuration appears to be written for newer Scarlett 18i20 generations (2nd/3rd/4th Gen) which may have 20 channels due to additional loopback channels. The configuration includes generation detection logic but doesn't properly identify USB ID 1235:800c as a 1st Gen device, causing it to use the default 20-channel configuration. ## Proposed Solution The UCM configuration should detect Gen 1 devices and set the appropriate channel count: 1. Add detection for USB ID 1235:800c as Gen 1 2. Set `DirectCaptureChannels` to 18 for Gen 1 devices 3. Keep 20 channels for newer generations if applicable ## Impact - Device fails to initialize properly - Users cannot use the audio interface without manual workarounds - Affects all Linux distributions using this UCM configuration ## Additional Information - This is a regression - older ALSA versions worked correctly - The issue affects PipeWire/WirePlumber but not JACK when using Pro Audio profile (which bypasses UCM) - Other Gen 1 Scarlett devices (6i6, 8i8) may have similar issues ## References - Device specifications: https://support.focusrite.com/hc/en-gb/articles/206849839-Scarlett-1st-Gen-… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/588 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

tascam us 322 / 366
by GitHub issues - edited 29 Jun '25

29 Jun '25

alsa-project/alsa-firmware issue #12 was edited from riramau: Can someone add support for tascam 322 / 366 ? Issue URL : https://github.com/alsa-project/alsa-firmware/issues/12 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

tascam us 322 / 366
by GitHub issues - opened 29 Jun '25

29 Jun '25

alsa-project/alsa-firmware issue #12 was opened from riramau: Can someone add support of tascam 322 / 366 ? Issue URL : https://github.com/alsa-project/alsa-firmware/issues/12 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

[PATCH v2 0/4] ASoc: tac5x1x: mixer-test report
by Niranjan H Y 28 Jun '25

28 Jun '25

mixer-test report: TAP version 13 # Card 0/titac5x1xsoundc - ti,tac5x1x-soundcard (ti,tac5x1x-soundcard) 1..553 ok 1 get_value.titac5x1xsoundc.78 # titac5x1xsoundc.78 ASI_TX_CH4_EN Capture Switch ok 2 name.titac5x1xsoundc.78 ok 3 write_default.titac5x1xsoundc.78 ok 4 write_valid.titac5x1xsoundc.78 ok 5 write_invalid.titac5x1xsoundc.78 ok 6 event_missing.titac5x1xsoundc.78 ok 7 event_spurious.titac5x1xsoundc.78 ok 8 get_value.titac5x1xsoundc.77 # titac5x1xsoundc.77 ASI_TX_CH3_EN Capture Switch ok 9 name.titac5x1xsoundc.77 ok 10 write_default.titac5x1xsoundc.77 ok 11 write_valid.titac5x1xsoundc.77 ok 12 write_invalid.titac5x1xsoundc.77 ok 13 event_missing.titac5x1xsoundc.77 ok 14 event_spurious.titac5x1xsoundc.77 ok 15 get_value.titac5x1xsoundc.76 # titac5x1xsoundc.76 PDM ch3 & ch4 Datain Select ok 16 name.titac5x1xsoundc.76 ok 17 write_default.titac5x1xsoundc.76 ok 18 write_valid.titac5x1xsoundc.76 ok 19 write_invalid.titac5x1xsoundc.76 ok 20 event_missing.titac5x1xsoundc.76 ok 21 event_spurious.titac5x1xsoundc.76 ok 22 get_value.titac5x1xsoundc.75 # titac5x1xsoundc.75 PDM ch1 & ch2 Datain Select ok 23 name.titac5x1xsoundc.75 ok 24 write_default.titac5x1xsoundc.75 ok 25 write_valid.titac5x1xsoundc.75 ok 26 write_invalid.titac5x1xsoundc.75 ok 27 event_missing.titac5x1xsoundc.75 ok 28 event_spurious.titac5x1xsoundc.75 ok 29 get_value.titac5x1xsoundc.74 # titac5x1xsoundc.74 IN2 Source Mux ok 30 name.titac5x1xsoundc.74 ok 31 write_default.titac5x1xsoundc.74 ok 32 write_valid.titac5x1xsoundc.74 ok 33 write_invalid.titac5x1xsoundc.74 ok 34 event_missing.titac5x1xsoundc.74 ok 35 event_spurious.titac5x1xsoundc.74 ok 36 get_value.titac5x1xsoundc.73 # titac5x1xsoundc.73 IN1 Source Mux ok 37 name.titac5x1xsoundc.73 ok 38 write_default.titac5x1xsoundc.73 ok 39 write_valid.titac5x1xsoundc.73 ok 40 write_invalid.titac5x1xsoundc.73 ok 41 event_missing.titac5x1xsoundc.73 ok 42 event_spurious.titac5x1xsoundc.73 ok 43 get_value.titac5x1xsoundc.72 # titac5x1xsoundc.72 ADC2 Config ok 44 name.titac5x1xsoundc.72 ok 45 write_default.titac5x1xsoundc.72 ok 46 write_valid.titac5x1xsoundc.72 ok 47 write_invalid.titac5x1xsoundc.72 ok 48 event_missing.titac5x1xsoundc.72 ok 49 event_spurious.titac5x1xsoundc.72 ok 50 get_value.titac5x1xsoundc.71 # titac5x1xsoundc.71 ADC2 Full-Scale ok 51 name.titac5x1xsoundc.71 ok 52 write_default.titac5x1xsoundc.71 ok 53 write_valid.titac5x1xsoundc.71 ok 54 write_invalid.titac5x1xsoundc.71 ok 55 event_missing.titac5x1xsoundc.71 ok 56 event_spurious.titac5x1xsoundc.71 ok 57 get_value.titac5x1xsoundc.70 # titac5x1xsoundc.70 ASI_RX_CH2_EN Switch ok 58 name.titac5x1xsoundc.70 ok 59 write_default.titac5x1xsoundc.70 ok 60 write_valid.titac5x1xsoundc.70 ok 61 write_invalid.titac5x1xsoundc.70 ok 62 event_missing.titac5x1xsoundc.70 ok 63 event_spurious.titac5x1xsoundc.70 ok 64 get_value.titac5x1xsoundc.69 # titac5x1xsoundc.69 OUT2x Driver ok 65 name.titac5x1xsoundc.69 ok 66 write_default.titac5x1xsoundc.69 ok 67 write_valid.titac5x1xsoundc.69 ok 68 write_invalid.titac5x1xsoundc.69 ok 69 event_missing.titac5x1xsoundc.69 ok 70 event_spurious.titac5x1xsoundc.69 ok 71 get_value.titac5x1xsoundc.68 # titac5x1xsoundc.68 OUT2x Config ok 72 name.titac5x1xsoundc.68 ok 73 write_default.titac5x1xsoundc.68 ok 74 write_valid.titac5x1xsoundc.68 ok 75 write_invalid.titac5x1xsoundc.68 ok 76 event_missing.titac5x1xsoundc.68 ok 77 event_spurious.titac5x1xsoundc.68 ok 78 get_value.titac5x1xsoundc.67 # titac5x1xsoundc.67 OUT2x Source ok 79 name.titac5x1xsoundc.67 ok 80 write_default.titac5x1xsoundc.67 ok 81 write_valid.titac5x1xsoundc.67 ok 82 write_invalid.titac5x1xsoundc.67 ok 83 event_missing.titac5x1xsoundc.67 ok 84 event_spurious.titac5x1xsoundc.67 ok 85 get_value.titac5x1xsoundc.66 # titac5x1xsoundc.66 ASI_RX_CH1_EN Switch ok 86 name.titac5x1xsoundc.66 ok 87 write_default.titac5x1xsoundc.66 ok 88 write_valid.titac5x1xsoundc.66 ok 89 write_invalid.titac5x1xsoundc.66 ok 90 event_missing.titac5x1xsoundc.66 ok 91 event_spurious.titac5x1xsoundc.66 ok 92 get_value.titac5x1xsoundc.65 # titac5x1xsoundc.65 OUT1x Driver ok 93 name.titac5x1xsoundc.65 ok 94 write_default.titac5x1xsoundc.65 ok 95 write_valid.titac5x1xsoundc.65 ok 96 write_invalid.titac5x1xsoundc.65 ok 97 event_missing.titac5x1xsoundc.65 ok 98 event_spurious.titac5x1xsoundc.65 ok 99 get_value.titac5x1xsoundc.64 # titac5x1xsoundc.64 OUT1x Config ok 100 name.titac5x1xsoundc.64 ok 101 write_default.titac5x1xsoundc.64 ok 102 write_valid.titac5x1xsoundc.64 ok 103 write_invalid.titac5x1xsoundc.64 ok 104 event_missing.titac5x1xsoundc.64 ok 105 event_spurious.titac5x1xsoundc.64 ok 106 get_value.titac5x1xsoundc.63 # titac5x1xsoundc.63 OUT1x Source ok 107 name.titac5x1xsoundc.63 ok 108 write_default.titac5x1xsoundc.63 ok 109 write_valid.titac5x1xsoundc.63 ok 110 write_invalid.titac5x1xsoundc.63 ok 111 event_missing.titac5x1xsoundc.63 ok 112 event_spurious.titac5x1xsoundc.63 ok 113 get_value.titac5x1xsoundc.62 # titac5x1xsoundc.62 ASI_TX_CH2_EN Capture Switch ok 114 name.titac5x1xsoundc.62 ok 115 write_default.titac5x1xsoundc.62 ok 116 write_valid.titac5x1xsoundc.62 ok 117 write_invalid.titac5x1xsoundc.62 ok 118 event_missing.titac5x1xsoundc.62 ok 119 event_spurious.titac5x1xsoundc.62 ok 120 get_value.titac5x1xsoundc.61 # titac5x1xsoundc.61 ASI_TX_CH1_EN Capture Switch ok 121 name.titac5x1xsoundc.61 ok 122 write_default.titac5x1xsoundc.61 ok 123 write_valid.titac5x1xsoundc.61 ok 124 write_invalid.titac5x1xsoundc.61 ok 125 event_missing.titac5x1xsoundc.61 ok 126 event_spurious.titac5x1xsoundc.61 ok 127 get_value.titac5x1xsoundc.60 # titac5x1xsoundc.60 ADC1 Config ok 128 name.titac5x1xsoundc.60 ok 129 write_default.titac5x1xsoundc.60 ok 130 write_valid.titac5x1xsoundc.60 ok 131 write_invalid.titac5x1xsoundc.60 ok 132 event_missing.titac5x1xsoundc.60 ok 133 event_spurious.titac5x1xsoundc.60 ok 134 get_value.titac5x1xsoundc.59 # titac5x1xsoundc.59 ADC1 Full-Scale ok 135 name.titac5x1xsoundc.59 ok 136 write_default.titac5x1xsoundc.59 ok 137 write_valid.titac5x1xsoundc.59 ok 138 write_invalid.titac5x1xsoundc.59 ok 139 event_missing.titac5x1xsoundc.59 ok 140 event_spurious.titac5x1xsoundc.59 ok 141 get_value.titac5x1xsoundc.58 # titac5x1xsoundc.58 IN_CH4_EN Capture Switch ok 142 name.titac5x1xsoundc.58 ok 143 write_default.titac5x1xsoundc.58 ok 144 write_valid.titac5x1xsoundc.58 ok 145 write_invalid.titac5x1xsoundc.58 ok 146 event_missing.titac5x1xsoundc.58 ok 147 event_spurious.titac5x1xsoundc.58 ok 148 get_value.titac5x1xsoundc.57 # titac5x1xsoundc.57 IN_CH3_EN Capture Switch ok 149 name.titac5x1xsoundc.57 ok 150 write_default.titac5x1xsoundc.57 ok 151 write_valid.titac5x1xsoundc.57 ok 152 write_invalid.titac5x1xsoundc.57 ok 153 event_missing.titac5x1xsoundc.57 ok 154 event_spurious.titac5x1xsoundc.57 ok 155 get_value.titac5x1xsoundc.56 # titac5x1xsoundc.56 IN_CH2_EN Capture Switch ok 156 name.titac5x1xsoundc.56 ok 157 write_default.titac5x1xsoundc.56 ok 158 write_valid.titac5x1xsoundc.56 ok 159 write_invalid.titac5x1xsoundc.56 ok 160 event_missing.titac5x1xsoundc.56 ok 161 event_spurious.titac5x1xsoundc.56 ok 162 get_value.titac5x1xsoundc.55 # titac5x1xsoundc.55 IN_CH1_EN Capture Switch ok 163 name.titac5x1xsoundc.55 ok 164 write_default.titac5x1xsoundc.55 ok 165 write_valid.titac5x1xsoundc.55 ok 166 write_invalid.titac5x1xsoundc.55 ok 167 event_missing.titac5x1xsoundc.55 ok 168 event_spurious.titac5x1xsoundc.55 ok 169 get_value.titac5x1xsoundc.54 # titac5x1xsoundc.54 ASI_TX_CH8_CFG ok 170 name.titac5x1xsoundc.54 ok 171 write_default.titac5x1xsoundc.54 ok 172 write_valid.titac5x1xsoundc.54 ok 173 write_invalid.titac5x1xsoundc.54 ok 174 event_missing.titac5x1xsoundc.54 ok 175 event_spurious.titac5x1xsoundc.54 ok 176 get_value.titac5x1xsoundc.53 # titac5x1xsoundc.53 ASI_TX_CH7_CFG ok 177 name.titac5x1xsoundc.53 ok 178 write_default.titac5x1xsoundc.53 ok 179 write_valid.titac5x1xsoundc.53 ok 180 write_invalid.titac5x1xsoundc.53 ok 181 event_missing.titac5x1xsoundc.53 ok 182 event_spurious.titac5x1xsoundc.53 ok 183 get_value.titac5x1xsoundc.52 # titac5x1xsoundc.52 ASI_TX_CH6_CFG ok 184 name.titac5x1xsoundc.52 ok 185 write_default.titac5x1xsoundc.52 ok 186 write_valid.titac5x1xsoundc.52 ok 187 write_invalid.titac5x1xsoundc.52 ok 188 event_missing.titac5x1xsoundc.52 ok 189 event_spurious.titac5x1xsoundc.52 ok 190 get_value.titac5x1xsoundc.51 # titac5x1xsoundc.51 ASI_TX_CH5_CFG ok 191 name.titac5x1xsoundc.51 ok 192 write_default.titac5x1xsoundc.51 ok 193 write_valid.titac5x1xsoundc.51 ok 194 write_invalid.titac5x1xsoundc.51 ok 195 event_missing.titac5x1xsoundc.51 ok 196 event_spurious.titac5x1xsoundc.51 ok 197 get_value.titac5x1xsoundc.50 # titac5x1xsoundc.50 ADC1 Phase Capture Volume ok 198 name.titac5x1xsoundc.50 ok 199 write_default.titac5x1xsoundc.50 ok 200 write_valid.titac5x1xsoundc.50 ok 201 write_invalid.titac5x1xsoundc.50 ok 202 event_missing.titac5x1xsoundc.50 ok 203 event_spurious.titac5x1xsoundc.50 ok 204 get_value.titac5x1xsoundc.49 # titac5x1xsoundc.49 ADC1 Fine Capture Volume ok 205 name.titac5x1xsoundc.49 ok 206 write_default.titac5x1xsoundc.49 ok 207 write_valid.titac5x1xsoundc.49 ok 208 write_invalid.titac5x1xsoundc.49 ok 209 event_missing.titac5x1xsoundc.49 ok 210 event_spurious.titac5x1xsoundc.49 ok 211 get_value.titac5x1xsoundc.48 # titac5x1xsoundc.48 ADC1 Digital Capture Volume ok 212 name.titac5x1xsoundc.48 ok 213 write_default.titac5x1xsoundc.48 ok 214 write_valid.titac5x1xsoundc.48 ok 215 write_invalid.titac5x1xsoundc.48 ok 216 event_missing.titac5x1xsoundc.48 ok 217 event_spurious.titac5x1xsoundc.48 ok 218 get_value.titac5x1xsoundc.47 # titac5x1xsoundc.47 ADC1 Audio BW ok 219 name.titac5x1xsoundc.47 ok 220 write_default.titac5x1xsoundc.47 ok 221 write_valid.titac5x1xsoundc.47 ok 222 write_invalid.titac5x1xsoundc.47 ok 223 event_missing.titac5x1xsoundc.47 ok 224 event_spurious.titac5x1xsoundc.47 ok 225 get_value.titac5x1xsoundc.46 # titac5x1xsoundc.46 Record Decimation Filter ok 226 name.titac5x1xsoundc.46 ok 227 write_default.titac5x1xsoundc.46 ok 228 write_valid.titac5x1xsoundc.46 ok 229 write_invalid.titac5x1xsoundc.46 ok 230 event_missing.titac5x1xsoundc.46 ok 231 event_spurious.titac5x1xsoundc.46 ok 232 get_value.titac5x1xsoundc.45 # titac5x1xsoundc.45 GPI1 GPI ok 233 name.titac5x1xsoundc.45 # GPI1 GPI is not writeable ok 234 # SKIP write_default.titac5x1xsoundc.45 # GPI1 GPI is not writeable ok 235 # SKIP write_valid.titac5x1xsoundc.45 # GPI1 GPI is not writeable ok 236 # SKIP write_invalid.titac5x1xsoundc.45 ok 237 event_missing.titac5x1xsoundc.45 ok 238 event_spurious.titac5x1xsoundc.45 ok 239 get_value.titac5x1xsoundc.44 # titac5x1xsoundc.44 GPIO2 GPI ok 240 name.titac5x1xsoundc.44 # GPIO2 GPI is not writeable ok 241 # SKIP write_default.titac5x1xsoundc.44 # GPIO2 GPI is not writeable ok 242 # SKIP write_valid.titac5x1xsoundc.44 # GPIO2 GPI is not writeable ok 243 # SKIP write_invalid.titac5x1xsoundc.44 ok 244 event_missing.titac5x1xsoundc.44 ok 245 event_spurious.titac5x1xsoundc.44 ok 246 get_value.titac5x1xsoundc.43 # titac5x1xsoundc.43 PDM4 Phase Capture Volume ok 247 name.titac5x1xsoundc.43 ok 248 write_default.titac5x1xsoundc.43 ok 249 write_valid.titac5x1xsoundc.43 ok 250 write_invalid.titac5x1xsoundc.43 ok 251 event_missing.titac5x1xsoundc.43 ok 252 event_spurious.titac5x1xsoundc.43 ok 253 get_value.titac5x1xsoundc.42 # titac5x1xsoundc.42 PDM3 Phase Capture Volume ok 254 name.titac5x1xsoundc.42 ok 255 write_default.titac5x1xsoundc.42 ok 256 write_valid.titac5x1xsoundc.42 ok 257 write_invalid.titac5x1xsoundc.42 ok 258 event_missing.titac5x1xsoundc.42 ok 259 event_spurious.titac5x1xsoundc.42 ok 260 get_value.titac5x1xsoundc.41 # titac5x1xsoundc.41 PDM4 Fine Capture Volume ok 261 name.titac5x1xsoundc.41 ok 262 write_default.titac5x1xsoundc.41 ok 263 write_valid.titac5x1xsoundc.41 ok 264 write_invalid.titac5x1xsoundc.41 ok 265 event_missing.titac5x1xsoundc.41 ok 266 event_spurious.titac5x1xsoundc.41 ok 267 get_value.titac5x1xsoundc.40 # titac5x1xsoundc.40 PDM3 Fine Capture Volume ok 268 name.titac5x1xsoundc.40 ok 269 write_default.titac5x1xsoundc.40 ok 270 write_valid.titac5x1xsoundc.40 ok 271 write_invalid.titac5x1xsoundc.40 ok 272 event_missing.titac5x1xsoundc.40 ok 273 event_spurious.titac5x1xsoundc.40 ok 274 get_value.titac5x1xsoundc.39 # titac5x1xsoundc.39 PDM4 Digital Capture Volume ok 275 name.titac5x1xsoundc.39 ok 276 write_default.titac5x1xsoundc.39 ok 277 write_valid.titac5x1xsoundc.39 ok 278 write_invalid.titac5x1xsoundc.39 ok 279 event_missing.titac5x1xsoundc.39 ok 280 event_spurious.titac5x1xsoundc.39 ok 281 get_value.titac5x1xsoundc.38 # titac5x1xsoundc.38 PDM3 Digital Capture Volume ok 282 name.titac5x1xsoundc.38 ok 283 write_default.titac5x1xsoundc.38 ok 284 write_valid.titac5x1xsoundc.38 ok 285 write_invalid.titac5x1xsoundc.38 ok 286 event_missing.titac5x1xsoundc.38 ok 287 event_spurious.titac5x1xsoundc.38 ok 288 get_value.titac5x1xsoundc.37 # titac5x1xsoundc.37 PDM2 Phase Capture Volume ok 289 name.titac5x1xsoundc.37 ok 290 write_default.titac5x1xsoundc.37 ok 291 write_valid.titac5x1xsoundc.37 ok 292 write_invalid.titac5x1xsoundc.37 ok 293 event_missing.titac5x1xsoundc.37 ok 294 event_spurious.titac5x1xsoundc.37 ok 295 get_value.titac5x1xsoundc.36 # titac5x1xsoundc.36 PDM1 Phase Capture Volume ok 296 name.titac5x1xsoundc.36 ok 297 write_default.titac5x1xsoundc.36 ok 298 write_valid.titac5x1xsoundc.36 ok 299 write_invalid.titac5x1xsoundc.36 ok 300 event_missing.titac5x1xsoundc.36 ok 301 event_spurious.titac5x1xsoundc.36 ok 302 get_value.titac5x1xsoundc.35 # titac5x1xsoundc.35 PDM2 Fine Capture Volume ok 303 name.titac5x1xsoundc.35 ok 304 write_default.titac5x1xsoundc.35 ok 305 write_valid.titac5x1xsoundc.35 ok 306 write_invalid.titac5x1xsoundc.35 ok 307 event_missing.titac5x1xsoundc.35 ok 308 event_spurious.titac5x1xsoundc.35 ok 309 get_value.titac5x1xsoundc.34 # titac5x1xsoundc.34 PDM1 Fine Capture Volume ok 310 name.titac5x1xsoundc.34 ok 311 write_default.titac5x1xsoundc.34 ok 312 write_valid.titac5x1xsoundc.34 ok 313 write_invalid.titac5x1xsoundc.34 ok 314 event_missing.titac5x1xsoundc.34 ok 315 event_spurious.titac5x1xsoundc.34 ok 316 get_value.titac5x1xsoundc.33 # titac5x1xsoundc.33 PDM2 Digital Capture Volume ok 317 name.titac5x1xsoundc.33 ok 318 write_default.titac5x1xsoundc.33 ok 319 write_valid.titac5x1xsoundc.33 ok 320 write_invalid.titac5x1xsoundc.33 ok 321 event_missing.titac5x1xsoundc.33 ok 322 event_spurious.titac5x1xsoundc.33 ok 323 get_value.titac5x1xsoundc.32 # titac5x1xsoundc.32 PDM1 Digital Capture Volume ok 324 name.titac5x1xsoundc.32 ok 325 write_default.titac5x1xsoundc.32 ok 326 write_valid.titac5x1xsoundc.32 ok 327 write_invalid.titac5x1xsoundc.32 ok 328 event_missing.titac5x1xsoundc.32 ok 329 event_spurious.titac5x1xsoundc.32 ok 330 get_value.titac5x1xsoundc.31 # titac5x1xsoundc.31 PDM Clk Divider ok 331 name.titac5x1xsoundc.31 ok 332 write_default.titac5x1xsoundc.31 ok 333 write_valid.titac5x1xsoundc.31 ok 334 write_invalid.titac5x1xsoundc.31 ok 335 event_missing.titac5x1xsoundc.31 ok 336 event_spurious.titac5x1xsoundc.31 ok 337 get_value.titac5x1xsoundc.30 # titac5x1xsoundc.30 DAC2 Audio BW ok 338 name.titac5x1xsoundc.30 ok 339 write_default.titac5x1xsoundc.30 ok 340 write_valid.titac5x1xsoundc.30 ok 341 write_invalid.titac5x1xsoundc.30 ok 342 event_missing.titac5x1xsoundc.30 ok 343 event_spurious.titac5x1xsoundc.30 ok 344 get_value.titac5x1xsoundc.29 # titac5x1xsoundc.29 DAC2 CHB Gain Calibration Playback Volume ok 345 name.titac5x1xsoundc.29 ok 346 write_default.titac5x1xsoundc.29 ok 347 write_valid.titac5x1xsoundc.29 ok 348 write_invalid.titac5x1xsoundc.29 ok 349 event_missing.titac5x1xsoundc.29 ok 350 event_spurious.titac5x1xsoundc.29 ok 351 get_value.titac5x1xsoundc.28 # titac5x1xsoundc.28 DAC2 CHA Gain Calibration Playback Volume ok 352 name.titac5x1xsoundc.28 ok 353 write_default.titac5x1xsoundc.28 ok 354 write_valid.titac5x1xsoundc.28 ok 355 write_invalid.titac5x1xsoundc.28 ok 356 event_missing.titac5x1xsoundc.28 ok 357 event_spurious.titac5x1xsoundc.28 ok 358 get_value.titac5x1xsoundc.27 # titac5x1xsoundc.27 DAC2 CHB Digital Playback Volume ok 359 name.titac5x1xsoundc.27 ok 360 write_default.titac5x1xsoundc.27 ok 361 write_valid.titac5x1xsoundc.27 ok 362 write_invalid.titac5x1xsoundc.27 ok 363 event_missing.titac5x1xsoundc.27 ok 364 event_spurious.titac5x1xsoundc.27 ok 365 get_value.titac5x1xsoundc.26 # titac5x1xsoundc.26 DAC2 CHA Digital Playback Volume ok 366 name.titac5x1xsoundc.26 ok 367 write_default.titac5x1xsoundc.26 ok 368 write_valid.titac5x1xsoundc.26 ok 369 write_invalid.titac5x1xsoundc.26 ok 370 event_missing.titac5x1xsoundc.26 ok 371 event_spurious.titac5x1xsoundc.26 ok 372 get_value.titac5x1xsoundc.25 # titac5x1xsoundc.25 OUT2M Analog Level Playback Volume ok 373 name.titac5x1xsoundc.25 ok 374 write_default.titac5x1xsoundc.25 ok 375 write_valid.titac5x1xsoundc.25 ok 376 write_invalid.titac5x1xsoundc.25 ok 377 event_missing.titac5x1xsoundc.25 ok 378 event_spurious.titac5x1xsoundc.25 ok 379 get_value.titac5x1xsoundc.24 # titac5x1xsoundc.24 OUT2P Analog Level Playback Volume ok 380 name.titac5x1xsoundc.24 ok 381 write_default.titac5x1xsoundc.24 ok 382 write_valid.titac5x1xsoundc.24 ok 383 write_invalid.titac5x1xsoundc.24 ok 384 event_missing.titac5x1xsoundc.24 ok 385 event_spurious.titac5x1xsoundc.24 ok 386 get_value.titac5x1xsoundc.23 # titac5x1xsoundc.23 ADC2 Phase Capture Volume ok 387 name.titac5x1xsoundc.23 ok 388 write_default.titac5x1xsoundc.23 ok 389 write_valid.titac5x1xsoundc.23 ok 390 write_invalid.titac5x1xsoundc.23 ok 391 event_missing.titac5x1xsoundc.23 ok 392 event_spurious.titac5x1xsoundc.23 ok 393 get_value.titac5x1xsoundc.22 # titac5x1xsoundc.22 ADC2 Fine Capture Volume ok 394 name.titac5x1xsoundc.22 ok 395 write_default.titac5x1xsoundc.22 ok 396 write_valid.titac5x1xsoundc.22 ok 397 write_invalid.titac5x1xsoundc.22 ok 398 event_missing.titac5x1xsoundc.22 ok 399 event_spurious.titac5x1xsoundc.22 ok 400 get_value.titac5x1xsoundc.21 # titac5x1xsoundc.21 ADC2 Digital Capture Volume ok 401 name.titac5x1xsoundc.21 ok 402 write_default.titac5x1xsoundc.21 ok 403 write_valid.titac5x1xsoundc.21 ok 404 write_invalid.titac5x1xsoundc.21 ok 405 event_missing.titac5x1xsoundc.21 ok 406 event_spurious.titac5x1xsoundc.21 ok 407 get_value.titac5x1xsoundc.20 # titac5x1xsoundc.20 ADC2 Audio BW ok 408 name.titac5x1xsoundc.20 ok 409 write_default.titac5x1xsoundc.20 ok 410 write_valid.titac5x1xsoundc.20 ok 411 write_invalid.titac5x1xsoundc.20 ok 412 event_missing.titac5x1xsoundc.20 ok 413 event_spurious.titac5x1xsoundc.20 ok 414 get_value.titac5x1xsoundc.19 # titac5x1xsoundc.19 OUT_CH4_EN Playback Switch ok 415 name.titac5x1xsoundc.19 ok 416 write_default.titac5x1xsoundc.19 ok 417 write_valid.titac5x1xsoundc.19 ok 418 write_invalid.titac5x1xsoundc.19 ok 419 event_missing.titac5x1xsoundc.19 ok 420 event_spurious.titac5x1xsoundc.19 ok 421 get_value.titac5x1xsoundc.18 # titac5x1xsoundc.18 OUT_CH3_EN Playback Switch ok 422 name.titac5x1xsoundc.18 ok 423 write_default.titac5x1xsoundc.18 ok 424 write_valid.titac5x1xsoundc.18 ok 425 write_invalid.titac5x1xsoundc.18 ok 426 event_missing.titac5x1xsoundc.18 ok 427 event_spurious.titac5x1xsoundc.18 ok 428 get_value.titac5x1xsoundc.17 # titac5x1xsoundc.17 OUT_CH2_EN Playback Switch ok 429 name.titac5x1xsoundc.17 ok 430 write_default.titac5x1xsoundc.17 ok 431 write_valid.titac5x1xsoundc.17 ok 432 write_invalid.titac5x1xsoundc.17 ok 433 event_missing.titac5x1xsoundc.17 ok 434 event_spurious.titac5x1xsoundc.17 ok 435 get_value.titac5x1xsoundc.16 # titac5x1xsoundc.16 OUT_CH1_EN Playback Switch ok 436 name.titac5x1xsoundc.16 ok 437 write_default.titac5x1xsoundc.16 ok 438 write_valid.titac5x1xsoundc.16 ok 439 write_invalid.titac5x1xsoundc.16 ok 440 event_missing.titac5x1xsoundc.16 ok 441 event_spurious.titac5x1xsoundc.16 ok 442 get_value.titac5x1xsoundc.15 # titac5x1xsoundc.15 ASI_RX_CH8_EN Playback ok 443 name.titac5x1xsoundc.15 ok 444 write_default.titac5x1xsoundc.15 ok 445 write_valid.titac5x1xsoundc.15 ok 446 write_invalid.titac5x1xsoundc.15 ok 447 event_missing.titac5x1xsoundc.15 ok 448 event_spurious.titac5x1xsoundc.15 ok 449 get_value.titac5x1xsoundc.14 # titac5x1xsoundc.14 ASI_RX_CH7_EN Playback ok 450 name.titac5x1xsoundc.14 ok 451 write_default.titac5x1xsoundc.14 ok 452 write_valid.titac5x1xsoundc.14 ok 453 write_invalid.titac5x1xsoundc.14 ok 454 event_missing.titac5x1xsoundc.14 ok 455 event_spurious.titac5x1xsoundc.14 ok 456 get_value.titac5x1xsoundc.13 # titac5x1xsoundc.13 ASI_RX_CH6_EN Playback ok 457 name.titac5x1xsoundc.13 ok 458 write_default.titac5x1xsoundc.13 ok 459 write_valid.titac5x1xsoundc.13 ok 460 write_invalid.titac5x1xsoundc.13 ok 461 event_missing.titac5x1xsoundc.13 ok 462 event_spurious.titac5x1xsoundc.13 ok 463 get_value.titac5x1xsoundc.12 # titac5x1xsoundc.12 ASI_RX_CH5_EN Playback ok 464 name.titac5x1xsoundc.12 ok 465 write_default.titac5x1xsoundc.12 ok 466 write_valid.titac5x1xsoundc.12 ok 467 write_invalid.titac5x1xsoundc.12 ok 468 event_missing.titac5x1xsoundc.12 ok 469 event_spurious.titac5x1xsoundc.12 ok 470 get_value.titac5x1xsoundc.11 # titac5x1xsoundc.11 ASI_RX_CH4_EN Playback Switch ok 471 name.titac5x1xsoundc.11 ok 472 write_default.titac5x1xsoundc.11 ok 473 write_valid.titac5x1xsoundc.11 ok 474 write_invalid.titac5x1xsoundc.11 ok 475 event_missing.titac5x1xsoundc.11 ok 476 event_spurious.titac5x1xsoundc.11 ok 477 get_value.titac5x1xsoundc.10 # titac5x1xsoundc.10 ASI_RX_CH3_EN Playback Switch ok 478 name.titac5x1xsoundc.10 ok 479 write_default.titac5x1xsoundc.10 ok 480 write_valid.titac5x1xsoundc.10 ok 481 write_invalid.titac5x1xsoundc.10 ok 482 event_missing.titac5x1xsoundc.10 ok 483 event_spurious.titac5x1xsoundc.10 ok 484 get_value.titac5x1xsoundc.9 # titac5x1xsoundc.9 DAC1 CHB Gain Calibration Playback Volume ok 485 name.titac5x1xsoundc.9 ok 486 write_default.titac5x1xsoundc.9 ok 487 write_valid.titac5x1xsoundc.9 ok 488 write_invalid.titac5x1xsoundc.9 ok 489 event_missing.titac5x1xsoundc.9 ok 490 event_spurious.titac5x1xsoundc.9 ok 491 get_value.titac5x1xsoundc.8 # titac5x1xsoundc.8 DAC1 CHA Gain Calibration Playback Volume ok 492 name.titac5x1xsoundc.8 ok 493 write_default.titac5x1xsoundc.8 ok 494 write_valid.titac5x1xsoundc.8 ok 495 write_invalid.titac5x1xsoundc.8 ok 496 event_missing.titac5x1xsoundc.8 ok 497 event_spurious.titac5x1xsoundc.8 ok 498 get_value.titac5x1xsoundc.7 # titac5x1xsoundc.7 DAC1 CHB Digital Playback Volume ok 499 name.titac5x1xsoundc.7 ok 500 write_default.titac5x1xsoundc.7 ok 501 write_valid.titac5x1xsoundc.7 ok 502 write_invalid.titac5x1xsoundc.7 ok 503 event_missing.titac5x1xsoundc.7 ok 504 event_spurious.titac5x1xsoundc.7 ok 505 get_value.titac5x1xsoundc.6 # titac5x1xsoundc.6 DAC1 CHA Digital Playback Volume ok 506 name.titac5x1xsoundc.6 ok 507 write_default.titac5x1xsoundc.6 ok 508 write_valid.titac5x1xsoundc.6 ok 509 write_invalid.titac5x1xsoundc.6 ok 510 event_missing.titac5x1xsoundc.6 ok 511 event_spurious.titac5x1xsoundc.6 ok 512 get_value.titac5x1xsoundc.5 # titac5x1xsoundc.5 OUT1M Analog Level Playback Volume ok 513 name.titac5x1xsoundc.5 ok 514 write_default.titac5x1xsoundc.5 ok 515 write_valid.titac5x1xsoundc.5 ok 516 write_invalid.titac5x1xsoundc.5 ok 517 event_missing.titac5x1xsoundc.5 ok 518 event_spurious.titac5x1xsoundc.5 ok 519 get_value.titac5x1xsoundc.4 # titac5x1xsoundc.4 OUT1P Analog Level Playback Volume ok 520 name.titac5x1xsoundc.4 ok 521 write_default.titac5x1xsoundc.4 ok 522 write_valid.titac5x1xsoundc.4 ok 523 write_invalid.titac5x1xsoundc.4 ok 524 event_missing.titac5x1xsoundc.4 ok 525 event_spurious.titac5x1xsoundc.4 ok 526 get_value.titac5x1xsoundc.3 # titac5x1xsoundc.3 DAC1 Audio BW ok 527 name.titac5x1xsoundc.3 ok 528 write_default.titac5x1xsoundc.3 ok 529 write_valid.titac5x1xsoundc.3 ok 530 write_invalid.titac5x1xsoundc.3 ok 531 event_missing.titac5x1xsoundc.3 ok 532 event_spurious.titac5x1xsoundc.3 ok 533 get_value.titac5x1xsoundc.2 # titac5x1xsoundc.2 Playback Decimation Filter ok 534 name.titac5x1xsoundc.2 ok 535 write_default.titac5x1xsoundc.2 ok 536 write_valid.titac5x1xsoundc.2 ok 537 write_invalid.titac5x1xsoundc.2 ok 538 event_missing.titac5x1xsoundc.2 ok 539 event_spurious.titac5x1xsoundc.2 ok 540 get_value.titac5x1xsoundc.1 # titac5x1xsoundc.1 ADC2 Common-mode Tolerance ok 541 name.titac5x1xsoundc.1 ok 542 write_default.titac5x1xsoundc.1 ok 543 write_valid.titac5x1xsoundc.1 ok 544 write_invalid.titac5x1xsoundc.1 ok 545 event_missing.titac5x1xsoundc.1 ok 546 event_spurious.titac5x1xsoundc.1 ok 547 get_value.titac5x1xsoundc.0 # titac5x1xsoundc.0 ADC1 Common-mode Tolerance ok 548 name.titac5x1xsoundc.0 ok 549 write_default.titac5x1xsoundc.0 ok 550 write_valid.titac5x1xsoundc.0 ok 551 write_invalid.titac5x1xsoundc.0 ok 552 event_missing.titac5x1xsoundc.0 ok 553 event_spurious.titac5x1xsoundc.0 # 6 skipped test(s) detected. Consider enabling relevant config options to improve coverage. # Totals: pass:547 fail:0 xfail:0 xpass:0 skip:6 error:0 Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> Niranjan H Y (4): ASoc: tac5x1x: add codec driver tac5x1x family Asoc: tac5x1x: document tac5x1x codec Asoc: pcm6240: remove support for taac5x1x family ASoc: pcm6240: dt-bindings device support list update .../devicetree/bindings/sound/ti,pcm6240.yaml | 10 - .../devicetree/bindings/sound/ti,tac5x1x.yaml | 248 ++ sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm6240.c | 124 +- sound/soc/codecs/pcm6240.h | 4 - sound/soc/codecs/tac5x1x-i2c.c | 2225 +++++++++++++++++ sound/soc/codecs/tac5x1x.h | 300 +++ 8 files changed, 2790 insertions(+), 135 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tac5x1x.yaml create mode 100644 sound/soc/codecs/tac5x1x-i2c.c create mode 100644 sound/soc/codecs/tac5x1x.h -- 2.45.2

4 8

ucm2: Qualcomm: add Dell XPS 9345
by GitHub pull_request - opened 28 Jun '25

28 Jun '25

alsa-project/alsa-ucm-conf pull request #587 was opened from alexVinarskis: Add a Regex string, seems to be mostly compatible with Yoga Slim7x. x4 speakers, x2 DMICs (unlike Yoga's 4), x2 DP, no headphone jack. DMI_info evaluates to `Dell Inc.-XPS-0DXPNM`. It appears to be okay to re-use Yoga, as [only 2 microphone channels are considered](https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/…. Audio part of Dell XPS 9345 DT is available [out of tree](https://github.com/alexVinarskis/linux-x1e80100-dell-tributo), will be submitted to the lists shortly. audioreach-topology was [merged a while ago](https://github.com/linux-msm/audioreach-topology/pull/20). Tested that speakers, MICs are working. **Known issues:** L and R sides are swapped due to kernel driver not taking DT labels into account just yet, and particular platform having WSAs connected to different soundwires interfaces than the rest of x1e80100 devices. Identified by Abel Vesa at Linaro. Fix will be on the kernel and/or DT side. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/587 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/587.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v3 0/4] ASoc: tac5x1x: mixer-test report
by Niranjan H Y 28 Jun '25

28 Jun '25

mixer-test report: TAP version 13 # Card 0/titac5x1xsoundc - ti,tac5x1x-soundcard (ti,tac5x1x-soundcard) 1..525 ok 1 get_value.titac5x1xsoundc.74 # titac5x1xsoundc.74 ASI_TX_CH4_EN Capture Switch ok 2 name.titac5x1xsoundc.74 ok 3 write_default.titac5x1xsoundc.74 ok 4 write_valid.titac5x1xsoundc.74 ok 5 write_invalid.titac5x1xsoundc.74 ok 6 event_missing.titac5x1xsoundc.74 ok 7 event_spurious.titac5x1xsoundc.74 ok 8 get_value.titac5x1xsoundc.73 # titac5x1xsoundc.73 ASI_TX_CH3_EN Capture Switch ok 9 name.titac5x1xsoundc.73 ok 10 write_default.titac5x1xsoundc.73 ok 11 write_valid.titac5x1xsoundc.73 ok 12 write_invalid.titac5x1xsoundc.73 ok 13 event_missing.titac5x1xsoundc.73 ok 14 event_spurious.titac5x1xsoundc.73 ok 15 get_value.titac5x1xsoundc.72 # titac5x1xsoundc.72 IN2 Source Mux ok 16 name.titac5x1xsoundc.72 ok 17 write_default.titac5x1xsoundc.72 ok 18 write_valid.titac5x1xsoundc.72 ok 19 write_invalid.titac5x1xsoundc.72 ok 20 event_missing.titac5x1xsoundc.72 ok 21 event_spurious.titac5x1xsoundc.72 ok 22 get_value.titac5x1xsoundc.71 # titac5x1xsoundc.71 IN1 Source Mux ok 23 name.titac5x1xsoundc.71 ok 24 write_default.titac5x1xsoundc.71 ok 25 write_valid.titac5x1xsoundc.71 ok 26 write_invalid.titac5x1xsoundc.71 ok 27 event_missing.titac5x1xsoundc.71 ok 28 event_spurious.titac5x1xsoundc.71 ok 29 get_value.titac5x1xsoundc.70 # titac5x1xsoundc.70 ADC2 Config ok 30 name.titac5x1xsoundc.70 ok 31 write_default.titac5x1xsoundc.70 ok 32 write_valid.titac5x1xsoundc.70 ok 33 write_invalid.titac5x1xsoundc.70 ok 34 event_missing.titac5x1xsoundc.70 ok 35 event_spurious.titac5x1xsoundc.70 ok 36 get_value.titac5x1xsoundc.69 # titac5x1xsoundc.69 ADC2 Full-Scale ok 37 name.titac5x1xsoundc.69 ok 38 write_default.titac5x1xsoundc.69 ok 39 write_valid.titac5x1xsoundc.69 ok 40 write_invalid.titac5x1xsoundc.69 ok 41 event_missing.titac5x1xsoundc.69 ok 42 event_spurious.titac5x1xsoundc.69 ok 43 get_value.titac5x1xsoundc.68 # titac5x1xsoundc.68 ASI_RX_CH2_EN Switch ok 44 name.titac5x1xsoundc.68 ok 45 write_default.titac5x1xsoundc.68 ok 46 write_valid.titac5x1xsoundc.68 ok 47 write_invalid.titac5x1xsoundc.68 ok 48 event_missing.titac5x1xsoundc.68 ok 49 event_spurious.titac5x1xsoundc.68 ok 50 get_value.titac5x1xsoundc.67 # titac5x1xsoundc.67 OUT2x Driver ok 51 name.titac5x1xsoundc.67 ok 52 write_default.titac5x1xsoundc.67 ok 53 write_valid.titac5x1xsoundc.67 ok 54 write_invalid.titac5x1xsoundc.67 ok 55 event_missing.titac5x1xsoundc.67 ok 56 event_spurious.titac5x1xsoundc.67 ok 57 get_value.titac5x1xsoundc.66 # titac5x1xsoundc.66 OUT2x Config ok 58 name.titac5x1xsoundc.66 ok 59 write_default.titac5x1xsoundc.66 ok 60 write_valid.titac5x1xsoundc.66 ok 61 write_invalid.titac5x1xsoundc.66 ok 62 event_missing.titac5x1xsoundc.66 ok 63 event_spurious.titac5x1xsoundc.66 ok 64 get_value.titac5x1xsoundc.65 # titac5x1xsoundc.65 OUT2x Source ok 65 name.titac5x1xsoundc.65 ok 66 write_default.titac5x1xsoundc.65 ok 67 write_valid.titac5x1xsoundc.65 ok 68 write_invalid.titac5x1xsoundc.65 ok 69 event_missing.titac5x1xsoundc.65 ok 70 event_spurious.titac5x1xsoundc.65 ok 71 get_value.titac5x1xsoundc.64 # titac5x1xsoundc.64 ASI_RX_CH1_EN Switch ok 72 name.titac5x1xsoundc.64 ok 73 write_default.titac5x1xsoundc.64 ok 74 write_valid.titac5x1xsoundc.64 ok 75 write_invalid.titac5x1xsoundc.64 ok 76 event_missing.titac5x1xsoundc.64 ok 77 event_spurious.titac5x1xsoundc.64 ok 78 get_value.titac5x1xsoundc.63 # titac5x1xsoundc.63 OUT1x Driver ok 79 name.titac5x1xsoundc.63 ok 80 write_default.titac5x1xsoundc.63 ok 81 write_valid.titac5x1xsoundc.63 ok 82 write_invalid.titac5x1xsoundc.63 ok 83 event_missing.titac5x1xsoundc.63 ok 84 event_spurious.titac5x1xsoundc.63 ok 85 get_value.titac5x1xsoundc.62 # titac5x1xsoundc.62 OUT1x Config ok 86 name.titac5x1xsoundc.62 ok 87 write_default.titac5x1xsoundc.62 ok 88 write_valid.titac5x1xsoundc.62 ok 89 write_invalid.titac5x1xsoundc.62 ok 90 event_missing.titac5x1xsoundc.62 ok 91 event_spurious.titac5x1xsoundc.62 ok 92 get_value.titac5x1xsoundc.61 # titac5x1xsoundc.61 OUT1x Source ok 93 name.titac5x1xsoundc.61 ok 94 write_default.titac5x1xsoundc.61 ok 95 write_valid.titac5x1xsoundc.61 ok 96 write_invalid.titac5x1xsoundc.61 ok 97 event_missing.titac5x1xsoundc.61 ok 98 event_spurious.titac5x1xsoundc.61 ok 99 get_value.titac5x1xsoundc.60 # titac5x1xsoundc.60 ASI_TX_CH2_EN Capture Switch ok 100 name.titac5x1xsoundc.60 ok 101 write_default.titac5x1xsoundc.60 ok 102 write_valid.titac5x1xsoundc.60 ok 103 write_invalid.titac5x1xsoundc.60 ok 104 event_missing.titac5x1xsoundc.60 ok 105 event_spurious.titac5x1xsoundc.60 ok 106 get_value.titac5x1xsoundc.59 # titac5x1xsoundc.59 ASI_TX_CH1_EN Capture Switch ok 107 name.titac5x1xsoundc.59 ok 108 write_default.titac5x1xsoundc.59 ok 109 write_valid.titac5x1xsoundc.59 ok 110 write_invalid.titac5x1xsoundc.59 ok 111 event_missing.titac5x1xsoundc.59 ok 112 event_spurious.titac5x1xsoundc.59 ok 113 get_value.titac5x1xsoundc.58 # titac5x1xsoundc.58 ADC1 Config ok 114 name.titac5x1xsoundc.58 ok 115 write_default.titac5x1xsoundc.58 ok 116 write_valid.titac5x1xsoundc.58 ok 117 write_invalid.titac5x1xsoundc.58 ok 118 event_missing.titac5x1xsoundc.58 ok 119 event_spurious.titac5x1xsoundc.58 ok 120 get_value.titac5x1xsoundc.57 # titac5x1xsoundc.57 ADC1 Full-Scale ok 121 name.titac5x1xsoundc.57 ok 122 write_default.titac5x1xsoundc.57 ok 123 write_valid.titac5x1xsoundc.57 ok 124 write_invalid.titac5x1xsoundc.57 ok 125 event_missing.titac5x1xsoundc.57 ok 126 event_spurious.titac5x1xsoundc.57 ok 127 get_value.titac5x1xsoundc.56 # titac5x1xsoundc.56 IN_CH4_EN Capture Switch ok 128 name.titac5x1xsoundc.56 ok 129 write_default.titac5x1xsoundc.56 ok 130 write_valid.titac5x1xsoundc.56 ok 131 write_invalid.titac5x1xsoundc.56 ok 132 event_missing.titac5x1xsoundc.56 ok 133 event_spurious.titac5x1xsoundc.56 ok 134 get_value.titac5x1xsoundc.55 # titac5x1xsoundc.55 IN_CH3_EN Capture Switch ok 135 name.titac5x1xsoundc.55 ok 136 write_default.titac5x1xsoundc.55 ok 137 write_valid.titac5x1xsoundc.55 ok 138 write_invalid.titac5x1xsoundc.55 ok 139 event_missing.titac5x1xsoundc.55 ok 140 event_spurious.titac5x1xsoundc.55 ok 141 get_value.titac5x1xsoundc.54 # titac5x1xsoundc.54 IN_CH2_EN Capture Switch ok 142 name.titac5x1xsoundc.54 ok 143 write_default.titac5x1xsoundc.54 ok 144 write_valid.titac5x1xsoundc.54 ok 145 write_invalid.titac5x1xsoundc.54 ok 146 event_missing.titac5x1xsoundc.54 ok 147 event_spurious.titac5x1xsoundc.54 ok 148 get_value.titac5x1xsoundc.53 # titac5x1xsoundc.53 IN_CH1_EN Capture Switch ok 149 name.titac5x1xsoundc.53 ok 150 write_default.titac5x1xsoundc.53 ok 151 write_valid.titac5x1xsoundc.53 ok 152 write_invalid.titac5x1xsoundc.53 ok 153 event_missing.titac5x1xsoundc.53 ok 154 event_spurious.titac5x1xsoundc.53 ok 155 get_value.titac5x1xsoundc.52 # titac5x1xsoundc.52 ASI_TX_CH8_CFG ok 156 name.titac5x1xsoundc.52 ok 157 write_default.titac5x1xsoundc.52 ok 158 write_valid.titac5x1xsoundc.52 ok 159 write_invalid.titac5x1xsoundc.52 ok 160 event_missing.titac5x1xsoundc.52 ok 161 event_spurious.titac5x1xsoundc.52 ok 162 get_value.titac5x1xsoundc.51 # titac5x1xsoundc.51 ASI_TX_CH7_CFG ok 163 name.titac5x1xsoundc.51 ok 164 write_default.titac5x1xsoundc.51 ok 165 write_valid.titac5x1xsoundc.51 ok 166 write_invalid.titac5x1xsoundc.51 ok 167 event_missing.titac5x1xsoundc.51 ok 168 event_spurious.titac5x1xsoundc.51 ok 169 get_value.titac5x1xsoundc.50 # titac5x1xsoundc.50 ASI_TX_CH6_CFG ok 170 name.titac5x1xsoundc.50 ok 171 write_default.titac5x1xsoundc.50 ok 172 write_valid.titac5x1xsoundc.50 ok 173 write_invalid.titac5x1xsoundc.50 ok 174 event_missing.titac5x1xsoundc.50 ok 175 event_spurious.titac5x1xsoundc.50 ok 176 get_value.titac5x1xsoundc.49 # titac5x1xsoundc.49 ASI_TX_CH5_CFG ok 177 name.titac5x1xsoundc.49 ok 178 write_default.titac5x1xsoundc.49 ok 179 write_valid.titac5x1xsoundc.49 ok 180 write_invalid.titac5x1xsoundc.49 ok 181 event_missing.titac5x1xsoundc.49 ok 182 event_spurious.titac5x1xsoundc.49 ok 183 get_value.titac5x1xsoundc.48 # titac5x1xsoundc.48 ADC1 Phase Capture Volume ok 184 name.titac5x1xsoundc.48 ok 185 write_default.titac5x1xsoundc.48 ok 186 write_valid.titac5x1xsoundc.48 ok 187 write_invalid.titac5x1xsoundc.48 ok 188 event_missing.titac5x1xsoundc.48 ok 189 event_spurious.titac5x1xsoundc.48 ok 190 get_value.titac5x1xsoundc.47 # titac5x1xsoundc.47 ADC1 Fine Capture Volume ok 191 name.titac5x1xsoundc.47 ok 192 write_default.titac5x1xsoundc.47 ok 193 write_valid.titac5x1xsoundc.47 ok 194 write_invalid.titac5x1xsoundc.47 ok 195 event_missing.titac5x1xsoundc.47 ok 196 event_spurious.titac5x1xsoundc.47 ok 197 get_value.titac5x1xsoundc.46 # titac5x1xsoundc.46 ADC1 Digital Capture Volume ok 198 name.titac5x1xsoundc.46 ok 199 write_default.titac5x1xsoundc.46 ok 200 write_valid.titac5x1xsoundc.46 ok 201 write_invalid.titac5x1xsoundc.46 ok 202 event_missing.titac5x1xsoundc.46 ok 203 event_spurious.titac5x1xsoundc.46 ok 204 get_value.titac5x1xsoundc.45 # titac5x1xsoundc.45 ADC1 Audio BW ok 205 name.titac5x1xsoundc.45 ok 206 write_default.titac5x1xsoundc.45 ok 207 write_valid.titac5x1xsoundc.45 ok 208 write_invalid.titac5x1xsoundc.45 ok 209 event_missing.titac5x1xsoundc.45 ok 210 event_spurious.titac5x1xsoundc.45 ok 211 get_value.titac5x1xsoundc.44 # titac5x1xsoundc.44 Record Decimation Filter ok 212 name.titac5x1xsoundc.44 ok 213 write_default.titac5x1xsoundc.44 ok 214 write_valid.titac5x1xsoundc.44 ok 215 write_invalid.titac5x1xsoundc.44 ok 216 event_missing.titac5x1xsoundc.44 ok 217 event_spurious.titac5x1xsoundc.44 ok 218 get_value.titac5x1xsoundc.43 # titac5x1xsoundc.43 PDM4 Phase Capture Volume ok 219 name.titac5x1xsoundc.43 ok 220 write_default.titac5x1xsoundc.43 ok 221 write_valid.titac5x1xsoundc.43 ok 222 write_invalid.titac5x1xsoundc.43 ok 223 event_missing.titac5x1xsoundc.43 ok 224 event_spurious.titac5x1xsoundc.43 ok 225 get_value.titac5x1xsoundc.42 # titac5x1xsoundc.42 PDM3 Phase Capture Volume ok 226 name.titac5x1xsoundc.42 ok 227 write_default.titac5x1xsoundc.42 ok 228 write_valid.titac5x1xsoundc.42 ok 229 write_invalid.titac5x1xsoundc.42 ok 230 event_missing.titac5x1xsoundc.42 ok 231 event_spurious.titac5x1xsoundc.42 ok 232 get_value.titac5x1xsoundc.41 # titac5x1xsoundc.41 PDM4 Fine Capture Volume ok 233 name.titac5x1xsoundc.41 ok 234 write_default.titac5x1xsoundc.41 ok 235 write_valid.titac5x1xsoundc.41 ok 236 write_invalid.titac5x1xsoundc.41 ok 237 event_missing.titac5x1xsoundc.41 ok 238 event_spurious.titac5x1xsoundc.41 ok 239 get_value.titac5x1xsoundc.40 # titac5x1xsoundc.40 PDM3 Fine Capture Volume ok 240 name.titac5x1xsoundc.40 ok 241 write_default.titac5x1xsoundc.40 ok 242 write_valid.titac5x1xsoundc.40 ok 243 write_invalid.titac5x1xsoundc.40 ok 244 event_missing.titac5x1xsoundc.40 ok 245 event_spurious.titac5x1xsoundc.40 ok 246 get_value.titac5x1xsoundc.39 # titac5x1xsoundc.39 PDM4 Digital Capture Volume ok 247 name.titac5x1xsoundc.39 ok 248 write_default.titac5x1xsoundc.39 ok 249 write_valid.titac5x1xsoundc.39 ok 250 write_invalid.titac5x1xsoundc.39 ok 251 event_missing.titac5x1xsoundc.39 ok 252 event_spurious.titac5x1xsoundc.39 ok 253 get_value.titac5x1xsoundc.38 # titac5x1xsoundc.38 PDM3 Digital Capture Volume ok 254 name.titac5x1xsoundc.38 ok 255 write_default.titac5x1xsoundc.38 ok 256 write_valid.titac5x1xsoundc.38 ok 257 write_invalid.titac5x1xsoundc.38 ok 258 event_missing.titac5x1xsoundc.38 ok 259 event_spurious.titac5x1xsoundc.38 ok 260 get_value.titac5x1xsoundc.37 # titac5x1xsoundc.37 PDM2 Phase Capture Volume ok 261 name.titac5x1xsoundc.37 ok 262 write_default.titac5x1xsoundc.37 ok 263 write_valid.titac5x1xsoundc.37 ok 264 write_invalid.titac5x1xsoundc.37 ok 265 event_missing.titac5x1xsoundc.37 ok 266 event_spurious.titac5x1xsoundc.37 ok 267 get_value.titac5x1xsoundc.36 # titac5x1xsoundc.36 PDM1 Phase Capture Volume ok 268 name.titac5x1xsoundc.36 ok 269 write_default.titac5x1xsoundc.36 ok 270 write_valid.titac5x1xsoundc.36 ok 271 write_invalid.titac5x1xsoundc.36 ok 272 event_missing.titac5x1xsoundc.36 ok 273 event_spurious.titac5x1xsoundc.36 ok 274 get_value.titac5x1xsoundc.35 # titac5x1xsoundc.35 PDM2 Fine Capture Volume ok 275 name.titac5x1xsoundc.35 ok 276 write_default.titac5x1xsoundc.35 ok 277 write_valid.titac5x1xsoundc.35 ok 278 write_invalid.titac5x1xsoundc.35 ok 279 event_missing.titac5x1xsoundc.35 ok 280 event_spurious.titac5x1xsoundc.35 ok 281 get_value.titac5x1xsoundc.34 # titac5x1xsoundc.34 PDM1 Fine Capture Volume ok 282 name.titac5x1xsoundc.34 ok 283 write_default.titac5x1xsoundc.34 ok 284 write_valid.titac5x1xsoundc.34 ok 285 write_invalid.titac5x1xsoundc.34 ok 286 event_missing.titac5x1xsoundc.34 ok 287 event_spurious.titac5x1xsoundc.34 ok 288 get_value.titac5x1xsoundc.33 # titac5x1xsoundc.33 PDM2 Digital Capture Volume ok 289 name.titac5x1xsoundc.33 ok 290 write_default.titac5x1xsoundc.33 ok 291 write_valid.titac5x1xsoundc.33 ok 292 write_invalid.titac5x1xsoundc.33 ok 293 event_missing.titac5x1xsoundc.33 ok 294 event_spurious.titac5x1xsoundc.33 ok 295 get_value.titac5x1xsoundc.32 # titac5x1xsoundc.32 PDM1 Digital Capture Volume ok 296 name.titac5x1xsoundc.32 ok 297 write_default.titac5x1xsoundc.32 ok 298 write_valid.titac5x1xsoundc.32 ok 299 write_invalid.titac5x1xsoundc.32 ok 300 event_missing.titac5x1xsoundc.32 ok 301 event_spurious.titac5x1xsoundc.32 ok 302 get_value.titac5x1xsoundc.31 # titac5x1xsoundc.31 PDM Clk Divider ok 303 name.titac5x1xsoundc.31 ok 304 write_default.titac5x1xsoundc.31 ok 305 write_valid.titac5x1xsoundc.31 ok 306 write_invalid.titac5x1xsoundc.31 ok 307 event_missing.titac5x1xsoundc.31 ok 308 event_spurious.titac5x1xsoundc.31 ok 309 get_value.titac5x1xsoundc.30 # titac5x1xsoundc.30 DAC2 Audio BW ok 310 name.titac5x1xsoundc.30 ok 311 write_default.titac5x1xsoundc.30 ok 312 write_valid.titac5x1xsoundc.30 ok 313 write_invalid.titac5x1xsoundc.30 ok 314 event_missing.titac5x1xsoundc.30 ok 315 event_spurious.titac5x1xsoundc.30 ok 316 get_value.titac5x1xsoundc.29 # titac5x1xsoundc.29 DAC2 CHB Gain Calibration Playback Volume ok 317 name.titac5x1xsoundc.29 ok 318 write_default.titac5x1xsoundc.29 ok 319 write_valid.titac5x1xsoundc.29 ok 320 write_invalid.titac5x1xsoundc.29 ok 321 event_missing.titac5x1xsoundc.29 ok 322 event_spurious.titac5x1xsoundc.29 ok 323 get_value.titac5x1xsoundc.28 # titac5x1xsoundc.28 DAC2 CHA Gain Calibration Playback Volume ok 324 name.titac5x1xsoundc.28 ok 325 write_default.titac5x1xsoundc.28 ok 326 write_valid.titac5x1xsoundc.28 ok 327 write_invalid.titac5x1xsoundc.28 ok 328 event_missing.titac5x1xsoundc.28 ok 329 event_spurious.titac5x1xsoundc.28 ok 330 get_value.titac5x1xsoundc.27 # titac5x1xsoundc.27 DAC2 CHB Digital Playback Volume ok 331 name.titac5x1xsoundc.27 ok 332 write_default.titac5x1xsoundc.27 ok 333 write_valid.titac5x1xsoundc.27 ok 334 write_invalid.titac5x1xsoundc.27 ok 335 event_missing.titac5x1xsoundc.27 ok 336 event_spurious.titac5x1xsoundc.27 ok 337 get_value.titac5x1xsoundc.26 # titac5x1xsoundc.26 DAC2 CHA Digital Playback Volume ok 338 name.titac5x1xsoundc.26 ok 339 write_default.titac5x1xsoundc.26 ok 340 write_valid.titac5x1xsoundc.26 ok 341 write_invalid.titac5x1xsoundc.26 ok 342 event_missing.titac5x1xsoundc.26 ok 343 event_spurious.titac5x1xsoundc.26 ok 344 get_value.titac5x1xsoundc.25 # titac5x1xsoundc.25 OUT2M Analog Level Playback Volume ok 345 name.titac5x1xsoundc.25 ok 346 write_default.titac5x1xsoundc.25 ok 347 write_valid.titac5x1xsoundc.25 ok 348 write_invalid.titac5x1xsoundc.25 ok 349 event_missing.titac5x1xsoundc.25 ok 350 event_spurious.titac5x1xsoundc.25 ok 351 get_value.titac5x1xsoundc.24 # titac5x1xsoundc.24 OUT2P Analog Level Playback Volume ok 352 name.titac5x1xsoundc.24 ok 353 write_default.titac5x1xsoundc.24 ok 354 write_valid.titac5x1xsoundc.24 ok 355 write_invalid.titac5x1xsoundc.24 ok 356 event_missing.titac5x1xsoundc.24 ok 357 event_spurious.titac5x1xsoundc.24 ok 358 get_value.titac5x1xsoundc.23 # titac5x1xsoundc.23 ADC2 Phase Capture Volume ok 359 name.titac5x1xsoundc.23 ok 360 write_default.titac5x1xsoundc.23 ok 361 write_valid.titac5x1xsoundc.23 ok 362 write_invalid.titac5x1xsoundc.23 ok 363 event_missing.titac5x1xsoundc.23 ok 364 event_spurious.titac5x1xsoundc.23 ok 365 get_value.titac5x1xsoundc.22 # titac5x1xsoundc.22 ADC2 Fine Capture Volume ok 366 name.titac5x1xsoundc.22 ok 367 write_default.titac5x1xsoundc.22 ok 368 write_valid.titac5x1xsoundc.22 ok 369 write_invalid.titac5x1xsoundc.22 ok 370 event_missing.titac5x1xsoundc.22 ok 371 event_spurious.titac5x1xsoundc.22 ok 372 get_value.titac5x1xsoundc.21 # titac5x1xsoundc.21 ADC2 Digital Capture Volume ok 373 name.titac5x1xsoundc.21 ok 374 write_default.titac5x1xsoundc.21 ok 375 write_valid.titac5x1xsoundc.21 ok 376 write_invalid.titac5x1xsoundc.21 ok 377 event_missing.titac5x1xsoundc.21 ok 378 event_spurious.titac5x1xsoundc.21 ok 379 get_value.titac5x1xsoundc.20 # titac5x1xsoundc.20 ADC2 Audio BW ok 380 name.titac5x1xsoundc.20 ok 381 write_default.titac5x1xsoundc.20 ok 382 write_valid.titac5x1xsoundc.20 ok 383 write_invalid.titac5x1xsoundc.20 ok 384 event_missing.titac5x1xsoundc.20 ok 385 event_spurious.titac5x1xsoundc.20 ok 386 get_value.titac5x1xsoundc.19 # titac5x1xsoundc.19 OUT_CH4_EN Playback Switch ok 387 name.titac5x1xsoundc.19 ok 388 write_default.titac5x1xsoundc.19 ok 389 write_valid.titac5x1xsoundc.19 ok 390 write_invalid.titac5x1xsoundc.19 ok 391 event_missing.titac5x1xsoundc.19 ok 392 event_spurious.titac5x1xsoundc.19 ok 393 get_value.titac5x1xsoundc.18 # titac5x1xsoundc.18 OUT_CH3_EN Playback Switch ok 394 name.titac5x1xsoundc.18 ok 395 write_default.titac5x1xsoundc.18 ok 396 write_valid.titac5x1xsoundc.18 ok 397 write_invalid.titac5x1xsoundc.18 ok 398 event_missing.titac5x1xsoundc.18 ok 399 event_spurious.titac5x1xsoundc.18 ok 400 get_value.titac5x1xsoundc.17 # titac5x1xsoundc.17 OUT_CH2_EN Playback Switch ok 401 name.titac5x1xsoundc.17 ok 402 write_default.titac5x1xsoundc.17 ok 403 write_valid.titac5x1xsoundc.17 ok 404 write_invalid.titac5x1xsoundc.17 ok 405 event_missing.titac5x1xsoundc.17 ok 406 event_spurious.titac5x1xsoundc.17 ok 407 get_value.titac5x1xsoundc.16 # titac5x1xsoundc.16 OUT_CH1_EN Playback Switch ok 408 name.titac5x1xsoundc.16 ok 409 write_default.titac5x1xsoundc.16 ok 410 write_valid.titac5x1xsoundc.16 ok 411 write_invalid.titac5x1xsoundc.16 ok 412 event_missing.titac5x1xsoundc.16 ok 413 event_spurious.titac5x1xsoundc.16 ok 414 get_value.titac5x1xsoundc.15 # titac5x1xsoundc.15 ASI_RX_CH8_EN Playback ok 415 name.titac5x1xsoundc.15 ok 416 write_default.titac5x1xsoundc.15 ok 417 write_valid.titac5x1xsoundc.15 ok 418 write_invalid.titac5x1xsoundc.15 ok 419 event_missing.titac5x1xsoundc.15 ok 420 event_spurious.titac5x1xsoundc.15 ok 421 get_value.titac5x1xsoundc.14 # titac5x1xsoundc.14 ASI_RX_CH7_EN Playback ok 422 name.titac5x1xsoundc.14 ok 423 write_default.titac5x1xsoundc.14 ok 424 write_valid.titac5x1xsoundc.14 ok 425 write_invalid.titac5x1xsoundc.14 ok 426 event_missing.titac5x1xsoundc.14 ok 427 event_spurious.titac5x1xsoundc.14 ok 428 get_value.titac5x1xsoundc.13 # titac5x1xsoundc.13 ASI_RX_CH6_EN Playback ok 429 name.titac5x1xsoundc.13 ok 430 write_default.titac5x1xsoundc.13 ok 431 write_valid.titac5x1xsoundc.13 ok 432 write_invalid.titac5x1xsoundc.13 ok 433 event_missing.titac5x1xsoundc.13 ok 434 event_spurious.titac5x1xsoundc.13 ok 435 get_value.titac5x1xsoundc.12 # titac5x1xsoundc.12 ASI_RX_CH5_EN Playback ok 436 name.titac5x1xsoundc.12 ok 437 write_default.titac5x1xsoundc.12 ok 438 write_valid.titac5x1xsoundc.12 ok 439 write_invalid.titac5x1xsoundc.12 ok 440 event_missing.titac5x1xsoundc.12 ok 441 event_spurious.titac5x1xsoundc.12 ok 442 get_value.titac5x1xsoundc.11 # titac5x1xsoundc.11 ASI_RX_CH4_EN Playback Switch ok 443 name.titac5x1xsoundc.11 ok 444 write_default.titac5x1xsoundc.11 ok 445 write_valid.titac5x1xsoundc.11 ok 446 write_invalid.titac5x1xsoundc.11 ok 447 event_missing.titac5x1xsoundc.11 ok 448 event_spurious.titac5x1xsoundc.11 ok 449 get_value.titac5x1xsoundc.10 # titac5x1xsoundc.10 ASI_RX_CH3_EN Playback Switch ok 450 name.titac5x1xsoundc.10 ok 451 write_default.titac5x1xsoundc.10 ok 452 write_valid.titac5x1xsoundc.10 ok 453 write_invalid.titac5x1xsoundc.10 ok 454 event_missing.titac5x1xsoundc.10 ok 455 event_spurious.titac5x1xsoundc.10 ok 456 get_value.titac5x1xsoundc.9 # titac5x1xsoundc.9 DAC1 CHB Gain Calibration Playback Volume ok 457 name.titac5x1xsoundc.9 ok 458 write_default.titac5x1xsoundc.9 ok 459 write_valid.titac5x1xsoundc.9 ok 460 write_invalid.titac5x1xsoundc.9 ok 461 event_missing.titac5x1xsoundc.9 ok 462 event_spurious.titac5x1xsoundc.9 ok 463 get_value.titac5x1xsoundc.8 # titac5x1xsoundc.8 DAC1 CHA Gain Calibration Playback Volume ok 464 name.titac5x1xsoundc.8 ok 465 write_default.titac5x1xsoundc.8 ok 466 write_valid.titac5x1xsoundc.8 ok 467 write_invalid.titac5x1xsoundc.8 ok 468 event_missing.titac5x1xsoundc.8 ok 469 event_spurious.titac5x1xsoundc.8 ok 470 get_value.titac5x1xsoundc.7 # titac5x1xsoundc.7 DAC1 CHB Digital Playback Volume ok 471 name.titac5x1xsoundc.7 ok 472 write_default.titac5x1xsoundc.7 ok 473 write_valid.titac5x1xsoundc.7 ok 474 write_invalid.titac5x1xsoundc.7 ok 475 event_missing.titac5x1xsoundc.7 ok 476 event_spurious.titac5x1xsoundc.7 ok 477 get_value.titac5x1xsoundc.6 # titac5x1xsoundc.6 DAC1 CHA Digital Playback Volume ok 478 name.titac5x1xsoundc.6 ok 479 write_default.titac5x1xsoundc.6 ok 480 write_valid.titac5x1xsoundc.6 ok 481 write_invalid.titac5x1xsoundc.6 ok 482 event_missing.titac5x1xsoundc.6 ok 483 event_spurious.titac5x1xsoundc.6 ok 484 get_value.titac5x1xsoundc.5 # titac5x1xsoundc.5 OUT1M Analog Level Playback Volume ok 485 name.titac5x1xsoundc.5 ok 486 write_default.titac5x1xsoundc.5 ok 487 write_valid.titac5x1xsoundc.5 ok 488 write_invalid.titac5x1xsoundc.5 ok 489 event_missing.titac5x1xsoundc.5 ok 490 event_spurious.titac5x1xsoundc.5 ok 491 get_value.titac5x1xsoundc.4 # titac5x1xsoundc.4 OUT1P Analog Level Playback Volume ok 492 name.titac5x1xsoundc.4 ok 493 write_default.titac5x1xsoundc.4 ok 494 write_valid.titac5x1xsoundc.4 ok 495 write_invalid.titac5x1xsoundc.4 ok 496 event_missing.titac5x1xsoundc.4 ok 497 event_spurious.titac5x1xsoundc.4 ok 498 get_value.titac5x1xsoundc.3 # titac5x1xsoundc.3 DAC1 Audio BW ok 499 name.titac5x1xsoundc.3 ok 500 write_default.titac5x1xsoundc.3 ok 501 write_valid.titac5x1xsoundc.3 ok 502 write_invalid.titac5x1xsoundc.3 ok 503 event_missing.titac5x1xsoundc.3 ok 504 event_spurious.titac5x1xsoundc.3 ok 505 get_value.titac5x1xsoundc.2 # titac5x1xsoundc.2 Playback Decimation Filter ok 506 name.titac5x1xsoundc.2 ok 507 write_default.titac5x1xsoundc.2 ok 508 write_valid.titac5x1xsoundc.2 ok 509 write_invalid.titac5x1xsoundc.2 ok 510 event_missing.titac5x1xsoundc.2 ok 511 event_spurious.titac5x1xsoundc.2 ok 512 get_value.titac5x1xsoundc.1 # titac5x1xsoundc.1 ADC2 Common-mode Tolerance ok 513 name.titac5x1xsoundc.1 ok 514 write_default.titac5x1xsoundc.1 ok 515 write_valid.titac5x1xsoundc.1 ok 516 write_invalid.titac5x1xsoundc.1 ok 517 event_missing.titac5x1xsoundc.1 ok 518 event_spurious.titac5x1xsoundc.1 ok 519 get_value.titac5x1xsoundc.0 # titac5x1xsoundc.0 ADC1 Common-mode Tolerance ok 520 name.titac5x1xsoundc.0 ok 521 write_default.titac5x1xsoundc.0 ok 522 write_valid.titac5x1xsoundc.0 ok 523 write_invalid.titac5x1xsoundc.0 ok 524 event_missing.titac5x1xsoundc.0 ok 525 event_spurious.titac5x1xsoundc.0 # Totals: pass:525 fail:0 xfail:0 xpass:0 skip:0 error:0 Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> Niranjan H Y (4): ASoc: tac5x1x: add codec driver tac5x1x family dt-bindings: sound: bindings for tac5x1x family of codecs Asoc: pcm6240: remove support for taac5x1x family dt-bindings: sound: device support list for pcm6240 .../devicetree/bindings/sound/ti,pcm6240.yaml | 10 - .../devicetree/bindings/sound/ti,tac5x1x.yaml | 262 ++ sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm6240.c | 126 +- sound/soc/codecs/pcm6240.h | 4 - sound/soc/codecs/tac5x1x-i2c.c | 2137 +++++++++++++++++ sound/soc/codecs/tac5x1x.h | 300 +++ 8 files changed, 2716 insertions(+), 137 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tac5x1x.yaml create mode 100644 sound/soc/codecs/tac5x1x-i2c.c create mode 100644 sound/soc/codecs/tac5x1x.h -- 2.45.2

5 12

Zoom R24 USB multichannel audio interface no longer recognized
by Johan Vromans 27 Jun '25

27 Jun '25

I'm new here so my apologies if I'm asking questions at the wrong place. Someone told me this could be a regression in ALSA. I'm running Fedora and have used a Zoom R24 as a USB multichannel audio interface (8-in/2-out) for many years. Recently I upgraded to Fedora 42 and now the device is presented as dumb 2-in/2-out USB audio interface. "pavucontrol" does not present multi-channel profiles for the device. The command "alsamixer -c 1" (to select the R24) gives "This device does not have any controls.". "qjackctl" also shows 2-in/2-out channels. I use the device with "ardour" for multi-channel recording but that no longer works, only 2-in/2-out channels are available. How can I get the R24 fully functional again? Thanks,

2 2

Change description of HiFi verbs
by GitHub issues - opened 27 Jun '25

27 Jun '25

alsa-project/alsa-ucm-conf issue #586 was opened from ford-prefect: I wanted to raise this issue before creating a PR -- the string "Play HiFi quality Music" is not very good as a user-facing description for the output of their device's audio capability. The configuration enables both playback and capture (so "Play" is incomplete). There is nothing "Music"-specific about the output. And finally, "HiFi quality" feels like an anachronism at best. Historically, we've used "Analog Stereo Duplex" for playback/capture and "Analog Stereo Input" or "Analog Stereo Output" for unidirectional configuration (these strings are also well translated at this point). Another option is to use "Analog Audio", I suppose. Open to other options, if there are ideas. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/586 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Handling Lost Audio IRQs in ALSA: Triggering XRUN on Timer Detection?
by Sandeep Verma 27 Jun '25

27 Jun '25

Hi all,I’m currently investigating a sporadic issue in our system where an audio interrupt is occasionally lost. When this happens, ALSA detects the anomaly (presumably through its internal timestamp or delay detection), but it doesn’t appear to take corrective action—leading to persistent audio jitter afterward.To address this, I’m considering implementing a mechanism in our driver or platform layer that uses a timer to check if the expected audio interrupt hasn’t arrived within a certain timeframe (e.g., 2x the period size). If this condition is met, I plan to explicitly trigger an XRUN to reset the pipeline and recover cleanly.My questions are: 1. Is this an acceptable and “ALSA-friendly” way to handle lost IRQs? 2. If this is a reasonable approach, why doesn’t ALSA do this by default? 3. Is there a better or recommended way within the ALSA framework to detect and recover from such missed interrupts? I’d appreciate any guidance or suggestions. If this kind of timer-based recovery would be broadly useful, I’d also be happy to explore whether it could be proposed upstream. Thanks, Sandeep

2 1

Re: [PATCH v3 2/4] dt-bindings: sound: bindings for tac5x1x family of codecs
by Holalu Yogendra, Niranjan 27 Jun '25

27 Jun '25

Thanks for quick review. > From: Rob Herring (Arm) <robh(a)kernel.org> > Sent: Friday, June 27, 2025 1:09 AM > To: Holalu Yogendra, Niranjan <niranjan.hy(a)ti.com> > Subject: Re: [PATCH v3 2/4] dt-bindings: sound: bindings for > tac5x1x family of codecs > On Thu, 26 Jun 2025 23:43:31 +0530, Niranjan H Y wrote: > > Documentation/devicetree/bindings/sound/ti,tac5x1x.yaml ... > yamllint warnings/errors: > > dtschema/dtc warnings/errors: > Warning: Duplicate compatible "ti,taa5412" found in schemas matching > Warning: Duplicate compatible "ti,taa5212" found in schemas matching > Warning: Duplicate compatible "ti,tad5212" found in schemas matching The duplicate compatible are already removed as part of the patch series in patch [v3,4/4] dt-bindings: sound: device support list for pcm6240 . https://patchwork.ozlabs.org/project/devicetree-bindings/patch/202506261813… Please let me know if still additional action is required. Regards Niranjan H Y

1 1

Re: [PATCH v2 2/4] Asoc: tac5x1x: document tac5x1x codec
by Holalu Yogendra, Niranjan 26 Jun '25

26 Jun '25

Thank you for the time and review comment. > From: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> > Sent: Wednesday, June 25, 2025 8:01 PM > Subject: Re: [PATCH v2 2/4] Asoc: tac5x1x: document tac5x1x codec > > On Tue, Jun 24, 2025 at 09:07:40PM +0530, Niranjan H Y wrote: > > tac5x1x family are series of low-power and high performance > > mono/stereo audio codecs consists of ADC and DAC combinations. > > Aren't DT bindings changes supposed to follow the common template in the > Subject? Is it okay to change the subject in the next patch or keep using the same subject tracking? please suggest. > > More important, the Cc list missed DT mailing lists and maintainers. > What's going on here, please? Pardon my ignorance. I will address in the next patch. Regards Niranjan

1 0

[GIT PULL] ASoC fixes for v6.16-rc3
by Mark Brown 26 Jun '25

26 Jun '25

The following changes since commit 86731a2a651e58953fc949573895f2fa6d456841: Linux 6.16-rc3 (2025-06-22 13:30:08 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.16-rc3 for you to fetch changes up to 6c038b58a2dc5a008c7e7a1297f5aaa4deaaaa7e: ASoC: SOF: Intel: hda: Use devm_kstrdup() to avoid memleak. (2025-06-24 16:39:42 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.16 A small collection of fixes, the main one being a fix for resume from hibernation on AMD systems, plus a few new quirk entries for AMD systems. ---------------------------------------------------------------- Jack Yu (1): ASoC: rt721-sdca: fix boost gain calculation error Luca Weiss (1): ASoC: qcom: sm8250: Fix possibly undefined reference Oliver Schramm (1): ASoC: amd: yc: Add DMI quirk for Lenovo IdeaPad Slim 5 15 Tamura Dai (1): ASoC: SOF: Intel: hda: Use devm_kstrdup() to avoid memleak. Vijendar Mukunda (1): ASoC: amd: ps: fix for soundwire failures during hibernation exit sequence Yuzuru10 (1): ASoC: amd: yc: add quirk for Acer Nitro ANV15-41 internal mic sound/soc/amd/ps/acp63.h | 4 ++++ sound/soc/amd/ps/ps-common.c | 18 ++++++++++++++++++ sound/soc/amd/yc/acp6x-mach.c | 14 ++++++++++++++ sound/soc/codecs/rt721-sdca.c | 23 +++++++++++++++++++---- sound/soc/qcom/Kconfig | 1 + sound/soc/sof/intel/hda.c | 6 +++--- 6 files changed, 59 insertions(+), 7 deletions(-)

2 1

[PATCH 0/2] Add SoundWire machines for ACP7.0/ACP7.1 sof stack
by Vijendar Mukunda 25 Jun '25

25 Jun '25

Add SoundWire machines as alternate machines for ACP7.0 & ACP7.1 platforms with the below machine configuration. Link 0: RT722 codec with three endpoints: Headset, Speaker, and DMIC. Vijendar Mukunda (2): ASoC: amd: acp: add soundwire machine for ACP7.0 and ACP7.1 sof stack ASoC: SOF: amd: add alternate machines for ACP7.0 and ACP7.1 platforms sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 12 ++++++++++++ sound/soc/amd/mach-config.h | 1 + sound/soc/sof/amd/pci-acp70.c | 1 + 4 files changed, 15 insertions(+) -- 2.45.2

2 3

ucm2: codecs: lpass-rx-macro: move mixers that do not belong
by GitHub pull_request - opened 25 Jun '25

25 Jun '25

alsa-project/alsa-ucm-conf pull request #585 was opened from Srinivas-Kandagatla: Some of the mixers ended up in rx-macro which actually are part of WCD codecs. Move such mixers to their respective codecs. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/585 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/585.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH REQUEST] No speaker output on HP Spectre with ALC245 unless warm-booted from Windows
by Erik Johnsson 25 Jun '25

25 Jun '25

* PCI-ID: |SUBSYS_103C89DA| * Codec: Realtek ALC245 * Symptom: "No audio via internal speakers unless first booting into Windows" * |alsa-info-sh|: http://alsa-project.org/db/?f=01ca91462e838cf9d54472011a745914ea112c37

1 0

Improvements to Steinberg UR-44 device support
by GitHub pull_request - opened 24 Jun '25

24 Jun '25

alsa-project/alsa-ucm-conf pull request #584 was opened from peterix: - Relabeled the inputs and outputs to make sense considering the device's physical layout and internals - Added one more USB device ID for the UR44 (0x1704 is a non-C variant) - Added stereo inputs in sensible combinations for the various inputs on the device - 1 for inputs 5 & 6 in the back - 3 'adjacent ones' for the 4 inputs in the front - 1 & 2 - 2 & 3 - 3 & 4 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/584 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/584.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: rt721-sdca: fix boost gain calculation error
by Jack Yu 24 Jun '25

24 Jun '25

Fix the boost gain calculation error in rt721_sdca_set_gain_get. This patch is specific for "FU33 Boost Volume". Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt721-sdca.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c index 1c9f32e405cf..ba080957e933 100644 --- a/sound/soc/codecs/rt721-sdca.c +++ b/sound/soc/codecs/rt721-sdca.c @@ -430,6 +430,7 @@ static int rt721_sdca_set_gain_get(struct snd_kcontrol *kcontrol, unsigned int read_l, read_r, ctl_l = 0, ctl_r = 0; unsigned int adc_vol_flag = 0; const unsigned int interval_offset = 0xc0; + const unsigned int tendA = 0x200; const unsigned int tendB = 0xa00; if (strstr(ucontrol->id.name, "FU1E Capture Volume") || @@ -439,9 +440,16 @@ static int rt721_sdca_set_gain_get(struct snd_kcontrol *kcontrol, regmap_read(rt721->mbq_regmap, mc->reg, &read_l); regmap_read(rt721->mbq_regmap, mc->rreg, &read_r); - if (mc->shift == 8) /* boost gain */ + if (mc->shift == 8) { + /* boost gain */ ctl_l = read_l / tendB; - else { + } else if (mc->shift == 1) { + /* FU33 boost gain */ + if (read_l == 0x8000 || read_l == 0xfe00) + ctl_l = 0; + else + ctl_l = read_l / tendA + 1; + } else { if (adc_vol_flag) ctl_l = mc->max - (((0x1e00 - read_l) & 0xffff) / interval_offset); else @@ -449,9 +457,16 @@ static int rt721_sdca_set_gain_get(struct snd_kcontrol *kcontrol, } if (read_l != read_r) { - if (mc->shift == 8) /* boost gain */ + if (mc->shift == 8) { + /* boost gain */ ctl_r = read_r / tendB; - else { /* ADC/DAC gain */ + } else if (mc->shift == 1) { + /* FU33 boost gain */ + if (read_r == 0x8000 || read_r == 0xfe00) + ctl_r = 0; + else + ctl_r = read_r / tendA + 1; + } else { /* ADC/DAC gain */ if (adc_vol_flag) ctl_r = mc->max - (((0x1e00 - read_r) & 0xffff) / interval_offset); else -- 2.34.1

2 1

[PATCH V2] ASoC: amd: ps: fix for soundwire failures during hibernation exit sequence
by Vijendar Mukunda 24 Jun '25

24 Jun '25

During the hibernate entry sequence, ACP registers will be reset to default values and acp ip will be completely powered off including acp SoundWire pads. During resume sequence, if acp SoundWire pad keeper enable register is not restored along with pad pulldown control register value, then SoundWire manager links won't be powered on correctly results in peripheral register access failures and completely audio function is broken. Add code to store the acp SoundWire pad keeper enable register and acp pad pulldown ctrl register values before entering into suspend state and restore the register values during resume sequence based on condition check for acp SoundWire pad keeper enable register for ACP6.3, ACP7.0 & ACP7.1 platforms. Fixes: 491628388005 ("ASoC: amd: ps: add callback functions for acp pci driver pm ops") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- Changes since v1: removed extra macro ACP_SW_PAD_KEEPER_EN and used existing macro ACP_SW0_PAD_KEEPER_EN in the code. sound/soc/amd/ps/acp63.h | 4 ++++ sound/soc/amd/ps/ps-common.c | 18 ++++++++++++++++++ 2 files changed, 22 insertions(+) diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index 85feae45c44c..d7c994e26e4d 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -334,6 +334,8 @@ struct acp_hw_ops { * @addr: pci ioremap address * @reg_range: ACP reigister range * @acp_rev: ACP PCI revision id + * @acp_sw_pad_keeper_en: store acp SoundWire pad keeper enable register value + * @acp_pad_pulldown_ctrl: store acp pad pulldown control register value * @acp63_sdw0-dma_intr_stat: DMA interrupt status array for ACP6.3 platform SoundWire * manager-SW0 instance * @acp63_sdw_dma_intr_stat: DMA interrupt status array for ACP6.3 platform SoundWire @@ -367,6 +369,8 @@ struct acp63_dev_data { u32 addr; u32 reg_range; u32 acp_rev; + u32 acp_sw_pad_keeper_en; + u32 acp_pad_pulldown_ctrl; u16 acp63_sdw0_dma_intr_stat[ACP63_SDW0_DMA_MAX_STREAMS]; u16 acp63_sdw1_dma_intr_stat[ACP63_SDW1_DMA_MAX_STREAMS]; u16 acp70_sdw0_dma_intr_stat[ACP70_SDW0_DMA_MAX_STREAMS]; diff --git a/sound/soc/amd/ps/ps-common.c b/sound/soc/amd/ps/ps-common.c index 1c89fb5fe1da..7b4966b75dc6 100644 --- a/sound/soc/amd/ps/ps-common.c +++ b/sound/soc/amd/ps/ps-common.c @@ -160,6 +160,8 @@ static int __maybe_unused snd_acp63_suspend(struct device *dev) adata = dev_get_drvdata(dev); if (adata->is_sdw_dev) { + adata->acp_sw_pad_keeper_en = readl(adata->acp63_base + ACP_SW0_PAD_KEEPER_EN); + adata->acp_pad_pulldown_ctrl = readl(adata->acp63_base + ACP_PAD_PULLDOWN_CTRL); adata->sdw_en_stat = check_acp_sdw_enable_status(adata); if (adata->sdw_en_stat) { writel(1, adata->acp63_base + ACP_ZSC_DSP_CTRL); @@ -197,6 +199,7 @@ static int __maybe_unused snd_acp63_runtime_resume(struct device *dev) static int __maybe_unused snd_acp63_resume(struct device *dev) { struct acp63_dev_data *adata; + u32 acp_sw_pad_keeper_en; int ret; adata = dev_get_drvdata(dev); @@ -209,6 +212,12 @@ static int __maybe_unused snd_acp63_resume(struct device *dev) if (ret) dev_err(dev, "ACP init failed\n"); + acp_sw_pad_keeper_en = readl(adata->acp63_base + ACP_SW0_PAD_KEEPER_EN); + dev_dbg(dev, "ACP_SW0_PAD_KEEPER_EN:0x%x\n", acp_sw_pad_keeper_en); + if (!acp_sw_pad_keeper_en) { + writel(adata->acp_sw_pad_keeper_en, adata->acp63_base + ACP_SW0_PAD_KEEPER_EN); + writel(adata->acp_pad_pulldown_ctrl, adata->acp63_base + ACP_PAD_PULLDOWN_CTRL); + } return ret; } @@ -408,6 +417,8 @@ static int __maybe_unused snd_acp70_suspend(struct device *dev) adata = dev_get_drvdata(dev); if (adata->is_sdw_dev) { + adata->acp_sw_pad_keeper_en = readl(adata->acp63_base + ACP_SW0_PAD_KEEPER_EN); + adata->acp_pad_pulldown_ctrl = readl(adata->acp63_base + ACP_PAD_PULLDOWN_CTRL); adata->sdw_en_stat = check_acp_sdw_enable_status(adata); if (adata->sdw_en_stat) { writel(1, adata->acp63_base + ACP_ZSC_DSP_CTRL); @@ -445,6 +456,7 @@ static int __maybe_unused snd_acp70_runtime_resume(struct device *dev) static int __maybe_unused snd_acp70_resume(struct device *dev) { struct acp63_dev_data *adata; + u32 acp_sw_pad_keeper_en; int ret; adata = dev_get_drvdata(dev); @@ -459,6 +471,12 @@ static int __maybe_unused snd_acp70_resume(struct device *dev) if (ret) dev_err(dev, "ACP init failed\n"); + acp_sw_pad_keeper_en = readl(adata->acp63_base + ACP_SW0_PAD_KEEPER_EN); + dev_dbg(dev, "ACP_SW0_PAD_KEEPER_EN:0x%x\n", acp_sw_pad_keeper_en); + if (!acp_sw_pad_keeper_en) { + writel(adata->acp_sw_pad_keeper_en, adata->acp63_base + ACP_SW0_PAD_KEEPER_EN); + writel(adata->acp_pad_pulldown_ctrl, adata->acp63_base + ACP_PAD_PULLDOWN_CTRL); + } return ret; } -- 2.45.2

2 1

[PATCH] ASoC: amd: yc: add quirk for Acer Nitro ANV15-41 internal mic
by Yuzuru10 23 Jun '25

23 Jun '25

This patch adds DMI-based quirk for the Acer Nitro ANV15-41, allowing the internal microphone to be detected correctly on machines with "RB" as board vendor. Signed-off-by: Yuzuru <yuzuru_10(a)proton.me> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 98022e5fd428..499f9f7c76ee 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -353,6 +353,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "83Q3"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "RB"), + DMI_MATCH(DMI_PRODUCT_NAME, "Nitro ANV15-41"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.50.0

1 0

[PATCH] ASoC: amd: yc: Add DMI quirk for Lenovo IdeaPad Slim 5 15
by Oliver Schramm 23 Jun '25

23 Jun '25

It's smaller brother has already received the patch to enable the microphone, now add it too to the DMI quirk table. Cc: stable(a)vger.kernel.org Signed-off-by: Oliver Schramm <oliver.schramm97(a)gmail.com> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 98022e5fd428..c9e1b777b70e 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -360,6 +360,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "83J2"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "83J3"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.49.0

1 0

[PATCH] ASoC: amd: yc: add quirk for Acer Nitro ANV15-41 internal mic
by Yuzuru10 23 Jun '25

23 Jun '25

This patch adds DMI-based quirk for the Acer Nitro ANV15-41, allowing the internal microphone to be detected correctly on machines with "RB" as board vendor. --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 98022e5fd428..499f9f7c76ee 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -353,6 +353,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "83Q3"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "RB"), + DMI_MATCH(DMI_PRODUCT_NAME, "Nitro ANV15-41"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.50.0

1 0

Call to snd_pcm_hw_params_get_buffer_size causes memory corruption in calling program
by GitHub issues - opened 23 Jun '25

23 Jun '25

alsa-project/alsa-lib issue #461 was opened from harryrowe01: Calling snd_pcm_hw_params_get_buffer_size is causing memory corruption in the calling program. The following C program demonstrates the issue. /* This example opens the default PCM device, sets some parameters, and then displays the value of some of the hardware parameters. It does not perform any sound playback or recording. */ /* Use the newer ALSA API */ #define ALSA_PCM_NEW_HW_PARAMS_API /* All of the ALSA library API is defined * in this header */ #include <alsa/asoundlib.h> int main() { int rc; snd_pcm_t *handle; snd_pcm_hw_params_t *params, *save_params; unsigned int val, val2; int dir; snd_pcm_uframes_t frames; /* Open PCM device for playback. */ rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0); if (rc < 0) { fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc)); exit(1); } /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&params); /* Fill it in with default values. */ snd_pcm_hw_params_any(handle, params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Signed 16-bit little-endian format */ snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(handle, params, 2); /* 44100 bits/second sampling rate (CD quality) */ val = 44100; // val = 8000; snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir); /* Write the parameters to the driver */ rc = snd_pcm_hw_params(handle, params); if (rc < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(rc)); exit(1); } /* Display information about the PCM interface */ // Before calling snd_pcm_hw_params_get_buffer_size, save // a copy of params which points to the parameter block. save_params = params; // This call to snd_pcm_hw_params_get_buffer_size causes // the value of params in main to be corrupted. // Since the params argument to the function is passed // by value, the function itself should not have access to // the original variable in main. snd_pcm_hw_params_get_buffer_size(params, (snd_pcm_uframes_t *)&val); // Then if the pointer has been modified, flag the problem and // restore the pointer from the saved value. if( params != save_params ) { printf( "The pointer to the parameter block has been corrupted!\n" ); printf( "was %lx \n", save_params ); printf( "is now %lx \n", params ); printf( "So I'm fixing it.\n" ); params = save_params; } printf("buffer size = %u frames\n", val); snd_pcm_close(handle); return 0; } Compiled and run as follows: harry@idun:~/Documents/morse$ cc bad_alsa.c -lasound -o bad_alsa harry@idun:~/Documents/morse$ ./bad_alsa The pointer to the parameter block has been corrupted! was 7ffcd2011610 is now 7ffc00000000 So I'm fixing it. buffer size = 1048576 frames harry@idun:~/Documents/morse$ cc bad_alsa.c -lasound -o bad_alsa harry@idun:~/Documents/morse$ ./bad_alsa The pointer to the parameter block has been corrupted! was 7fffc7483030 is now 7fff00000000 So I'm fixing it. buffer size = 1048576 frames harry@idun:~/Documents/morse$ Host is an AMD64 version of Debian 12. Installed version of ALSA library is libasound2 1.2.8-1+b1 Issue URL : https://github.com/alsa-project/alsa-lib/issues/461 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Add UCM support for MediaTek mt8365-evk
by GitHub pull_request - opened 23 Jun '25

23 Jun '25

alsa-project/alsa-ucm-conf pull request #583 was opened from zoran055: Add support for MT8365 EVK with SOF enabled. When SOF is disabled, the "HiFi" use case is used; when SOF is enabled, the "SOF" use case is used. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/583 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/583.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe
by 이승빈 23 Jun '25

23 Jun '25

From: Seungbin Lee <seungbin.lee(a)samsung.com> Date: Wed, 18 Jun 2025 13:33:02 +0900 Subject: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe If the soc_compr_trigger_fe function is called during snd_soc_dpcm_runtime_update execution, the FE's DPCM runtime update variable may be changed. Then the BE list connected to the FE flies away, and the BE cannot be closed. This needs to be protected by the snd_soc_dpcm_mutex_lock. Change-Id: Ibfd1712e6baf41a30afa357a91ada83d205f968b Signed-off-by: Seungbin Lee <seungbin.lee(a)samsung.com> --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index fc8db54352541..1a2a3fe6a2340 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -286,6 +286,8 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) snd_soc_card_mutex_lock(fe->card); + snd_soc_dpcm_mutex_lock(fe); + ret = snd_soc_dai_compr_trigger(cpu_dai, cstream, cmd); if (ret < 0) goto out; @@ -315,6 +317,8 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) out: fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + snd_soc_dpcm_mutex_unlock(fe); + snd_soc_card_mutex_unlock(fe->card); return ret; } -- 2.34.1

1 0

FW: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe
by 이승빈 23 Jun '25

23 Jun '25

Add recipient addition and modification patch to the body of the transmission. From: 이승빈 <seungbin.lee(a)samsung.com <mailto:seungbin.lee@samsung.com> > Sent: Thursday, June 19, 2025 10:25 AM To: 'tiwai(a)suse.de' <tiwai(a)suse.de <mailto:tiwai@suse.de> > Subject: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe Hi I am sending this email to report an issue found in sound/soc/soc- compress.c. Scenario: Playing a mix of MP3 and WAV files repeatedly from YouTube Music by pressing "next". Issue occurs when snd_soc_dpcm_runtime_update is called and soc_compr_trigger_fe is invoked, causing the value of “fe- >dpcm[stream].runtime_update” to change mid-process. This results in a problem where, during the closing of the BE connected to the FE in snd_soc_dpcm_runtime_update, only the entry is removed from the list without actually being closed. The attached patch resolves this issue as confirmed through testing. Additionally, while performing DPCM related operations in soc-compress.c, snd_soc_dpcm_mutex_lock is used; however, soc_compr_trigger_fe does not have this lock. Please review the attached patch regarding this matter. Thank you. From 72a0340bf7dccc6f53dc2d02e04bb3f894d796e3 Mon Sep 17 00:00:00 2001 From: Seungbin Lee <seungbin.lee(a)samsung.com <mailto:seungbin.lee@samsung.com> > Date: Wed, 18 Jun 2025 13:33:02 +0900 Subject: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe If the soc_compr_trigger_fe function is called during snd_soc_dpcm_runtime_update execution, the FE's DPCM runtime update variable may be changed. Then the BE list connected to the FE flies away, and the BE cannot be closed. This needs to be protected by the snd_soc_dpcm_mutex_lock. Change-Id: Ibfd1712e6baf41a30afa357a91ada83d205f968b Signed-off-by: Seungbin Lee <seungbin.lee(a)samsung.com <mailto:seungbin.lee@samsung.com> > --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index fc8db54352541..1a2a3fe6a2340 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -286,6 +286,8 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) snd_soc_card_mutex_lock(fe->card); + snd_soc_dpcm_mutex_lock(fe); + ret = snd_soc_dai_compr_trigger(cpu_dai, cstream, cmd); if (ret < 0) goto out; @@ -315,6 +317,8 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) out: fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + snd_soc_dpcm_mutex_unlock(fe); + snd_soc_card_mutex_unlock(fe->card); return ret; } -- 2.34.1

1 0

RE: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe
by 이승빈 23 Jun '25

23 Jun '25

I am sending it again after changing it to plain text mail. Thanks From: 이승빈 <seungbin.lee(a)samsung.com> Sent: Monday, June 23, 2025 10:31 AM To: 'alsa-devel(a)alsa-project.org' <alsa-devel(a)alsa-project.org>; 'linux-sound(a)vger.kernel.org' <linux-sound(a)vger.kernel.org>; 'gregkh(a)linuxfoundation.org' <gregkh(a)linuxfoundation.org>; 'pierre-louis.bossart(a)linux.intel.com' <pierre-louis.bossart(a)linux.intel.com>; 'perex(a)perex.cz' <perex(a)perex.cz>; 'tiwai(a)suse.de' <tiwai(a)suse.de> Cc: 'pilsun.jang(a)samsung.com' <pilsun.jang(a)samsung.com>; 's47.kang(a)samsung.com' <s47.kang(a)samsung.com>; 'donghee.moon(a)samsung.com' <donghee.moon(a)samsung.com> Subject: FW: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe Add recipient addition and modification patch to the body of the transmission. From: 이승빈 <mailto:seungbin.lee@samsung.com> Sent: Thursday, June 19, 2025 10:25 AM To: 'tiwai(a)suse.de' <mailto:tiwai@suse.de> Subject: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe Hi I am sending this email to report an issue found in sound/soc/soc-compress.c. Scenario: Playing a mix of MP3 and WAV files repeatedly from YouTube Music by pressing "next". Issue occurs when snd_soc_dpcm_runtime_update is called and soc_compr_trigger_fe is invoked, causing the value of “fe->dpcm[stream].runtime_update” to change mid-process. This results in a problem where, during the closing of the BE connected to the FE in snd_soc_dpcm_runtime_update, only the entry is removed from the list without actually being closed. The attached patch resolves this issue as confirmed through testing. Additionally, while performing DPCM related operations in soc-compress.c, snd_soc_dpcm_mutex_lock is used; however, soc_compr_trigger_fe does not have this lock. Please review the attached patch regarding this matter. Thank you. From 72a0340bf7dccc6f53dc2d02e04bb3f894d796e3 Mon Sep 17 00:00:00 2001 From: Seungbin Lee <mailto:seungbin.lee@samsung.com> Date: Wed, 18 Jun 2025 13:33:02 +0900 Subject: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe If the soc_compr_trigger_fe function is called during snd_soc_dpcm_runtime_update execution, the FE's DPCM runtime update variable may be changed. Then the BE list connected to the FE flies away, and the BE cannot be closed. This needs to be protected by the snd_soc_dpcm_mutex_lock. Change-Id: Ibfd1712e6baf41a30afa357a91ada83d205f968b Signed-off-by: Seungbin Lee <mailto:seungbin.lee@samsung.com> --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index fc8db54352541..1a2a3fe6a2340 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -286,6 +286,8 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) snd_soc_card_mutex_lock(fe->card); + snd_soc_dpcm_mutex_lock(fe); + ret = snd_soc_dai_compr_trigger(cpu_dai, cstream, cmd); if (ret < 0) goto out; @@ -315,6 +317,8 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) out: fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + snd_soc_dpcm_mutex_unlock(fe); + snd_soc_card_mutex_unlock(fe->card); return ret; } -- 2.34.1

1 0

[PATCH] ALSA: hda/realtek: add fixup for HP Spectre x360 14t-ea100 (ALC245)
by Kaden Berger 22 Jun '25

22 Jun '25

This patch adds a quirk to enable proper speaker output on the HP Spectre x360 14t-ea100 with the Realtek ALC245 code. SIgned-off-by: Kaden Berger <kadenb816(a)gmail.com> diff -ura linux-6.15.1.orig/sound/pci/hda/patch_realtek.c linux-6.15.1.new/sound/pci/hda/patch_realtek.c --- linux-6.15.1.orig/sound/pci/hda/patch_realtek.c 2025-06-09 08:21:07.178183524 -0500 +++ linux-6.15.1.new/sound/pci/hda/patch_realtek.c 2025-06-09 08:26:16.859486062 -0500 @@ -10792,6 +10793,7 @@ SND_PCI_QUIRK(0x103c, 0x8be9, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c15, "HP Spectre x360 2-in-1 Laptop 14-eu0xxx", ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX), + SND_PCI_QUIRK(0x103c, 0x89da, "HP Spectre x360 14t-ea100", ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX), SND_PCI_QUIRK(0x103c, 0x8c16, "HP Spectre x360 2-in-1 Laptop 16-aa0xxx", ALC245_FIXUP_HP_SPECTRE_X360_16_AA0XXX), SND_PCI_QUIRK(0x103c, 0x8c17, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c21, "HP Pavilion Plus Laptop 14-ey0XXX", ALC245_FIXUP_HP_X360_MUTE_LEDS),

2 1

[PATCH v2] ALSA: hda/realtek: Add quirks for some Clevo laptops
by Tim Crawford 22 Jun '25

22 Jun '25

Add audio quirks to fix speaker output and headset detection on the following Clevo models: - V350ENC - V350WNPQ - V540TU - X560WNR - X580WNS Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cfcc204264aa..13bea7863bef 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2657,6 +2657,7 @@ static const struct hda_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1558, 0x3702, "Clevo X370SN[VW]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x50d3, "Clevo PC50[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x5802, "Clevo X58[05]WN[RST]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65d1, "Clevo PB51[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65d2, "Clevo PB51R[CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65e1, "Clevo PB51[ED][DF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), @@ -11131,6 +11132,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x2624, "Clevo L240TU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x28c1, "Clevo V370VND", ALC2XX_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x35a1, "Clevo V3[56]0EN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x35b1, "Clevo V3[57]0WN[MNP]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -11158,6 +11161,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x51b1, "Clevo NS50AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x51b3, "Clevo NS70AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x5630, "Clevo NP50RNJS", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x5700, "Clevo X560WN[RST]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f2, "Clevo NH79EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -11197,6 +11201,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0xa650, "Clevo NP[567]0SN[CD]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa741, "Clevo V54x_6x_TNE", ALC245_FIXUP_CLEVO_NOISY_MIC), + SND_PCI_QUIRK(0x1558, 0xa743, "Clevo V54x_6x_TU", ALC245_FIXUP_CLEVO_NOISY_MIC), SND_PCI_QUIRK(0x1558, 0xa763, "Clevo V54x_6x_TU", ALC245_FIXUP_CLEVO_NOISY_MIC), SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.49.0

2 1

ucm2: USB-Audio: Add Teufel CAGE PRO
by GitHub pull_request - opened 21 Jun '25

21 Jun '25

alsa-project/alsa-ucm-conf pull request #582 was opened from hsitter: two stereo outputs: one for "game" and one for "chat". one mono input Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/582 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/582.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: amd: ps: fix for soundwire failures during hibernation exit sequence
by Vijendar Mukunda 21 Jun '25

21 Jun '25

During the hibernate entry sequence, ACP registers will be reset to default values and acp ip will be completely powered off including acp SoundWire pads. During resume sequence, if acp SoundWire pad keeper enable register is not restored along with pad pulldown control register value, then SoundWire manager links won't be powered on correctly results in peripheral register access failures and completely audio function is broken. Add code to store the acp SoundWire pad keeper enable register and acp pad pulldown ctrl register values before entering into suspend state and restore the register values during resume sequence based on condition check for acp SoundWire pad keeper enable register for ACP6.3, ACP7.0 & ACP7.1 platforms. Fixes: 491628388005 ("ASoC: amd: ps: add callback functions for acp pci driver pm ops") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/acp63.h | 5 +++++ sound/soc/amd/ps/ps-common.c | 18 ++++++++++++++++++ 2 files changed, 23 insertions(+) diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index 85feae45c44c..babf2fa7ea27 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -82,6 +82,7 @@ #define ACP63_SDW0_DMA_MAX_STREAMS 6 #define ACP63_SDW1_DMA_MAX_STREAMS 2 #define ACP63_P1_AUDIO_TX_THRESHOLD 6 +#define ACP_SW_PAD_KEEPER_EN 0x0001454 /* * Below entries describes SDW0 instance DMA stream id and DMA irq bit mapping @@ -334,6 +335,8 @@ struct acp_hw_ops { * @addr: pci ioremap address * @reg_range: ACP reigister range * @acp_rev: ACP PCI revision id + * @acp_sw_pad_keeper_en: store acp SoundWire pad keeper enable register value + * @acp_pad_pulldown_ctrl: store acp pad pulldown control register value * @acp63_sdw0-dma_intr_stat: DMA interrupt status array for ACP6.3 platform SoundWire * manager-SW0 instance * @acp63_sdw_dma_intr_stat: DMA interrupt status array for ACP6.3 platform SoundWire @@ -367,6 +370,8 @@ struct acp63_dev_data { u32 addr; u32 reg_range; u32 acp_rev; + u32 acp_sw_pad_keeper_en; + u32 acp_pad_pulldown_ctrl; u16 acp63_sdw0_dma_intr_stat[ACP63_SDW0_DMA_MAX_STREAMS]; u16 acp63_sdw1_dma_intr_stat[ACP63_SDW1_DMA_MAX_STREAMS]; u16 acp70_sdw0_dma_intr_stat[ACP70_SDW0_DMA_MAX_STREAMS]; diff --git a/sound/soc/amd/ps/ps-common.c b/sound/soc/amd/ps/ps-common.c index 1c89fb5fe1da..f18d2a0d83aa 100644 --- a/sound/soc/amd/ps/ps-common.c +++ b/sound/soc/amd/ps/ps-common.c @@ -160,6 +160,8 @@ static int __maybe_unused snd_acp63_suspend(struct device *dev) adata = dev_get_drvdata(dev); if (adata->is_sdw_dev) { + adata->acp_sw_pad_keeper_en = readl(adata->acp63_base + ACP_SW_PAD_KEEPER_EN); + adata->acp_pad_pulldown_ctrl = readl(adata->acp63_base + ACP_PAD_PULLDOWN_CTRL); adata->sdw_en_stat = check_acp_sdw_enable_status(adata); if (adata->sdw_en_stat) { writel(1, adata->acp63_base + ACP_ZSC_DSP_CTRL); @@ -197,6 +199,7 @@ static int __maybe_unused snd_acp63_runtime_resume(struct device *dev) static int __maybe_unused snd_acp63_resume(struct device *dev) { struct acp63_dev_data *adata; + u32 acp_sw_pad_keeper_en; int ret; adata = dev_get_drvdata(dev); @@ -209,6 +212,12 @@ static int __maybe_unused snd_acp63_resume(struct device *dev) if (ret) dev_err(dev, "ACP init failed\n"); + acp_sw_pad_keeper_en = readl(adata->acp63_base + ACP_SW_PAD_KEEPER_EN); + dev_dbg(dev, "ACP_SW_PAD_KEEPER_EN:0x%x\n", acp_sw_pad_keeper_en); + if (!acp_sw_pad_keeper_en) { + writel(adata->acp_sw_pad_keeper_en, adata->acp63_base + ACP_SW_PAD_KEEPER_EN); + writel(adata->acp_pad_pulldown_ctrl, adata->acp63_base + ACP_PAD_PULLDOWN_CTRL); + } return ret; } @@ -408,6 +417,8 @@ static int __maybe_unused snd_acp70_suspend(struct device *dev) adata = dev_get_drvdata(dev); if (adata->is_sdw_dev) { + adata->acp_sw_pad_keeper_en = readl(adata->acp63_base + ACP_SW0_PAD_KEEPER_EN); + adata->acp_pad_pulldown_ctrl = readl(adata->acp63_base + ACP_PAD_PULLDOWN_CTRL); adata->sdw_en_stat = check_acp_sdw_enable_status(adata); if (adata->sdw_en_stat) { writel(1, adata->acp63_base + ACP_ZSC_DSP_CTRL); @@ -445,6 +456,7 @@ static int __maybe_unused snd_acp70_runtime_resume(struct device *dev) static int __maybe_unused snd_acp70_resume(struct device *dev) { struct acp63_dev_data *adata; + u32 acp_sw_pad_keeper_en; int ret; adata = dev_get_drvdata(dev); @@ -459,6 +471,12 @@ static int __maybe_unused snd_acp70_resume(struct device *dev) if (ret) dev_err(dev, "ACP init failed\n"); + acp_sw_pad_keeper_en = readl(adata->acp63_base + ACP_SW_PAD_KEEPER_EN); + dev_dbg(dev, "ACP_SW_PAD_KEEPER_EN:0x%x\n", acp_sw_pad_keeper_en); + if (!acp_sw_pad_keeper_en) { + writel(adata->acp_sw_pad_keeper_en, adata->acp63_base + ACP_SW0_PAD_KEEPER_EN); + writel(adata->acp_pad_pulldown_ctrl, adata->acp63_base + ACP_PAD_PULLDOWN_CTRL); + } return ret; } -- 2.45.2

2 1

[PATCH] ALSA: hda/realtek: Add quirks for some Clevo laptops
by Tim Crawford 20 Jun '25

20 Jun '25

Add audio quirks to fix speaker output and headset detection on the following Clevo models: - V350ENC - V350WNPQ - V540TU - X560WNR - X580WNS Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cfcc204264aa..54d6edf203a1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11131,6 +11131,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x2624, "Clevo L240TU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x28c1, "Clevo V370VND", ALC2XX_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x35a1, "Clevo V3[56]0EN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x35b1, "Clevo V3[57]0WN[MNP]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -11158,6 +11160,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x51b1, "Clevo NS50AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x51b3, "Clevo NS70AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x5630, "Clevo NP50RNJS", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x5700, "Clevo X560WN[RST]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f2, "Clevo NH79EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -11197,6 +11200,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0xa650, "Clevo NP[567]0SN[CD]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa741, "Clevo V54x_6x_TNE", ALC245_FIXUP_CLEVO_NOISY_MIC), + SND_PCI_QUIRK(0x1558, 0xa743, "Clevo V54x_6x_TU", ALC245_FIXUP_CLEVO_NOISY_MIC), SND_PCI_QUIRK(0x1558, 0xa763, "Clevo V54x_6x_TU", ALC245_FIXUP_CLEVO_NOISY_MIC), SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.49.0

1 0

aplay.1: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 20 Jun '25

20 Jun '25

Package: alsa-utils Version: 1.2.14-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use grep -n -e ' $' -e '\\~$' -e ' \\f.$' -e ' \\"' <file> to find (most) trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? troff:<stdin>:258: warning: trailing space in the line * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: 13.0 APT prefers testing-security APT policy: (500, 'testing-security'), (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.32-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 34.2-2 ii libasound2t64 1.2.14-1 ii libatopology2t64 1.2.14-1 ii libc6 2.41-8 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5+20250216-2 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5+20250216-2 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

alsactl.1: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 20 Jun '25

20 Jun '25

Package: alsa-utils Version: 1.2.14-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use grep -n -e ' $' -e '\\~$' -e ' \\f.$' -e ' \\"' <file> to find (most) trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? troff:<stdin>:224: warning: trailing space in the line * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: 13.0 APT prefers testing-security APT policy: (500, 'testing-security'), (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.32-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 34.2-2 ii libasound2t64 1.2.14-1 ii libatopology2t64 1.2.14-1 ii libc6 2.41-8 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5+20250216-2 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5+20250216-2 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

Handling Lost Audio IRQs in ALSA: Triggering XRUN on Timer Detection?
by Verma, Sandeep 20 Jun '25

20 Jun '25

Hi all, I’m currently investigating a sporadic issue in our system where an audio interrupt is occasionally lost. When this happens, ALSA detects the anomaly (presumably through its internal timestamp or delay detection), but it doesn’t appear to take corrective action—leading to persistent audio jitter afterward. To address this, I’m considering implementing a mechanism in our driver or platform layer that uses a timer to check if the expected audio interrupt hasn’t arrived within a certain timeframe (e.g., 2x the period size). If this condition is met, I plan to explicitly trigger an XRUN to reset the pipeline and recover cleanly. My questions are: a. Is this an acceptable and “ALSA-friendly” way to handle lost IRQs? b. If this is a reasonable approach, why doesn’t ALSA do this by default? c. Is there a better or recommended way within the ALSA framework to detect and recover from such missed interrupts? I’d appreciate any guidance or suggestions. If this kind of timer-based recovery would be broadly useful, I’d also be happy to explore whether it could be proposed upstream. Thanks, Sandeep

1 0

[PATCH v2] ALSA: hda/realtek: Fixup ft alc257 rename alc3328
by wangdicheng 20 Jun '25

20 Jun '25

Audio ALC3328 recognized as ALC257, updated PCI ID0x10EC12F0 to rename it to 3328 Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- 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 cd0d7ba7320e..2a08115d8c17 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1088,6 +1088,7 @@ static const struct alc_codec_rename_pci_table rename_pci_tbl[] = { { 0x10ec0668, 0x103c, 0, "ALC3662" }, { 0x10ec0283, 0x17aa, 0, "ALC3239" }, { 0x10ec0292, 0x17aa, 0, "ALC3232" }, + { 0x10ec0257, 0x12f0, 0, "ALC3328" }, { } /* terminator */ }; -- 2.25.1

2 1

[GIT PULL] ASoC fixes for v6.16-rc2
by Mark Brown 20 Jun '25

20 Jun '25

The following changes since commit 19272b37aa4f83ca52bdf9c16d5d81bdd1354494: Linux 6.16-rc1 (2025-06-08 13:44:43 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.16-rc2 for you to fetch changes up to 16ea4666bbb7f5bd1130fa2d75631ccf8b62362e: ASoC: Intel: sof-function-topology-lib: Print out the unsupported dmic count (2025-06-19 15:44:42 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.16 A relatively large collection of fixes and updates that came in since the merge window. Of note are a couple of Cirrus ones which change the firmware naming for some newly added devices, and a fix from Laurentiu for issues booting firmwares on the DSPs on i.MX8 SoCs. ---------------------------------------------------------------- Gabriel Santese (1): ASoC: amd: yc: Add quirk for MSI Bravo 17 D7VF internal mic Harshit Mogalapalli (1): ASoC: cs48l32: Fix a signedness bug in cs48l32_hw_params() Huacai Chen (1): ASoC: loongson: Fix build warnings about export.h Laurentiu Mihalcea (1): ASoC: SOF: imx8: add core shutdown operation for imx8/imx8x Mark Brown (1): ASoC: cs35l56: Change firmware filenames for SoundWire Peter Ujfalusi (1): ASoC: Intel: sof-function-topology-lib: Print out the unsupported dmic count Raven Black (1): ASoC: amd: yc: update quirk data for HP Victus Richard Fitzgerald (4): ASoC: cs35l56: Use SoundWire address as firmware name suffix for new silicon ASoC: cs35l56: Use SoundWire address as alternate firmware suffix on L56 B0 ASoC: doc: cs35l56: Update to add new SoundWire firmware filename suffix ASoC: doc: cs35l56: Add CS35L63 to the list of supported devices Sven Peter (1): ASoC: apple: mca: Drop default ARCH_APPLE in Kconfig Thorsten Blum (1): ASoC: sdw_utils: Fix potential NULL pointer deref in is_sdca_endpoint_present() Zhang Yi (1): ASoC: codecs: ES8326: Modify initialization configuration Documentation/sound/codecs/cs35l56.rst | 24 ++++++-- sound/soc/amd/yc/acp6x-mach.c | 14 +++++ sound/soc/apple/Kconfig | 1 - sound/soc/codecs/cs35l56-sdw.c | 18 +++--- sound/soc/codecs/cs35l56.c | 72 +++++++++++++++++++--- sound/soc/codecs/cs35l56.h | 3 + sound/soc/codecs/cs48l32.c | 4 ++ sound/soc/codecs/es8326.c | 3 +- sound/soc/codecs/wm_adsp.c | 27 +++++--- sound/soc/codecs/wm_adsp.h | 2 + sound/soc/intel/common/sof-function-topology-lib.c | 3 +- sound/soc/loongson/loongson_i2s.c | 1 + sound/soc/sdw_utils/soc_sdw_utils.c | 2 + sound/soc/sof/imx/imx8.c | 15 +++++ 14 files changed, 154 insertions(+), 35 deletions(-)

2 1

Unable to see my Focusrite Scarlett Solo 4th gen
by GitHub issues - opened 18 Jun '25

18 Jun '25

alsa-project/alsa-lib issue #460 was opened from thereticle: Hi, a few days ago I bought a Focusrite Scarlett Solo 4th gen. It worked for a while, but since I had problems with Davinci Resolve (crackling sound) I tried to fix it without success until I could not see the Scarlett anymore. Actually lsusb sees the card Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 001 Device 002: ID 0db0:0076 Micro Star International MYSTIC LIGHT Bus 001 Device 007: ID 1235:8218 Focusrite-Novation Scarlett Solo 4th Gen Bus 002 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub [...] but alsa does not see it. After many attempts with the help of ChatGPT and other IA, I thought I had solved it by installing the alsa-ucm-conf package, but after the next reboot the card disappeared again. I use Xubuntu 25.04 with Pipewire (but I also tried with pulseaudio without success). Any suggestions? Issue URL : https://github.com/alsa-project/alsa-lib/issues/460 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ALSA: hda/conexant: Renaming the codec with device ID 0x1f86 and 0x1f87
by wangdicheng 18 Jun '25

18 Jun '25

From: wangdicheng <wangdicheng(a)kylinos.cn> Due to changes in the manufacturer's plan, all 0x14f11f86 will be named CX11880, and 0x14f11f87 will be named SN6140 Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/pci/hda/patch_conexant.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4985e72b9094..1366a33de8d3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -42,7 +42,7 @@ struct conexant_spec { unsigned int gpio_led; unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; - bool is_cx8070_sn6140; + bool is_cx11880_sn6140; }; @@ -195,7 +195,7 @@ static int cx_auto_init(struct hda_codec *codec) cxt_init_gpio_led(codec); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); - if (spec->is_cx8070_sn6140) + if (spec->is_cx11880_sn6140) cx_fixup_headset_recog(codec); return 0; @@ -247,7 +247,7 @@ static void cx_update_headset_mic_vref(struct hda_codec *codec, struct hda_jack_ { unsigned int mic_present; - /* In cx8070 and sn6140, the node 16 can only be configured to headphone or disabled, + /* In cx11880 and sn6140, the node 16 can only be configured to headphone or disabled, * the node 19 can only be configured to microphone or disabled. * Check hp&mic tag to process headset plugin & plugout. */ @@ -1191,11 +1191,11 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->spec = spec; codec->patch_ops = cx_auto_patch_ops; - /* init cx8070/sn6140 flag and reset headset_present_flag */ + /* init cx11880/sn6140 flag and reset headset_present_flag */ switch (codec->core.vendor_id) { case 0x14f11f86: case 0x14f11f87: - spec->is_cx8070_sn6140 = true; + spec->is_cx11880_sn6140 = true; snd_hda_jack_detect_enable_callback(codec, 0x19, cx_update_headset_mic_vref); break; } @@ -1283,7 +1283,7 @@ static int patch_conexant_auto(struct hda_codec *codec) */ static const struct hda_device_id snd_hda_id_conexant[] = { - HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f11f86, "CX11880", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f11f87, "SN6140", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f120d0, "CX11970", patch_conexant_auto), -- 2.25.1

2 1

[alsa-devel] ASoC TDM channel re-mapping
by Edward Cragg 17 Jun '25

17 Jun '25

Hi, We have an ASoC driver in which we have added support for TDM over an I2S interface. We would like to have the capability of re-mapping the channel ordering, for which i know there is a callback in place to start this process, set_channel_map. The only backend implementation of this is using the ALSA API and the .copy handler in an ALSA PCM driver to mutate data copied out of ALSA before it's DMA'd to hardware. This is in sound/soc/blackfin/bf5xx-i2s-pcm.c The problem we have is that the driver we're using currently uses a dmaengine based PCM driver, and i'm not sure whether this is compatible with using the .copy callback, without reimplementing the driver entirely as an ALSA PCM driver. Since there aren't very many examples in mainline, I was hoping to ask for advice for the best way of doing this. I wondered: - Would it be possible to add a functional copy callback using a dmaengine based PCM driver? - Otherwise, would there be a good way to implement channel remapping in a TDM frame using the dmaengine based PCM driver? or - Would the best way indeed be to reimplement the PCM driver as a normal ALSA PCM driver? Thanks, Ed Cragg

2 2

[PATCH] ASoC: qcom: soundwire: Add support for controlling audio CGCR from HLOS
by Srinivasa Rao Mandadapu 17 Jun '25

17 Jun '25

Add support for controlling soundwire audio CGCR interface using clock framework to make hclk ungating with software. As per new hardware changes, software has to always ungate hclk if soundwire is operational and keep it running. This requirement is for latest LPASS chipsets for RX, TX and WSA path to work. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> --- This patch set depends on: -- Clock driver patches with reset control support, yet to be posted. drivers/soundwire/qcom.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index fb183bd..440aad8 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -12,6 +12,7 @@ #include <linux/of_irq.h> #include <linux/of_device.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <linux/slab.h> #include <linux/slimbus.h> #include <linux/soundwire/sdw.h> @@ -129,6 +130,7 @@ struct qcom_swrm_ctrl { struct device *dev; struct regmap *regmap; void __iomem *mmio; + struct reset_control *audio_cgcr; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs; #endif @@ -618,6 +620,8 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) val = FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_ROW_CTRL_BMSK, ctrl->rows_index); val |= FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_COL_CTRL_BMSK, ctrl->cols_index); + reset_control_reset(ctrl->audio_cgcr); + ctrl->reg_write(ctrl, SWRM_MCP_FRAME_CTRL_BANK_ADDR(0), val); /* Enable Auto enumeration */ @@ -1269,6 +1273,9 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.ops = &qcom_swrm_ops; ctrl->bus.port_ops = &qcom_swrm_port_ops; ctrl->bus.compute_params = &qcom_swrm_compute_params; + ctrl->audio_cgcr = devm_reset_control_get_exclusive(dev, "swr_audio_cgcr"); + if (IS_ERR(ctrl->audio_cgcr)) + dev_err(dev, "Failed to get audio_cgcr reset required for soundwire-v1.6.0\n"); ret = qcom_swrm_get_port_config(ctrl); if (ret) -- 2.7.4

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[PATCH] ASoC: amd: acp: Fix support for a Huawei Matebook laptop
by Marian Postevca 17 Jun '25

17 Jun '25

Previous commit that added support for Huawei MateBook D16 2021 with Ryzen 4600H (HVY-WXX9 M1010) was incomplete. To activate support for this laptop, the DMI table in acp3x-es83xx machine driver must also be updated. Fixes: b5338b1b901e ("ASoC: amd: acp: Add support for a new Huawei Matebook laptop") Signed-off-by: Marian Postevca <posteuca(a)mutex.one> --- sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c index f85b85ea4be9..2b0aa270a3e9 100644 --- a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c +++ b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c @@ -354,6 +354,14 @@ static const struct dmi_system_id acp3x_es83xx_dmi_table[] = { }, .driver_data = (void *)(ES83XX_ENABLE_DMIC|ES83XX_48_MHZ_MCLK), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "HVY-WXX9"), + DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "M1010"), + }, + .driver_data = (void *)(ES83XX_ENABLE_DMIC), + }, { .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"), -- 2.43.0

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[PATCH v1] ASoC: tas2781: Move the "include linux/debugfs.h" into tas2781.h
by Shenghao Ding 16 Jun '25

16 Jun '25

Move the include linux/debugfs.h into tas2781.h for code clean. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2781.h | 4 ++++ sound/soc/codecs/tas2781-i2c.c | 3 --- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 40cd3bd079b5..3875e92f1ec5 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -17,6 +17,10 @@ #ifndef __TAS2781_H__ #define __TAS2781_H__ +#ifdef CONFIG_SND_SOC_TAS2781_ACOUST_I2C +#include <linux/debugfs.h> +#endif + #include "tas2781-dsp.h" /* version number */ diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index c40d8f754d89..9f4d965a1335 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -14,9 +14,6 @@ // #include <linux/crc8.h> -#ifdef CONFIG_SND_SOC_TAS2781_ACOUST_I2C -#include <linux/debugfs.h> -#endif #include <linux/firmware.h> #include <linux/gpio/consumer.h> #include <linux/i2c.h> -- 2.43.0

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add UCM2 support for MSI MAG B850M Mortar Wifi (ALC4080)
by GitHub pull_request - opened 16 Jun '25

16 Jun '25

alsa-project/alsa-ucm-conf pull request #581 was opened from DanielDecker: Solves problems with front panel audio and S/PDIF sound output for MSI MAG B850M Mortar Wifi, which uses the Realtek ALC4080 audio chip. I also tested line in (back side) and microphone input (front panel). [alsa-info.txt](https://alsa-project.org/db/?f=70ed25ea247c3ac8c253755a5c70e… Thanks in advance. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/581 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/581.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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